| 00:04.10 | jamko | Can someone get me a yum command for all the dependencies? |
| 00:06.21 | drmessano | Huh? |
| 00:06.49 | jamko | sorry wrong channel |
| 00:09.38 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-197-9.mia.bellsouth.net) |
| 00:13.40 | jamko | However not such an outlandish question, even with it being in the wrong channel.. ie - yum -y install gcc gcc-c++ kernel-devel bison openssl-devel \ |
| 00:13.41 | jamko | libtermcap-devel ncurses-devel doxygen curl-devel newt-devel |
| 00:13.56 | jamko | etc etc etc |
| 00:15.09 | raden | ask in #suse |
| 00:15.19 | raden | use zypper |
| 00:16.28 | jamko | nevermind. |
| 00:16.52 | leifmadsen | jamko: this is the minimum I've found for compiling basic asterisk |
| 00:16.53 | leifmadsen | yum install gcc gcc-c++ make wget subversion \ |
| 00:16.53 | leifmadsen | libxml2-devel ncurses-devel openssl-devel \ |
| 00:16.53 | leifmadsen | vim-enhanced |
| 00:17.07 | leifmadsen | vim-enhanced is optional |
| 00:19.39 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 00:21.17 | jamko | @leifmadsen: thanks, I'm good on asterisk.. My question was meant for #opensips .. BUT if you have a yum command for opensips dependencies on centos 5.3, I would really appreciate it.. : ) |
| 00:21.57 | b14ck | <3 vim-enhanced |
| 00:27.51 | leifmadsen | I don't use OpenSIPS |
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| 00:38.03 | ghostnik11 | dogboy: I know it has been a long time but i did, i was the guy that asked you about the info for how to get free outbound calls through google voice and asterisk + dialplan etc. |
| 00:38.19 | ghostnik11 | dogboy: i did it |
| 00:38.35 | DogBoy | neat ghostnik11 |
| 00:39.15 | ghostnik11 | dogboy: i don't have to use cellular minutes, thank you and guess what i didn't even need the dial plan because there is an app that does what the dial plan was going to do |
| 00:43.18 | ghostnik11 | dogboy: thank you again, and if you ever need to information on how i did it through a cell phone just let me know, because without your help i would have never figured it out, thank you agian |
| 00:44.15 | *** part/#asterisk ghostnik11 (~king@pool-71-125-20-221.nycmny.fios.verizon.net) |
| 00:44.20 | DogBoy | I don't have a data plan on my G1 ghostnik11 |
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| 00:53.47 | *** join/#asterisk LemensTS (~LemensTS@70.238.154.222) |
| 00:56.29 | LemensTS | if i call out using xlite to my sip provider to a phone, and click hold on xlite after the call is answered, moh works for like 20-30s then the call is hung up automatically. not seeing anything helpful in the cli...this is asterisk 1.6.2.10 |
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| 01:05.52 | pabelanger | LemensTS: Enable SIP debugs and see what happens |
| 01:08.42 | b14ck | Yo, quick question. Since you can do includes like: #include <*.conf> , is it somehow possible to do wildcard includes and specify a specific file to exclude? |
| 01:08.52 | b14ck | I want to include all .conf files, except for a file named config.conf. |
| 01:08.58 | b14ck | Any ideas? :) |
| 01:12.24 | pabelanger | b14ck: create a subfolder and symlink the .conf files? |
| 01:13.43 | b14ck | pabelanger, hrm |
| 01:13.57 | b14ck | I suppose I could. |
| 01:14.08 | b14ck | Actually, I should probably just re-do the directory structure. |
| 01:14.15 | b14ck | Doesn't make perfect sense to have it the way I do. |
| 01:14.17 | b14ck | Thanks. |
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| 02:30.29 | Micc | Is OrderlyQ a hosted only service or can you install it on your own server? |
| 02:33.08 | [TK]D-Fender | micc: yes you install it on your own server |
| 02:33.55 | Micc | TKD-Fender, any idea how much that option costs? I can't find pricing anywhere. And what kinds of dependancies does it have? I really didn't like queue metrics because it required java. |
| 02:34.19 | pabelanger | It is per agent, and on his website |
| 02:34.56 | pabelanger | http://orderlyq.com/orderlystatsse/pricing.html |
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| 02:37.59 | [TK]D-Fender | Micc: You know they have a nice contact page... it also alludes to having resellers which I suspect they may refer you to |
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| 02:44.08 | Micc | TKD-Fender, I've sent them multiple messages. Still waiting to hear back from them. |
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| 02:48.23 | [TK]D-Fender | Micc: they clearly don't want your money... |
| 02:49.35 | Micc | I kinda thought that when the free trial signup page broke and it gave me an out of space on device error. |
| 02:49.49 | Micc | then their international number wasn't reachable. |
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| 03:17.25 | stphn | any ss7 gurus around? |
| 03:20.14 | bougyman | you mean: people intimately familiar with the asterisk codebase? |
| 03:20.20 | bougyman | yes, some of those people are here from time to time. |
| 03:20.44 | bougyman | rarely do the questions presented in here need such level of support, though. |
| 03:21.17 | bougyman | stphn: is this on digium hardware? |
| 03:23.42 | [TK]D-Fender | bougyman: One does not have to know all about the * codebase to know about SS7 |
| 03:23.58 | bougyman | no, that's just what I consider a 'guru' |
| 03:23.58 | [TK]D-Fender | stphn: What in particular do you want to know? |
| 03:24.08 | bougyman | i suppose everyone has their own perspective on that. |
| 03:24.15 | [TK]D-Fender | bougyman: * codebase != SS7 |
| 03:24.19 | bougyman | i was more trying to drive to a specific question. |
| 03:24.21 | stphn | well, i need to send TNS info to the switch I am interfacing with |
| 03:24.29 | [TK]D-Fender | bougyman: There is some "perspective" for you to consider |
| 03:24.30 | stphn | and I am using asterisk of course |
| 03:24.59 | [TK]D-Fender | stphn: Which driver, and what hardware? |
| 03:25.09 | stphn | i looked in isup.c from libss7, and it seems that the TNS piece in not complete |
| 03:25.16 | bougyman | ^^ [TK]D-Fender the what hardware was my specific query to him. |
| 03:25.21 | stphn | I've got a Sangoma A104 installed |
| 03:25.26 | bougyman | there we go. |
| 03:25.29 | stphn | and working, I might add, except for the TNS portion |
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| 03:25.54 | stphn | asterisk 1.6 and the latest dahdi |
| 03:25.59 | stphn | from the Centos Repos |
| 03:26.07 | stphn | not trixbox or anything like that |
| 03:26.10 | stphn | just vanilla asterisk |
| 03:26.35 | [TK]D-Fender | stphn: There is generally very little by way of SS7 user in here normally... have you already posted this on the mailing lists? |
| 03:26.47 | stphn | I looked at chan_ss7, but it seems that its mostly for the ITU variant |
| 03:27.05 | stphn | I haven't, thought I would check here first |
| 03:27.29 | stphn | well, perhaps I can fill that gap when I become more comfortable with it |
| 03:27.35 | [TK]D-Fender | stphn: Not a bad idea, but statistically unfavourable as SS7 goes |
| 03:27.42 | stphn | gotcha |
| 03:27.52 | bougyman | stphn: the guys in #sangoma will assist with sangoma-specific ss7 stuff from ~9-5 eastern, m-f |
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| 03:28.21 | stphn | I noticed, but I imagine they'll want you to use their stack, as opposed to libss7 |
| 03:28.30 | bougyman | likely. |
| 03:28.33 | stphn | surprisingly, their ss7 stack is pricy |
| 03:28.45 | stphn | pricey |
| 03:29.00 | bougyman | the smg-ss7? |
| 03:29.09 | stphn | well, thanks for you time guys |
| 03:29.14 | stphn | i appreciate it |
| 03:29.40 | bougyman | it's on their ftp site for download i didn't know it had a cost if you had their hardware. |
| 03:30.01 | stphn | Yeah, i believe you have to license it |
| 03:30.05 | stphn | but I am not entirely sure |
| 03:30.18 | stphn | i like how clean and integrated libss7 is, too |
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| 03:30.21 | bougyman | oh, the ss7boxd binary is licensed |
| 03:30.28 | stphn | so i would prefer it, were it complete |
| 03:31.13 | stphn | chan_ss7 looks like a confusing mess with scant documentation, plus it seems like itu is the only variant it supports |
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| 04:03.07 | golikwid|mac | anyone know off hand the asterisk command for highest number of simultaneous calls so far the system has handled |
| 04:03.30 | stphn | not sure there is a cli command |
| 04:03.34 | stphn | have you looked into snmp? |
| 04:03.56 | golikwid|mac | whats that |
| 04:04.06 | stphn | simple network management protocol |
| 04:04.15 | stphn | there are MIBs for asterisk |
| 04:04.15 | golikwid|mac | hm |
| 04:04.24 | golikwid|mac | sounds complicated lol |
| 04:04.40 | golikwid|mac | i think i have an app for that on my iphone |
| 04:04.43 | stphn | it's really pretty simple when you get your head around in |
| 04:04.43 | stphn | lol |
| 04:05.19 | stphn | quick google search turned this up |
| 04:05.19 | stphn | http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131 |
| 04:05.43 | stphn | and here |
| 04:05.44 | stphn | http://www.voip-info.org/wiki/view/Asterisk+monitoring |
| 04:05.49 | golikwid|mac | i love yu |
| 04:05.51 | golikwid|mac | yum |
| 04:06.00 | stphn | I used MRTG on my network |
| 04:06.10 | golikwid|mac | that was a typo not a confession of love lol |
| 04:07.06 | golikwid|mac | im always scared of screwing up a production system installing new things... |
| 04:07.06 | stphn | bed time for me |
| 04:07.15 | golikwid|mac | this is a weekend project for sure |
| 04:07.19 | stphn | ah yeah, but that's the best way to learn |
| 04:07.20 | stphn | lol |
| 04:07.29 | golikwid|mac | i have learned alot than |
| 04:07.29 | stphn | good luck! |
| 04:07.33 | golikwid|mac | thanks |
| 04:15.06 | jamko | Anyone experimented sending SIP URI calls with TCP transport? |
| 04:35.29 | *** join/#asterisk TTT_Travis (~Travis@cle-bb-cable2-ws-85.dsl.airstreamcomm.net) |
| 04:37.32 | TTT_Travis | My phone system runs both of my businesses -- looking for an ip phone that would allow me to connect to two extensions -- to identify which business the call is coming in for and to set the caller id when I call to the right business---possible? |
| 04:41.00 | Corydon76-dig | TTT_Travis: PRI circuit or SIP trunking? |
| 04:41.15 | TTT_Travis | SIP trunking |
| 04:41.39 | Corydon76-dig | Do you have internal extensions? |
| 04:43.08 | TTT_Travis | Yeah currently just 1 extension per phone...I have a rigged up dial plan so I dial a 9 before outgoing calls for biz1 and 8 for outgoing calls for biz2 so it switches to a different trunk which sets the caller id correctly |
| 04:43.42 | TTT_Travis | But it seems like there has to be a better way |
| 04:43.45 | Corydon76-dig | That's exactly what I'd suggest |
| 04:43.53 | *** join/#asterisk KingDavidNYC (~Chris1232@ool-435554a8.dyn.optonline.net) |
| 04:43.59 | KingDavidNYC | Hello everybody |
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| 04:44.39 | TTT_Travis | ok well the problem we have is with incoming calls..the receptionist doesn't know which business the call is coming in for --- for example if someone calls for Biz1 she anwers -- hello Biz1 -- but currently she doesn't have any indication |
| 04:46.08 | Corydon76-dig | I'd suggest changing the CallerID to prefix a letter for each business |
| 04:54.10 | TTT_Travis | Is there an IP phone that could say set line 1 to Extension 5001 and line 2 to Extension 5002 just for the receptionist phone? so then calls would come in for Biz1 on Line1 and Biz2 on Line 2 -- this is how are old dinosaur phone system worked |
| 04:54.28 | TTT_Travis | our* |
| 04:54.30 | KingDavidNYC | friends, I have to write a program that has to give the caller the option, at the end of the call, to press 1 if he wants to recharge, and then continue the call... The way I would do this is to write a php script that runs every minute in linux, but maybe there is an easier way to this, can anyboby please tell me if I am overworking it? |
| 04:55.05 | KingDavidNYC | ...or if I am right :) |
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| 05:53.54 | AliRezaTaleghani | ChannelZ: hi, do u have time, i had a problem with AGI-perl script |
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| 05:55.04 | metfan2007 | Hi all! anyone successfullly tested SSML and CEPSTRAL in the dial plan? |
| 05:58.10 | AliRezaTaleghani | L-) hi all, can anyone give me a help about this AGI problem... my AGI-perl script is this:http://paste.ubuntu.com/479250/ and where i use the, this is the log http://paste.ubuntu.com/479251/ |
| 05:58.12 | AliRezaTaleghani | it do nothing... |
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| 06:28.45 | Bendbanks | hi everyone just done a new asterisk now install with the asterisk gui option and when I go to the http://ip address nothing happens anythoughts |
| 06:29.25 | TTT_Travis | try http://ipaddresshere:8088 |
| 06:30.29 | TTT_Travis | I think it's actually http://ipaddresshere:8088/asterisk/static/config/index.html |
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| 06:31.39 | Bendbanks | no to either of those TTT_Travis |
| 06:31.52 | TTT_Travis | do you get 404 or no response? |
| 06:31.53 | Bendbanks | server is up I can ssh in |
| 06:32.14 | Bendbanks | unable to establish connection |
| 06:32.20 | Bendbanks | is the error |
| 06:32.23 | TTT_Travis | telnet localhost 8088 |
| 06:32.31 | TTT_Travis | see if that ^ connects |
| 06:34.03 | Bendbanks | no connection refused |
| 06:34.32 | TTT_Travis | hmm that means the gui server isn't running |
| 06:34.44 | TTT_Travis | restart asterisk? |
| 06:38.07 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:38.08 | Bendbanks | mmm i actually can not do that it want let me in asterisk stop will stop it asterisk -r to restart right |
| 06:38.23 | Bendbanks | none will work |
| 06:38.32 | TTT_Travis | what distro are you on? |
| 06:38.50 | Bendbanks | centos |
| 06:39.40 | Bendbanks | which is what comes from asterisknow iso |
| 06:40.11 | TTT_Travis | hmmm I'm a debian guy |
| 06:40.23 | TTT_Travis | I think in centos you type "service asterisk stop" and start etc. |
| 06:45.17 | Igneous | does anyone in here happen to have experience with CAC Adits? |
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| 06:56.25 | metfan2007 | any one with app_swift? |
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| 07:01.18 | AliRezaTaleghani | L-) hi |
| 07:01.54 | AliRezaTaleghani | which application, i can use to Say the text" Hello"! not the Say Phonetic |
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| 07:49.55 | klashniv | hullo all; quick question: whats the asterisk 1.6 equivalent command for 'g729 show licenses'? thanks |
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| 07:59.08 | tzafrir | klashniv, should be the same |
| 07:59.24 | tzafrir | I suspect that the g729 codec module is not loaded |
| 07:59.57 | klashniv | thanks |
| 08:00.11 | klashniv | wanted to check if the license was installed |
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| 08:03.12 | mechbangirc | asterisk is automatically generating incoming calls. |
| 08:03.35 | mechbangirc | [Aug 17 12:33:00] NOTICE[26464] chan_dahdi.c: Got event 18 (Ring Begin)...[Aug 17 12:33:01] NOTICE[26464] chan_dahdi.c: Got event 2 (Ring/Answered)... |
| 08:03.53 | tzafrir | klashniv, the module can load regardless of licenses |
| 08:04.08 | mechbangirc | any idea? |
| 08:04.21 | klashniv | thanks, decided to use another codec, cheers |
| 08:04.29 | tzafrir | mechbangirc, it got an event from the low-level driver |
| 08:04.33 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 08:05.03 | tzafrir | mechbangirc, what device is connected there? |
| 08:05.05 | mechbangirc | tzafrir, you mean dahdi? |
| 08:05.10 | tzafrir | yes |
| 08:05.55 | mechbangirc | its tdm400p |
| 08:06.11 | tzafrir | is it a FXS module? Is there a phone connected to it? |
| 08:06.36 | mechbangirc | yeah it has 2 fxs and 2 fxo modules |
| 08:07.02 | tzafrir | was this event on a FXO or on a FXS? |
| 08:08.02 | mechbangirc | i'm not quite sure, but i think its on FXO |
| 08:08.31 | tzafrir | Did it only happen once? Does it happen occasionally? All the time? |
| 08:08.52 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 08:09.25 | mechbangirc | it happens all the time, normally after every 2 to 3 hours |
| 08:11.25 | ChannelZ | qwest used to reset our lines or do something crazy every night and I'd get weird crap like that |
| 08:11.28 | *** join/#asterisk BitMonkey (~bitmonkey@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 08:11.31 | *** join/#asterisk ruied (~ruied@bl14-240-129.dsl.telepac.pt) |
| 08:11.40 | BitMonkey | greets all |
| 08:11.57 | BitMonkey | was wondering, is there an asterisk-addons that can be used with asterisk1.8beta? |
| 08:12.50 | tzafrir | BitMonkey, no. It has been merged into Asterisk ( addons/ ) |
| 08:12.58 | BitMonkey | ah excellent, thanks! |
| 08:12.58 | tzafrir | Modules are not built by default |
| 08:13.00 | BitMonkey | :) |
| 08:13.02 | BitMonkey | gotcha |
| 08:13.04 | BitMonkey | good to know |
| 08:13.07 | BitMonkey | cheers! |
| 08:15.25 | EmleyMoor | Is there a way to make asterisk accept SIP call requests from a registered peer without further authentication? |
| 08:18.58 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 08:19.52 | ChannelZ | insecure=invite |
| 08:20.43 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:20.59 | EmleyMoor | Hmmm... tried that - obviously not the answer to my problem - still waiting on responses from elsewhere |
| 08:21.52 | klashniv | ok, another dumb question, how do i load the g729 codec in asterisk 1.6? |
| 08:22.16 | hrhrhr | EmleyMoor: explain your full problem to ChannelZ. he is pretty good |
| 08:22.27 | ChannelZ | module load codec_g729a |
| 08:23.41 | klashniv | module load codec_g729a Unable to load module codec_g729a |
| 08:24.07 | klashniv | module load format_g729 Module 'format_g729' already exists |
| 08:24.35 | klashniv | tried the format_g729 as it was the only file with g729 in its name |
| 08:24.53 | ChannelZ | then you're missing half of it |
| 08:24.59 | Micc_ | I think there is something wrong with T38 ECM. I set it to none, sip show peer shows none, but it still manages to negotiate it and try to use it. |
| 08:25.04 | EmleyMoor | When I am connected from my N97 over 3G to my asterisk box, I can only receive calls. If I try to make one, the asterisk box sends a 407, and then a 401 when it rejects the phone's response. The same phone functions just fine over WiFi, at least at home. I am seeking information to help me track down the cause of the problem. Please see my paste at http://asterisk.pastey.net/139642 |
| 08:25.19 | klashniv | thing is, if its already loaded, why wont g729 show licenses work? |
| 08:26.53 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 08:26.58 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-uhkoiwhwwmwqjrwi) |
| 08:27.02 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:27.58 | ChannelZ | klashniv: because it's not loaded. You pasted the failure yourself |
| 08:28.15 | ChannelZ | do 'module show like g729' |
| 08:28.16 | *** join/#asterisk Secret_Hamster (~pdiddly@host81-149-30-66.in-addr.btopenworld.com) |
| 08:29.12 | klashniv | format_g729.so Raw G729 data |
| 08:29.36 | pif | hi, what file contains command aliases again? |
| 08:29.44 | ChannelZ | that's it? So that's only the file format handler, but not the codec. You're missing the codec. |
| 08:29.47 | hrhrhr | ChannelZ: off the top of your head, common reasons for the following? WARNING[3304]: chan_iax2.c:7771 socket_process: Call rejected by 192.168.1.9: <Unknown> |
| 08:29.48 | pif | found |
| 08:30.07 | klashniv | doesn't asterisk 1.6 come with a g729a codec? |
| 08:30.15 | ChannelZ | no |
| 08:30.17 | hrhrhr | klashniv: mine doesn't seem to |
| 08:30.26 | ChannelZ | you have to go download it |
| 08:30.45 | klashniv | ok, cool, thanks |
| 08:30.53 | *** join/#asterisk hariom (~hariom@122.170.17.246) |
| 08:31.03 | ChannelZ | You need to get the one that matches your asterisk major version, and you can also get them built/optimized for different architectures |
| 08:32.00 | ChannelZ | hrhrhr: not sure. is 192.168.1.9 a remote system than the one that spit out that message? |
| 08:32.44 | hrhrhr | yes |
| 08:33.07 | hrhrhr | one is native asterisk 1.4 and the other is asterisknow (1.6) |
| 08:33.14 | *** join/#asterisk Tim_Toady (~moi@178.128.17.183.dsl.dyn.forthnet.gr) |
| 08:33.20 | ChannelZ | so the other end rejected the call, if it's your box look on its console and see why |
| 08:33.35 | ChannelZ | EmleyMoor: try insecure=invite,port for that peer |
| 08:33.38 | hrhrhr | iax debug output is lacking somewhat |
| 08:33.54 | hrhrhr | as far as i can see tho, the box that produces that error is not authing to the other side |
| 08:34.53 | ChannelZ | well without seeing any configs or dials I can't guess much |
| 08:35.21 | hrhrhr | ok cheers |
| 08:35.34 | hrhrhr | it just seems that i must figure out the way freepbx does stuff |
| 08:35.50 | hrhrhr | bit of a steep learning curve which im sure its not sposed to be :P |
| 08:37.11 | ChannelZ | I guess on paper freepbx is supposed to make things easier but when I looked at it it was a little confusing and the documentation beyond the very basics was pretty much non-existant |
| 08:37.37 | EmleyMoor | ChannelZ: Will try that shortly |
| 08:38.22 | ChannelZ | EmleyMoor: Whatever SIP client you're using seems to be kind of crap, it sends the INVITE, Asterisk rejects and tells it how to authenticate, but it just ignores it and starts registering again. |
| 08:39.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:39.58 | hrhrhr | agreed. |
| 08:42.06 | EmleyMoor | ChannelZ: Still the same with that insecure setting. I think you are coming to the same conclusion as me - blame Nokia |
| 08:42.35 | Micc_ | t38 is such a pain in the ass. |
| 08:44.34 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:46.53 | ChannelZ | I dont understand why you're getting a 'proxy authentication required' as opposed to just an 'unauthorized' |
| 08:46.57 | *** join/#asterisk ankur_6997 (~Dev_1@122.177.231.250) |
| 08:47.03 | hrhrhr | and it works for him over wifi... |
| 08:47.40 | hrhrhr | pastebin a successful call over wifi EmleyMoor? |
| 08:47.44 | ankur_6997 | hi can i connect PSTN telephone line to my asterix box so that the calls comming in that lines can be terminated at an IVR prompt ? |
| 08:48.20 | ankur_6997 | what kind of hardware will be required ? |
| 08:48.52 | EmleyMoor | hrhrhr: Cannot do that until this evening at least - but may well, so that the difference can be seen. |
| 08:49.24 | ChannelZ | ankur_6997: didn't we answer this question a week or two ago? |
| 08:49.46 | EmleyMoor | I have posted the information to Nokia - I'll see if it gets me anywhere. |
| 08:49.48 | ankur_6997 | yes but chat was intruppted |
| 08:49.54 | *** join/#asterisk lftsy (~lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 08:50.03 | ankur_6997 | ok using linksys s3102 ? |
| 08:50.43 | hariom | Hi, I am getting error while playing a prompt: WARNING[2621]: format_ogg_vorbis.c:521 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! |
| 08:51.44 | hariom | ankur_6997: do you have pstn line connected with asterisk? |
| 08:52.04 | ankur_6997 | not yet |
| 08:52.51 | ankur_6997 | but soon if some one clearly tell me that linksys s3102 will be fine |
| 08:53.42 | ChannelZ | spa3102 |
| 08:54.07 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 08:54.11 | ChannelZ | It has 1 FXS (plugging a telephone into) and 1 FXO (plugging into the wall to your telco) |
| 08:55.01 | ChannelZ | It's slightly confusing to configure but works and is cheaper than a TDM card if all you need is one line |
| 08:56.39 | ChannelZ | EmleyMoor: what version of asterisk is this |
| 08:56.50 | *** join/#asterisk chasing`Sol (~rc4@62.114.241.91) |
| 08:57.47 | EmleyMoor | 1.4.21.2~dfsg-3+lenny1 |
| 08:58.26 | ankur_6997 | thanks channelz |
| 08:59.33 | hrhrhr | if we have any iax gurus here, i'd be interested to know why AUTHREQ is not sent in some instances |
| 08:59.42 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 09:00.18 | *** join/#asterisk newasterx (~asda@114.199.103.188) |
| 09:01.19 | EmleyMoor | An upgrade to 1.6.2.9-1 is reasonably possible in the fairly near future and can probably be brought forward if it will help. |
| 09:01.58 | ChannelZ | it might but I really dont know |
| 09:02.49 | EmleyMoor | As it is, I may be upgrading the hardware on that box soon - if so, the new installation will be that version |
| 09:03.12 | chasing`Sol | hi guys, if i want to write an application such that when someone calls in and start interacting with asterisk, asterisk starts interacting with a database fetching, updating the caller's records, where do i start? |
| 09:03.48 | ChannelZ | chasing`Sol: AGI |
| 09:03.51 | chasing`Sol | shall i write a dial plan, or shall i interface asterisk with an agi? |
| 09:03.52 | ChannelZ | more than likely |
| 09:04.54 | *** part/#asterisk klashniv (~klashniv@langw.roketelkom.co.ug) |
| 09:05.38 | ChannelZ | ugh 3am.. way past my bed time |
| 09:05.51 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 09:08.14 | *** join/#asterisk newasterx (dasdasdsad@114.199.103.188) |
| 09:08.26 | newasterx | Hellow all... |
| 09:09.41 | newasterx | may i drop a question ? |
| 09:11.03 | newasterx | Hello |
| 09:11.25 | EmleyMoor | newasterx: No need to ask to ask - just ask |
| 09:11.40 | newasterx | oke emley... |
| 09:12.03 | newasterx | I am implementing ToIP by using asterisk 1.6.12 |
| 09:12.16 | newasterx | using SIPcon1 for textphone |
| 09:12.45 | newasterx | my question is why i can not sent a text message while voice message is very clear |
| 09:12.47 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 09:16.17 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 09:19.57 | ankur_6997 | can agi be used to set multiple varibles that can be used in asterisk ? in just one agi call from dialplan? |
| 09:21.18 | WIMPy | ankur_6997: Sure |
| 09:21.47 | WIMPy | You can do as much as you like before returning. |
| 09:23.40 | ankur_6997 | i can't find a example using fastagi do you have one ? |
| 09:25.18 | WIMPy | Just repeat to 'set variable'. |
| 09:26.20 | Secret_Hamster | can anyone think what would cause a phone to reboot on call pick (and sometimes just being called) |
| 09:26.43 | EmleyMoor | What sort of phone? |
| 09:27.07 | Secret_Hamster | it's a cisco 7912, it's something either in the configs. It also reboots if another phone in the huntgroup picks up the call to the huntgroup |
| 09:27.15 | Secret_Hamster | however, outbound calls are fine |
| 09:27.23 | WIMPy | Secret_Hamster: Buggy Software? Or someone who doesn't like you configured something nasty. |
| 09:27.53 | EmleyMoor | So basically it reboots when it stops ringing? |
| 09:28.03 | Secret_Hamster | the thing is I'm abit of newb to this, and I've taken over the system from the previous admin, but left pretty much alone |
| 09:28.29 | Secret_Hamster | it reboots if you call it directly (sometimes only when you try to pick the call up) |
| 09:28.49 | Secret_Hamster | it reboots if someone else in the hunt group picks the call to the huntgroup up |
| 09:29.23 | EmleyMoor | Does it reboot if the caller hangs up before it's answered? |
| 09:30.08 | Secret_Hamster | let me check |
| 09:30.20 | Secret_Hamster | yes |
| 09:30.51 | EmleyMoor | Secret_Hamster: I guess you have other similar phones about - am I right? |
| 09:31.30 | Secret_Hamster | yep, mine is the same, also tested with another one |
| 09:31.42 | Secret_Hamster | it just seems to be something to do with that login/exten |
| 09:31.52 | EmleyMoor | Have you tried exchanging the configs? |
| 09:32.02 | Secret_Hamster | yep, works with other configs |
| 09:32.12 | Secret_Hamster | other phones do not work with that config |
| 09:32.18 | EmleyMoor | The problem moves with the config? |
| 09:32.29 | Secret_Hamster | yep |
| 09:32.48 | Secret_Hamster | but as far as I can tell, it has the same setup as all the others |
| 09:33.11 | EmleyMoor | In that case I'd be looking te make a new config for that phone - preferably starting from one you know works. |
| 09:33.23 | Secret_Hamster | done that already |
| 09:33.42 | Secret_Hamster | used the same script to create other test ones, they work fine |
| 09:34.13 | Secret_Hamster | I presume it has something to do with the database info in the asterisk server, but I'm abit light on that area |
| 09:34.37 | Secret_Hamster | I've looked at the show sip peers, that seems to be okay, not sure what else to look at |
| 09:36.55 | hrhrhr | Secret_Hamster: im sure there's an option to reboot fones via sip invite |
| 09:36.58 | hrhrhr | you might wanna look into that |
| 09:37.19 | Secret_Hamster | it rings sometimes |
| 09:37.32 | Secret_Hamster | but then cuts out on pickup |
| 09:37.48 | Secret_Hamster | I presume that is in the server config? |
| 10:06.24 | *** join/#asterisk ruyo (~psantos@a83-132-248-161.cpe.netcabo.pt) |
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| 10:23.14 | newasterx | hello |
| 10:23.46 | newasterx | anybody have experience of implementing text over IP ? |
| 10:25.21 | ruyo | newasterx, like mIRC? |
| 10:25.33 | ruyo | :P |
| 10:25.55 | WIMPy | Or e-mail? |
| 10:27.51 | Secret_Hamster | XMPP? |
| 10:28.33 | WIMPy | Oh, wait. I think we're actually sending text over IP right here. |
| 10:36.04 | ruyo | Does a Wi-Fi AP with WMM read DSCP values? |
| 10:46.55 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 10:50.32 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:52.03 | Secret_Hamster | Okay found the answer out to my issue, the rebooting phone. The name of the user appeared to be too long |
| 10:52.05 | Secret_Hamster | agg |
| 10:52.13 | Secret_Hamster | two days I've been looking at this |
| 10:53.45 | WIMPy | wonders if that would work with the callerid as well. |
| 10:54.47 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
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| 11:32.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:33.35 | newasterx | hai |
| 11:33.54 | newasterx | i means like this |
| 11:33.56 | newasterx | http://www.voip-info.org/wiki/view/Real-time+text |
| 11:34.09 | newasterx | as asterisk support video, voice |
| 11:34.26 | newasterx | and now we can combine with text tooo |
| 11:34.33 | newasterx | realtime text |
| 11:35.30 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 11:39.04 | newasterx | ruyo |
| 11:40.58 | tzafrir | newasterx, you mean: instant messaging? |
| 11:41.14 | tzafrir | There is a IM protocol for SIP called SIMPLE |
| 11:42.05 | tzafrir | Likewise XMPP is known to be used as a IM protocol of Jingle (a slight understatement) |
| 11:42.31 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 11:43.12 | newasterx | instant messaging and realtime is little bit different |
| 11:43.30 | newasterx | http://www.voip-info.org/wiki/view/Real-time+text |
| 11:43.34 | newasterx | just read that |
| 11:44.00 | newasterx | curently i am using asterisk 1.6.12 for SIP server |
| 11:44.13 | newasterx | and SIPCon1 as textphone |
| 11:45.37 | ruyo | Apparently, dialplan-wise, it's pretty much the same as a regular phone call. |
| 11:45.40 | newasterx | but i am still can not send text to another SIPCon1 |
| 11:45.48 | ruyo | Never tried the real-time text though. |
| 11:46.05 | newasterx | ruyo : try this |
| 11:46.12 | newasterx | its pretty cool |
| 11:46.39 | ruyo | newasterx, try forcing the codec to t140 to see what happens. |
| 11:46.51 | ruyo | Only allow=t140 |
| 11:47.11 | newasterx | yes.. |
| 11:47.13 | ruyo | Oh, and allow=h263. |
| 11:47.14 | newasterx | already |
| 11:47.15 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 11:47.26 | ruyo | "at least one video codec as H.261, H.263 or H.263+ is needed" |
| 11:47.32 | newasterx | but simple as what u say |
| 11:47.45 | newasterx | but not so simple as what u say :) |
| 11:48.16 | ruyo | Does * actually makes the call? |
| 11:48.34 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 11:48.42 | newasterx | yes |
| 11:48.51 | newasterx | voice is very clear |
| 11:49.01 | newasterx | using SIPCon1 |
| 11:49.44 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 11:50.05 | ruyo | With only t140 and h263 you can have voice? |
| 11:56.24 | newasterx | yes |
| 11:59.47 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
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| 12:32.26 | Kobaz | man this is frusterating |
| 12:32.35 | drmessano | Spelling? |
| 12:32.38 | Kobaz | that too |
| 12:33.04 | Kobaz | any time i update this one customer to any 1.6.2... there's horrendous problems after it's been running for more than a day |
| 12:33.29 | Kobaz | i think it's probably related to this bug that causes moh to get corrupted |
| 12:33.47 | Kobaz | i think if i can find and fix that... it might be good |
| 12:36.37 | Kobaz | http://pastebin.com/GCAvpDdS |
| 12:36.43 | Kobaz | that just happend about 10 minutes ago |
| 12:37.05 | Kobaz | it doesn't usually crash when i do that... but that time it did |
| 12:41.15 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 12:47.42 | *** join/#asterisk hariom (~hariom@122.169.90.109) |
| 12:48.08 | hariom | D-Fender, was dissconnected. Did you write any reply? |
| 12:48.26 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:48.42 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:49.15 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 12:49.39 | hariom | anybody else to offer help? |
| 12:50.15 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 12:51.32 | Chainsaw | Kobaz: Have you spoken to jkroon? He had a pretty significant 1.6.2 crasher at one point. |
| 12:52.33 | [TK]D-Fender | hariom: You did not write anything since I arrived until that question. |
| 12:53.14 | Kobaz | Chainsaw: i haven't |
| 12:53.27 | Kobaz | Chainsaw: i just posted it to -dev... which i should have done ages ago |
| 12:53.37 | Chainsaw | Kobaz: I believe he solved it in the end; I've been keeping a 1.6.1 release in the portage tree just for him. |
| 12:53.46 | zamba | how does asterisk work in a virtualized environment? |
| 12:53.56 | hariom | This I wrote before, in case if didn't reach you. My net got disconnected for few minutes. [TK]D-Fender: taking the yesterday conversation forward, I am not able to play ogg/vorbis file properly. First, it is confirm that it is required to resample to 16bit 8Khz. Second after doing that if I run, I get error that seeking is not supported on ogg/vorbis streams |
| 12:53.58 | Chainsaw | zamba: Not very well I would say. |
| 12:54.17 | [TK]D-Fender | hariom: And I see NOTHING |
| 12:54.18 | zamba | Chainsaw: why's that? |
| 12:54.24 | Chainsaw | zamba: VoIP has low latency requirements. It does not deal well with high latency or unpredictable latency. |
| 12:54.31 | hariom | didn't get |
| 12:54.36 | Chainsaw | zamba: Especially the latter can be problematic in a virtualised environment. |
| 12:54.37 | hariom | what do you want to convey? |
| 12:55.17 | hariom | How to play ogg vorbis files on playback? |
| 12:55.26 | Kobaz | Chainsaw: ah... i'm stuck on 1.6.0.19 because of this |
| 12:55.42 | Kobaz | Chainsaw: there's other issues when i try and bump to a newer 1.6.0 too |
| 12:55.48 | [TK]D-Fender | hariom: show me the FAILED ATTEMPT, and the file. |
| 12:55.55 | Kobaz | Chainsaw: this customer just has the crazyest problems i've ever seen |
| 12:56.52 | Chainsaw | Kobaz: And there are no funky third-party driver stacks involved? |
| 12:57.59 | hariom | [TK]D-Fender, http://pastebin.com/0xc9gnGY |
| 12:58.40 | *** join/#asterisk kpettit (~keith@99-172-37-26.lightspeed.tblltx.sbcglobal.net) |
| 12:58.48 | [TK]D-Fender | hariom: [Aug 17 15:44:26] WARNING[2786]: format_ogg_vorbis.c:521 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! <-- seems to say your file is bad |
| 12:58.59 | Kobaz | Chainsaw: other than sangoma... no |
| 12:59.13 | Chainsaw | Kobaz: I consider pipewan a third-party driver stack. |
| 12:59.25 | Kobaz | i don't consider it a funky one |
| 12:59.49 | hariom | [TK]D-Fender: I can play this file using "play" on my system |
| 12:59.55 | Kobaz | that could be related though |
| 13:00.06 | Kobaz | i can't produce the problem in the lab with just using sip |
| 13:00.09 | Chainsaw | hariom: It wants to seek. It could be something as mundane as seeking around metadata. Try to keep to a bare stream. |
| 13:00.16 | Kobaz | maybe i'll try and load up a t1 with the sangoma drivers |
| 13:00.46 | [TK]D-Fender | [08:59]<hariom>[TK]D-Fender: I can play this file using "play" on my system <- irrelevant |
| 13:00.49 | *** join/#asterisk hrhrhr (~c1@213.1.224.2) |
| 13:00.58 | Chainsaw | Kobaz: *nod* I'm not convinced it is a pure SIP problem. I have had a lot less crashers then others here and that seems to be because I have my ISDN behind SIP gateways. |
| 13:01.07 | hariom | [TK]D-Fender: GOT IT. |
| 13:01.08 | [TK]D-Fender | hariom: * does not like stream formatted OGG/Vorbis |
| 13:01.15 | Kobaz | Chainsaw: i've been moving to those... it's been much better on those boxes |
| 13:01.24 | Chainsaw | Kobaz: So SIP-wise it's fairly solid. DAHDI & third-party drivers... not so much. |
| 13:01.32 | [TK]D-Fender | hariom: Just like it does not like VBR MP3's, ID3 tags, etc |
| 13:01.43 | Kobaz | Chainsaw: i've gotten really burned by dahdi this year... so none of the new installations are using line cards |
| 13:02.01 | Kobaz | everything from t1 randomly going down to complete box lockups |
| 13:02.02 | Chainsaw | Kobaz: What brand have you gone with? I ended up with Patton on the system I inherited. |
| 13:02.11 | Kobaz | once dahdi was gone... perfect operation |
| 13:02.14 | Kobaz | adtran |
| 13:02.23 | Chainsaw | Kobaz: They're quirky telco gear with useless documentation. But they are very stable. |
| 13:02.51 | Chainsaw | Kobaz: Adtran, okay. I'll put them on the list as well. |
| 13:03.07 | Kobaz | i really like them |
| 13:03.14 | Kobaz | they have 10 year support on boxes |
| 13:04.51 | Kobaz | http://www.voipgorilla.com/ProductDetails.asp?ProductCode=Patton%20Smartnode%204960 |
| 13:04.54 | Kobaz | you use something like that? |
| 13:05.27 | Chainsaw | I have 4118 for my analog gear and 4634 to talk to ISDN BRI. |
| 13:05.31 | Kobaz | it says multi port... is that one of those stupid boxes where you buy the box with only one t1 activated and have to buy licenses to use the rest? |
| 13:05.51 | Chainsaw | I don't do T1/ISDN PRI. My deployment is not that big. |
| 13:06.09 | drmessano | VoIP Gorilla? |
| 13:06.27 | Kobaz | i found an 8 span pri for cheap... but firmware wise you can only use one port... and then you pay like 300 dollars each to activate the extra ports |
| 13:06.32 | Kobaz | drmessano: first hit in froogle |
| 13:06.38 | Kobaz | drmessano: dunno |
| 13:06.44 | bougyman | crazy |
| 13:06.44 | drmessano | Ook ook? |
| 13:06.59 | bougyman | we pay about $4k for our 8 port (108ds) but they're rock solid. |
| 13:07.14 | bougyman | that's with EC. |
| 13:07.30 | bougyman | for the ones not facing the telco it's abotu $1600k for 8 ports |
| 13:07.40 | bougyman | er just $1600 |
| 13:07.43 | Kobaz | interesting |
| 13:08.12 | Kobaz | a 108ds of what |
| 13:08.17 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 13:08.21 | Kobaz | froogle wants to sell me hilighters |
| 13:08.42 | Kobaz | PENSXS159 - Highlighters, Retractable, 108/DS, Assorted amazon.com |
| 13:08.53 | drmessano | Retractable? Sweet |
| 13:08.56 | drmessano | BRB then |
| 13:09.41 | hariom | [TK]D-Fender: http://pastebin.com/L2EyktsV |
| 13:10.10 | [TK]D-Fender | hariom: Don't know what to tell you.... |
| 13:10.12 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 13:10.24 | [TK]D-Fender | hariom: Why are you even using OGG again? |
| 13:10.40 | hariom | Whats the meaning of stream file? It is a normal ogg file without any ID3 tags |
| 13:10.58 | bougyman | Kobaz: http://www.sangomacards.com/Sangoma-A108X-p/sangoma%20a108x.htm |
| 13:10.58 | Kobaz | Chainsaw: the weird thing... is same dahdi drivers... same sangoma drivers... 1.6.0 on this box is just fine... switch to 1.6.2 and it's trouble |
| 13:11.08 | drmessano | You're banging a SQUARE peg into a ROUND hole.. Use WAV or MP3 |
| 13:11.12 | Kobaz | bougyman: oh yeah... but i'm staying away from line cards |
| 13:11.17 | Kobaz | bougyman: i know sangoma is cheap |
| 13:11.30 | *** part/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 13:11.35 | bougyman | i don't think it's that cheap |
| 13:11.44 | bougyman | they're usually a bit higher than rhino and digium. |
| 13:11.59 | Kobaz | yeah but rhino sucks |
| 13:12.12 | Kobaz | and digium... well... i haven't used their t1.. but i've had problems with the analogs |
| 13:12.13 | hariom | So that I don't have to keep 2 formats, one for playing on * and one on my system. I will convert all to ogg so that I can play on both the systems |
| 13:12.14 | doolittlework | hi there need some help with recordings, i have two files xxx-out.ulaw and xx-in.ulaw does anyone know how to convert them so i can hear whats on them? |
| 13:12.21 | bougyman | but the warranty is great on the sangoma and we've got two ds3s worth of PRI running on them for 3 years now. |
| 13:12.32 | [TK]D-Fender | hariom: What "other system"? |
| 13:12.45 | bougyman | plus they do most everything on-card (one interrupt/card) so CPU needed to run 8 spans is negligible. |
| 13:12.53 | hariom | My own laptop |
| 13:12.58 | [TK]D-Fender | hariom: VLC handles GSm just fine, along with probably jsuta bout everything else |
| 13:13.21 | Kobaz | bougyman: if i could slap an asterisk box together with a sangoma card in a SFF machine.. cheaper than i can buy a sip-pri gateway from a vendor... then i'll start using them |
| 13:13.22 | [TK]D-Fender | hariom: and "laptop" is a SIZE/FORMAT of computer... it doe s not denote any specific functionality |
| 13:13.52 | bougyman | Kobaz: you could, baraccuda does (cudatel) but minus the asterisk. |
| 13:14.28 | hariom | [TK]D-Fender: See, if one can keep a single format for all the future needs then why not? I can play ogg on any play but not gsm. I can play it on my mobile as well. I can share it with anybody I like without worrying about format. |
| 13:14.28 | coppice | [TK]D-Fender: well, it implies fly swatting potential, so it does give some indication of functionality |
| 13:14.37 | Kobaz | bougyman: cheaper than i can get an adtran904? |
| 13:14.48 | bougyman | Kobaz: i haven't priced the adtran |
| 13:14.56 | [TK]D-Fender | hariom: then MP3 it is |
| 13:15.00 | bougyman | i was looking at mediatrix and audiocodes before I found my current solution |
| 13:15.09 | bougyman | they were 16k+ for 8 port PRI gateway |
| 13:15.14 | Kobaz | yeah |
| 13:15.16 | bougyman | i'm sure that's all come down now (maybe?) |
| 13:15.19 | Kobaz | i pay about 750 per t1, on adtran |
| 13:15.29 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 13:15.48 | Kobaz | and... i call them up... and they fix my problems |
| 13:15.52 | bougyman | that's pretty solid. |
| 13:15.58 | bougyman | though same on the support side. |
| 13:15.59 | Kobaz | if i build it myself (dahdi)... well... then i'm out of luck |
| 13:16.06 | bougyman | cept I irc them and they log in. |
| 13:16.11 | bougyman | (the sangoma folks) |
| 13:16.19 | Kobaz | yeah sangoma has good support |
| 13:16.25 | Kobaz | but they wont help you with asterisk obviouslty |
| 13:16.28 | hariom | 1) MP3 may have license problem. 2) To me, it seems like Mp3 can consume more resources than ogg. 3) Ogg is open source. And as you said in your previous msg, ogg and mp3 and id3 tags |
| 13:16.31 | bougyman | sure they will |
| 13:16.41 | bougyman | they help with asterisk more than anything. |
| 13:16.52 | bougyman | they've set up asterisk, yate, and fs for me. |
| 13:16.52 | Kobaz | so yes.. sangoma has great hardware... but the sangoma<->asterisk integration is what kills the deal |
| 13:17.00 | Kobaz | bougyman: they charge you for it? |
| 13:17.03 | bougyman | Kobaz: no. |
| 13:17.05 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 13:17.06 | Kobaz | weird |
| 13:17.08 | bougyman | they do it all day in #sangoma, too |
| 13:17.11 | OlafsenM | hello, need help |
| 13:17.15 | Kobaz | they've never helped me with asterisk |
| 13:17.18 | bougyman | far more asterisk questions than anything else in there. |
| 13:17.27 | *** join/#asterisk Orentet (~Orentet@bzq-218-138-39.cablep.bezeqint.net) |
| 13:17.37 | Kobaz | they say "sangoma card is working fine.. the problem is in your sip software.. that's all we can do" |
| 13:17.39 | hariom | [TK]D-Fender: I want to know where I am doing wrong. Need for ogg Vs MP3 Vs GSm can change anytime so better to know the process and pitfalls. |
| 13:17.42 | OlafsenM | chan_dahdi.c: -- Channel 0/2, span 1 got hangup ACK |
| 13:17.59 | OlafsenM | i get this message without first hanging up the channel |
| 13:18.11 | OlafsenM | and after this message the channel stays active in Asterisk |
| 13:18.13 | OlafsenM | ? |
| 13:18.17 | OlafsenM | how can that be? |
| 13:18.19 | bougyman | OlafsenM: yuck. |
| 13:18.23 | bougyman | which asterisk version? |
| 13:18.28 | OlafsenM | 1.6.2.0 |
| 13:18.38 | OlafsenM | i don't see hangup request, only ACK |
| 13:18.42 | OlafsenM | wtf |
| 13:18.52 | OlafsenM | is maybe something wrong with telco? |
| 13:18.54 | Kobaz | bougyman: can they help me with libpri crashes? |
| 13:19.02 | OlafsenM | who requested the hangup in first place |
| 13:19.14 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 13:19.33 | bougyman | Kobaz: perhaps... i stopped using libpri about a year and a half ago when i was seeing crashes caused by it. |
| 13:19.39 | [TK]D-Fender | hariom: * is not great with OGG. What is this "mobile" you're talking about? a PHONE? What phone likes OGG? Will your next one? Thinking you can use one format for everything often leads you to picking a crappy choice. And you have not mentioned ANY of the specifics of these various devices you are using. |
| 13:19.50 | bougyman | using sangoma_prid now. |
| 13:20.19 | Kobaz | bougyman: oh... i didn't know they had their own pri stack |
| 13:20.47 | bougyman | they built it after they took over maintenance of openzap (now freetdm) |
| 13:20.53 | Kobaz | interesting |
| 13:21.05 | Kobaz | that's the biggest problem i've had with line cards... is bugs in libpri |
| 13:21.41 | hariom | [TK]D-Fender: ok so you say that * is not great with ogg so there might be problems. Ogg Vorbis is one of the best choice today when it comes to Free and Open source audio files. Google WebM supports Vorbis too. |
| 13:22.18 | Kobaz | good stuff |
| 13:22.34 | [TK]D-Fender | hariom: I love blind idealists. Try not to place your ideals to far above what WORKS. |
| 13:22.39 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 13:23.06 | [TK]D-Fender | Open != supported |
| 13:24.10 | hariom | [TK]D-Fender: Hey I will call that as exagration. Sorry but I am not offending. I guess I was asking for help to resolve the issue I have in hand. Not to change the work plan because there is nobody to solve that issue. |
| 13:24.55 | kuku | IF I have 1.4.22-4 running, and I would like to have func_audiohookinherit.so added, do I need to upgrade asterisk, or can I just download that SO from somewhere and attach it ? |
| 13:25.11 | [TK]D-Fender | hariom: Well perhaps you could contact the author of format_ogg and see about having it cleaned up to your taste. Thing is that NOW doesn't seem to make using it a good idea. |
| 13:25.39 | hariom | Ok, now it is getting little clear that Ogg is not very good with *. What do you suggest to go with Mp3. |
| 13:26.15 | hariom | will playback support that? Is there format_mp3 like thing? |
| 13:26.25 | hariom | or fork mpg123? |
| 13:26.25 | [TK]D-Fender | hariom: I personally agree that OGG should take over its segment. I'd love to see proprietary formats all die to comparable open equivalents, but I also won't pidgin-hole myself with "dreaming" either. |
| 13:26.55 | hariom | hmm... right |
| 13:26.55 | [TK]D-Fender | [09:26]<hariom>will playback support that? Is there format_mp3 like thing? <- yes... perhaps you've heard of ASTERISK-ADDONS |
| 13:27.09 | hariom | :) yea. |
| 13:27.20 | hariom | Need to check that out now again. |
| 13:30.38 | *** join/#asterisk shapr (~shapr@nat/digium/x-aohtmvrgqpxqulpw) |
| 13:32.11 | OlafsenM | guys: Channel 0/26, span 1 got hangup, cause 81 |
| 13:32.13 | OlafsenM | ?? |
| 13:32.20 | OlafsenM | invalid call reference value. |
| 13:32.21 | OlafsenM | ? |
| 13:32.26 | OlafsenM | what is happening? |
| 13:32.27 | OlafsenM | :) |
| 13:32.57 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:33.25 | pabelanger | OlafsenM: This cause indicates that the equipment sending this cause has received a message with a call reference which is not currently in use on the user-network interface. |
| 13:33.49 | OlafsenM | lol, i know that |
| 13:33.55 | OlafsenM | http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 13:33.55 | OlafsenM | :) |
| 13:34.11 | OlafsenM | but why is asterisk sending wrong call reference? |
| 13:34.13 | pabelanger | So, now you know what is happening |
| 13:34.23 | OlafsenM | no, I don't :) |
| 13:36.31 | WIMPy | Maybe it's trying to clear a channel that has already been cleared? |
| 13:36.51 | WIMPy | What happened before? |
| 13:38.29 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 13:39.00 | pabelanger | OlafsenM: Enable an ISDN trace and see what is going on. |
| 13:39.55 | OlafsenM | [Aug 4 12:51:01] VERBOSE[4082] chan_dahdi.c: -- Accepting call from 'xxxx' to 'xxx' on channel 0/4, span 5 |
| 13:40.14 | OlafsenM | [Aug 4 12:51:03] VERBOSE[4082] chan_dahdi.c: -- Channel 0/4, span 5 got hangup, cause 81 |
| 13:41.24 | WIMPy | That's why I was already tempted to correct myself, speakting like asterisk. It's about a call, not a channel, off course. |
| 13:41.46 | WIMPy | So the logged information is not particularly usefull. |
| 13:42.24 | pabelanger | OlafsenM: *CLI> pri set debug on |
| 13:45.05 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 13:46.01 | *** join/#asterisk KingDavidNYC (~Chris1232@ool-435554a8.dyn.optonline.net) |
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| 13:59.54 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:01.53 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
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| 14:06.23 | *** join/#asterisk gnarf (d15e2a04@gateway/web/freenode/ip.209.94.42.4) |
| 14:07.25 | gnarf | having some serious trouble getting dahdi up and running on my machine, everything was working fine a few days ago, but I had to recompile the wanrouter drivers for my card... here are some commands that might be useful if you can think of anything that might help (error message included in pastie as well ) http://pastie.org/private/djv3qb74m39vizz3mbku5q |
| 14:08.13 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 14:09.11 | leifmadsen | pabelanger: ohai! |
| 14:09.58 | [TK]D-Fender | gnarf: do "dahdi_cfg -vvvv" and then restart * |
| 14:10.47 | pabelanger | salutes leifmadsen |
| 14:11.07 | leifmadsen | pabelanger: better not be the middle finger kind! |
| 14:11.53 | [TK]D-Fender | prepares a 21 gun salute ... and aims it at pabelanger |
| 14:12.14 | gnarf | [TK]D-Fender: worked... should dahdi_cfg be in my startup scripts? |
| 14:12.32 | [TK]D-Fender | gnarf: Yes. You failed to initialize DAHDI first. |
| 14:13.22 | pabelanger | rolls a six sided dice. +4 dodge |
| 14:13.32 | gnarf | [TK]D-Fender: i don't need the v's in the startup script do i? |
| 14:14.05 | [TK]D-Fender | applies the modifiers and hits pabelanger for 3D12 + 5,000,000 damage! |
| 14:14.17 | [TK]D-Fender | *b00m* |
| 14:14.33 | [TK]D-Fender | gnarf: No, but a visual readout is nice |
| 14:14.47 | [TK]D-Fender | gnarf: What are your running it on? |
| 14:14.52 | gnarf | cool, lifesafer... was getting really stressed out over that one... |
| 14:15.15 | gnarf | the wanrouter startup script took care of it before... |
| 14:16.58 | gnarf | seems that when i recompiled those drivers, it installed the /dev/zap -- ztcfg script instead of the dahdi / dadhi_cfg script |
| 14:18.34 | KingDavidNYC | friends, I have to write a program which, in the middle of a call, just before the balance is up, sends caller to a script. The way I would do this, as a way of handling the timing, is to write a php script that runs every minute in linux, but maybe there is an easier way to this, can anyboby please tell me if I am overworking it? |
| 14:18.35 | [TK]D-Fender | gnarf: You could symlink it if you felt like it |
| 14:19.15 | [TK]D-Fender | KingDavidNYC: What happens then? |
| 14:20.02 | KingDavidNYC | the program asks the user if he wants to recharge his balance with his credit card, and if approved, joints the 2 people again |
| 14:20.03 | WIMPy | KingDavidNYC: Do you allow multiple calls for the same account? If not just check the balance at the beginning and use Dial option L. |
| 14:21.28 | KingDavidNYC | WIMPy: I dont see in option L the ability to send one channel to a dialplan |
| 14:22.44 | KingDavidNYC | I am looking for an option where I can start a process just 5 seconds before the end of the call |
| 14:23.33 | myster | KingDavidNYC, how will you know when the call is going to end? |
| 14:23.33 | fenrus | a psychic option ? |
| 14:23.36 | WIMPy | KingDavidNYC: I think it stays in the dial plan, but if you want to hold the 2nd leg to eventually reconnect the two it's definitely for something more sophisticated. |
| 14:23.44 | KingDavidNYC | fenrus: yes! |
| 14:23.45 | myster | psychic hotline. |
| 14:23.49 | fenrus | awsome |
| 14:24.02 | KingDavidNYC | myster: yes! |
| 14:24.52 | KingDavidNYC | I have to put the pyschic on hold, while I send the customer to recharge his balance |
| 14:25.07 | KingDavidNYC | I can do all that with the AMI |
| 14:25.42 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 14:25.47 | KingDavidNYC | it is just the part of starting a process just before time is up |
| 14:26.17 | *** join/#asterisk pbrunnen (~pbrunnen@74-44-78-42.dsl1-field.roch.ny.frontiernet.net) |
| 14:26.34 | *** part/#asterisk pbrunnen (~pbrunnen@74-44-78-42.dsl1-field.roch.ny.frontiernet.net) |
| 14:26.46 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 14:27.09 | KingDavidNYC | again, if I use crontab for a process to run every 60 seconds to check whose time is up, I get the feeling I am overkilling the computer |
| 14:28.07 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 14:28.19 | KingDavidNYC | and by the way, those psychic women, it is like 8 of them, and they are hot |
| 14:28.27 | [TK]D-Fender | KingDavidNYC: M()+ spawn script that sleeps for the timeout period. Use AMI redirect the channels. |
| 14:28.29 | crowb4r | lol |
| 14:29.24 | kpettit | I'm using 1.8 asterisk and trying to get jabber.conf working. I get a successfull jabber connect and can see user go online but I then get a asterisk Segmentation fault. I don't get a seg fault if user cannot connect, only if user can connect. |
| 14:29.41 | *** part/#asterisk gnarf (d15e2a04@gateway/web/freenode/ip.209.94.42.4) |
| 14:30.04 | kpettit | I've read about some iksemel issues and have build that from source with different methods show on voip-info.org and rebuild asterisk as well with the same problems. Any ideas? |
| 14:31.28 | *** join/#asterisk mr-m (~russellm@S01060014c25cd036.cg.shawcable.net) |
| 14:31.28 | KingDavidNYC | [TK]D-Fender: I see... man, that is genious |
| 14:31.33 | mr-m | does anyone use aastra 6757i's? i can't get it to speak only rfc3261, the phone keeps sending a Record-Route/Route header in my dialogs... |
| 14:32.32 | KingDavidNYC | [TK]D-Fender: Thanks man, that's awsome!... you are great |
| 14:35.31 | [TK]D-Fender | KingDavidNYC: You're welcome |
| 14:35.48 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:41.20 | raden | Good morning KingDavidNYC |
| 14:41.58 | KingDavidNYC | raden: what's up! |
| 14:43.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:46.49 | *** join/#asterisk newasterx (~dasdasdsa@114.199.101.35) |
| 14:46.54 | newasterx | Hello |
| 14:47.26 | newasterx | why in my callee i got this message "No m=text found in SDP" |
| 14:51.24 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
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| 16:03.21 | Diffen2 | Hello, im planning on installing a couple of patches on my asterisk. Can someone link a guide on how to do that? i havent found anyone. im planning on going from 1.4.24 -> 1.4.27.2 |
| 16:04.31 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 16:05.08 | pabelanger | Custom patches? or just upgrading from .24 to 27.2? |
| 16:06.09 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:06.43 | Diffen2 | official digium pathers |
| 16:06.45 | Diffen2 | patches |
| 16:07.37 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 16:08.18 | pabelanger | download Asterisk 1.4.27.2, un tar asterisk, apply patches, ./configure, make install. Restart asterisk |
| 16:08.26 | p3nguin | If you want to use the official patches, that will be the same as using the official releases. |
| 16:09.22 | p3nguin | There's no real good reason to patch up to 1.4.27.2 using official patches... just get the tar ball of the version you need and get to work building/installing. |
| 16:09.27 | Diffen2 | ok so i can go from 1.4.24 directly to 1.4.27.2? p3nguin i just want to patch and touch as little as possible in the system |
| 16:09.49 | p3nguin | There's no good reason to patch up to that version. |
| 16:10.07 | pabelanger | Diffen2: Usually yes, however read UPGRADE.txt and CHANGES in the source folder |
| 16:10.19 | p3nguin | It is no less work to patch several versions as compared to simply installing the version you want. |
| 16:10.23 | pabelanger | p3nguin: not for you, but maybe him |
| 16:10.45 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:11.56 | p3nguin | If you apply patches to a current version, you still have to recompile it and then reinstall it. That's essentially the same work as downloading the version you want to use and compiling and installing it. But without the patching work. |
| 16:12.33 | asteriskATmarmuD | hi guys, I can't get numbers following after my telephone number (00000000XX) into asterisk. is it possible that my NTBA doesn't forward them to my isdn-interface which is connected to asterisk? |
| 16:12.44 | Diffen2 | ok hmm im confused here, what version should i update to? i havent read all the patchnotes so i havent got the hole situation. i read that 1.4.24 is not a safe version so thats why i want to update to a couple of patches. |
| 16:12.46 | asteriskATmarmuD | if I dial 00000000XX only 00000000 gets to asterisk |
| 16:13.12 | p3nguin | Your dialplan could be blocking it. |
| 16:14.20 | *** join/#asterisk Faithful (~Faithful@nat76.mia.three.co.uk) |
| 16:14.33 | WIMPy | asteriskATmarmuD: 1. The NT is dumb and only forwards messages. 2. It is normal that you get a call to your base number, it's up to you to request more digits. |
| 16:14.52 | WIMPy | Asterisk will do that for you when your dialplan is set up correctely. |
| 16:15.11 | pabelanger | Diffen2: I would recommend setting up a staging box and copy your production settings to it. Then you can test any version of Asterisk before moving it into production. |
| 16:16.26 | Diffen2 | pabelanger sounds like a plan |
| 16:17.07 | garymc | Anyone willing to hold my hand? I need help with "core show application gotoif" commands |
| 16:17.13 | garymc | setting and understanding them |
| 16:17.20 | pabelanger | ~ask |
| 16:17.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:17.39 | asteriskATmarmuD | WIMPy: thx, so if I dial 00000000XX the NTBA will get that call (even if the "correct" number is 0000000) and forward all digits |
| 16:19.24 | newasterx | why in my callee i got this message "No m=text found in SDP" |
| 16:20.02 | WIMPy | asteriskATmarmuD: Forget about the NT. It's just an interface converter. You're talking tou the switch in the CO. And it depends how you call that number. |
| 16:21.02 | WIMPy | If you use overlap dialling you get a setup to your base number and the extension digits follow as they are dialled. When dialling en-block you get the whole number at once. |
| 16:21.25 | chazzam | oh noes |
| 16:22.29 | Diffen2 | pabelanger whats the "most secure" patch on 1.4 that doesent have a lot of bugs in it? I know its a stupid question but im asking anyway |
| 16:22.52 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:24.02 | pabelanger | Diffen2: Couldn't tell you. I usually go with the latest version, that way if there are problems, you can report them to the issue tracker. Otherwise, if you are using an older version of Asterisk, first thing they will tell you is to upgrade and retest with the latest version. |
| 16:24.05 | asteriskATmarmuD | <PROTECTED> |
| 16:24.25 | Diffen2 | pabelanger ok thanks man |
| 16:25.10 | *** join/#asterisk evilgeenius (5b663e22@gateway/web/freenode/ip.91.102.62.34) |
| 16:25.19 | WIMPy | asteriskATmarmuD: If you dial interactively, it's overlap. If you enter the number first and lift the reveicer afterwards, like on your mobile, it's en-block. |
| 16:25.32 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:26.01 | asteriskATmarmuD | <PROTECTED> |
| 16:26.22 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 16:27.21 | [TK]D-Fender | Diffen2: As always.. the altest.... |
| 16:27.27 | [TK]D-Fender | latest* |
| 16:27.30 | evilgeenius | Hello All, I am a complete and total nubee at all of this. I have just signed up for a sipgate account, so I have my own number. What do i need to do to get started with asterisk and connect it to my number? I plan on using the Ruby RAGI framework to play about with it. I am just installing asterisk but I have no idea where to go from here. I will look into it offcourse but Is there any useful tips you could give me to get |
| 16:28.02 | [TK]D-Fender | evilgeenius: ~book |
| 16:28.04 | [TK]D-Fender | ~book |
| 16:28.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:28.06 | [TK]D-Fender | ^^^ |
| 16:28.20 | Diffen2 | ok TK d-fender hmm so 1.4.35 then :) |
| 16:28.33 | Diffen2 | i dont dare upgrade the system |
| 16:28.35 | [TK]D-Fender | evilgeenius: And "connect it to my number" is completely vague and what the number leads to |
| 16:28.47 | [TK]D-Fender | Diffen2: Nowhere to go but DOWN then |
| 16:28.52 | evilgeenius | test |
| 16:28.56 | [TK]D-Fender | Diffen2: Let us know when you hit rock-bottom |
| 16:29.19 | [TK]D-Fender | evilgeenius: how does a NUMBER talk to a SOFTWARE PROGRAM? |
| 16:29.22 | jamko | Here is what I am getting after reinstall asterisk, thinking I should just reinstall again, but: |
| 16:29.27 | jamko | service asterisk st/usr/sbin/safe_asterisk: line 145: 20852 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} |
| 16:29.28 | jamko | Asterisk ended with exit status 139 |
| 16:29.28 | jamko | Asterisk exited on signal 11. |
| 16:29.32 | evilgeenius | [TK]D-Fender: like I said, im a complete nubee |
| 16:30.01 | [TK]D-Fender | evilgeenius: You can't even tell what a "number" leads to? I have a number... when it gets called my CELL PHONE RINGS. |
| 16:30.07 | evilgeenius | [TK]D-Fender: All I can tell you is that i created an account at sipgate.co.uk and now have a number. |
| 16:30.37 | [TK]D-Fender | evilgeenius: then you are receiving calls to it over **SIP** |
| 16:30.55 | evilgeenius | [TK]D-Fender: cool, that is a start |
| 16:31.01 | [TK]D-Fender | evilgeenius: So go set up your sip.conf and minimal dialplan to process the call. Sipgate should ahve minimal samples for you |
| 16:31.03 | evilgeenius | [TK]D-Fender: woohoo |
| 16:31.55 | evilgeenius | [TK]D-Fender: Ok, when I have modified sip.conf, what is the next step |
| 16:31.56 | evilgeenius | ? |
| 16:32.20 | [TK]D-Fender | evilgeenius: TEST IT and see what happens. |
| 16:32.54 | [TK]D-Fender | evilgeenius: before we start lets just get another item out of the way. Are you running your * server behind a NAT router? |
| 16:33.35 | evilgeenius | [TK]D-Fender: yeah it will be. It is behind my router provided by my ISP. The server is setup as a DMZ so it receives all traffic. |
| 16:34.28 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:34.37 | evilgeenius | [TK]D-Fender: So, after I have installed asterisk, and changed my sip.conf, will the system do anything? |
| 16:35.06 | evilgeenius | [TK]D-Fender: how will I know if it has worked at all? I mean, its a fresh install of asterisk. |
| 16:35.06 | *** join/#asterisk Faithful (~Faithful@nat76.mia.three.co.uk) |
| 16:35.45 | [TK]D-Fender | evilgeenius: fOLLOW THIS GUIDE FOR nat SETTINGS. iT IS required. |
| 16:35.55 | [TK]D-Fender | evilgeenius: THEN test it. pLACE A CALL AND SEE WHAT HAPPENS. |
| 16:36.08 | evilgeenius | [TK]D-Fender: what should happen? |
| 16:36.19 | [TK]D-Fender | evilgeenius: what did you TELL it to do? |
| 16:36.35 | evilgeenius | [TK]D-Fender: nothing yet |
| 16:36.52 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:36.56 | p3nguin | I bet that's what it will do, then. |
| 16:36.59 | evilgeenius | I plan on installing RAGI and playing around with ruby to get something working. Have you heard of RAGI? |
| 16:37.11 | evilgeenius | Has anyone heard of RAGI? is it commonly used? |
| 16:37.23 | [TK]D-Fender | evilgeenius: No, it isn't |
| 16:37.33 | evilgeenius | It works in a simular way to the RubyOnRails web framework that i use a lot. |
| 16:37.41 | [TK]D-Fender | evilgeenius: And your call will do nothing if you don't configure it to do otherwise |
| 16:40.07 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.7.160.242) |
| 16:40.41 | GhOnDiE | hi, anybody here with experiance using voip for broadcast radio stations? |
| 16:41.54 | [TK]D-Fender | GhOnDiE: VoIP != broadcost. |
| 16:42.23 | WIMPy | GhOnDiE: What's the relation? |
| 16:42.57 | p3nguin | Lots of people use VoIP at their radio stations. |
| 16:43.38 | evilgeenius | [TK]D-Fender: what's the simplest thing I can do after I have changed my sip.conf to test if it is setup correctly? |
| 16:43.55 | GhOnDiE | looking for a studio clock software that can interface effectively to sip to display line status. |
| 16:44.07 | [TK]D-Fender | evilgeenius: Place a call and see if it fails at the point where * accepts the call and tries to dump it into the dialplan. |
| 16:44.53 | [TK]D-Fender | GhOnDiE: "clock software"? huh? What sense of "time" relates toa call processor like *? |
| 16:45.06 | *** join/#asterisk pgarcia (~chatzilla@yoda.kanatek.com) |
| 16:45.16 | GhOnDiE | well i use asterisk and wondered if somebody may have any ideas? |
| 16:45.28 | [TK]D-Fender | GhOnDiE: "lines status"? What lines? Connected to what? |
| 16:45.54 | GhOnDiE | studio incoming line, phones dont ring in broadcast studios |
| 16:46.03 | GhOnDiE | normally you have a phone light that flashes |
| 16:46.17 | drmessano | GhOnDiE: Look for a BetaBrite or similar |
| 16:46.42 | pgarcia | Hi... I'm trying to get Libpri working with BRI in asterisk 1.6.xxx, but I cannot make it dial through a specific span/channel. Is this possible? Anybody knows it? |
| 16:46.45 | [TK]D-Fender | GhOnDiE: park your calls. Enable presence for your parking lots. The end |
| 16:46.50 | drmessano | GhOnDiE: All purpose LED display.. then with some scripting can interface with Asterisk via the Serial port |
| 16:47.29 | WIMPy | pgarcia: What's happening? |
| 16:47.39 | drmessano | [TK]D-Fender: He's looking for some way to get an overhead/on wall call display |
| 16:48.06 | GhOnDiE | ive solved the other little problem i had in freepbx chan earlier |
| 16:48.12 | GhOnDiE | now on to the next problem |
| 16:48.15 | [TK]D-Fender | drmessano: He said "clock". Hold on.. I have to set my dishwasher to BROIL. |
| 16:48.51 | pgarcia | well, I'm using something like exten => 600,1,Dial(DAHDI/2/7000) and I would expect that the second span should be used... but it just return error "app_dial.c:1747 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)" |
| 16:49.11 | drmessano | GhOnDiE: We have BetaBrite displays going on that I am planning to do just as I described.. one in each studio, scroller for the hotline, request line, etc |
| 16:49.23 | pabelanger | pgarcia: Is DAHDI installed and loaded? |
| 16:49.23 | drmessano | going in* |
| 16:49.24 | *** join/#asterisk titter (~titter@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:49.59 | WIMPy | pgarcia: That's 2nd channel, not 2nd span. Are you sure your interfaces are working?. dahdi show status |
| 16:50.01 | pgarcia | yes, I can receive calls though.. this is a test extension that receives a BRI call and places another one... |
| 16:50.06 | drmessano | GhOnDiE: They can be the "clock" 99% of the time, then change to a HOTLINE scroller or REQUEST scroller when theres a call, if you like |
| 16:50.13 | evilgeenius | I spoke to a phone engineer who just came to fit our new telephone system at my work. He said he only uses Asterisk as a utility thing that lets him add simple services to an existing full fledged off-the-shelf pbx system. Is this true? Would you not use Asterisk as a full phone system that does everything? |
| 16:50.27 | pgarcia | so, the number after "/" is the channel number? not the span? |
| 16:50.33 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:51.03 | drmessano | evilgeenius: He's an idiot.. Asterisk CAN be used as a gateway to a standard PBX to access services it doesn't natively support, but Asterisk is used for PBX's worldwide, call centers, ITSPs, etc |
| 16:51.12 | pabelanger | pgarcia: correct, however you can setup groups. IE: Dial(DAHDI/g1/12345) |
| 16:51.15 | ChannelZ | yes. The channel numbers are sequential |
| 16:51.37 | titter | Can someone explain to me why this call is answered by a Shoretel hunt group, but kicks back to Asterisk and drops it into MOH ... and then once the call is answered goes to the following warning? I am stumped lol http://pastebin.com/8JMWJmF4 |
| 16:51.37 | pgarcia | maybe it is a newbe error... As fxo and fxs, BRI also uses channel numbering, for example 1 is the first span, first channel, 2 is first span, second channel? |
| 16:51.41 | pabelanger | so, group your spans together. g0, g1, g2, etc |
| 16:51.55 | drmessano | evilgeenius: Most other PBX's are feature limited and overpriced by about $25,000 to $250,000 or so |
| 16:51.57 | pgarcia | grouping are working for me |
| 16:51.57 | [TK]D-Fender | evilgeenius: I use Asterisk to make me COFFEE and as a jukebox |
| 16:52.16 | drmessano | evilgeenius: Asterisk can pretty much do anything |
| 16:52.31 | drmessano | evilgeenius: Especially replace a large or small PXB |
| 16:52.34 | drmessano | PBX* |
| 16:52.36 | ChannelZ | It saved me 30% or more on my car insurance |
| 16:52.59 | WIMPy | pgarcia: Yes, it's a little unfortunate for different types of interface with completely different context to share the same namespace, IMHO, but that's the way it is. |
| 16:53.25 | coppice | ChannelZ: I found a way to save 100% on my car insurance |
| 16:53.35 | WIMPy | s/context/concepts. |
| 16:53.39 | pabelanger | steal the car? |
| 16:53.39 | drmessano | Asterisk makes me coffee, turns my lights on an off, wakes me in the morning, provides Magic 8-Ball like services for decision making, and provides PSTN service for family and friends.. and that's just what I am doing at HOME. |
| 16:53.40 | WIMPy | s/context/concepts/ |
| 16:53.43 | WIMPy | gni |
| 16:53.59 | pgarcia | hmmm, I see... that's make sense to me. Let me test using this numbering.... |
| 16:54.00 | ChannelZ | coppice: Ahh, the 'undocumented worker' route |
| 16:54.35 | evilgeenius | drmessano: He said off the shelf systems were just easier to setup. Would you have to build a complete system from scratch or does all the functionality already exists in modules that you can just add to it? |
| 16:54.46 | coppice | nah. I live in a place with working public transport, and few people want cars |
| 16:55.49 | pabelanger | evilgeenius: Asterisk is not a product (out of the box). |
| 16:56.03 | [TK]D-Fender | pabelanger: We ship boxes now :) |
| 16:56.23 | drmessano | evilgeenius: You can start from scratch or use something like AsteriskNOW which gets you most of the way there |
| 16:56.25 | [TK]D-Fender | evilgeenius: Asterisk is jsut about complete control over your calls. |
| 16:56.36 | [TK]D-Fender | evilgeenius: It will jump through whatever hoops you set up for it. |
| 16:56.42 | [TK]D-Fender | evilgeenius: What do you WANT to do? |
| 16:56.52 | titter | evilgeenius: I find Asterisk very simple to setup ... I can compile and compose a basic dialplan in around 30 minutes now. |
| 16:57.04 | drmessano | evilgeenius: But Asterisk with a GUI is a complete blank slate for you to do EXACTLY what you want, maybe to more of an extent than you need or want, but it's there |
| 16:57.12 | drmessano | Sorry |
| 16:57.13 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 16:57.22 | drmessano | evilgeenius: But Asterisk WITHOUT a GUI is a complete blank slate for you to do EXACTLY what you want, maybe to more of an extent than you need or want, but it's there |
| 16:57.29 | pgarcia | BTW, anybody knows whether libpri is supporting Overlap dial in Network mode? |
| 16:58.01 | drmessano | There's many ways to get Asterisk going.. From scratch, AsteriskNOW ISO, other GUI's or dialplan building apps, etc |
| 16:59.08 | WIMPy | pgarcia: Yes. |
| 16:59.13 | evilgeenius | drmessano: what do you do? |
| 16:59.29 | WIMPy | pgarcia: But for NT ptmp you need 1.8. |
| 16:59.32 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 16:59.35 | [TK]D-Fender | [12:56]<[TK]D-Fender>evilgeenius: What do you WANT to do? |
| 16:59.42 | evilgeenius | drmessano: Do you start from scratch or use AsteriskNOW? |
| 17:00.09 | titter | evilgeenius: Most in here probably start with Asterisk and not the GUI |
| 17:00.16 | evilgeenius | [TK]D-Fender: How do you program it? What language/framework? |
| 17:00.47 | drmessano | evilgeenius: Both... Sometimes I may need a simple Asterisk box for a simple task, other times I build a FreePBX system up by hand (which gets you nearly the same end result as an AsteriskNOW install) |
| 17:00.58 | [TK]D-Fender | evilgeenius: If you are looking at RAGI then you are not doing a GUI at all... you are learning one programming language with great limitations instead of learning how to program * yourself |
| 17:01.09 | [TK]D-Fender | evilgeenius: and neither is "using a GUI" |
| 17:01.12 | p3nguin | evilgeenius: You have so much more control over things when you start at the bottom and build Asterisk up yourself. |
| 17:01.26 | [TK]D-Fender | evilgeenius: So again, what do YOU want to do exactly? |
| 17:02.05 | evilgeenius | [TK]D-Fender: so * is a language itself? |
| 17:02.25 | drmessano | Asterisk dialplan is unique to Asterisk |
| 17:02.25 | [TK]D-Fender | evilgeenius: yes, the dialplan (what processes your calls) is its own language. |
| 17:03.02 | [TK]D-Fender | evilgeenius: 99% of * = dialplan. getting a call in is virtually nothing. DOING something with it is another matter. |
| 17:03.03 | evilgeenius | [TK]D-Fender: if there is a programming language framework that makes creating applications easier then I will use it, that is how you get things done easier/more efficiently. |
| 17:03.14 | drmessano | If it was written in PHP or Ruby, it would be called Astrsk (note the missing vowels) and the website would be PINK and GREEN |
| 17:03.32 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 17:03.39 | ruyo | Anyone knows if tos_sip option (sip.conf) change all SIP messages? |
| 17:03.40 | evilgeenius | [TK]D-Fender: for example, I use the RubyOnRails web framework that allows me to create applications at blazing speed that are easy to maintain. |
| 17:03.54 | *** join/#asterisk fofware (~fabian@190.225.15.129) |
| 17:04.32 | drmessano | evilgeenius: Asterisk dialplan is more akin to scripting level programming vs some higher level devel |
| 17:04.50 | [TK]D-Fender | evilgeenius: Don't throw generic words like "application" around so much. Its a worthless buzzword right now. |
| 17:04.56 | [TK]D-Fender | evilgeenius: Get SPECIFIC . Fast. |
| 17:04.57 | drmessano | You don't need a framework.. Just google, a look at the book, and some REAL motivation |
| 17:05.08 | evilgeenius | drmessano: So how do external languages interface with asterisk? is there an api or something? |
| 17:05.15 | [TK]D-Fender | evilgeenius: Several |
| 17:05.17 | drmessano | ~book |
| 17:05.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:05.22 | [TK]D-Fender | evilgeenius: For those who have any NEED of it |
| 17:06.03 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 17:06.04 | drmessano | Controlling Asterisk via the API is really silly.. The API is there for interfacing externally, but you really should write your dialplan and the core of your call control IN Asterisk |
| 17:06.42 | *** join/#asterisk fofware (~fabian@190.225.15.129) |
| 17:06.45 | evilgeenius | drmessano: why? if using a higher level language can help you create your applications quicker? with less code.. an easier maintainability? |
| 17:07.12 | drmessano | Some have set up a VERY basic config and used their own preferred language to do EVERYTHING via AMI to Asterisk, but those people also burn ants with magnifiers and drive around in scary red vans with "free candy" spraypainted on the side. |
| 17:08.06 | evilgeenius | drmessano: lol, you're right, i think assembly should be the only language used when writing programmes, those darn people who use higher level languages and get things done quickly!! damn them! |
| 17:08.13 | drmessano | evilgeenius: I think once you read the book you'll realize how silly this route is.. Asterisk dialplan isn't that hard |
| 17:08.33 | drmessano | evilgeenius: No, the comparison is not the same |
| 17:08.48 | evilgeenius | drmessano: How do you interface with a backend database using the dialplan? |
| 17:08.58 | drmessano | You're talking about driving a car with a remote control and some servos attached to the wheel vs learning how to sit in the car and just fscking drive it |
| 17:09.34 | evilgeenius | So how do you integrate a backend DB with your dialplan? |
| 17:09.38 | drmessano | evilgeenius: ODBC or there's an addon MySQL connector |
| 17:09.46 | evilgeenius | drmessano: lol |
| 17:10.06 | evilgeenius | drmessano: You need to get out of the dark ages.... |
| 17:10.14 | drmessano | I do? |
| 17:10.40 | evilgeenius | drmessano: Actually I prefer it when there are plenty of old-schoolers around, they make me lood good when i do things x10 quicker :-) |
| 17:10.53 | drmessano | WTF? |
| 17:11.10 | drmessano | You asked how Asterisk interfaces with a database and I told you |
| 17:11.15 | drmessano | Whats with the other shit? |
| 17:11.16 | WIMPy | evilgeenius: Dou you also write a module for Apache instead of looking into httpd.conf? |
| 17:11.21 | evilgeenius | drmessano: I rarely use SQL to interface with backend DBs |
| 17:11.32 | drmessano | Good for you |
| 17:11.43 | p3nguin | This was entertaining earlier, but now you're starting to look like an asshole. |
| 17:11.56 | p3nguin | No one cares if you're some elite high-level programmer. |
| 17:12.04 | evilgeenius | p3nguin: im not |
| 17:12.06 | evilgeenius | p3nguin: at all |
| 17:12.22 | *** join/#asterisk nova911 (~Adium@122.182.0.38) |
| 17:12.24 | evilgeenius | p3nguin: I just know how to get things done quickly using the latest technologies |
| 17:12.31 | ruyo | Asterisk's dialplan is pretty high level programming, tbh. |
| 17:12.34 | [TK]D-Fender | evilgeenius: And you are still in Generic Land (population : YOU) |
| 17:12.34 | p3nguin | hence the "elite" |
| 17:12.38 | evilgeenius | p3nguin: which is always a lot easier than working low level |
| 17:12.46 | p3nguin | except when it's not. |
| 17:12.47 | drmessano | p3nguin: Sounds like another one of those tools thats gonna show us all up and write a framework for language-of-the-week to interface with Asterisk and then realize he should have just written some dialplan and went on with life |
| 17:13.00 | [TK]D-Fender | evilgeenius: So what do you want to do EXACTLY? |
| 17:13.00 | titter | evilgeenius: You are making yourself look like a fool |
| 17:13.04 | WIMPy | evilgeenius: But you don't seem to know the difference between programming new software and configuring existing software. |
| 17:13.25 | p3nguin | You don't have to WRITE Asterisk... just use it! |
| 17:13.25 | evilgeenius | WIMPy: explain |
| 17:13.39 | *** join/#asterisk fofware (~fabian@190.225.15.129) |
| 17:13.47 | evilgeenius | p3nguin: do you not write the dialplans? |
| 17:13.56 | p3nguin | Sure, and it takes a mere few minutes. |
| 17:14.11 | p3nguin | It's pretty much like basic scripting. |
| 17:14.14 | drmessano | p3nguin: I can do with with PHP-On-Perl-C++-On-Rails-.NET.. AKA POPCORN |
| 17:14.26 | pabelanger | don't feed the trolls |
| 17:14.33 | p3nguin | heh, popcorn. |
| 17:14.33 | [TK]D-Fender | evilgeenius: You don't evenknow what it LOOKS like yet let alone have abasis of comparison based on what HE is doing with it and you can't even get off your ass to tell us what YOU want to do with it. |
| 17:14.36 | evilgeenius | p3nguin: what about when you integrate it with a backed DB? |
| 17:14.56 | p3nguin | Asterisk has DB support included. |
| 17:15.10 | evilgeenius | [TK]D-Fender: im just gona play around with it at first. Then maybe work into integrating it with our DB |
| 17:15.16 | [TK]D-Fender | evilgeenius: You are just spewing generic crap about added higher-level frameworks to lower stuff = better and necessary at all times. Worthless unvalidated rhetoric |
| 17:15.19 | evilgeenius | p3nguin: not using sql though |
| 17:15.43 | [TK]D-Fender | evilgeenius: and "integrating" is again more generic junk. |
| 17:15.53 | drmessano | p3nguin: POPCORN or PEANUTS framework all the way |
| 17:15.56 | titter | evilgeenius: What DB technology |
| 17:17.26 | [TK]D-Fender | titter: BINARY! |
| 17:17.27 | drmessano | He's googling for one |
| 17:17.34 | drmessano | Give him a moment |
| 17:17.35 | evilgeenius | [TK]D-Fender: Well, we have a large customer database... we have several project managers in several different branches that deal with groups of customers.... all this info is stored in a DB and managed through a web interface. When a call comes in, id like to look up the client in our database using the number, then find out which project manager(s) deal with that client, then forward it on to the project managers mobile..... |
| 17:17.37 | drmessano | Lynx is a little slow |
| 17:18.09 | [TK]D-Fender | evilgeenius: So far about 2 lines fo dialplan. Go install your framework to circumvent that now :p |
| 17:18.15 | p3nguin | haha |
| 17:18.17 | drmessano | LOL yeah |
| 17:18.25 | drmessano | Simple DB lookup? |
| 17:18.26 | drmessano | HA |
| 17:18.29 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:18.29 | titter | lol |
| 17:18.40 | drmessano | Break out the Rubies and the Rails |
| 17:18.44 | [TK]D-Fender | drActually... you could probably do it in the DIAL command itself in 1 line with FUNC_ODBC :p |
| 17:18.49 | [TK]D-Fender | drmessano: ^^ |
| 17:18.52 | drmessano | lol |
| 17:19.10 | evilgeenius | all the info about which project managers are associated with customers is stored in a DB. 2 lines? |
| 17:19.17 | drmessano | [TK]D-Fender: I demand a better API.. the line is over 45 characters |
| 17:19.17 | p3nguin | or 1 |
| 17:19.46 | drmessano | Yeah, 1 or 2.. Depending on how much of the book you read |
| 17:19.54 | evilgeenius | 1 very big line? |
| 17:19.57 | [TK]D-Fender | evilgeenius: To get the manager's # is probably 1 field in the customer record. that is 1 line fo dialplan to pull it from the DB |
| 17:20.02 | drmessano | No, 1 well written line |
| 17:20.08 | drmessano | 2 if you cant nest |
| 17:20.21 | [TK]D-Fender | evilgeenius: And dialing it it pure #. You can only COMPLICATE that action with "frameworK" |
| 17:20.30 | evilgeenius | [TK]D-Fender: The manager' # is stored in the managers table. |
| 17:20.32 | [TK]D-Fender | evilgeenius: Because you have no concept of how calls coming in/out |
| 17:20.56 | evilgeenius | [TK]D-Fender: the association between managers and clients is stored in another |
| 17:21.02 | [TK]D-Fender | evilgeenius: Yes, and I suspect its a single SQL query to chain the 2 files together <- |
| 17:21.11 | [TK]D-Fender | ONE LINE |
| 17:21.13 | [TK]D-Fender | SQL |
| 17:21.22 | [TK]D-Fender | LEARN IT BITCHES |
| 17:21.33 | *** join/#asterisk BANSAL (~bansal@117.199.121.13) |
| 17:21.41 | evilgeenius | i know it |
| 17:21.49 | drmessano | s/Where is your god now?/Where is your framework now?/ |
| 17:21.50 | pabelanger | Ok, next topic |
| 17:22.03 | ChannelZ | Let's talk about chicks, man |
| 17:22.10 | evilgeenius | There can be several project managers per client, so if one project manager doesn't answer, then it needs to ring the next... |
| 17:22.31 | pabelanger | drmessano: I actually find this pretty funny, as I'm working on a framework ATM ;) |
| 17:22.43 | evilgeenius | The client also has the option to listen to information about their current jobs... |
| 17:22.57 | asteriskATmarmuD | ChannelZ: woohoo, my topic, finally! ;) |
| 17:22.59 | evilgeenius | This information is stored in about 300 tables... |
| 17:23.15 | drmessano | OMG.. is this isn't a troll.. |
| 17:23.19 | drmessano | if* |
| 17:23.30 | drmessano | We're up to what, 3 lines.. maybe 4 |
| 17:23.34 | evilgeenius | The logic for this is currently within the application logic in a ruby on rails application |
| 17:24.18 | [TK]D-Fender | drmessano: Correct, the lure has struck bottom and he's ANCHORED :p |
| 17:24.18 | drmessano | The logic is Asterisk 101 |
| 17:24.19 | evilgeenius | im just telling you what my situation is, im not trying to be a troll |
| 17:24.19 | drmessano | Couple simple lines of dialplan |
| 17:24.19 | evilgeenius | Dont be so defensive |
| 17:24.19 | drmessano | I'm not |
| 17:24.19 | [TK]D-Fender | evilgeenius: Go learn *. |
| 17:24.19 | [TK]D-Fender | evilgeenius: install it. Get your call in. DO SHIT |
| 17:24.45 | [TK]D-Fender | smirks at his homonym humour |
| 17:25.00 | titter | How about someone help me figure out why this gets kicked into MOH! http://pastebin.com/8JMWJmF4 |
| 17:25.30 | drmessano | Asterisk isn't my project.. But if you're gonna open your mouth, you should open your ears too.. Asterisk doesn't need a bunch of crap piled on to execute the very SIMPLE demands you've made thus far.. Which is maybe why it's so successful. |
| 17:25.31 | pabelanger | titter: enable Verbose on your console |
| 17:25.38 | titter | pabelanger: it is. |
| 17:25.56 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 17:26.00 | pabelanger | ~collectdebug |
| 17:26.00 | infobot | i guess collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 17:27.06 | evilgeenius | drmessano: Surely using a very powerful high level language has its advantages? Are you saying there's no point at all to using anything other than *? |
| 17:27.09 | [TK]D-Fender | titter: If get pu on hold because you bridged it to a channel that PUT him on hold |
| 17:27.19 | [TK]D-Fender | titter: which looks to be this "shoretel" |
| 17:27.32 | evilgeenius | drmessano: is * really a brilliant well-designed language? |
| 17:27.47 | [TK]D-Fender | evilgeenius: You are spouting framework merits with even having a relative understanding on what it SITS on top of. |
| 17:27.56 | evilgeenius | drmessano: Is it simple and easy to understand and maintain? Do you have to be a * expert to understand it? |
| 17:28.27 | drmessano | evilgeenius: Asterisk is very good at making itself work, evilgeenius.. Here you go again with the framework stuff, and completely ignoring that what you want to execute here is like 3 lines of dialplan.. 5 minutes work.. |
| 17:28.28 | titter | [TK]D-Fender: That is what I was thinking, the dialplan is as simple as exten => _X.,1,Dial(SIP/shoresip/${EXTEN},45,r) ... it rings to the Shoretel system (a very awful PBX), and a queue on that system answers ... didn't think it would put the call into MOH on the * side |
| 17:28.36 | drmessano | Do you like making things harder than they really are? |
| 17:28.36 | evilgeenius | [TK]D-Fender: im just trying to get an understanding |
| 17:28.46 | [TK]D-Fender | titter: It clearly does. |
| 17:29.13 | titter | [TK]D-Fender: I see that it does, but I was confused if it could rather. Thanks for the clarification. |
| 17:29.15 | [TK]D-Fender | evilgeenius: No, you want to generically argue the merits of a framewaork while having no understanding of what it is sitting on. |
| 17:29.26 | [TK]D-Fender | evilgeenius: So jsut stop. NOW. And go learn it for yourself |
| 17:29.29 | evilgeenius | drmessano: no, easier, that's why i use ruby. |
| 17:29.35 | evilgeenius | [TK]D-Fender: I will |
| 17:30.07 | evilgeenius | If the api is really bad then I could totally understand where you're coming from |
| 17:30.12 | evilgeenius | is it really bad? |
| 17:30.18 | titter | [TK]D-Fender: Actually the real issue is once the call is answered by another SIP phone on the Shoretel system, * goes to that Warning, has a hissy fit, and no audio is transmitted to either side |
| 17:31.16 | drmessano | evilgeenius: You can insist all you want on how great your powerful, high level languages are.. but fact is, I would feel pretty stupid putting REAL TIME into making something work in my language of choice when native language does all you need and is less than a few minutes work.. As someone asked earlier.. do you use httpd.conf to configure Apache or do you waste your time configuring it using some framework? |
| 17:31.18 | *** join/#asterisk SirLouen (sir.louen@84.122.192.145.dyn.user.ono.com) |
| 17:31.40 | [TK]D-Fender | evilgeenius: STOP. NOW. |
| 17:31.48 | [TK]D-Fender | evilgeenius: Go work with *. |
| 17:31.52 | evilgeenius | drmessano: Most of the systems I interface with ruby have their own language. But ruby makes it easier to work with them all. Are you saying * is different? |
| 17:32.02 | [TK]D-Fender | evilgeenius: The rest of this framework conjecture is pointless. |
| 17:32.02 | pabelanger | Ok, off-topic, anybody recommend a virtual whiteboard software? |
| 17:32.14 | [TK]D-Fender | evilgeenius: Just. Fucking. DROP. IT. |
| 17:32.21 | evilgeenius | [TK]D-Fender: Chill |
| 17:32.47 | [TK]D-Fender | evilgeenius: Every time you jump back to it you're just going to keep getting more of the same.. |
| 17:33.48 | SirLouen | i have an analog telephone conected to a tdm410p to a fxs port. the card has been configured in the kernel but the telephone doesn't receive power |
| 17:33.57 | SirLouen | any ideas on how can i check what can be happening? |
| 17:34.33 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 17:34.36 | pabelanger | SirLouen: did you attach power to the molex connect on the tdm410p? |
| 17:34.37 | titter | ^ |
| 17:34.51 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 17:34.52 | cusco | hi |
| 17:34.56 | drmessano | evilgeenius: Are you really that inept that there's only one language you can use to make this work? MANY people write dialplans in Asterisk and go on with their life. You seem to insist on using the API here, and we've told you how senseless it is.. What you want to do with Asterisk doesnt SCRATCH the surface of what it can do.. you have BASIC, SIMPLE requirements that are 5 MINUTES work.. Why insist on the API? |
| 17:35.05 | evilgeenius | [TK]D-Fender: I didn't realise you'd be so senstive on the issue of using an external language. Its like me going into #sql and them getting pissed of because i've told them that using an Object Relational Mapper (ORM) means i don't have to waste my time writing raw sql anymore. |
| 17:35.13 | SirLouen | pabelanger uhm not so sure the card was into the server when i received |
| 17:35.17 | SirLouen | i much check that |
| 17:35.27 | drmessano | evilgeenius: No one is being sensitive.. you're just not listening to common sense. |
| 17:35.28 | SirLouen | thanks for the guidance |
| 17:35.30 | cusco | having the following dial: Dial(SIP/6542${EXTEN:4}@gateway,,g); -- allows me to execute more statements following the dialplan, right? the g flag does that, right?? |
| 17:36.16 | cusco | it doesn't seem to happen |
| 17:36.18 | drmessano | evilgeenius: You're going to spend more time learning the API than you will these 4 lines of code to create a solution to what you presented |
| 17:36.18 | SirLouen | is just a single molex just like the case fans received isn't it? |
| 17:36.19 | [TK]D-Fender | evilgeenius: Perhaps they understand that for the GOAL such extra scrap is pointless? that is a "validated" answer". You however hype merit without having the basis of comparison. What do you call a person who compares things they know nothing about? |
| 17:36.23 | evilgeenius | drmessano: I'll use any language. But im thinking long term, and creating solutions that can integrate with many other external services. |
| 17:36.25 | [TK]D-Fender | GLEN BECK |
| 17:36.40 | [TK]D-Fender | +N |
| 17:36.42 | [TK]D-Fender | :p |
| 17:36.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:37.05 | ruyo | SirLouen, like HDDs. |
| 17:37.41 | [TK]D-Fender | evilgeenius: If you're thinking ":logn term" think about how long that frame work will follow *'s development and the added dependency. All while not having a qualified comparison as to what the different implementations would look like |
| 17:37.42 | pabelanger | SirLouen: Yes |
| 17:38.12 | evilgeenius | there is nothing on the RAGI site :-( http://www.snapvine.com/code/ragi |
| 17:38.53 | cusco | I have the following dialplan for outbounds: http://paste.debian.net/83938/ |
| 17:39.01 | jamko | Asterisk crashing when starting after new install.... Could someone take a look at this pastebin please: http://pastebin.com/CJqnHJj2 |
| 17:39.02 | p3nguin | Why is this still an issue? Let evilgeenius waste his own time while we continue doing things in a more sensible manner. |
| 17:39.06 | cusco | the NoOp() before the Hangup doesn't print... |
| 17:39.14 | cusco | so im assuming dial,,g ain't working ? |
| 17:39.30 | pabelanger | jamko: Drop safe_asterisk and start asterisk directly. asterisk -vvvvvcg |
| 17:39.35 | [TK]D-Fender | cusco: How would we know.. you aren't showing us the CALL |
| 17:39.49 | evilgeenius | All the interfaces we create at my work are web interfaces, so the management of the phone system would be a web interface. Does it not make sense to use the same language for everything? |
| 17:39.56 | EmleyMoor | I have a pastebin of debug from a good call from my N97 (over WiFi) - hrhrhr and ChannelZ suggested it may help - either of them about? |
| 17:39.58 | cusco | [TK]D-Fender: ok hold... |
| 17:40.00 | evilgeenius | [TK]D-Fender: nice talking to you, youve been a real help |
| 17:40.09 | pabelanger | jamko: Also read doc/backtrace.txt on how to get an unoptimized core dump, then pb the backtrace |
| 17:40.11 | bougyman | evilgeenius: it does if you have a team that focuses on one language. |
| 17:40.17 | bougyman | but watch out for the sledge hammer. |
| 17:40.33 | bougyman | for instance, you probably use whatever language you specialize in plus javascript, right? |
| 17:40.35 | [TK]D-Fender | evilgeenius: as long as it leads you to actually seeing what you're dealing with with *, sure. |
| 17:40.38 | cusco | [TK]D-Fender: there is a lot happening, it is just hard to filter out the call... |
| 17:40.59 | bougyman | evilgeenius: you looking at ruby as your language? |
| 17:41.09 | evilgeenius | bougyman: well, if all the logic is contained in the web app/db, id have to re-create/duplicate the logic in *. Does that make sense? |
| 17:41.09 | bougyman | did you not like adhearsion? |
| 17:41.18 | evilgeenius | bougyman: yeah i was thinking of using ruby |
| 17:41.19 | [TK]D-Fender | bougyman: He hasn't even INSTALLED * yet |
| 17:41.24 | [TK]D-Fender | bougyman: And knows nothing of it. |
| 17:41.28 | bougyman | [TK]D-Fender: so he's jsut doing research. |
| 17:41.40 | cusco | ok got it. [TK]D-Fender the cli output is here: http://paste.debian.net/83940/ |
| 17:41.41 | [TK]D-Fender | bougyman: Spare yourself the net result of the last half hour of insanity here |
| 17:41.43 | bougyman | i was right where he was 3.5 years ago. |
| 17:41.52 | bougyman | so I can relate. |
| 17:42.00 | evilgeenius | bougyman: is adhearsion any good? |
| 17:42.11 | bougyman | we had a team of ruby, java, lisp coders, and two network engineers. |
| 17:42.21 | bougyman | but little to no voip know-how. |
| 17:42.30 | bougyman | evilgeenius: yes, it's good at what it does. |
| 17:42.40 | bougyman | but I don't like dsl's so much for what I do. |
| 17:42.46 | bougyman | for what you are needing it seems to fit the bill. |
| 17:43.04 | evilgeenius | bougyman: i dont know exactly what ill be using it for yet, im just looking into it |
| 17:43.18 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 17:43.19 | evilgeenius | bougyman: is it asterisk on the backend? |
| 17:43.32 | bougyman | evilgeenius: yes, adhearsion is (for-now) asterisk-only |
| 17:43.39 | bougyman | it's being ported to yate and freeswitch, accd to the devs. |
| 17:43.56 | evilgeenius | bougyman: did you look into RAGI? |
| 17:44.11 | bougyman | yes, i met the ragi author at a railsconf in '07 or '08 |
| 17:44.15 | bougyman | that's what led me down this road. |
| 17:44.26 | evilgeenius | oh cool |
| 17:44.36 | evilgeenius | Which road? adhearsion? |
| 17:44.44 | bougyman | ruby controlling phone switches. |
| 17:44.51 | bougyman | had to learn voip basics first. |
| 17:45.14 | evilgeenius | bougyman: so you're all for it then? |
| 17:45.29 | bougyman | then tdm/PRI, then finally got in to controlling them with external processes (ruby) |
| 17:45.52 | cusco | so it seems the g flag ins't working... |
| 17:45.52 | bougyman | we used adhearsion for about a year but then ran into scaling problems so wrote an Event Socket library in ruby for our apps. |
| 17:45.58 | *** join/#asterisk clintc (~clintc@n128-227-41-106.xlate.ufl.edu) |
| 17:46.08 | bougyman | evilgeenius: yes, it's a fine venture. |
| 17:46.12 | evilgeenius | So do you use RAGI or adhearsion? |
| 17:46.18 | evilgeenius | Is ragi still going? |
| 17:46.28 | evilgeenius | bougyman: there is nothing on its web page |
| 17:46.41 | bougyman | i don't use either anymore. |
| 17:46.59 | evilgeenius | bougyman: what do you use? |
| 17:47.04 | cusco | Dial(SIP/6542${EXTEN:4}@gateway,,g); is syntaxt correctly isn't it? dial(bla,timeout,options) |
| 17:47.17 | bougyman | an event socket library we wrote, it's not asterisk based. |
| 17:47.52 | evilgeenius | bougyman: so its a ruby library? |
| 17:48.02 | bougyman | yessir. |
| 17:48.17 | evilgeenius | Is this a commercial product? |
| 17:48.33 | bougyman | no, open source, githubbed. |
| 17:48.37 | jamko | pabelanger: http://pastebin.com/H6CTd2Ta |
| 17:48.41 | pabelanger | cusco: what version of asterisk? |
| 17:48.48 | evilgeenius | What's wrong with adhearsion/RAGI? why did you write your own? |
| 17:48.55 | evilgeenius | bougyman: got the link? :-) |
| 17:49.03 | bougyman | evilgeenius: off-topic in here, pm. |
| 17:49.17 | pabelanger | jamko: Ok, now generate a backtrace from the core dump. Read doc/backtrace.txt |
| 17:49.43 | pabelanger | [Aug 17 13:40:43] WARNING[9194]: config.c:1102 process_text_line: parse error: No category context for line 14 of /etc/asterisk/cdr_mysql.conf |
| 17:50.04 | [TK]D-Fender | cusco: So what NoOp don't you see that you feel you should? |
| 17:50.48 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 17:50.50 | a1fa | hey |
| 17:50.57 | a1fa | what is the 10-digit dialing compliance called? |
| 17:51.06 | a1fa | ;) |
| 17:51.36 | [TK]D-Fender | a1fa: You're probably thinking of ... |
| 17:51.38 | [TK]D-Fender | ~nanpa |
| 17:51.39 | infobot | somebody said nanpa was North American Numbering Plan Administration; the organization responsible for administering the integrated telephone numbering plan serving 19 North American countries. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.com/ |
| 17:52.16 | a1fa | somebody just implemented 11 digit dial and disabled 10 dial |
| 17:52.19 | a1fa | so i have to complain |
| 17:52.20 | a1fa | :( |
| 17:52.44 | [TK]D-Fender | a1fa: could you be a little more generic please? |
| 17:52.55 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com) |
| 17:52.59 | a1fa | the phone company requires 11 digits vs 10 |
| 17:53.08 | a1fa | 1+XXX+XXX+XXXX |
| 17:53.10 | drmessano | We're not the phone company |
| 17:53.14 | drmessano | Tyr #tacobell |
| 17:53.15 | a1fa | i know :) |
| 17:53.17 | drmessano | Try |
| 17:53.29 | a1fa | drmessano : i knew D-Fender would know the answer |
| 17:53.33 | [TK]D-Fender | drmessano: All I got was "CONGESTION" :p |
| 17:53.47 | a1fa | hehe |
| 17:53.53 | [TK]D-Fender | orders a quadruple bypass |
| 17:53.57 | jamko | pabelanger... thanks... stupid oversite.. pesky ; in the wrong place. |
| 17:54.06 | a1fa | [TK]D-Fender : speaking of Congestion.. I had this number that calls me every month, and stays stuck in the IVR for 31minutes |
| 17:54.11 | *** part/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 17:54.17 | a1fa | :) |
| 17:54.18 | [TK]D-Fender | ~cds |
| 17:54.19 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 17:54.23 | [TK]D-Fender | ^^^ |
| 17:54.44 | [TK]D-Fender | a1fa: Or stop looping your damn IVR forever like a schmuck :p |
| 17:54.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:54.49 | a1fa | i just set the absolute timeout :) |
| 17:54.57 | a1fa | there, i fixxxd it |
| 17:55.00 | [TK]D-Fender | a1fa: No, stop looking forever like a schmuck :p |
| 17:55.28 | [TK]D-Fender | cusco: Well? Which NoOp? |
| 17:55.32 | [TK]D-Fender | looping* |
| 17:55.54 | a1fa | :X |
| 17:56.02 | a1fa | but absoulte timeout fixes everything |
| 17:56.09 | a1fa | har har har |
| 17:56.19 | a1fa | it even stops lousy complaint calls i used to get |
| 17:56.22 | a1fa | :( |
| 17:56.31 | EmleyMoor | Oh, so that's why Asterisk and my N97 aren't getting on... |
| 17:56.34 | EmleyMoor | <g> |
| 17:57.37 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 18:02.37 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 18:04.29 | rustyclarkson | Any suggestions for a good way to run a command ASAP after Asterisk is started? [without getting "Unable to connect to remote asterisk (does /var/run/asterisk//asterisk.ctl exist?")] |
| 18:04.29 | rustyclarkson | Is "sleep/wait" the most sensible solution? |
| 18:05.05 | p3nguin | Depends what you want to run and why. |
| 18:06.03 | rustyclarkson | I would like to run '/usr/sbin/asterisk -rx "originate Local/hack@moh_hack application Echo"', and because I want my MOH stream to always be active |
| 18:09.02 | *** part/#asterisk newasterx (~dasdasdsa@114.199.101.35) |
| 18:10.06 | [TK]D-Fender | rustyclarkson: Mod whatever script you use that loads it in the first place |
| 18:11.51 | rustyclarkson | yea, I'm using /etc/init.d/asterisk, which runs start-stop-daemon, after start-stop-daemon, if I run '/usr/bin/asterisk -rx "<command>"', I get "unable to connect to remote asterisk...", my current solution is sleeping for 5 seconds before running the command |
| 18:12.18 | rustyclarkson | i was just wondering if there was a better way for knowing when i'll be able to run the command :p |
| 18:12.37 | ruben23 | hi guys i have asterisk using voip trunk, but when calling even a single call there is a great echo on the line, is there any work around for this..? |
| 18:13.00 | Naikrovek | rustyclarkson: after asterisk has started (not necessarily when the startup script returns) is when you can run -rx commands |
| 18:13.26 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:14.13 | [TK]D-Fender | rustWell you know you need to wait... so WAIT. You wanted the TRIGGER POINT. You have it. |
| 18:18.45 | cusco | [TK]D-Fender: the noop() before the hangup on http://paste.debian.net/83938/ |
| 18:19.11 | Naikrovek | rustyclarkson: it'll probably suffice to look for the .pid file to know if asterisk is running |
| 18:19.49 | pabelanger | rustyclarkson: asterisk -rx "core waitfullybooted" |
| 18:20.23 | pabelanger | then: asterisk -rx "your command" |
| 18:20.46 | WIMPy | Hmm. Since 1.6 I din't have any trouble doing asterisk;asterisk -rx. |
| 18:21.58 | rustyclarkson | Thanks Naikrovek, [TK]D-Fender and pabelanger. I'll try and work with waitfullybooted. ( I really like that ) |
| 18:22.15 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 18:25.32 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 18:27.37 | pgarcia | WIMPy, I was in a meeting... sorry... for NT PtMP I have to use 1.8? You know why? |
| 18:27.45 | *** join/#asterisk op_tech (~optech@96.234.232.223) |
| 18:28.27 | WIMPy | pgarcia: It wasn't available before. |
| 18:28.33 | op_tech | hi, I just joined this IRC channel...is anyone here perhaps able to help me with matching channel state with extensions? |
| 18:28.37 | WIMPy | At least not using dahdi. |
| 18:28.59 | op_tech | show channels and show channels verbose don't provide hold status; sip show channels does |
| 18:29.16 | [TK]D-Fender | cusco: Line 37. Looks like * killed the entire CHANNEL. Why else do you think "h" got called? |
| 18:29.17 | p3nguin | drmessano: Did you record your own sounds for your 8-ball? I use SetCalledParty() and just display the answer in text on my phone display, but if there are sound clips available, that would be nicer. |
| 18:29.29 | op_tech | however, I can't figure out a way to get full channel ID from show channels to match with the sip channel |
| 18:29.51 | pgarcia | WIMPy, I see.. so using Dahdi, this feature is available only on 1.8 ... Other than that, everything is supported in 1.6 (nt, te, pp with/without overlap)? |
| 18:30.23 | pabelanger | op_tech: I think your looking for device hints. |
| 18:30.27 | WIMPy | pgarcia: yes |
| 18:30.34 | op_tech | no, not device hints |
| 18:30.42 | op_tech | I'm trying to implement chanspy |
| 18:30.58 | jamko | I know this is an obvious message but, new to putting * in DB, sooo: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 18:31.04 | op_tech | but a user can be on multiple channels, with someone on hold |
| 18:31.18 | op_tech | I'm trying to find the correct channel by looking at hold messages on the console |
| 18:31.25 | op_tech | and spying on the channel not on hold |
| 18:31.31 | pgarcia | WIMPy, interesting... but probably a back-porting is possible... or there are core changes that make this very hard to do it ?(maybe this is more a question to #asterisk-dev...) |
| 18:31.46 | op_tech | the user can't loop through the channels using * for privacy reasons (no permission to listen in) |
| 18:32.45 | WIMPy | pgarcia: You can use other channels, depending on what versions of other stuff you require and what hardware you're using. |
| 18:32.54 | op_tech | right now I have a perl script which takes in an extension, finds any channels associated with that extension/username, and I need to match those channels with sip channels to find hold status |
| 18:32.56 | Micc_ | I get these errors a few times a day and I've read up on this error and done everything it says to fix it but it still seems to happen. Is it something I can ignore? Here is the error Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.1.18 |
| 18:33.25 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 18:34.33 | pgarcia | I see.. I'd like to stick with Dahdi, though... I'm not sure where I'm gonna need NT/PTMPso I can keep going ... Are you aware when 1.8 is due to be released? |
| 18:34.47 | op_tech | jamko, does your mysql client connect? |
| 18:35.07 | op_tech | (did you set up odbc?) |
| 18:35.14 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 18:35.18 | WIMPy | pgarcia: No idea. Just tried the beta3. |
| 18:35.44 | WIMPy | And after I tried, I'm not sure I want to change to dahdi soon. |
| 18:36.03 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 18:36.08 | hardwire | you jerks kicked me. |
| 18:36.14 | hardwire | :P |
| 18:36.20 | hardwire | stupid nickserv. |
| 18:36.20 | pgarcia | WIMPy, hmm... could you tell me the main reasons for that? |
| 18:37.33 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:37.35 | op_tech | no one here has experience with channel identification? |
| 18:37.39 | WIMPy | pgarcia: I managed to kill it immediately and I see configuration issues. And otherwise chan_lcr has worked very well for me since I changed there. |
| 18:38.06 | WIMPy | But let's see how 1.8 develops. |
| 18:39.22 | pgarcia | that's the good of having different options... I'll give 1.8 a try, just for fun..... |
| 18:39.42 | jamko | op_tech: I followed this article to a T: The mysql and odbc configs are at the bottom: http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation |
| 18:40.47 | hardwire | anybody other than me experienced a problem with sip requests blocking eachother while one request waits for an ACK? |
| 18:41.01 | op_tech | did you do the optional stuff also? |
| 18:41.29 | op_tech | hardwire, have you experimented with canreinvite? |
| 18:41.42 | op_tech | did you do the optional stuff also? that's to jamko |
| 18:41.45 | hardwire | it's kind of strange.. if my DSL is congested chan_sip becomes unresponsive if it's attempting to route a call through it .. completely blocking all internal SIP traffic as well while it waits. |
| 18:41.52 | jamko | op_tech: everything except for libpri |
| 18:41.53 | hardwire | op_tech: yeh.. not really related. |
| 18:42.22 | jamko | and no speex |
| 18:42.45 | hardwire | op_tech: what sort of channel identification (I just rejoined) |
| 18:42.48 | op_tech | jamko: have you tried mysql command line? |
| 18:43.03 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:43.08 | op_tech | hardwire: basically, I'm trying to match up channels from show channels (show channels verbose) with sip show channels to get the hold state |
| 18:43.15 | jamko | I can connect to the mysql and see the database. |
| 18:43.33 | op_tech | show channels doesn't show the hold state, and sip show channels doesn't have the same information I need to get from matching extensions and adapters |
| 18:43.45 | hardwire | op_tech: core show channel xyz.. get the channel id.. look for the ID in sip show channels |
| 18:43.53 | hardwire | op_tech: or write a quick manager program to handle it. |
| 18:44.00 | hardwire | since it will see all the information as it happens |
| 18:44.09 | op_tech | hardwire: the problem is in an agi, it's not getting the full channel name |
| 18:44.21 | hardwire | only the first x chars right? |
| 18:44.36 | op_tech | hardwire: it cuts off the full channel name, and I can't then find out more information about it. On the console, I can tab to complete the channel name |
| 18:44.52 | hardwire | op_tech: set a variable before you call the AGI with the currentl channel id. |
| 18:45.01 | hardwire | that way it magically appears as an agi variable |
| 18:45.05 | hardwire | use the CHANNEL function |
| 18:45.18 | op_tech | hardwire: won't help; it's an agi to chanspy on a given extension |
| 18:45.44 | hardwire | op_tech: I don't see how seeing it in sip show channels would help with that application |
| 18:46.04 | hardwire | is it an IVR to chanspy? |
| 18:46.22 | op_tech | hardwire: basically, a user can be on multiple calls, and I need to find which one is not on hold. As far as I can tell, only sip show channels shows hold state |
| 18:46.49 | op_tech | it's not an IVR as much as a perl script to take an extension, loop through channels that use that extension/username, and find the one that is not on hold |
| 18:47.46 | op_tech | If you can think of a better way to do it, I'm all ears. extenspy doesn't work; it doesn't capture outgoing calls. chanspy lets me loop through with *, but the specs don't allow for the party spying to listen in, only to whisper |
| 18:48.03 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 18:48.11 | hardwire | op_tech: I think it's in your best interest to make a manager daemon that collects this information and puts it somewhere |
| 18:48.21 | hardwire | because using the CLI won't help much.. too much overlap |
| 18:48.51 | op_tech | the manager will return the proper channel variable? |
| 18:49.18 | hardwire | op_tech: either use manager api and log events and make a note of them.. or use it to query using getvar |
| 18:49.23 | hardwire | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+GetVar |
| 18:49.44 | hardwire | getvar, sip/channel-blah, SIPCALLID |
| 18:50.19 | op_tech | but how do I get the full sip/channel? |
| 18:50.41 | hardwire | op_tech: do me a favor and look at "core show channel SIP/blah-xyz" on an active channel |
| 18:50.49 | hardwire | any of those vars should be fetchable from the Manager API |
| 18:50.57 | hardwire | you can also list sip channels afaik |
| 18:51.30 | hardwire | hmm.. maybe not on the sip channels. |
| 18:51.32 | op_tech | I don't think you understand. |
| 18:51.44 | op_tech | show channels |
| 18:51.44 | op_tech | Channel Location State Application(Data) |
| 18:51.44 | op_tech | SIP/7277-021-094875b (None) Up Bridged Call(SIP/7277-061-09cc |
| 18:51.49 | op_tech | The channel is cut off |
| 18:51.54 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 18:51.58 | op_tech | core show channel on that WON'T find the channel |
| 18:52.05 | op_tech | it needs me to tab to get the rest of the channel name |
| 18:52.18 | hardwire | hehe |
| 18:52.34 | hardwire | core show channels consise |
| 18:52.35 | hardwire | or |
| 18:52.38 | hardwire | core show channels verbose |
| 18:52.41 | hardwire | does that help? |
| 18:53.07 | TTT_Travis | Who do you you guys recomnmend for SIP termination? I only use about 1000 minutes outbound a month so looking for something pay-as-you-go probably |
| 18:53.28 | op_tech | no, I am already using verbose |
| 18:53.30 | bougyman | i've been very satisfied with flowroute. |
| 18:53.31 | p3nguin | I like VoIP.ms or Flowroute. |
| 18:53.39 | op_tech | It gives me more information, but not a longer channel name |
| 18:53.44 | hardwire | op_tech: use concise |
| 18:53.56 | hardwire | sorry.. spelled it wrong earlier |
| 18:53.59 | TTT_Travis | haven't heard of flowroute so I'll have to check them out |
| 18:54.33 | hardwire | op_tech: or.. like I said earlier.. use the manager API to log events as they happen .. no polling the CLI needed |
| 18:54.40 | p3nguin | You're looking about about 1 cent per minute for calls in the US. |
| 18:54.44 | op_tech | so concise gives it to me, thanks. But it's almost impossible to find the correct channel I need |
| 18:54.47 | hardwire | once you tap into the manager API via telnet and watch the call flow.. you'll understand |
| 18:54.47 | Nugget | telnet is eeeeeeevil! |
| 18:55.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:55.21 | op_tech | I feel like using a manager is overkill; there is a lot of traffic, and it would require a lot of sorting...but I can play around with it |
| 18:55.36 | hardwire | op_tech: computers are really.. really.. good at sorting. |
| 18:55.36 | TTT_Travis | is there a way to just add a few dollars on voip.ms to test? or does it require $25 minimum? |
| 18:56.01 | hardwire | they freaking live for it |
| 18:56.03 | p3nguin | You can apply the $25 minimum and test. If you don't like the service, ask for a refund of your unused funds. |
| 18:56.04 | op_tech | lol |
| 18:58.02 | p3nguin | I can't imagine why you'd want to get a refund, though. Check the termination rates on the web site before you apply funds to your account. |
| 18:58.48 | p3nguin | Their prices for DIDs and unlimited DIDs are very low, too, so you can even get a phone number from them if you want. |
| 19:00.52 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 19:02.33 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:03.29 | *** join/#asterisk ebroad (~EB@72.11.213.195) |
| 19:03.43 | hardwire | I'm thinking to get around the chan_sip problems I'm having I should probably use openser as a registration server as well as outbound proxy.. should be fun |
| 19:03.49 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 19:04.05 | hardwire | at least for any traffic using a circuit that may ever be congested.. ever. |
| 19:05.41 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 19:07.28 | Naikrovek | howcome everyone is so much more expensive than my provider |
| 19:08.05 | Naikrovek | right now i have unlimited trunks. $44/mo for unlimited everything. US, Canada, most of europe = free, part of the cost of the trunk |
| 19:08.09 | uqlev | Naikrovek, then your provider is cheating |
| 19:08.20 | bougyman | Naikrovek: who the heck is that? |
| 19:08.25 | bougyman | oh.. $44/month per trunk? |
| 19:08.32 | bougyman | i can beat that with a bunch of providers. |
| 19:08.43 | bougyman | if a trunk == one concurrent call. |
| 19:08.49 | Naikrovek | and the per-minute guys i'm about to switch to are $20/trunk, up to 20k minutes, then $0.009/minute after |
| 19:08.57 | Naikrovek | yes |
| 19:09.03 | Naikrovek | s/trunk/channel/ |
| 19:09.16 | bougyman | flowroute virtual pris are 17.95/month, but I haven't seen how much usage that is. |
| 19:09.22 | bougyman | it's supposedly unlimited. |
| 19:09.26 | bougyman | i'm sure there must be caps, though. |
| 19:09.29 | Naikrovek | unlimited, but not free |
| 19:09.42 | Naikrovek | per-minute on everything (even local) |
| 19:09.49 | Naikrovek | which makes sense |
| 19:10.05 | Naikrovek | the provider i'm about to switch to does free local though, even when i go over 20k minutes |
| 19:10.16 | Naikrovek | local calls still use bandwidth, so it makes sense that they cost a little something |
| 19:10.26 | bougyman | which provider is taht? |
| 19:10.36 | bougyman | the only sip-local providers i've found are airespring and verizon. |
| 19:10.43 | tuxxie | I am going to create an IVR that has five options that will be numbered 1-5. I have extensions that start start with 1,2 and 4's. Will this create an issue for my ivr? |
| 19:10.54 | fauxalliance | no |
| 19:10.56 | fauxalliance | not at all |
| 19:10.58 | Naikrovek | let me look it up. i'm going through a reseller - let me get the name of the provider itself |
| 19:11.23 | Naikrovek | voxitas |
| 19:11.47 | crowb4r | I like vitelity personally |
| 19:11.50 | tuxxie | fauxalliance: was no to me? |
| 19:12.11 | fauxalliance | tuxxie thats affirmative. |
| 19:12.24 | tuxxie | thanks :) |
| 19:12.24 | fauxalliance | personally, link2voip is swell. |
| 19:16.23 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 19:16.57 | jsidhu | is there a commandline tool to normalize audio or change amplitude of a .sln file? if not, how can i convert it to a wav? |
| 19:17.13 | fauxalliance | jsidhu, sox |
| 19:17.51 | jsidhu | i see examples of converting sln to wav, but when i try to reverse the process, it says unknown format, .sln |
| 19:18.28 | fauxalliance | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk @ jsidhu |
| 19:18.54 | jsidhu | yes, that only talks about wav to sln, not the reverse |
| 19:19.27 | fauxalliance | the point is now moot. |
| 19:19.38 | bougyman | is macro-stdexten some sort of standard? |
| 19:19.44 | bougyman | like... will it get used by default? |
| 19:19.56 | bougyman | i don't have it anywhere in extensions.conf but it seems it is being used. |
| 19:20.23 | jsidhu | sox: Failed reading test.sln: Do not understand format type: sln |
| 19:21.09 | fauxalliance | sox -t raw -r 8000 -s -w -c 1 {inputfile}.sln {outputfile}.wav |
| 19:21.10 | fauxalliance | p00t |
| 19:21.31 | fauxalliance | did you 'man sox |
| 19:21.38 | fauxalliance | ' jsidhu |
| 19:21.53 | jsidhu | yeah reading thru the man page, got lost a bit |
| 19:22.08 | jsidhu | thanks for the tip |
| 19:22.13 | fauxalliance | no sweat |
| 19:22.28 | p3nguin | $44 per month? I haven't spent $44 on my phone service in an ENTIRE YEAR. |
| 19:23.35 | cusco | tk- |
| 19:23.41 | cusco | er... |
| 19:23.57 | bougyman | is it ; Create voicemail mailbox and use use macro-stdexten |
| 19:24.12 | bougyman | that's the comment on hasvoicemail = yes |
| 19:24.31 | p3nguin | Sounds like some sample crap. |
| 19:24.42 | bougyman | so if we get our phones out of users.conf and into sip.conf we can alleviate that? |
| 19:24.51 | bougyman | or just change that to hasvoicemail = no? |
| 19:25.03 | bougyman | it's in [general] in users.conf |
| 19:25.04 | p3nguin | Stop using sample configs. |
| 19:26.03 | bougyman | that's exactly what i'm in the process of doing. |
| 19:26.26 | p3nguin | When you stop using sample configs, macro-stdexten will only exist if you create it. |
| 19:26.44 | bougyman | i've removed all references to it, we have to do this iteratively |
| 19:26.49 | p3nguin | And there will be no reference to Macro(stdexten) unless you create them. |
| 19:26.56 | bougyman | i've come behind another * engineer who set this up approx two years ago. |
| 19:27.20 | bougyman | we had to strip out the astguiclient and then his own special agi stuff first. |
| 19:27.25 | bougyman | now we're on this stdexten stuff. |
| 19:28.25 | bougyman | p3nguin: seems Dial will drop to macro-stdexten if it exists, too. |
| 19:28.46 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 19:28.48 | p3nguin | Not unless you tell it to (or don't tell it not to when using sample configs). |
| 19:29.19 | p3nguin | In a default config, nothing exists until you create it. |
| 19:35.27 | jsidhu | how can i tell file is playing for music on hold? ive got the logging turned up pretty high and still dont see any mention of which file is selected to play.. |
| 19:37.54 | jamko | If I don't have any peers setup for asterisk realtime, but peers are attempting to register, would I get this message: [Aug 17 14:32:01] WARNING[10081]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 19:38.35 | p3nguin | jsidhu: I would probably run something like lsof -u asterisk|grep -i "sln\|ulaw\|gsm\|wav\|mp3" so see what files are being used. |
| 19:38.59 | p3nguin | to see, rather. |
| 19:39.22 | KavanS | oh snap |
| 19:40.45 | jsidhu | thanks p3nguin |
| 19:43.01 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 19:45.09 | hardwire | exten => h,1,Hangup |
| 19:45.13 | hardwire | incredibly redundant.. right? |
| 19:45.26 | hardwire | that's not even a failsafe of any kind.. |
| 19:45.56 | Naikrovek | p3nguin: yes, $44/mo with all the fees and taxes included, and about 20k minutes of usage per month |
| 19:47.21 | p3nguin | If you use enough minutes that you need an unmetered service, $44/mo. is probably a great deal. I don't use that much, though, so it wouldn't be a wise choice for me to go to an unmetered service. Is it Voice Spring that has that rate? |
| 19:47.22 | *** join/#asterisk fraudory (d56be9a9@gateway/web/freenode/ip.213.107.233.169) |
| 19:47.41 | Naikrovek | p3nguin: agreed. yes. |
| 19:47.45 | fraudory | Hi all, quick question and wondered if anyone might be able to help me. I'm running System() to execute a couple of perl scripts, but they take a while to perform and this seems to 'hold' up the dialplan execution... is there any similar function that doesn't care about the result of execution? |
| 19:48.35 | Naikrovek | fraudory: i forget - can you have perl fork, then return, and do the work in the fork? |
| 19:49.32 | p3nguin | If I ever need a crapload of minutes, hopefully I will remember that rate from that company. For now, however, I'll continue with my few dollars per month services. |
| 19:49.47 | Naikrovek | p3nguin: yeah for home use unlimited trunks are silly |
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| 19:50.49 | fraudory | Naikrovek: Ahh, thanks for the suggestion yes I'll take a look into that |
| 19:51.10 | crowb4r | Naikrovek: I think you can |
| 19:51.27 | Naikrovek | fraudory: you can easily write a perl script wrapper that calls your script and returns immediately, ignoring the output |
| 19:51.33 | crowb4r | fork and return, but continue to work in the fork. |
| 19:51.43 | Naikrovek | yeah i think you can too, but it's been a while since i've done it |
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| 20:21.32 | TTT_Travis | whats the best cheap IP phone for SoHo users? |
| 20:24.13 | keith4 | define "cheap" |
| 20:24.41 | Micc_ | aastra 6730i |
| 20:25.06 | Micc_ | I've heard snom is pretty good and similar price range I think. |
| 20:25.09 | WIMPy | Does it have to work? |
| 20:25.18 | keith4 | heh. yah... check ebay |
| 20:25.18 | Micc_ | don't get grandstream |
| 20:25.28 | keith4 | polycom's low-end stuff is decent |
| 20:25.38 | Naikrovek | polycom's everything is awesome |
| 20:25.42 | Naikrovek | ftfy |
| 20:25.52 | WIMPy | Snom 320/360/370 is good, yes. |
| 20:26.07 | keith4 | isn't there a 330 in there, too? |
| 20:26.15 | keith4 | i can't keep their models straight |
| 20:26.27 | keith4 | ... or is that polycom? ;-) |
| 20:26.33 | WIMPy | There is a 300, but thats different and I don't know how good that is. |
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| 20:27.02 | Naikrovek | polycom has a 320,321,330,331,335,450,550,650,670 |
| 20:27.14 | WIMPy | There is th 8xx series, but that's solid gold. |
| 20:27.19 | keith4 | i have a few snom 300s. they're okay |
| 20:27.28 | Naikrovek | 320 and 330 are discontinued but mentioned because they were effin' popular |
| 20:27.29 | keith4 | very basic functionality |
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| 20:32.20 | Naikrovek | i used snom way back in the day |
| 20:32.24 | Naikrovek | liked them when i used them |
| 20:32.36 | Naikrovek | but it's been oh 8 years |
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| 20:41.18 | titter | Any reason the callerid number on my Polycom would show as sip:1112223333@111.111.111.111 instead of just 1112223333? The name shows as just 1112223333 |
| 20:41.27 | titter | Incoming calls^ |
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| 20:50.10 | p3nguin | Is there a function or application to force lower case letters in a given string? |
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| 20:53.49 | p3nguin | I saw something about TOLOWER() and TOUPPER() to change case, but I guess it doesn't exist in 1.4. |
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| 20:54.41 | Corydon76-dig | p3nguin: there are an ever-increasing number of features which are not in 1.4 |
| 20:55.06 | p3nguin | I'll keep holding my breath for a 1.8 release. |
| 20:55.42 | Corydon76-dig | p3nguin: I'd say about 4 weeks or so |
| 20:55.53 | Corydon76-dig | but then again, it's not my say |
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| 20:58.14 | p3nguin | Are 1.5 and 1.7 nonexistent, or reserved similar to the way the Linux kernel reserves the odd numbers? |
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| 21:06.03 | drmessano | Do the current 1.6 applications work on 1.8... Fax, G729, SFA, etc> |
| 21:06.04 | drmessano | ? |
| 21:06.42 | chazzam | the 1.6.2 ones might |
| 21:07.00 | chazzam | but I dunno =/ |
| 21:07.11 | drmessano | Well, I was looking for an authoritative answer.. I could guess too :) |
| 21:07.13 | chazzam | I haven't had a chance to test that out yet |
| 21:09.39 | chazzam | actually, I might could do that now |
| 21:11.01 | WIMPy | I'd be more interested if it's possible to connect chan_capi to capi4hylafax via a capi loopback device. Has anyone tried that? |
| 21:11.54 | WIMPy | Or is there another solution for G4 fax, I haven't found yet? |
| 21:12.59 | *** join/#asterisk lesouvage (~lesouvage@cc741036-b.groni1.gr.home.nl) |
| 21:13.27 | chazzam | hmm, well, the G.729 generic_32 module loaded, but didn't load my license for G.729 |
| 21:13.47 | chazzam | but then, I haven't tried loading that in probably 6 months or more, so I dunno if the license is still valid =/ |
| 21:13.58 | chazzam | so ... non-conclusive |
| 21:18.43 | chazzam | hmm, re-registered, and I still just get failed to initialize copy protection |
| 21:19.19 | chazzam | I am running svn though r282366 |
| 21:23.13 | leifmadsen | I'd call Digium then because that's a commercial module |
| 21:23.23 | leifmadsen | drmessano: the answer is no they do not |
| 21:24.18 | leifmadsen | p3nguin: 1.1, 1.3, 1.5, and 1.7 are non-existent |
| 21:24.18 | drmessano | leifmadsen: Thank you |
| 21:24.18 | drmessano | I guess I will wait before I blow up my home system for 1.8 |
| 21:24.19 | leifmadsen | drmessano: you'll have to wait until Asterisk 1.8.0-RC1 for new commerical modules to be built |
| 21:24.36 | leifmadsen | during the beta it is possible the API/ABI could change (although at this point unlikely) |
| 21:24.38 | drmessano | Ah ok |
| 21:24.43 | drmessano | Makes sense |
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| 21:38.39 | TTT_Travis | Polycom 320 or aastra 6730i? |
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| 22:18.19 | jkroon | hi guys, i'm looking to package the g729 codec for gentoo. However, due to slight limitations in the ebuild naming format (specifically the version part) makes this difficult without some help from upstream. Is there somebody here I can speak with with regards to possible help from Digium's side or should I contact support? |
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| 22:34.57 | Micc_ | TTT_Travis, only thing I don't like about the 6730i is that you can't send it a alert-info header to silent the ringer. |
| 22:35.23 | Micc_ | If you need that feature then get the Polycom, if not, the 6730i is a good bet for the money. |
| 22:36.56 | Diffen2 | Hello, im trying to connect a asterisk to another asterisk. if i do a register => uname:password@ip the first asterisk/1000 it works really good. but if i add a second line to the same ip but a different username and tries to call that number i get username mismatch, have <1901-iptelefonibolaget>, digest has <1902-iptelefonibolaget> |
| 22:37.26 | Diffen2 | isnt it possible to have two type=friend users registered to the same ip of the first asterisk? |
| 22:37.56 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 22:38.08 | Micc_ | Diffen2, I can't think of a reason why you would want to do that. |
| 22:38.32 | Micc_ | Diffen2, you can pass as many calls as you want through the same peer. |
| 22:38.41 | jkroon | Micc_, there are reasons for doing that (accounting/billing purposes usually) |
| 22:39.00 | jkroon | Diffen2, create a single type=user that auths based on host and two type=peer entries. |
| 22:40.31 | Diffen2 | jkroon hmm ok so in the sip.conf only one register line? and that should be a type=user? |
| 22:42.10 | jkroon | no, you end up with two register lines, one for each username+pass, a [prov] type=user; host=i.p.a.d and two [user] type=peer; host=i.p.a.d sections. |
| 22:42.51 | Diffen2 | ok i wil try that out |
| 22:43.57 | p3nguin | type=user does not authenticate based on host. |
| 22:44.43 | p3nguin | Register statements are only to tell the other system how to reach you. It is intended for dynamic IP addresses or mobile user agents. |
| 22:44.52 | Diffen2 | this is odd :D the type=user doesnt work but the type=peer work. the type=peer didnt work before but now it works fine. |
| 22:47.00 | p3nguin | Each of the two asterisk systems would only need a single register statement and single peer definition. |
| 22:47.37 | jkroon | ok, i had a trick around it, the above def works with IAX/2. p3nguin - a sip user is always first located on host= line before attempting a lookup by username if I'm not mistaken. |
| 22:48.40 | p3nguin | type=user authenticate by username, type=peer authenticates by IP/port. type=user only allows calls to go inbound to asterisk. |
| 22:49.05 | Diffen2 | hmm i dont get it, if i set the type=peer on both the users in sip.conf only the first register => line works. not the second one. |
| 22:49.08 | jkroon | hmm, so sip and iax/2 differs on that. |
| 22:49.28 | jkroon | type=peer on one and type=friend on the other? |
| 22:49.46 | p3nguin | You only need ONE register statement. Stop trying to complicate things. |
| 22:49.56 | jkroon | i recall having issues with registering two SIP accounts on a remote server. |
| 22:50.14 | jkroon | Diffen2, what is the purpose (for you) of having two registers ? |
| 22:50.23 | jkroon | two different accounts from the same provider? |
| 22:50.48 | p3nguin | I thought he said he was interconnecting a pair of Asterisk systems. |
| 22:52.32 | Diffen2 | since the second asterisk is not connected directly to a pstn gw im using the first asterisk to send the calls to the second one. the second one is a bigbluebutton server so there will never be outgoing calls, just incoming to a specific conference room. so after the register=> info i have done a /1000 for conferenceroom number. |
| 22:52.35 | jkroon | in that case i agree with you, he can get away with only a single type=friend on both sides, and if he's on static IPs no register statements at all. |
| 22:53.25 | Diffen2 | works great with one regitster but not two, if there is a smart way to separate the calls to the second asterisk (im sure there is) i would love to learn. I just thought that my solution was pretty neat :D |
| 22:53.55 | jkroon | you could just on the first asterisk server do Dial(SIP/peername/1000) |
| 22:54.43 | jkroon | instead of Dial(SIP/peername) ... |
| 22:55.10 | p3nguin | Regardless how many calls you want to push between the systems, only a single register would be required on each system to locate itself to the other system. |
| 22:55.13 | Diffen2 | hmm cool |
| 22:55.56 | Diffen2 | ok p3nguin, should it be a peer register then |
| 22:56.19 | p3nguin | register is register is register. |
| 22:56.34 | p3nguin | The register statement tells another system how to contact you. |
| 22:59.18 | jkroon | p3nguin, i'd still like to know how you would handle the case where you obtain multiple SIP accounts from a provider, ie you can't control what they use, and you HAVE to register each of those accounts to them, as well as be able to dial out with the different accounts (for example based on who the "internal" user is) |
| 23:00.13 | p3nguin | I would use a register for every user that needs its own account. I would also probably include the username in the Dial() command. |
| 23:00.56 | p3nguin | But that's just a first guess since I don't have any reason to do that myself. |
| 23:01.49 | p3nguin | The point was that to interconnect two asterisk systems together only one account is required on each system. |
| 23:04.39 | rustyclarkson | Naikrovek: thanks for the idea of waiting for the pid, wasn't long enough so i decided to use the /var/run/asterisk/asterisk.ctl to wait for. Turns out it's long enough to submit commands, but some applications haven't registered yet, so I still have to wait longer. |
| 23:05.41 | rustyclarkson | pabelanger: thanks for the idea of waitfullybooted, it's pretty neat, unfortunately I'm still getting "No such command 'originate Local/hack@moh_hack application Echo'..." if I run the command after it returns from "core waitfullybooted", so i still end up having to wait another few seconds |
| 23:09.26 | TTT_Travis | is there any mac software that integrates with Asterisk to dial calls etc.? I would use hudlite but my server is debian and I can't find hudlite-server software for debian |
| 23:11.16 | bougyman | most ofthe mac guys I know use adhearsion for that |
| 23:12.33 | TTT_Travis | yeah were mostly just looking for a way to initate calls from Address Book etc. and maybe show incoming calls caller ID on screen or something |
| 23:13.07 | bougyman | i bet the #adhearsion guys have done all of that stuffs. |
| 23:13.53 | TTT_Travis | ok thanks |
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| 23:33.57 | Corydon76-dig | bougyman: only thing I really don't like about Adhearsion is how their people have seem to have really drunk their own KoolAid |
| 23:34.36 | Corydon76-dig | In essence, "Adhearsion is the bestest, so everybody else should stop trying." |
| 23:35.32 | Corydon76-dig | (Seriously, they told Asterisk core developers to stop trying.) |
| 23:38.06 | carrar | Why haven't they stopped!! |
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| 23:46.41 | Diffen2 | p3nguin and jkroon thanks for you help :) |
| 23:46.44 | Diffen2 | now its time for bed |
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