| 00:00.39 | jakehow | details on phone setups |
| 00:00.59 | jakehow | i searched around on voip-info for the 941 but nothing really provides this level of detail |
| 00:01.16 | hads | You don't need to set anything else up... |
| 00:02.08 | [TK]D-Fender | jakehow : This is something you just have to open your eyes to realize. You try a setting, and the emprical evidence tells you what it means, and you just extrapolate the rest. |
| 00:02.49 | [TK]D-Fender | jakehow : I don't think there is a book out there to truely change the way you look at things so you can better realize what's right in front of you. |
| 00:03.29 | jakehow | [TK]D-Fender: i have not been "on the ground" so to speak w/ this project so that is probably part of the problem |
| 00:03.30 | [TK]D-Fender | jakehow : Real techies don't need manuals, just buttons to push :) |
| 00:03.40 | jakehow | but it seems to me there is a serious lack of documentation in this space |
| 00:03.45 | [TK]D-Fender | jakehow : Got to get your hands dirty :) |
| 00:04.06 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 00:04.23 | jakehow | articles and docs help to prepare you to go push buttons certainly though... |
| 00:04.27 | [TK]D-Fender | jakehow : Or maybe its just not worded in ways you were prepared to process to your satisfaction. |
| 00:04.34 | jakehow | quite possibly |
| 00:04.47 | jakehow | i am going to be messing w/ this all next week so just doing any research i can now |
| 00:04.56 | [TK]D-Fender | jakehow : this is as much a "trade" as say carpentry. Its hard to put in writing. |
| 00:06.06 | jakehow | well thanks for the help guys |
| 00:06.31 | [TK]D-Fender | jakehow : no prob, and good luck |
| 00:07.34 | Druken | [TK]D-Fender: evening... |
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| 00:08.54 | [TK]D-Fender | Druken : y0 |
| 00:09.04 | Druken | [TK]D-Fender: wut up ? |
| 00:12.50 | [TK]D-Fender | not much, just getting ready to head out to play some pool |
| 00:13.19 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 00:14.46 | *** join/#asterisk xAD (n=xAD@62.152.98.66) |
| 00:19.11 | [TK]D-Fender | ok, checkout time, bbiab (maybe) |
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| 00:22.48 | kink0 | anyone can tell me why DTMF from a Mexico PSTN phone does not works with my Asterisk ? |
| 00:23.15 | kink0 | of course works with some USA IVR systems, we have try to call to Networksolutions and works fine. |
| 00:23.25 | kink0 | but my ASterisk ignores her DTMF |
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| 00:46.44 | rogier | Can somebody tell me if asterisk can do a codec translation on the fly ? So let's say I my softphone (sip) that is registered to asterisk does not have the codec that is necessary to use a sip endpoint. |
| 00:48.51 | kink0 | rogier: yes |
| 00:50.12 | aXanaXa | hey guys can anyone point me to a howto on setting up multiple phonelines in a rollover type setup. I have 4 VoIP lines I want coming into an asterisk box and I want the incoming calls to ring everyone unless someone is on another line. |
| 00:50.14 | rogier | kink0, okay, good to know |
| 00:50.41 | kink0 | but you will consume cpu |
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| 00:51.04 | rogier | kink0, yes ofcourse, that is unavoidable |
| 00:52.20 | rogier | But I unfortunately have registered to a voip provider that uses a codec that is not available on linux, except in asterisk. Now I know that I can at least make up the limited credit I buyed from them. |
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| 00:55.34 | kink0 | rogier: most carriers uses g729 |
| 00:57.26 | rogier | kink0, that's exactly the one. Unfortunately proprietary. |
| 01:02.03 | kink0 | yes, but the most extended in use |
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| 01:45.51 | ashafi | ping! anyone home |
| 01:46.20 | bkw__ | NO COMMENT |
| 01:46.40 | macTijn | oh noes |
| 01:46.44 | macTijn | it's bkw__! |
| 01:46.57 | bkw__ | apparently so |
| 01:53.04 | mitcheloc | bkw__ just isn't quite as cool as the real bkw eh |
| 01:54.11 | bkw__ | what? |
| 01:54.15 | bkw__ | I am the real bkw boi |
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| 02:56.13 | vooduhal | Hey guys. I've got a strange problem. What is the site where I should post code, output, etc? |
| 02:56.16 | davidcsi | question: I just compiled asterisk 1.2.13, libpri 1.2.4, zaptel 1.2.10 and wanpipe 2.3.3-7 on a debian with kernel 2.6.8, everything went fine, zttools shows the sangoma E1 as UP/OK but there is no CHAN_ZAP.SO and no zap command on asterisk CLI... anyone knows why |
| 03:01.07 | vooduhal | Is anyone alive? |
| 03:02.41 | davidcsi | i don't think so |
| 03:12.46 | benjk | yes |
| 03:12.53 | benjk | you need a newer kernel |
| 03:12.59 | benjk | 2.6.15 or higher |
| 03:13.13 | file | is that the order you built in davidcsi? |
| 03:13.17 | benjk | the kernel API changed from .14 to .15 |
| 03:14.14 | benjk | Zaptel has been adjusted to use the newer kernel API |
| 03:14.43 | benjk | your zaptel.so build most likely failed as a result |
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| 03:18.27 | davidcsi | no |
| 03:18.30 | davidcsi | i got it... |
| 03:18.56 | davidcsi | wanpipe patches zaptel, so you gotta recompile zaptel AND asterisk... |
| 03:19.06 | davidcsi | its all up now... thanks anyway. |
| 03:19.54 | davidcsi | just put a call through and all ;) |
| 03:20.04 | vooduhal | Can someone tell me where I can post the output for my question? |
| 03:20.13 | *** part/#asterisk lule (i=lule@host203.200-117-175.telecom.net.ar) |
| 03:20.19 | davidcsi | pastebin.ca i think |
| 03:20.46 | davidcsi | yeah, there |
| 03:22.08 | vooduhal | Is there anyway to close these channels? http://pastebin.ca/227224 |
| 03:22.15 | vooduhal | Soft hangup doesn't do anything. |
| 03:22.55 | vooduhal | The only thing I've found is to restart asterisk but this is a 24/7 production ACD system. |
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| 03:46.14 | krapper | any word on when the asterisk gui framework will be available? |
| 03:46.36 | Qwell | krapper: it already is... |
| 03:46.48 | Qwell | krapper: "asterisk gui framework" is really just manager and static-http |
| 03:51.11 | krapper | when you say manager you're referring to AMS? So is asterisk gui framework just digium's version of those two items with a clean interface? |
| 03:51.23 | Qwell | AMD? |
| 03:51.25 | Qwell | erm, AMS? |
| 03:51.30 | amdtech | not me |
| 03:51.31 | amdtech | :) |
| 03:51.38 | krapper | http://www.intuitivecreations.com/contributions/AMS/ |
| 03:51.53 | Qwell | krapper: Now, there *is* a Digium GUI, but it's not the "asterisk gui framework" |
| 03:52.15 | krapper | ok so maybe i'm confusing terms then |
| 03:52.36 | krapper | i'm referring to what will be include digium's asterisk appliance |
| 03:52.42 | Qwell | the framework is what the GUI is built on |
| 03:53.34 | Qwell | krapper: the answer, however, is still "yes", but it's a bit...alpha |
| 03:53.41 | krapper | mark spencer was showing off screenshots of a gui interface via a browser created by digium at the ITC conference in san diego a few weeks ago |
| 03:54.05 | krapper | and he said it would be open source |
| 03:54.16 | Qwell | that it is |
| 03:54.22 | hads | asterisk-gui |
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| 03:55.53 | amdtech | svn co http://svn.digium.com/svn/asterisk-gui/trunk |
| 03:57.22 | krapper | yup |
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| 04:23.18 | docelmo | Man its good to be home |
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| 04:29.57 | Qwell | docelmo: What, got something against Dallas? :p |
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| 04:35.48 | DasTech | evening |
| 04:36.02 | DasTech | anyone here got sphinx working with asterisk |
| 04:37.29 | docelmo | yes.. Read the docs.. |
| 04:43.32 | DasTech | I tried the wiki and it does not work |
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| 04:45.02 | *** join/#asterisk kerlnel20 (n=kerlnel2@203.160.223.26) |
| 04:45.06 | kerlnel20 | hi there |
| 04:45.23 | kerlnel20 | can asterisk hold multivideo conferencing? |
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| 04:50.14 | kerlnel20 | can asterisk hold multivideo conferencing? |
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| 04:52.15 | kerlnel20 | can asterisk hold multivideo conferencing? |
| 04:55.38 | wunderkin | kerlnel .. ? |
| 05:02.47 | kerlnel20 | can asterisk hold multivideo conferencing? |
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| 05:07.34 | coppice | let's see. he asks about video, and repeats endlessly - must work in daytime TV |
| 05:08.07 | hads | :) |
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| 05:24.54 | Marquel | morning |
| 05:26.06 | kerlnel20 | coppice: .|. |
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| 06:04.58 | [TK]D-Fender | unload chan_moron.so |
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| 06:26.15 | wepy | hey |
| 06:26.39 | wepy | do i need asterisk if i just want to make free phone calls to my friend online? |
| 06:26.59 | wepy | like no ptsn, just purely internet voice communication.. |
| 06:28.06 | wepy | i mean, i know the other options are like skype and stuff, but they aren't open source.. and i'm wondering if my server with a static IP might be useful for this anyway... |
| 06:29.53 | hads | Depends what you want to do... |
| 06:30.19 | wepy | i just want to talk to my friend for free since international rates are high.. |
| 06:30.31 | wepy | i figure we'll both use softphones or something |
| 06:30.52 | hads | You just answered your own question. |
| 06:31.00 | wepy | although, if it's easy, i might want to setup voicemail or something on an asterisk box.. |
| 06:31.15 | wepy | hads: but to use softphones, i would need an asterisk server right? |
| 06:31.43 | hads | SO, you could call directly between the two, or use a SIP service. |
| 06:31.59 | wepy | directly... does that work? |
| 06:32.12 | wepy | what's a SIP service? |
| 06:32.21 | hads | No, I just said it for the fun of it ;) |
| 06:32.54 | wepy | hah.. so like linphone, the SIP phone i downloaded.. if i run that, maybe my friend can just use sip://wepy@my.ip.right.now ? |
| 06:33.17 | hads | Indeed, if the correct port is open. |
| 06:33.23 | wepy | interesting! |
| 06:33.25 | wepy | ok.. |
| 06:33.32 | wepy | but my IP changes all the time.. |
| 06:33.52 | wepy | is there a way to let my friend contact my static IP server somewhere, and have it automagically call me here? |
| 06:34.07 | hads | So you could use a service provider and register your softphone to them. |
| 06:34.25 | wepy | or, i could be the service provider, right? |
| 06:34.50 | wepy | hm |
| 06:34.56 | hads | Correct. |
| 06:34.59 | wepy | the problem is bandwidth i think.. |
| 06:35.13 | wepy | i can only get about 25 Kbyte/sec upload... |
| 06:35.18 | wepy | is that enough for a conversation? |
| 06:36.01 | hads | It's not a lot |
| 06:37.55 | wepy | also.. would asterisk work OK over a vpn? |
| 06:38.08 | hads | It can. |
| 06:38.28 | wepy | i have a feeling the latency of the internationalness of this all will ne the biggest problem |
| 06:39.07 | wepy | ok.. so basically, i can have my computer here, register it's SIP phone at my server, then people can call me@the.server.address, and it will redirect the call to sip://me@current.ip.address ? |
| 06:39.17 | wepy | or would it all go through the server? |
| 06:40.44 | wepy | bah |
| 06:40.48 | wepy | thanks for the help |
| 06:40.49 | wepy | gtg |
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| 08:06.50 | Assid | man.. this DST on polycoms driving me crazy |
| 08:09.00 | shellshark | DST-- ;) |
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| 09:00.51 | [shodan] | anyone knows the syntax for the sipura/linksys dialplans ? |
| 09:01.34 | [shodan] | I have this spa-3102 and I want that as soon as it's ringing , it picks up then forward to my * server (s extension ideally) |
| 09:02.31 | [shodan] | right now it picks up after 4 rings, gives the caller a dialtone and sends the caller to the extension he dialed |
| 09:05.01 | [shodan] | ok now it picks up after 1 second.. (just had to set PSTN Answer Delay:) |
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| 09:12.08 | iskandaar | hi |
| 09:12.50 | RayJWPi | morning iskandaar |
| 09:13.05 | iskandaar | morning channel |
| 09:13.56 | iskandaar | is it possible to configure software raid during aah\trixbox installation? |
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| 09:19.04 | RayJWPi | cu ... channel I go to channel asterisk.de |
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| 09:19.37 | X-Gen | hey freaks |
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| 10:17.51 | [shodan] | this just in, SPA-3*0* OWN :) |
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| 10:52.14 | obi1964 | hey all... wish you a wonderful day from sunny germany and hope I'm doin' nothing wrong here... I'm new to IRC :) |
| 10:52.43 | obi1964 | Have a question about asterisk and hope that somebody out there can give me a clue |
| 10:54.30 | obi1964 | Is it possible to play a soundfile instead of the standard ringtone in the time between the dial and the pickup? |
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| 10:59.53 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 11:04.47 | qdk | [shodan]: i have been looking at that box too... will it quickly detect if the internet is down and dial the locale phone instead? |
| 11:06.26 | qdk | [shodan]: and if you dial out using the phone in it, will it the dial out through VoIP if its connected and switch to PSTN if/when the internet is down? (not talking about live call failover) |
| 11:18.56 | *** join/#asterisk vaddineni (n=vaddinen@59.93.91.42) |
| 11:29.47 | *** join/#asterisk adorah (n=admin@84.94.208.102.cable.012.net.il) |
| 11:37.29 | kumamoto | anyone use the BT-200 grandstream phone? |
| 11:43.39 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 11:53.03 | *** join/#asterisk Dr__Frank (n=drf_35@adsl-70-234-221-13.dsl.rcsntx.sbcglobal.net) |
| 11:55.31 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 12:02.13 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:21.36 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 12:24.06 | Druken | icky..... snowing this morning..... |
| 12:25.46 | PakiPenguin | ah snow!! |
| 12:25.48 | PakiPenguin | send us some! |
| 12:28.54 | *** join/#asterisk apardo (n=apardo@87.217.146.43) |
| 12:34.39 | *** join/#asterisk PKv6 (i=uppal@2001:5c0:8fff:fffe:0:0:0:5ae7) |
| 12:38.47 | *** join/#asterisk prttp (i=Ftv@140.Red-83-38-109.dynamicIP.rima-tde.net) |
| 12:57.59 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 13:00.58 | *** join/#asterisk tdawgpharaoh (n=chatzill@196.205.196.1) |
| 13:00.59 | *** join/#asterisk test34- (n=test34@unaffiliated/test34) |
| 13:03.44 | *** join/#asterisk un_j (n=un_j@24-148-29-13.grn-bsr1.chi-grn.il.static.cable.rcn.com) |
| 13:04.15 | un_j | need some help with cisco 7960 |
| 13:04.36 | un_j | I will pay.....I am despreate |
| 13:06.36 | Druken | what's your problem ? |
| 13:08.13 | un_j | I got 4 old cisco |
| 13:08.22 | un_j | with sccp firmware |
| 13:08.36 | un_j | versions are around 3 |
| 13:09.22 | Druken | well, rather then paying someone to get skinny working, why not pay for the SIP firmware |
| 13:09.34 | un_j | I donwloaded the new version of firmware but theres no bin file just (P0S3-08-2-00.sb2) |
| 13:10.02 | un_j | phone request just bin file that don't exist |
| 13:10.11 | un_j | can buy the old firmware? |
| 13:10.43 | Druken | well, in that case, i'd call cisco about getting a stepup firmware... |
| 13:10.55 | Druken | i had to do that with an aastra phone i had.... |
| 13:11.12 | un_j | I will do thne |
| 13:11.19 | un_j | thank you |
| 13:11.23 | un_j | do live in chicago? |
| 13:11.34 | Druken | me? god no... ontario canada |
| 13:11.52 | un_j | I would get you a beer :-) |
| 13:11.54 | un_j | thank you |
| 13:12.25 | Druken | hehehe don't think i've really done much but give you an opinion... hehe |
| 13:12.46 | un_j | :-) |
| 13:26.00 | *** join/#asterisk burnproof (n=hellrace@210.213.244.169) |
| 13:26.02 | *** join/#asterisk nonickname123 (n=noname@85.217.194.15) |
| 13:26.55 | burnproof | hello guys, whenever i get svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk i always get revision 43628 ? |
| 13:27.19 | burnproof | 46328 rather |
| 13:27.39 | burnproof | ?? |
| 13:27.40 | benjk | un_j, the procedure is somewhat cumbersome, you probably have to go incremental |
| 13:28.14 | un_j | seems like I just need to get the older version |
| 13:28.25 | un_j | and then 8.X |
| 13:28.42 | benjk | last time I did this I had to go like SCCP -> SIP v2 ... 2.1, 2.2, 2.3, 3.0 .... 6.0, 7.5 |
| 13:28.57 | un_j | I called cisco there are closed |
| 13:31.17 | nonickname123 | http://pastebin.ca/227677 searched google, for such a problem, nothing there helps |
| 13:33.15 | un_j | cisco is the only source whre I can get those bin files from? |
| 13:43.35 | Corydon76-home | Cisco or a Cisco authorized reseller |
| 13:46.27 | un_j | thnx |
| 13:46.56 | Corydon76-home | That was odd. Usually this channel doesn't stay quiet for 10 minutes |
| 13:47.22 | ruskie | lol |
| 13:47.57 | *** join/#asterisk nin1 (n=zorman@mut38-2-82-67-67-190.fbx.proxad.net) |
| 13:48.44 | nin1 | what the best solution for fax server and asterisk |
| 13:48.50 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:48.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:01.10 | nin1 | spandsp does not seem to be ready for production shoud if use hylafax with IAXmodem |
| 14:03.22 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:05.54 | *** join/#asterisk dacleric (n=dacleric@p54823CEE.dip0.t-ipconnect.de) |
| 14:06.37 | Corydon76-home | spandsp works just fine |
| 14:06.58 | Corydon76-home | You need merely pay attention to the installation instructions |
| 14:11.12 | coppice | nin1: you do realise that iaxmodem is just a layer around spandsp, don't you? |
| 14:13.52 | *** join/#asterisk prttp (i=Ftv@140.Red-83-38-109.dynamicIP.rima-tde.net) |
| 14:16.23 | *** join/#asterisk anthonyl (n=anthonyl@ip70-185-8-160.ma.dl.cox.net) |
| 14:17.03 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:21.03 | nin1 | ok so scandsp is the base thing |
| 14:21.53 | benjk | scandsp? |
| 14:21.58 | nin1 | sorry if i ask stupid question but I'm just beginning with asterisk manager to make it work |
| 14:23.09 | nin1 | Just another question there is destar and freepbx for managing asterisk what is the advantage of each one |
| 14:23.20 | nin1 | so it help me choose one |
| 14:27.48 | *** join/#asterisk olor1n (n=olorin@rke75-4-82-234-109-130.fbx.proxad.net) |
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| 14:36.09 | *** part/#asterisk nin1 (n=zorman@mut38-2-82-67-67-190.fbx.proxad.net) |
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| 14:56.30 | robin_sz | never get your kids an ant farm. oh no. |
| 14:57.27 | Damin | Actually, if you do get an ant-farm, better get an ant-eater as well.. |
| 14:59.32 | Druken | hehehe |
| 14:59.41 | Druken | aren't ant farms sealed? |
| 15:01.03 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:01.05 | olor1n | guys you have idea why dtmf is unreliable with the ECHO_CAN_MG2 |
| 15:01.42 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 15:03.10 | *** part/#asterisk kumamoto (n=kumamoto@24-178-2-212.dhcp.ftwo.tx.charter.com) |
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| 15:24.43 | robin_sz | Druken: in theory, yes. |
| 15:24.46 | robin_sz | in theory. |
| 15:26.45 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 15:41.19 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 15:41.41 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:42.05 | *** join/#asterisk af_ (n=af@ip-170-152.sn1.eutelia.it) |
| 15:42.46 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 15:43.34 | PakiPenguin | hmms |
| 15:58.46 | *** join/#asterisk nonickname123 (n=noname@digsys34-217.pip.digsys.bg) |
| 16:02.28 | *** join/#asterisk cjmoya (n=cjmoya@201.236.246.77) |
| 16:03.56 | cjmoya | hello |
| 16:04.18 | lennard | it probably isnt any use asking about bugs in chan_modem, is it? what with it being obsolete and all that |
| 16:05.33 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:07.25 | benjk | chan_modem is of very limited use, it's half duplex only |
| 16:07.54 | nonickname123 | http://pastebin.ca/227790 |
| 16:07.59 | nonickname123 | anynone any clue |
| 16:12.57 | cjmoya | who can help me with sip agents? |
| 16:20.11 | un_j | how to reinitiate the upgrade process on 7960 cisco? (I already got the sip software (v3) on it and want to upgrade tov 4) |
| 16:21.58 | JunK-Y | nonickname123: make sure you zapata.conf is configured well and you have channeltype zap |
| 16:22.22 | *** join/#asterisk fbnts (n=thomas@87.253.4.10) |
| 16:23.02 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:23.49 | fbnts | hi, I've recently set up OH323 with asterisk. Its all working fine except I am not using a gatekeeper. I have set gatekeeper = DISABLE in h323.conf |
| 16:24.00 | fbnts | but I am still getting: chan_oh323.c:4249 oh323_gk_check: Gatekeeper discovery failed. in the console |
| 16:24.31 | fbnts | any ideas? |
| 16:34.56 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
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| 17:34.17 | *** join/#asterisk mmarcos_ (n=uutytuy@r190-64-27-230.dialup.adsl.anteldata.net.uy) |
| 17:34.40 | mmarcos_ | how can i set the remote console to display colors? |
| 17:34.48 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 17:38.48 | *** join/#asterisk fholmes (n=fholmes@cpe-68-201-200-58.houston.res.rr.com) |
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| 17:55.26 | JunK-Y | mmarcos_: just set ur terminal correctly. |
| 17:57.05 | mmarcos_ | hmm any particular thing you remember using putty? |
| 17:58.15 | *** join/#asterisk Math` (n=privmath@modemcable238.43-82-70.mc.videotron.ca) |
| 18:03.36 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.15) |
| 18:04.07 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
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| 18:08.08 | pifiu | mornin everyone |
| 18:09.51 | *** join/#asterisk af_ (n=af@ip-170-152.sn1.eutelia.it) |
| 18:11.36 | *** join/#asterisk rene1 (n=rene1@189.140.214.199) |
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| 18:26.57 | *** join/#asterisk packetman (n=trevor@d57-225-189.home.cgocable.net) |
| 18:27.41 | packetman | Hi Guys, anyone here? |
| 18:29.14 | DarKnesS_WolF | yes |
| 18:29.58 | packetman | I have a strange issue. Running asterisk 1.2.12 and freePBX. Using a TDM400 2 fxo and 2 fxs. When calling out or in is # is pressed during a conversation the line goes dead air |
| 18:30.18 | packetman | I have a strange issue. Running asterisk 1.2.12 and freePBX. Using a TDM400 2 fxo and 2 fxs. When calling out or in if # is pressed during a conversation the line goes dead air |
| 18:30.27 | packetman | sorry grammer error |
| 18:30.44 | rene1 | any news of pci-express hardware from diguim @ astricon? |
| 18:30.58 | packetman | The dead air is on both users. |
| 18:31.05 | packetman | caller and callee |
| 18:31.24 | packetman | Asterisk CLI shows nothing when the # is pressed |
| 18:32.28 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 18:32.28 | packetman | I've changed my features.conf to ## for blind transfer, *2 for attended transfer, and *1 for on demand recording so it could not be the system detecting # for blind transfer as ## is now working |
| 18:32.31 | DarKnesS_WolF | packetman: check the feuter.conf file see what dose # do |
| 18:32.45 | packetman | yup |
| 18:32.50 | packetman | look above |
| 18:33.13 | packetman | Its weird, cause the asterisk CLI shows nothing |
| 18:34.55 | packetman | say if I call out to my cellphone. We are connected. I push # on my asterisk extention. The line goes dead air on both my extention and my cellphone, when I hang up my asterisk extention, on my cell I hear a quick blip of a tone |
| 18:49.11 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-1-243.bflony.east.verizon.net) |
| 18:49.20 | SuPrSluG | hello all |
| 18:51.47 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 18:54.28 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 18:54.30 | TheCops | Hi |
| 18:55.10 | TheCops | someone know how to change the name "asterisk" as unknown caller ID? |
| 18:55.27 | DasTech | in the cid fileds |
| 18:55.32 | DasTech | you add for each user |
| 18:55.40 | DasTech | and on outbound routes |
| 18:56.00 | TheCops | How from a Zap channel^ |
| 18:56.02 | DasTech | and it depends on if your provider passes full cid or not |
| 18:56.29 | DasTech | look inthe zaptel.conf |
| 18:56.29 | TheCops | this is doing this when it is a confidential caller ID from my carrier |
| 18:56.30 | TheCops | ok |
| 18:56.46 | DasTech | they block the cid then |
| 18:56.55 | DasTech | there is no way to chabge this |
| 18:57.25 | DasTech | if the carrier is blocking it there is no way to change it] |
| 18:57.44 | TheCops | I guess this is not my carrier that is sending asterisk on my phone |
| 18:58.06 | Qwell | TheCops: never know |
| 18:58.11 | DasTech | then look in zaptel.conf |
| 18:58.12 | TheCops | lol |
| 18:58.23 | DasTech | cop this copper |
| 18:58.45 | DasTech | you will never get me see it like this see I am better then you see |
| 18:59.05 | DasTech | I run this town copper so packup and get out |
| 18:59.13 | DasTech | you see |
| 18:59.22 | DasTech | <== Buggs Malone |
| 19:00.36 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 19:00.40 | DasTech | get pig your not wanted here see |
| 19:00.43 | DasTech | lol |
| 19:01.10 | DasTech | the fuzz is not welcome here see. this is a free and open source channel see |
| 19:01.12 | DasTech | lol |
| 19:02.05 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:02.27 | *** join/#asterisk tdawgpharaoh (n=chatzill@196.205.196.1) |
| 19:03.27 | DasTech | sorry got to much cleep jonzing now |
| 19:03.39 | DasTech | cleep/sleep |
| 19:18.22 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 19:18.44 | teknoprep | i need an ecuador did... anyone know where i can get one? |
| 19:19.02 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 19:22.48 | *** join/#asterisk MattB2 (i=user@c-68-59-181-202.hsd1.tn.comcast.net) |
| 19:22.56 | MattB2 | hi all, happy sunday! |
| 19:23.03 | MattB2 | got a dev question |
| 19:23.21 | MattB2 | triny to mod app_record to play a beep 10s and 5s before maxduration |
| 19:24.08 | MattB2 | taking into account i know C fairly well but nothing about developing in asterisk, i placed a simple ast_streamfile and ast_waitstream into the recording loop |
| 19:24.26 | MattB2 | this does the beeps fine but drops a small amount of the recording |
| 19:24.37 | MattB2 | is there some kind of asycn streamfile i can use to prevent this? |
| 19:25.22 | MattB2 | 257 users and not a single person speaking ;) |
| 19:27.55 | pifiu | anyone know a bit more about trunking? |
| 19:41.02 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 19:48.20 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:48.20 | *** mode/#asterisk [+o mog] by ChanServ |
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| 20:01.41 | *** join/#asterisk sundancer (n=akamai@esy.perkmandlc.org) |
| 20:01.52 | sundancer | Hello. Anyone using freeworlddialup? |
| 20:03.01 | Nivex | no, everyone switched to gizmo |
| 20:03.19 | sundancer | Why? |
| 20:05.00 | Qwell | because they're sheep :p |
| 20:05.16 | sahafeez | sheeple |
| 20:05.22 | sundancer | I guess its just another skypealike stupidity |
| 20:05.48 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:07.17 | docelmo | Anyone have a sec to answer a question on a TDM400P? |
| 20:07.24 | Qwell | docelmo: can try |
| 20:07.42 | docelmo | I am trying to compile Zaptel and its bitching about something not sure what.. |
| 20:07.48 | docelmo | I compiled libpri |
| 20:08.01 | docelmo | I set the link to the build directory of kernel 2.6 |
| 20:09.54 | docelmo | http://pastebin.ca/227996 |
| 20:10.00 | docelmo | this is a copy of the error |
| 20:12.02 | Qwell | let me guess |
| 20:12.04 | Qwell | ~centosbug |
| 20:12.09 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:12.25 | docelmo | yep |
| 20:12.35 | Qwell | run that, try again |
| 20:13.06 | docelmo | Can I run that anywhere in the shell? |
| 20:13.30 | Qwell | yeah, it's full path + wildcarded |
| 20:13.40 | *** join/#asterisk s0lid (n=jlq@210.213.199.28) |
| 20:14.13 | docelmo | yep that fixed it.. Why not do a check on the OS version and fix it automatically.. |
| 20:14.21 | docelmo | thanks dude |
| 20:14.22 | Qwell | because it's not really possible |
| 20:14.40 | Qwell | though, I guess it *MIGHT* be possible to check in the configure script |
| 20:14.48 | docelmo | yep |
| 20:14.54 | *** join/#asterisk beighto (n=Kry5ta1@adsl-75-8-225-140.dsl.scrm01.sbcglobal.net) |
| 20:14.59 | Qwell | for the struct itself.. not the OS/kernel version |
| 20:15.00 | docelmo | just check for redhat-release |
| 20:15.06 | docelmo | if there check and see if its centos |
| 20:15.44 | Qwell | all it really needs, is to compile a simple app that uses rwlock, and see if it compiles. If not...tell them to fix it |
| 20:18.00 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 20:22.30 | beighto | looking for a way to ignore all # keypresses in my dialplan, any ideas besides removing the trailing digit? |
| 20:25.25 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 20:32.11 | *** join/#asterisk backblue (n=moo@87-196-105-224.net.novis.pt) |
| 20:38.08 | *** join/#asterisk clive- (n=pirch@dsl-145-45-30.telkomadsl.co.za) |
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| 20:50.19 | packetman | <beighto> Are you having a problem with #? |
| 20:53.23 | beighto | packetman yes |
| 20:53.29 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 20:53.36 | packetman | What is the symptoms? |
| 20:54.20 | clive- | has anyone heard of this dsp board for asterisk: SigC5561 VoIP PCI card for Asterisk IP PBX server ?? |
| 20:55.23 | beighto | packetman When someone punches in a bunch of numbers and presses # my dialplan thinks the # key is just another digit. Where their intention was to press # to confirm the digits pressed and go on to the next context. I want to be able to ignore that # symbol. |
| 20:55.55 | packetman | Cause I am having a problem when if a caller presses # they receive dead air? |
| 20:56.06 | packetman | What is the settings of your feature.conf ? |
| 20:56.36 | packetman | It may be set so # = blindxfer |
| 20:57.55 | beighto | packetman: I tried that, and I don't recall it working correctly. My dialplan is set up so it gets _X. for the digits pressed and even with that set in features.conf it still adds the # to the end |
| 20:59.08 | packetman | <beighto> What is the setting there? blindxfer => # ? |
| 20:59.24 | beighto | yes, blindxfer => # |
| 20:59.33 | packetman | You shold change that |
| 20:59.40 | packetman | to something else |
| 20:59.44 | packetman | use ## |
| 20:59.57 | beighto | blindxfer => ##? |
| 21:00.01 | packetman | or comment it out buy putting ; infront |
| 21:00.03 | packetman | yes |
| 21:00.11 | packetman | this will allow # to be used |
| 21:00.26 | beighto | I will give that a try right now, thanks |
| 21:00.29 | packetman | as a normal DTMF tone passed to the runk |
| 21:00.36 | packetman | trunk |
| 21:01.26 | packetman | else asterisk will see # and not pass it to the trunk |
| 21:01.44 | packetman | you could also turn it off by commenting it out by using ; infront on blindxfer => # but this will mean your extention transfers features will not work |
| 21:01.46 | beighto | still, when I press something like 123# it registers as 123# |
| 21:02.14 | packetman | why is it you want to strip #? |
| 21:02.49 | packetman | if you do this how will your users access automated telephone systems IVR's that require #? |
| 21:03.12 | packetman | like "please enter your banking information followed by the # key" |
| 21:03.52 | beighto | it is for call conferencing, some people like to add # when they are done dialing a conference number, but when they do that it adds # to the end of the conference number and fails. There are no automated IVRs on this server, it is strictly call conferencing |
| 21:04.20 | packetman | ah |
| 21:04.32 | packetman | hmm |
| 21:04.38 | packetman | can't help ya there |
| 21:05.07 | beighto | the workaround I thought might work would be adding a _X# and stripping the last digit, then add a _XX#, strip the last digit and so on |
| 21:05.28 | packetman | hmm |
| 21:05.52 | packetman | I'm just reading lots about dialplan writing |
| 21:05.59 | packetman | not quite there yet |
| 21:06.28 | packetman | that would work as long as you cover all lenghts of numbers |
| 21:06.48 | packetman | there should be a cleaner way of doing it |
| 21:06.54 | beighto | yeah... I figure I could go 10 digits out and be good in 99% of all the conference numbers that might be entered |
| 21:06.57 | packetman | ah I just read something in my dialplan learning |
| 21:07.13 | packetman | hold up let me see if I can pull the example |
| 21:08.01 | packetman | I found something in the pattern matching area |
| 21:08.11 | packetman | let me read it and see if it will work for you |
| 21:08.26 | beighto | ok |
| 21:10.03 | packetman | You can use a period |
| 21:10.06 | packetman | . |
| 21:10.11 | packetman | wildcard |
| 21:10.20 | packetman | let me get you the link |
| 21:10.22 | *** part/#asterisk clive- (n=pirch@dsl-145-45-30.telkomadsl.co.za) |
| 21:10.24 | packetman | to the example |
| 21:10.43 | beighto | I am using the ., but I can't add a # after the . and strip the last digit |
| 21:11.15 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 21:11.38 | packetman | _X.# does snot work? |
| 21:11.39 | Strom_C | how about just using variables and substrings? |
| 21:11.51 | hads | Yeah, . is everything so characters after that won't work. |
| 21:12.07 | packetman | you can use [0-9] also |
| 21:12.20 | Strom_C | [0-9] == X |
| 21:12.21 | packetman | let me check the example |
| 21:12.41 | *** join/#asterisk RoyK (n=roy@ti211310a080-2264.bb.online.no) |
| 21:13.41 | hads | Use Strom_C's idea. |
| 21:13.54 | beighto | I would need a [0-9] that would apply to the whole string of digits entered that way |
| 21:14.12 | hads | Retrieve last digit, check if it's a # and remove it. |
| 21:14.13 | packetman | ya I just saw that It only allpies to the single |
| 21:14.19 | beighto | because a conference number could be 123 or 1234567890 in which case I would need a context of XXX and XXXXXXXXXX |
| 21:14.25 | packetman | ya use varables |
| 21:14.49 | Strom_C | or even better, use the Read() application, which terminates input with a # |
| 21:15.19 | beighto | Strom_C: now that might work |
| 21:15.27 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:16.13 | packetman | ya that would catch that # and put it in its place. Terminate :) |
| 21:16.53 | beighto | thats the answer! Thanks guys! |
| 21:17.03 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 21:20.36 | *** join/#asterisk emx961 (n=adrian@OL182-14.fibertel.com.ar) |
| 21:20.48 | packetman | I have a strange issue. Running asterisk 1.2.12 and freePBX. Using a TDM400 2 fxo and 2 fxs. When calling out or in if # is pressed during a conversation the line goes dead air |
| 21:21.16 | packetman | I've changed my features.conf to ## for blind transfer, *2 for attended transfer, and *1 for on demand recording so it could not be the system detecting # for blind transfer as ## is now working |
| 21:21.32 | packetman | Asterisk CLI shows nothing when the # is pressed |
| 21:21.58 | packetman | say if I call out to my cellphone. We are connected. I push # on my asterisk extention. The line goes dead air on both my extention and my cellphone, when I hang up my asterisk extention, on my cell I hear a quick blip of a tone |
| 21:24.05 | beighto | could there be a record option in there somewhere? |
| 21:25.39 | Qwell | <packetman> _X.# does snot work? |
| 21:25.49 | Qwell | Just wanted to clear this question up.. Anything after . is ignored. |
| 21:26.01 | packetman | thanks Qwell |
| 21:26.09 | Qwell | . and ! force an early return |
| 21:26.25 | hads | < hads> Yeah, . is everything so characters after that won't work. |
| 21:26.40 | hads | Although, yes, yours is clearer :) |
| 21:27.13 | Qwell | I'm still not entirely clear on what ! does though... |
| 21:27.20 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:27.49 | Qwell | ~lart Qwell |
| 21:32.12 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:36.07 | *** join/#asterisk aronofsky (n=aronofsk@AAmiens-157-1-102-116.w86-208.abo.wanadoo.fr) |
| 21:37.01 | aronofsky | hi , i got an asterisk server and i was wondering if there was a good apps that permits to use a computer as a virtual ip phone ( with mic and headphone ) |
| 21:37.32 | Qwell | aronofsky: sure, that would be called a softphone.. there are a bunch available |
| 21:37.34 | EyeCue | what os is the 'computer' ? |
| 21:37.35 | Qwell | ~softphones |
| 21:37.38 | Qwell | ~softphone |
| 21:37.40 | jbot | something that should be drug out into the street and shot |
| 21:37.40 | ariel_ | packetman, there is a channel here for help with freepbx it's #freepbx depending on the version your using there have been lots of updates an fixes in the pass few weeks for there dial rules. |
| 21:38.08 | aronofsky | ok ,and whate are the best free solutions ? |
| 21:38.20 | EyeCue | what os? |
| 21:38.22 | Qwell | aronofsky: one that people like is called x-lite. It isn't open source (it IS free however), but it's good |
| 21:38.23 | packetman | Hmm so It might be something freepbx did to the dialplan |
| 21:38.49 | Qwell | packetman: wouldn't doubt it, unfortunately. and unless you go through the whole thing (it's a rats nest), there's no way to tell for sure |
| 21:38.55 | ariel_ | packetman, yes they use allot of macro's and checks before dialing |
| 21:38.58 | EyeCue | aronofsky, my favourites for elegance and simplicity with simple and somewhat refined UI's are idefisk + miranda plus iax plugin |
| 21:39.12 | EyeCue | + = or. |
| 21:39.13 | Qwell | aronofsky: EyeCue: idefisk is really good too |
| 21:39.30 | Qwell | I also like iaxcomm, because it actually is open source |
| 21:39.35 | packetman | k I will ask in freePBX, ,aybe they can tell me hwo to open up some debugging so I can see what is happening when I press # |
| 21:39.38 | EyeCue | yeh, the ui needs some usability work, but its clean |
| 21:39.51 | ariel_ | set verbose 9 |
| 21:40.15 | EyeCue | as far as 'just works' is concerned, idefisk or miranda + iax plugin (a good friend dev'd this btw) |
| 21:40.17 | packetman | in cli, K I will try that, when it was at 5 it did not tell me anything when I press # |
| 21:42.16 | packetman | Any good linux based IAX softphones. I'm using unbuntu |
| 21:42.18 | Qwell | umm |
| 21:42.21 | Qwell | what time is it? |
| 21:42.31 | Qwell | all the clocks in my house are out of sync, heh |
| 21:42.34 | ariel_ | 4:42pm here EST |
| 21:42.40 | packetman | what part of the world do you live in? |
| 21:42.44 | Qwell | so, wtf...why didn't my phone change? |
| 21:42.50 | packetman | ya shows 4:42 |
| 21:42.53 | packetman | est |
| 21:43.09 | packetman | IP phone or cell |
| 21:43.16 | Qwell | cell, surprisingly |
| 21:43.22 | Qwell | I figured it would definitely change |
| 21:43.25 | packetman | cell, has a option recive time update from network |
| 21:43.33 | Qwell | yeah, and mine does when I switch timezones |
| 21:43.36 | packetman | atleast mine does |
| 21:43.38 | ariel_ | packetman, there is a xlite linux version that works well for sip. |
| 21:43.49 | Qwell | you know what...no |
| 21:43.55 | Qwell | I'm inclined to believe pool.ntp.org |
| 21:44.06 | packetman | <ariel_> no IAX though eh. Ah I'll google around and see whats out there |
| 21:44.23 | Qwell | So, this begs the question... What timezone is detroit? |
| 21:44.31 | packetman | est? |
| 21:44.37 | ariel_ | sip makes a far better protocal then iax2 |
| 21:44.37 | packetman | I think |
| 21:44.50 | Qwell | So, I obviously picked the wrong timezone ;) |
| 21:45.03 | packetman | ya but NAT is a wh*re with sip |
| 21:45.13 | Qwell | man, when you have 3 phones telling you different things... |
| 21:45.14 | ariel_ | packetman, yes that is correct |
| 21:45.16 | lennard | it is? |
| 21:45.20 | Qwell | erm, 3 clocks |
| 21:45.31 | packetman | So still sip is better that IAX2? |
| 21:45.40 | ariel_ | yes |
| 21:45.54 | pifiu | hey ariel! |
| 21:45.55 | pifiu | long time |
| 21:45.55 | packetman | what are some of the bennifits |
| 21:46.00 | ariel_ | just think of this everything goes via the same port via iax2 |
| 21:46.03 | ariel_ | pifiu, hi |
| 21:46.23 | packetman | same port yes better for NAT |
| 21:46.34 | ariel_ | yes better for not but not too scallable |
| 21:46.49 | packetman | whats its scalibility? |
| 21:46.54 | packetman | max? |
| 21:47.02 | packetman | for iax |
| 21:47.13 | *** join/#asterisk FrdPrefct (i=adamisp@whaddu.com) |
| 21:47.15 | FrdPrefct | Hello |
| 21:47.26 | FrdPrefct | I'm having problems getting zaptel to work in ubuntu |
| 21:47.30 | FrdPrefct | can someone please help? |
| 21:47.47 | ariel_ | packetman, it can handle allot but after 8 or 9 calls it starts to have issues with sound and jitters |
| 21:48.14 | packetman | I guess with sip as long as you use proper SIP and RTP port forwards in your router, and exterip and localnet settings, then a stun server for the client on the other side all should be whell |
| 21:48.16 | ariel_ | ubuntu makes a great desktop os I use Kubuntu. But for servers...humm don't use it at all. |
| 21:48.33 | ariel_ | packetman, yep |
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| 21:49.21 | packetman | I guess me just being a somewhat NooB until I really understood the settings, which are not that hard, I assumed IAX2 was better cause it was easier to config |
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| 21:50.18 | ariel_ | packetman, it's easyer yes but you need to know it's limitations I use it for voicepulse since I only have one inbound number with them and it can handle the 4 calls without issues. But for the rest it's sip all the way. |
| 21:51.17 | packetman | I still get stupid oneway audio issues and DTMF problems with sip, even with all the proper settings. I assume I'm still missing some sometimes. If I'm missing just one I guess things won't work. As I get better at this I'm sure I will change my mind that sip is better |
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| 22:10.49 | dovid | morning all |
| 22:13.34 | shellshark | yo |
| 22:14.38 | saftsack | dovid, wherer do you live that you can say morning? :> japan? |
| 22:14.44 | dovid | Israel |
| 22:14.52 | dovid | 0015 here now |
| 22:15.05 | saftsack | humm, ok ;) |
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| 22:35.53 | qdk | Qwell aronofsky: one that people like is called x-lite. It isn't open source (it IS free however), but it's good <- i beg to differ, so could you tell me whats good about it? |
| 22:36.36 | Qwell | qdk: I don't personally like it, so I can't comment |
| 22:40.02 | *** join/#asterisk grabowski (i=grabowsk@i.use.efnut.com) |
| 22:40.12 | qdk | Qwell: ok, fair enough. I dislike it as well due to its high latency. |
| 22:40.42 | grabowski | Is there a way to test if my timing source (ztdummy) is working? Or just try a meetme extension and watch for errors? |
| 22:40.50 | qdk | ariel_: what was that about IAX being crap with more than 10 concurrent calls? |
| 22:42.05 | mog | qdk, What are you talking about? |
| 22:42.48 | qdk | mog: Depends. |
| 22:46.30 | qdk | <PROTECTED> |
| 22:55.20 | docelmo | Anyone know how to enable H.264 in asterisk for Video phones? |
| 23:05.28 | *** part/#asterisk grabowski (i=grabowsk@i.use.efnut.com) |
| 23:14.34 | Damin | docelmo: Nope, but I can tell you that func_curl with AEL2 has solved 99% of my problems! :) |
| 23:20.47 | Qwell | Damin: example of problems? |
| 23:21.03 | Qwell | I'm genuinely curious.. |
| 23:22.22 | brookshire | qwell: i believe it had to do with sharing voicemail.conf |
| 23:22.34 | Qwell | That was Juggie :D |
| 23:23.46 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 23:23.49 | brookshire | oh yah |
| 23:24.13 | brookshire | qwell: i'm at bumpers :) |
| 23:24.24 | brookshire | *drinks his beer* |
| 23:24.35 | brookshire | damnit.. twisted just got an 8ball on the break |
| 23:24.40 | brookshire | that means i lose |
| 23:24.42 | Qwell | loser :p |
| 23:24.52 | brookshire | Qwell: come out here (this is twisted) |
| 23:24.59 | Qwell | meh |
| 23:25.03 | brookshire | dude |
| 23:25.07 | brookshire | my bday was saturday |
| 23:25.07 | Qwell | I'm tired :p |
| 23:25.09 | brookshire | come out for a little bit |
| 23:25.12 | brookshire | (back to matt) |
| 23:25.22 | lennard | now I REALLY don't understant why they removed chan_modem and chan_modem_i4l |
| 23:25.27 | lennard | it works perfectly fine |
| 23:25.40 | lennard | well, after fixing-or-maybe-breaking a few things |
| 23:25.56 | PakiPenguin | :) |
| 23:26.28 | brookshire | Qwell: you don't have to buy anything, just come out for a few |
| 23:26.34 | brookshire | and bring the wife i haven't met yet |
| 23:27.11 | Qwell | brookshire: You have |
| 23:27.15 | Qwell | well, "you" |
| 23:27.27 | brookshire | not twisted |
| 23:27.28 | brookshire | hah |
| 23:27.31 | brookshire | he typed that |
| 23:27.56 | twistedbrook | lol |
| 23:27.57 | twistedbrook | sorry |
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| 23:31.20 | kink0 | hi |
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| 23:36.12 | linagee | this is what cox cable uses for their "cox digital telephone": http://www.arrisi.com/product_catalog/_docs/_specsheet/060510_Touchstone_Telephony_Modem_TM502G.pdf |
| 23:36.19 | linagee | :) |
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| 23:40.09 | ftexcom | hi |
| 23:40.18 | ftexcom | I got this error when I try to start asterisk |
| 23:40.24 | ftexcom | ERROR[3439] chan_zap.c: Unable to open channel 2: No such device or address |
| 23:40.24 | ftexcom | <ftexcom> here = 0, tmp->channel = 2, channel = 2 |
| 23:40.27 | ftexcom | any ideas? |
| 23:44.29 | ftexcom | you talk a lot |
| 23:45.03 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 23:45.17 | JT | and you're impatient a lot, your point? :) |
| 23:45.51 | docelmo | What no one know how to setup GXV3000? |
| 23:46.16 | ftexcom | JT my point..asterisk dies after that message |
| 23:47.18 | benjk | asterisk == dead man walking |
| 23:48.21 | JT | ftexcom: you didn't provide enough information for anyone to do anything than to ask for more information |
| 23:48.30 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 23:49.37 | ftexcom | I do believe the problem comes from zapata.conf...but I have no idea what's that "channel 2" |
| 23:50.07 | *** join/#asterisk AtomicStack (n=matt@malibu.ecomtel.com.au) |
| 23:50.55 | JT | ftexcom: is zaptel.conf setup, did you run ztcfg? |
| 23:51.44 | ftexcom | no, it's not zaptel.conf...its /etc/asterisk/zapata.conf. Yes I run ztcfg |
| 23:51.47 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 23:52.20 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 23:52.30 | THX2000 | I'm having trouble dialing out on my zap channels, I'm thinking its starting to dial the number too soon, so sometimes it'll go through and sometimes it'll say stuff like "you need to dial one before calling this number"... |
| 23:52.32 | JT | ftexcom: are you trying to correct me? |
| 23:52.36 | THX2000 | Is there a way to delay it from dialing? |
| 23:52.50 | ftexcom | are you trying to correct me? what kind of question is that? |
| 23:52.59 | hads | LOL |
| 23:53.19 | JT | ftexcom: to get a Zap channel working with Asterisk you require both /etc/zaptel.conf and /etc/asterisk/zapata.conf configured correctly |
| 23:53.31 | JT | and you need to run ztcfg before starting asterisk |
| 23:54.45 | JT | ftexcom: your remark seemed to imply that i was wrong to bring up zaptel.conf |
| 23:54.46 | ftexcom | I can hardly start asterisk if it dies all the time |
| 23:55.05 | JT | so i will ask one more time |
| 23:55.17 | JT | have you configured /etc/zaptel.conf correctly? |
| 23:55.50 | ftexcom | JT genzaptelconf did the job for me. Everything was working fine. I added some musiconhold, installed mpg123, then..decided to reboot, and then all went to hell |
| 23:55.59 | hads | Ug. |
| 23:56.18 | hads | Maybe genzaptelconf didn't do the job for you... |
| 23:56.36 | hads | And, why mpg123? |
| 23:57.00 | hads | files mode > mpg123 |
| 23:57.25 | ftexcom | i was trying to make streaming musiconhold |
| 23:58.17 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |