| 00:05.50 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 00:06.53 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 00:07.21 | kombi | I read that you can use icecast for moh, but can you also stream a phone convo to icecast? |
| 00:07.59 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 00:13.08 | kombi | ok, found that you can. do you need ices by itself or just the module? |
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| 00:21.59 | *** part/#asterisk schorpp (n=schorpp@HSI-KBW-091-089-004-107.hsi2.kabelbw.de) |
| 00:22.06 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 00:23.32 | lee_is_me | hey all. If an extension has a MeetMe() followed by another priority Goto(), shouldn't call flow back to that GoTo() instead of hanging up? |
| 00:23.54 | lee_is_me | when the last "marked" caller leaves? |
| 00:37.04 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 00:37.54 | lee_is_me | never mind, got it...dumb |
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| 00:43.12 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 00:45.25 | Jon335 | Anyone have experience with Allworx IP PBXes? |
| 00:45.46 | Jon335 | I'm trying to connect one to Asterisk over the internet |
| 00:45.53 | Jon335 | I got SIP working, but they aren't transferring audio/RTP |
| 00:46.47 | *** join/#asterisk VoIPMasta (n=dopeshow@201.139.156.12.cable.dyn.cableonline.com.mx) |
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| 00:49.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:51.38 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 00:54.17 | VoIPMasta | Hello, I'm trying to build a small app that will show calls placed by a specific SIP peer/friend in "real-time", showing destination and call length (duration)... any ideas on where to start? |
| 00:54.43 | kolian123 | hi TK! |
| 00:54.50 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:54.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:54.56 | *** join/#asterisk r1sk (n=chatzill@69.12.242.130) |
| 00:55.05 | VoIPMasta | First I thought about doing a "show channels" and parsing the output, but it could cause some load on my small asterisk box |
| 00:55.15 | kolian123 | just wondering is attented transfer broken in 1.2.18 |
| 00:55.16 | kolian123 | ? |
| 00:55.43 | kolian123 | res_features.c:844 builtin_atxfer: Did not read data |
| 00:55.55 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 00:56.09 | kolian123 | this is for an inbound call on IAX2 and then ringing a station on SIP/ |
| 00:56.17 | kolian123 | station would try to transfer |
| 00:57.33 | kolian123 | VoipMasta, use Manager API |
| 00:57.58 | VoIPMasta | but it still has to query the asterisk server every X seconds... right? |
| 00:58.04 | kolian123 | no |
| 00:58.19 | kolian123 | it will send events once your application connect |
| 00:58.39 | kolian123 | there are applications available that do what you want though |
| 00:59.09 | VoIPMasta | do you have the url to any of them? |
| 01:00.07 | kolian123 | flash operator panel |
| 01:00.11 | kolian123 | just try google |
| 01:00.33 | VoIPMasta | ok I will give it a try |
| 01:00.35 | VoIPMasta | thank you very much |
| 01:00.54 | kolian123 | http://www.intuitivecreations.com/contributions/AMS/ama.php |
| 01:01.00 | kolian123 | that's another one, java |
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| 01:03.10 | r1sk | hi all... i am having a problem with zap failing to start after updating asterisk-addons, but if i manually unload and load the drivers it will start... any ideas? |
| 01:03.28 | r1sk | asterisk and zaptel 1.2.18 |
| 01:03.54 | *** part/#asterisk barretj (n=barretj@unaffiliated/barretj) |
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| 01:11.34 | ReDNeQ- | is tehre a reason tx and rxgains need to be set so high? |
| 01:11.39 | ReDNeQ- | is there somtehing im missing |
| 01:12.03 | Nugget | no gain, some pain. |
| 01:12.12 | ReDNeQ- | hah |
| 01:12.36 | ReDNeQ- | is rxgain=20.0 and txgain=10.0 normal? |
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| 01:17.36 | Hmmhesays | my heart inside is breaking, this shits gone way to far |
| 01:18.35 | ReDNeQ- | whats the range on the rxgain txgain |
| 01:20.21 | *** join/#asterisk Anonyblessed (n=chatzill@host-53.216-16-29.iw.net) |
| 01:20.51 | ManxPower | ReDNeQ: I've seen an install that need rxgain of something like 10 |
| 01:21.03 | ReDNeQ- | yes i had rxgain=20 |
| 01:21.06 | ReDNeQ- | txgain=10 |
| 01:21.12 | ReDNeQ- | but i have changed it |
| 01:21.20 | *** join/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com) |
| 01:21.22 | Anonyblessed | does freebsd support the Wildcard TE120P card? |
| 01:21.49 | Anonyblessed | zaptel installed but doesn't detect it |
| 01:22.08 | l2cache | Hello, I have phones that are on the same network as my asterisk box and they are registered, however, i get a 401 not authorized from the phones that are on NAT (still same location)...any ideas...I have tried nat=no and nat =yes |
| 01:25.24 | *** join/#asterisk SavageOne (n=savageon@206.53.71.224) |
| 01:25.26 | SavageOne | hello all |
| 01:25.32 | SavageOne | I'm having a zaptel issue |
| 01:25.46 | SavageOne | I'm did a clean make install of the current version, released on the 8th |
| 01:25.52 | SavageOne | but I can't get it to see my modules |
| 01:26.11 | SavageOne | it was working before but I kept having to re-run the genzaptelconf setup to kick it in the ass |
| 01:26.15 | SavageOne | and now that doesn't work at all |
| 01:26.41 | ReDNeQ | music on hold sounds stuttery |
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| 01:45.33 | Anonyblessed | does anyone have any recommendations for OS of choice using asterisk/Wildcard TE120P - T1 ? |
| 01:45.53 | Qwell | Linux |
| 01:46.31 | SavageOne | qwell: rofl |
| 01:46.36 | Anonyblessed | thx I'm amending my question to distro |
| 01:46.50 | Qwell | any you're familiar with |
| 01:47.14 | Anonyblessed | freebsd which apparently doesn't work with my card |
| 01:47.26 | r1sk | qwell suggest centos 4.4 which is rhel 4.4 |
| 01:47.26 | Qwell | freebsd isn't a distro |
| 01:47.32 | Qwell | no it isn't |
| 01:47.41 | Qwell | and no, I don't suggest centos |
| 01:47.41 | wunderkin | especially not of linux |
| 01:47.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:50.27 | SavageOne | I personally use centos |
| 01:50.30 | SavageOne | I dig it |
| 01:50.34 | SavageOne | ubuntu works I hear |
| 01:50.46 | Anonyblessed | I understand freebsd doesn't use linux's kernel making it not a linux distro. I've been using FBSD for quite some time but haven't delved much into linux. It's structure, um is in some ways much different. Perhaps I'm not asking the right question. Which linux distro would be the easiest.... |
| 01:51.17 | Qwell | the one you are already familiar with would be the easiest |
| 01:52.36 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
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| 01:55.53 | Anonyblessed | is astlinux maintained well? |
| 01:56.11 | Juggie | astlinux is for embedded devices i believe |
| 01:56.57 | Juggie | it will work on a pc, but it may not be the most advisible to use |
| 01:57.03 | Juggie | i would suggest CENTOS 4.5 or 5.0 |
| 01:57.57 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 01:58.01 | mmlj4 | Anonyblessed: if you're a BSD user, then slackware is probably right up your alley |
| 01:58.22 | mmlj4 | it uses BSD-style init, not SysV |
| 01:58.47 | Juggie | touche |
| 01:59.02 | Juggie | i find rhel/centos less messy |
| 01:59.05 | Juggie | but thats just my 2cents |
| 01:59.27 | Qwell | bsd == messy, so yeah |
| 01:59.27 | Qwell | :p |
| 02:00.28 | Anonyblessed | Juggie: appreciate the feedback mmlj4: thank you Qwell: um not so much |
| 02:00.31 | Juggie | does slack have nice lil utils for managing services like centos does? |
| 02:00.35 | Juggie | eg, chkconfig |
| 02:01.23 | *** join/#asterisk ledoktre (n=ledoktre@nhtn-02-0108.dsl.iowatelecom.net) |
| 02:01.26 | ledoktre | hello ! |
| 02:02.34 | *** part/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 02:03.42 | ledoktre | anyone here have any experience on the app "meetme"? |
| 02:06.41 | ledoktre | trying to install the dumb thing and it is being stubborn . :-( I have ztdummy loaded, and I am compiling - whats a good few steps to try to troubleshoot? |
| 02:08.44 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 02:09.49 | ledoktre | hello? |
| 02:10.08 | SavageOne | try freepbx channel |
| 02:10.17 | SavageOne | never worked w/ meetme personally |
| 02:11.00 | _DAW | ledoktre: You never said what is isn't doing exactly. |
| 02:14.11 | *** join/#asterisk saftsack (n=saftsack@pD9E044CC.dip.t-dialin.net) |
| 02:14.44 | ledoktre | _DAW : lsmod shows ztdummy and zaptel both, trying to do a standard ./configure;make;make install - which works fine. when trying to use MeetMe, it tells me it not a valid app. |
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| 02:23.58 | *** join/#asterisk davidcsi (n=davidcsi@190.84.250.60) |
| 02:24.27 | davidcsi | hello all: Anyone knows what perl AGI exec("DIAL","WHATEVER") returns? |
| 02:24.45 | davidcsi | sometimes its 0 or 1 |
| 02:25.42 | davidcsi | anyone? |
| 02:32.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:35.27 | VoIPMasta | Ok, I've a small project involving Flash, PHP and Asterisk (a very simplified version of Flash Operator Panel), I'm willing to pay for it... if anyone's interested please msg me |
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| 03:31.30 | VoIPMasta | Ok, I've a small project involving Flash, PHP and Asterisk (a very simplified version of Flash Operator Panel), I'm willing to pay for it... if anyone's interested please msg me |
| 03:33.01 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-33-245.dsl.tul2ok.sbcglobal.net) |
| 03:45.27 | De_Mon | VoIPMasta why not use fop? |
| 03:46.05 | VoIPMasta | It's just too complex for what I need |
| 03:46.18 | De_Mon | ? |
| 03:46.38 | VoIPMasta | I just need to see the dialed number by one extension (SIP peer), not all of them |
| 03:46.50 | De_Mon | fop will do that |
| 03:47.04 | VoIPMasta | a small flash file (SWF) that would display if there's an active call placed by a specific SIP peer and let me know when the call ends |
| 03:47.40 | VoIPMasta | De_Mon, but it will also display far more info, and the person that will "monitor" that SWF shouldn't see calls placed by other SIP peers |
| 03:47.46 | VoIPMasta | to be more specific... |
| 03:48.00 | De_Mon | you can let it display as much or as little as you want |
| 03:48.24 | VoIPMasta | There's a guy working here that places a lot of calls, I want my assistant to be able to see when he has dialed a number, what number he's dialed and for how long |
| 03:49.08 | VoIPMasta | but I want my assistant to be able to see his calls only, and not mine or the ones placed by other sip users here |
| 03:49.33 | VoIPMasta | Right now we keep auditing the CDRs every single day, but I want it to be more "real time" |
| 03:49.35 | De_Mon | fop only shows the peers *YOU* specify |
| 03:49.46 | De_Mon | be it one, two or one hundred |
| 03:49.48 | VoIPMasta | I didn't know that |
| 03:49.59 | De_Mon | I noticed |
| 03:50.24 | De_Mon | granted, if someone is nice enough to reinvent the wheel when fop does do what you want great |
| 03:50.30 | VoIPMasta | now, does FOP allow me to "insert" more info to be displayed? |
| 03:50.34 | De_Mon | otherwise, I suggest learning how to get fop to do what you want |
| 03:50.59 | VoIPMasta | For an instance, we have all the country/area codes in a database, should I be able to insert some text to FOP indicating where that call is going to? |
| 03:51.40 | De_Mon | duno |
| 03:52.00 | VoIPMasta | Ok, I will give it a try, although I was experimenting with another "toy" |
| 03:52.07 | VoIPMasta | We have an AGI that does all the call routing |
| 03:52.16 | *** join/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 03:52.33 | De_Mon | you could probably write a simple php page to do what you're after too |
| 03:52.34 | *** join/#asterisk bmg505 (n=leon@196.209.176.188) |
| 03:52.40 | VoIPMasta | What I'm trying to do right now is to insert a record in the database (not the CDR) when a call is dialed and removing it after they hang up |
| 03:53.05 | VoIPMasta | that way I can do a small SWF that reads from a PHP file, and that PHP file keeps querying the mysql database... |
| 03:53.12 | VoIPMasta | I think it *might* work |
| 03:54.02 | De_Mon | sounds a bit over complicated |
| 04:01.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:04.16 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:04.21 | vutamhoan | I have a problem with TE110P card, it can't use the sync source from far end as I request: span=1,1,0,cas,hdb3,yellow |
| 04:04.30 | vutamhoan | can anybody help me? |
| 04:05.29 | vutamhoan | I've just tested in a running system and result is still error :( |
| 04:09.20 | VoIPMasta | De_Mon, It's not really overcomplicated since we use this AGI script to determine whether a call is going to one of our branch offices (even when dialed using the PSTN), to one of our cell phones, to another SIP user or to the "external world" |
| 04:09.22 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-33-245.dsl.tul2ok.sbcglobal.net) |
| 04:13.06 | De_Mon | asterisk > mysql > php > flash is not more complicated than asterisk > flash ? we must live in different worlds. |
| 04:14.20 | VoIPMasta | De_Mon but asterisk > flash would mean keeping an open connection and constant queries to the asterisk server (which is a small box), on the other hand, we are already doing asterisk > mysql > php and the mysql db as well as the php scripts run on a far more powerful server |
| 04:15.03 | VoIPMasta | and I'm not really into actionscript to set everything from a SWF file ;) |
| 04:28.45 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:29.01 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:29.23 | [hC] | anyone tried out anthm's app_confcall? |
| 04:29.38 | [TK]D-Fender | VoIPMasta, ping |
| 04:33.16 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:34.29 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 04:44.11 | tuxd00d | I'm have a bad brain day. What do I need to do to access my voicemail remotely (call into asterisk from outside)? |
| 04:44.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:45.09 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 04:45.15 | De_Mon | tuxd00d show application voicemail/voicemailmain |
| 04:45.22 | VoIPMasta | tuxd00d, a DID? |
| 04:47.28 | [TK]D-Fender | tuxd00d, Its your dialplan. make yourself an option to get to voicemailmain. |
| 04:47.56 | [TK]D-Fender | no need for an special # to dial. Any that lead to your system can do it depending on how you want to set it up. |
| 04:50.42 | tuxd00d | I see, I see, thanks guys |
| 04:51.18 | bkruse_home | brainstorm.....if ztcfg does NOT like having a [default] context, but action: getconfig needs it to be parsed properly....what to do! |
| 04:52.10 | bkruse_home | ugh... :[ |
| 04:56.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 05:18.44 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 05:19.15 | Aces1Up | has anyone here engineered call back solutions with asterisk? i would like to ask you a few questions. |
| 05:21.25 | [TK]D-Fender | Aces1Up, ask fast |
| 05:22.24 | VoIPMasta | Aces1Up, call back... ANI based? SMS based? you need to be a little bit more specific |
| 05:26.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:29.36 | Aces1Up | via ANI. |
| 05:30.35 | Aces1Up | here is what i want to do, the user receives a call. want the user to hang-up and then call back on that number to be connected to someone. |
| 05:30.47 | Aces1Up | its strange i know. |
| 05:30.47 | *** join/#asterisk mdipalma (n=mdipalma@85-18-136-100.fastres.net) |
| 05:31.25 | VoIPMasta | it's not strange |
| 05:31.28 | [TK]D-Fender | Aces1Up, not at all. Great way to rape cell-phone "free incoming calls" setups |
| 05:31.30 | VoIPMasta | most callback services work that way |
| 05:32.21 | [TK]D-Fender | Aces1Up, they call in. you call a script (whatever kind you want). If makes a .call file and on hangup executes it. it should embed a minor delay. End of story. |
| 05:32.32 | VoIPMasta | I have something like that set up here... only that as my cellphone doesn't send any ANI/CID I use an "unannounced DID" |
| 05:32.49 | VoIPMasta | everytime it "rings" it calls me back to my cellphone and allows me to dial other numbers |
| 05:33.10 | VoIPMasta | the only "risk" with this scheme is that if someone starts dialing that DID every 5 minutes, my cell phone will go crazy :) |
| 05:35.22 | [TK]D-Fender | VoIPMasta, Thats why it should auth, |
| 05:35.39 | [TK]D-Fender | VoIPMasta, auto-accept on CID, IVR on no match. |
| 05:36.22 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 05:37.55 | Aces1Up | voipmasta, mine isn't quite like that. |
| 05:38.14 | Aces1Up | i ask the user to call back on a specific number, to a DID i own. |
| 05:38.19 | Aces1Up | so i can route the call. |
| 05:38.24 | Aces1Up | to my box. |
| 05:38.32 | mdipalma | hi guys... I have to install Asterisk for the first time: do you think is more convenient to start with AsteriskNow or TripBox??? And Why?? |
| 05:41.14 | VoIPMasta | But when you have no CID... |
| 05:41.18 | VoIPMasta | no ANI... |
| 05:41.21 | VoIPMasta | how do you auth? |
| 05:41.26 | VoIPMasta | without answering the call |
| 05:41.54 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:43.56 | [TK]D-Fender | Aces1Up, I would do something MEAN to you... like forge my CID to your DID and let your system infinite-loop itself into crashing :) |
| 05:44.20 | [TK]D-Fender | Aces1Up, Just consider the asumptions of the conditions of that call.... |
| 05:45.04 | [TK]D-Fender | and on that note... |
| 05:45.11 | [TK]D-Fender | goodnight/morning/whatever all.... |
| 05:45.17 | [TK]D-Fender | zzzzzzzzz |
| 05:59.38 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 06:05.34 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 06:19.23 | Aces1Up | hey voipmasta you get that last comment of TKD? I don't understand what he meant. |
| 06:19.58 | VoIPMasta | Yes |
| 06:20.02 | VoIPMasta | that ANI/DID can be forged |
| 06:21.37 | Aces1Up | hrmm... but.. the DID I give them they have to call back on... I own those DID's. |
| 06:21.50 | Aces1Up | as far as ani, I can see that being a problem. |
| 06:22.42 | Aces1Up | ok nevermind what i just said. |
| 06:22.43 | Aces1Up | lol |
| 06:22.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:22.57 | creativx | np. |
| 06:23.27 | Aces1Up | how bout this... a security check that checks the ani/did/CID to make sure it orginates in same country/city as access DID. |
| 06:23.35 | Aces1Up | would that help any? |
| 06:24.08 | creativx | it can still be faked |
| 06:25.36 | Aces1Up | what if the system doesn't just allow any old CID to access the system? all CID's are registered to paying users. |
| 06:26.29 | creativx | i can still fake the cid |
| 06:26.37 | creativx | so that it would always validate |
| 06:26.57 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 06:26.59 | Aces1Up | hrmm, what is the additional security measure i need? combo it with an sms message? |
| 06:27.15 | creativx | I'm not even sure what you are trying to accomplish, I just arrived @office :> |
| 06:27.28 | *** join/#asterisk mirco (n=mirco@87.139.74.16) |
| 06:27.29 | creativx | and i didnt feel like scrolling right now ;) |
| 06:27.31 | Aces1Up | creativx can i pm ya? |
| 06:27.48 | creativx | please dont, i wont be here the next 3 minutes. |
| 06:28.22 | Aces1Up | ok, dang. anyone know if there are any asterisk user group in las vegas? |
| 06:28.32 | VoIPMasta | Aces1Up, just let me see if I understood what you're trying to do |
| 06:28.40 | Aces1Up | ok doke |
| 06:28.45 | VoIPMasta | A regular "joe" dials one of your DIDs and hangs up before the call is answered |
| 06:28.55 | VoIPMasta | your asterisk server dials joe's number |
| 06:28.59 | *** join/#asterisk angryuser (n=aster@df01t2-195-36-137-125.d4.club-internet.fr) |
| 06:29.14 | VoIPMasta | and connects the call to "john" who is on another number (PSTN/SIP/whatever) |
| 06:29.15 | VoIPMasta | right? |
| 06:29.23 | angryuser | goog day ;) |
| 06:29.30 | Aces1Up | almost. |
| 06:29.34 | Aces1Up | second part is right. |
| 06:29.39 | Aces1Up | first part goes... |
| 06:30.02 | VoIPMasta | please avoid using "user" or "friend" to refer to any party |
| 06:30.14 | Aces1Up | system calls joe with Caller id to call back on... |
| 06:30.22 | Aces1Up | joe calls back on that caller id. |
| 06:30.30 | angryuser | when i do Set(Group(1)) and after execute a call , when call ends tha called quit the goup? |
| 06:30.40 | Aces1Up | and connect the call to an existing conference room. |
| 06:30.46 | VoIPMasta | angryuser, it should |
| 06:31.08 | angryuser | VoIPMasta ok thx |
| 06:31.19 | VoIPMasta | Aces1Up, if you don't mind me asking... what's the "benefit" of doing that? just to let Joe know that there's an ongoing conference? |
| 06:31.38 | Aces1Up | to take advantage of incoming minutes on DID. |
| 06:31.43 | Aces1Up | in another country. |
| 06:32.02 | Aces1Up | so i don't pay to terminate traffic. |
| 06:32.25 | VoIPMasta | but will Joe want to pay the toll fees to call your DID? |
| 06:32.43 | Aces1Up | did will be local to Joe. |
| 06:33.02 | VoIPMasta | ok, but in most countries even local calls have a charge |
| 06:33.08 | Aces1Up | sure.. |
| 06:33.26 | Aces1Up | but not as bad as dialing internatationally directly. |
| 06:33.27 | VoIPMasta | ok, let's see |
| 06:34.02 | VoIPMasta | but if joe picks up the call, you will have toll charges (termination) |
| 06:34.12 | VoIPMasta | how can you prevent that? |
| 06:34.25 | Aces1Up | voip not if the other part is calling in as well on a DID, hence the conference room. |
| 06:35.15 | VoIPMasta | now, what would trigger the call to Joe's phone? |
| 06:35.27 | VoIPMasta | I mean, how does the system know when to call Joe |
| 06:35.38 | angryuser | operatois it is ">=" or "=>" ? |
| 06:35.53 | angryuser | *operators |
| 06:36.01 | Aces1Up | there are mant ways to do it. |
| 06:36.27 | VoIPMasta | Ok, you will have to code an AGI script to do so |
| 06:36.33 | VoIPMasta | my best bet would be: |
| 06:36.41 | Aces1Up | yep, lots of codin. |
| 06:36.48 | VoIPMasta | Validate if a call should be placed (by whatever means you want to trigger the call) |
| 06:37.09 | Aces1Up | voip.. one sec.. got a paste for ya. |
| 06:37.14 | VoIPMasta | Set a ringing time of maybe 5 seconds (if Joe isn't going to pick up the call, it shouldn't ring long) |
| 06:37.17 | Aces1Up | call flow diagram a built. |
| 06:37.20 | VoIPMasta | use a pastebin |
| 06:39.27 | angryuser | can someone give me example of script based on ${DIALSTATUS} ? |
| 06:40.13 | creativx | goto(s-${DIALSTATUS}) |
| 06:40.24 | creativx | exten => s-BUSY,1,noop(oiii) |
| 06:40.31 | creativx | etc. |
| 06:41.16 | creativx | did that make sense angryuser ? :) |
| 06:41.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:41.33 | angryuser | not really, what s-BUSY does? |
| 06:41.48 | angryuser | what is s-BUSY? ;) |
| 06:42.00 | VoIPMasta | it's just an extension name |
| 06:42.12 | Aces1Up | Voip you get that paste? |
| 06:42.17 | angryuser | pk i got it |
| 06:42.19 | angryuser | ok |
| 06:42.21 | VoIPMasta | Looking at it roght now |
| 06:42.24 | VoIPMasta | right now |
| 06:43.37 | angryuser | <creativx> yes it is a nice way to do it, thanks |
| 06:44.13 | creativx | angryuser: it sure is. in CLI do "show application dial" to see all possible return statuses |
| 06:44.41 | Aces1Up | Voip Masta, I was lookin at adhearsion you use it? |
| 06:44.53 | Aces1Up | thinking of using it instead of AGI. |
| 06:44.57 | angryuser | <creativx> yea i got that, just need to think a bit, to mae a soup with existing scripts |
| 06:45.01 | angryuser | *make |
| 06:45.14 | creativx | remeber seasoning. |
| 06:45.25 | creativx | remember. damn friday typos |
| 06:46.45 | angryuser | it is Friday! yes! going to drink today |
| 06:46.58 | creativx | hell yes |
| 06:47.03 | creativx | sun and beer is great success. |
| 06:47.05 | creativx | very nice |
| 06:51.04 | Aces1Up | Voip what ya think? |
| 06:54.49 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:56.45 | Aces1Up | voip you there? |
| 06:56.59 | VoIPMasta | yes |
| 06:57.02 | VoIPMasta | was on the phone |
| 06:57.11 | Aces1Up | ahh sorry. |
| 06:57.27 | Aces1Up | what do you think? |
| 06:57.47 | VoIPMasta | I'm not completely understanding your scheme |
| 06:57.55 | Aces1Up | hrmmm. |
| 06:58.16 | Aces1Up | nevermind then. |
| 06:58.32 | VoIPMasta | is this some kind of dating application? |
| 06:58.39 | VoIPMasta | dating/matchmaking? |
| 06:58.46 | Aces1Up | no. lol. |
| 06:58.50 | Aces1Up | hehehe. |
| 06:59.32 | VoIPMasta | now I haven't use adhearsion (I don't even know what it is) |
| 06:59.40 | Aces1Up | ohh. |
| 06:59.54 | VoIPMasta | I do most things using AGI and PHP |
| 07:00.16 | VoIPMasta | now, on call processing |
| 07:00.33 | VoIPMasta | your 1st step is "receive incoming call" but wasn't the call to joe the first step? |
| 07:01.03 | Aces1Up | http://adhearsion.com/ |
| 07:01.14 | Aces1Up | heh lol. |
| 07:01.22 | *** join/#asterisk JoeMoes (n=JoeMoes@wolverine.vcc.de) |
| 07:02.43 | Aces1Up | the box basically recieves calls put them in conference rooms, then brings in the other callers as they call in on other DId's. |
| 07:02.56 | Aces1Up | it just accepts calls only really. |
| 07:03.06 | VoIPMasta | ok so to be a little bit more straigthforward... |
| 07:03.20 | VoIPMasta | You "invite" users to pre-existing conference rooms by calling them |
| 07:03.38 | Aces1Up | yeh, thats one way of looking at it yes. |
| 07:03.40 | VoIPMasta | they dial in and your system matches the DID and the CID to see what conference room they should join |
| 07:03.46 | Aces1Up | yes. |
| 07:04.05 | Aces1Up | and the system does it automatic.. no interaction on their part. least for the callee. |
| 07:04.55 | VoIPMasta | ok you should start by doing that, then add the other features |
| 07:05.16 | VoIPMasta | Dial (PROTO/DEST,15) so that it rings for just 15 seconds |
| 07:05.20 | Aces1Up | voip, yes thats what i was thinking, just wondering if that call flow looks good. |
| 07:05.33 | *** part/#asterisk tanacsdavid (n=david@office.axpnet.com) |
| 07:05.39 | Aces1Up | voip i see. |
| 07:05.44 | Aces1Up | good thing to start with. |
| 07:05.49 | VoIPMasta | Yes, even when I still don't see any practical use for it hehehe |
| 07:06.05 | Aces1Up | ok... |
| 07:06.10 | Aces1Up | well let me worry bout that. |
| 07:06.17 | VoIPMasta | and I still think that the average joe will pick up the call, instead of letting it ring and then dialing in |
| 07:06.55 | Aces1Up | these aren't average Joe's! these are Joe's Dad's. |
| 07:07.02 | Aces1Up | lol |
| 07:07.04 | VoIPMasta | and you should be aware (as you're talking about several countries) that in some countries CID from VoIP originated calls doesn't work |
| 07:07.11 | Aces1Up | Joe sr. |
| 07:07.18 | Aces1Up | uhh Sr. |
| 07:07.34 | VoIPMasta | Mexico is one of those countries |
| 07:07.45 | VoIPMasta | I sure can tell as I'm mexican |
| 07:08.22 | Aces1Up | voip, i was thinkin bout that, thas why i thought integrating SMS somehow might be a solutiom. |
| 07:08.26 | VoIPMasta | I know when one of my customers is calling me on my cellphone because I don't get any CID |
| 07:08.46 | VoIPMasta | CID doesn't work in most African countries |
| 07:08.48 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:09.03 | VoIPMasta | Central/South American countries do also have issues with "user set" caller IDs |
| 07:09.04 | Aces1Up | such like SMS: Your Mom is calling you, to recieve call dial this #. |
| 07:09.29 | VoIPMasta | yes, with SMS it would be easier |
| 07:09.42 | *** join/#asterisk Vec (n=Vec@dsl-243-106-186.telkomadsl.co.za) |
| 07:09.55 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:10.02 | Aces1Up | voip. well... that is the reason for the 1min termination call. |
| 07:10.16 | Aces1Up | to let the party know to call back. |
| 07:10.21 | Aces1Up | on certain number. |
| 07:10.37 | VoIPMasta | yes, that would work better |
| 07:10.42 | VoIPMasta | now, another way to do it... |
| 07:10.54 | VoIPMasta | to have the callee enter the number of the conference room |
| 07:11.08 | VoIPMasta | that way if they use another phone to dial in, they can still get in the conference room |
| 07:11.17 | Aces1Up | voip yeh good one. |
| 07:11.48 | Aces1Up | dang so CID isn't reliable then huh? |
| 07:12.02 | angryuser | * one week in production and no crashes ! yahoo ;) |
| 07:12.12 | VoIPMasta | let's say that the recording says: "You have a call waiting from [ insert caller's name here, it can be recorded by the caller ], to start a conference with [name again] please dial 123456789 and when prompted dial [conference room number]" |
| 07:12.35 | Aces1Up | voipmasta, yeh, but thats is awful lot. |
| 07:12.46 | VoIPMasta | not really |
| 07:12.48 | angryuser | <VoIPMasta> yea too much, let secretary do it |
| 07:12.50 | *** join/#asterisk svenna_ (n=svenna@p548D1445.dip0.t-ipconnect.de) |
| 07:12.56 | VoIPMasta | think about it... in most countries, cell phone calls are pretty expensive |
| 07:13.09 | VoIPMasta | let's say that Dad wants to call Joe and enters Joe's cell phone number |
| 07:13.25 | VoIPMasta | and Joe does really want to talk with Dad but is out of credit on his prepaid cell phone |
| 07:13.27 | *** join/#asterisk DEac- (n=deac@2001:6f8:1021:0:213:d4ff:fe7f:1e39) |
| 07:13.30 | DEac- | moin |
| 07:13.45 | VoIPMasta | so he uses a payphone or fixed phone to dial a local number (your DID) and enter the conference |
| 07:13.55 | Aces1Up | voip yep. |
| 07:14.00 | svenna_ | moin deac |
| 07:14.04 | Aces1Up | thas where i'm going with this. |
| 07:14.28 | angryuser | <VoIPMasta> he call he secretary => transfer ? |
| 07:14.40 | VoIPMasta | if you do a little bit of research, you'll see that there are more mobile phones than fixed lines in most countries, however most people use prepaid cell phones. |
| 07:15.04 | VoIPMasta | angryuser, if Joe's dad has a secretary... but if he doesn't then he would have to dial an international call himself |
| 07:15.14 | Aces1Up | voip even so, a local call on a pre-paid phone isn't that much. |
| 07:15.28 | VoIPMasta | it depends on which country we're talking about |
| 07:15.33 | Aces1Up | mexico. |
| 07:15.35 | Aces1Up | lol. |
| 07:15.41 | VoIPMasta | Spain = 0.50 eur/minute |
| 07:15.54 | VoIPMasta | Mexico = 0.40 USD / minute |
| 07:16.09 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:16.12 | VoIPMasta | Argentina = 0.60 USD / minute |
| 07:16.24 | VoIPMasta | Italy = 0.38 eur/minute |
| 07:16.30 | angryuser | i dont calculation today, with * we divided the cost of phoning by 5 |
| 07:16.31 | Aces1Up | voip 40cents a min/local call? |
| 07:16.35 | angryuser | *did |
| 07:16.42 | VoIPMasta | Aces1Up, where? in Mexico? |
| 07:16.49 | Aces1Up | yes |
| 07:16.58 | DEac- | i want to change the default language. i changed language=de in [general] in extensions.conf and i set languageprefix=yes in [general] in asterisk.conf . i also installed the german voices in /var/lib/asterisk/sounds/de |
| 07:17.07 | VoIPMasta | Yes, a local call from a prepaid cell phone in Mexico would cost about 0.40 USD /minute |
| 07:17.20 | DEac- | but if i call to my machine, i hear a english voice |
| 07:17.28 | Aces1Up | pre-paid phones get free incoming calls though? |
| 07:17.31 | angryuser | <DEac-> i did nothing but remplaced the defaul sounds ;) |
| 07:17.47 | VoIPMasta | Aces1Up, yes if you are within the same city (area code) where you got your mobile phone |
| 07:17.51 | DEac- | angryuser: this is unclean |
| 07:17.54 | VoIPMasta | otherwise they charge you a roaming fee |
| 07:18.00 | Aces1Up | aces how much does it cost from landline per min? |
| 07:18.05 | VoIPMasta | DEac-, I did also replace the original sounds ;) |
| 07:18.08 | angryuser | <DEac-> yea but i dont need en t all ,so whatever ? |
| 07:18.19 | VoIPMasta | Aces1Up, local call from a land line? to another fixed phone? |
| 07:18.21 | Aces1Up | landline local call. |
| 07:18.34 | VoIPMasta | Local calls are charged "per call" and not "per minute" |
| 07:18.45 | VoIPMasta | about 0.15 USD/call |
| 07:18.59 | VoIPMasta | that's a fixed phone to fixed phone |
| 07:20.05 | Aces1Up | voip when i purchase DID's in mexico, aren't those considered fixed phones? |
| 07:20.14 | VoIPMasta | yes |
| 07:20.27 | VoIPMasta | where did you purchase them (if you don't mind me asking) |
| 07:20.33 | Aces1Up | well it really comes down to me paying the .40c a min or the end user. |
| 07:20.57 | Aces1Up | voip, don't they have free nights and weekend plans? |
| 07:21.05 | VoIPMasta | not on prepaid mobile phones |
| 07:21.12 | DEac- | i installed it via portage (gentoo) and i want to managed by portage |
| 07:21.40 | Aces1Up | whats the charge per min to call from cell phone to usa? |
| 07:22.04 | VoIPMasta | I think it's something ~ 1 USD / min |
| 07:22.37 | Aces1Up | wow. |
| 07:22.37 | VoIPMasta | I haven't placed a call to the US from a cell phone since 2001 |
| 07:22.51 | DEac- | but there's a way to do it clean. in asterisk-buch there's a description, but it doesn't work |
| 07:23.08 | Aces1Up | voip, you know the percentage of cell-phone users in mexico use pre-paid? |
| 07:23.12 | VoIPMasta | DEac-, did you reload/restart asterisk after modifying the conf files? |
| 07:23.36 | VoIPMasta | Aces1Up, I have the EXACT statistics on my laptop, but my laptop is at home right now... from what I remember it's about 80% |
| 07:23.36 | angryuser | <DEac-> remplace the files and forget about the pain |
| 07:23.57 | Aces1Up | voip, if i shot you my e-mail would you mind sending me those statistics? |
| 07:24.34 | VoIPMasta | I wouldn't mind at all |
| 07:24.40 | VoIPMasta | do you know spanish? |
| 07:24.46 | Aces1Up | not much. |
| 07:24.48 | DEac- | VoIPMasta: yes, of course |
| 07:28.05 | VoIPMasta | DEac-, if you're not going to use english recordings, why keep them in your server? |
| 07:28.37 | DEac- | i love the clean way |
| 07:28.47 | DEac- | and right way |
| 07:31.29 | angryuser | <DEac-> yea , but when you work in the company you need the results sometimes clean way pass out |
| 07:31.42 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-48-203.dsl.tul2ok.sbcglobal.net) |
| 07:33.31 | *** join/#asterisk GMitre (n=qzblo@200-234-185-201.usr.microlinknet.com.br) |
| 07:34.08 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 07:34.17 | GMitre | Some one can say to me how can i specify in extension.conf thatดs only do that function if is the correct time, example: if is more than 06 PM than do this else to that ?? |
| 07:34.41 | Daejeo1 | has anyone used a sipp testing tool? |
| 07:35.05 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:35.35 | Daejeo1 | http://sipp.sourceforge.net/ |
| 07:36.51 | *** join/#asterisk Daniel_TechFlare (n=chatzill@203-206-217-91.perm.iinet.net.au) |
| 07:37.40 | creativx | GMitre: " show application GotoIfTime |
| 07:37.42 | creativx | " |
| 07:37.58 | GMitre | creativx thankดs very mutch |
| 07:38.45 | Daniel_Tech | just wondering if someone could give me some advice on what zaptel card i should choose. i have never used one before so im kind of stumped |
| 07:39.12 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 07:40.26 | VoIPMasta | DEac-, There's something you've to learn about IT... if it works, then it's the right way. Most tasks are accomplished using dirty hacks (even the software giants like microsoft do so) |
| 07:40.50 | DEac- | i don't like dirty hacks |
| 07:40.54 | DEac- | i love bugfixes |
| 07:41.21 | VoIPMasta | ok, you're not a programmer ;) |
| 07:41.28 | DEac- | i'm a programmer |
| 07:43.58 | VoIPMasta | and you hate dirty hacks? |
| 07:44.16 | DEac- | yes |
| 07:44.17 | VoIPMasta | have you ever customized a kernel? |
| 07:44.33 | DEac- | yes (the nt-kernel |
| 07:44.55 | VoIPMasta | nt-kernel? |
| 07:45.05 | DEac- | windows nt |
| 07:45.24 | VoIPMasta | nmmm and if you don't mind me asking... how did you customize microsoft's close-source kernel? |
| 07:45.58 | DEac- | there're some tools for it |
| 07:46.05 | DEac- | like reshack for example |
| 07:46.19 | DEac- | but this is long ago |
| 07:46.25 | VoIPMasta | but they add/remove services, don't modify the kernel |
| 07:46.33 | VoIPMasta | they tweak microsoft registry |
| 07:46.46 | VoIPMasta | which is also far away from kernel customization |
| 07:47.48 | DEac- | no, you can modify the kernel too |
| 07:48.07 | DEac- | but this is an other hist |
| 07:48.09 | VoIPMasta | does it get disassembled, decompressed, decompiled, recompiled, relinked? |
| 07:49.06 | DEac- | no, the machinecode was changed directly |
| 07:49.17 | VoIPMasta | yeah right, well I just have something to say |
| 07:49.29 | DEac- | like in netbsd |
| 07:49.33 | VoIPMasta | in the Open Source world, you'll find a lot of software that works far better with quick dirty hacks |
| 07:50.01 | DEac- | and slow |
| 07:50.07 | VoIPMasta | sometimes you have a library missing (because of a version mismatch) and it's easier and quicker to just create a symlink from your current library and "fake the required one" |
| 07:50.29 | DEac- | yes, the old dirty way :-D |
| 07:50.58 | DEac- | this is important if you must run propritary apps, like alladins etoken-lib |
| 07:51.14 | VoIPMasta | if you browse through bugzilla sites you'll see that most bug fixes start with a dirty hack :) |
| 07:53.45 | DEac- | ok, this dirty hack replacing english voices with german doesn't work |
| 07:53.59 | VoIPMasta | ? |
| 07:54.04 | VoIPMasta | do you still hear english voices? |
| 07:54.09 | DEac- | the problem is, that the voicemail needs some files, which doesn't exists in german voices |
| 07:54.53 | VoIPMasta | ok if you copy the files, they will overwrite the existing ones and leave the original ones (that aren't in the german voices) intact |
| 07:55.09 | DEac- | yes, i know |
| 07:55.45 | *** join/#asterisk d4rk_hun7er (n=adaom@201-048-044-245.xd-dynamic.ctbcnetsuper.com.br) |
| 07:56.08 | VoIPMasta | and by looking at the file dates, you can figure out which files weren't replaced so that you can record them yourself (if you want it to be 100% german) |
| 07:56.26 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:00.26 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 08:00.56 | DEac- | it's possible to hear mailbox with a cellphone, which is unknown in dialplan? |
| 08:01.22 | DEac- | remote enquiry |
| 08:03.11 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:16.15 | *** join/#asterisk FliTTi (n=chatzill@212.218.65.194) |
| 08:16.21 | FliTTi | hey@ll |
| 08:18.15 | FliTTi | i have a question: i whant to log if somebody is doing a 3way transfer. is this possible? I have tryied to see it in the cdr, but ic can't. The same if I look in cdr_mysql. Have anyone an idear, how it works? |
| 08:19.42 | angryuser | <FliTTi> have you checked if module is loaded ? |
| 08:20.05 | angryuser | <FliTTi> you need the mysql client aslo installed in you system |
| 08:20.51 | FliTTi | yes that's right. some date is allready put in the cdr, or in the mysqltable but, i can't see in this data if an call is transferd. |
| 08:22.15 | angryuser | a have some ptoblems in my mysql tables too, after upgrade from 1.4.0 to 1.4.4 a lot of blank or half compled fields started to appear |
| 08:23.35 | FliTTi | that's not my problem. i whant to regonize if a call is transfered. I whant to see it in the log |
| 08:25.11 | FliTTi | have somebody an idea? |
| 08:52.51 | *** join/#asterisk TJ` (n=freenode@bnc.crazybnc.com) |
| 08:52.52 | TJ` | :@ |
| 08:53.30 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 08:55.52 | *** join/#asterisk matsk (n=mk@194.68.102.163) |
| 08:56.50 | *** join/#asterisk cervi2 (n=cervicek@rhlx09.hs-esslingen.de) |
| 08:57.05 | *** join/#asterisk mirco (n=mirco@87.139.74.16) |
| 09:05.29 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 09:08.10 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:09.16 | deegan | I could use some help regarding passing arguments in a menu. I'm not terribly familiar with how asterisk handles this, but basicly what i want to do is have a dialin for chanspy() and have it do someting like this. |
| 09:09.37 | deegan | user dial in -> enter 4 digits -> chanspy(SIP/$ARG,q) |
| 09:14.42 | HarryR | Read(pin,enter-pin,4) |
| 09:14.54 | HarryR | ChanSpy(SIP/$pin) |
| 09:14.56 | HarryR | sort of thing |
| 09:15.26 | DEac- | oh, asterisk is segmentation fault |
| 09:16.04 | DEac- | exten => s,1,VoiceMail(${ARG1},${ARG2}) |
| 09:16.09 | DEac- | exten => a,1,VoiceMailMain(${ARG1}) |
| 09:16.29 | DEac- | and then press * while it spokes |
| 09:17.03 | DEac- | is this a bug or it's because i run it in foreground? |
| 09:18.54 | deegan | HarryR: Thank you very much, i didn't think i would get it done that bash-scripty. :) |
| 09:22.06 | *** join/#asterisk dreamind (n=dreamind@p54A7A1A6.dip0.t-ipconnect.de) |
| 09:22.08 | dreamind | Hi folks |
| 09:22.40 | dreamind | I have problems with a Macro called via M() option on dial |
| 09:23.03 | dreamind | I would like to read a digit from the person being called prior to connecting the called with the callee. |
| 09:23.23 | dreamind | this is what I do in that macro: |
| 09:23.24 | dreamind | exten => s,n,Read(RESULT,beep,1,,5,3) |
| 09:23.53 | dreamind | but Read() does always tell me (on the console) the person being called didn't enter a digit |
| 09:24.11 | dreamind | I already verified that DTMF per se (and Read()) does work fine. |
| 09:27.35 | creativx | are there any norwegianers in this channel? |
| 09:29.50 | dreamind | creativx: sorry not me... |
| 09:30.24 | creativx | i need some norwegian voice prompts.. arr |
| 09:30.41 | HarryR | You could hire some norwegian voice talent to do it? |
| 09:31.09 | HarryR | For under ฃ100 I'm sure you could get an hour of their time fairly easily |
| 09:31.12 | creativx | i could |
| 09:31.26 | creativx | the voop voicepack is too new norwegian for me |
| 09:31.38 | creativx | i just gotta find someone who offers those services |
| 09:31.42 | HarryR | ah |
| 09:32.48 | creativx | cause my boss hates the current voicepack hehe |
| 09:33.01 | creativx | it either sounds really comic or disasterous |
| 09:33.20 | HarryR | How many people in your company? |
| 09:33.40 | HarryR | It might be worth just trying out everybody to see if anybody in-house can do professional sounding stuff |
| 09:34.03 | dreamind | :( I just verified, Read() works in "standard" macros (being called via Macro()) |
| 09:34.23 | dreamind | but not in a macro which is called by Dial() on the line of the person being called :( |
| 09:39.10 | creativx | HarryR: we are 15 |
| 09:39.13 | creativx | its the time im worried about |
| 09:39.39 | creativx | could easily be time consuming.. but i guess we could try |
| 09:39.50 | creativx | i just dont have any sound editing software at hand |
| 09:39.55 | creativx | and ackk.. i'd rather pay some sucker to do it |
| 09:39.55 | creativx | hehe |
| 09:46.23 | angryuser | does somebody use misdn here? |
| 09:47.39 | angryuser | i woluld like to send a fax by misdn, do i need to check if port 1-4 is free and send to freer on on i just need top group them and let asterisk do the rest ? |
| 09:48.01 | angryuser | doe asterisk check if port n is free ? |
| 09:48.21 | angryuser | i have a lot of incoming call on that ports |
| 09:50.07 | angryuser | and if i call group of ports i receive 'BUSY' when i n reality the number is free |
| 09:53.59 | *** join/#asterisk scanna (n=scannach@81-174-16-211.staticnet.ngi.it) |
| 09:54.50 | scanna | hi all, can someone help me compiling chan_h323 with asterisk 1.2.18? |
| 09:55.48 | scanna | when i do "make opt" i get this |
| 09:55.50 | scanna | make: *** No rule to make target `opt'. Stop. |
| 09:55.58 | scanna | with asterisk 1.2.14 no problems |
| 09:56.40 | *** join/#asterisk rkr245 (n=lisp@59.144.92.213) |
| 09:56.49 | *** part/#asterisk rkr245 (n=lisp@59.144.92.213) |
| 10:09.27 | *** join/#asterisk saftsack (n=saftsack@pD9E04E71.dip.t-dialin.net) |
| 10:17.05 | cy303 | yo |
| 10:23.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:24.43 | *** join/#asterisk yassaccan (n=yassacca@admin147.hgo.se) |
| 10:28.17 | *** join/#asterisk achu (n=achu@61.17.217.185) |
| 10:29.49 | achu | DOVID, are u there ? |
| 10:32.20 | *** join/#asterisk vgster (n=vgster@h147170.navonline.net) |
| 10:33.12 | *** join/#asterisk A-Data (n=asd@196.218.120.252) |
| 10:34.24 | A-Data | Hello all . How can i make * query a database server the scenerio is like that (Customer enter his userid number) the * query another mysql DB server and return results my developlemt staff know mysql very well but we don`t know where to put every thing do we have to use PHP or there is special programming language |
| 10:37.38 | Pilko | A-Data - just look at Asterisk AGI |
| 10:37.53 | *** join/#asterisk matsk (n=mk@194.68.102.175) |
| 10:37.53 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 10:38.33 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 10:38.35 | zeeesh | hi |
| 10:39.25 | A-Data | Pilko any refrence to it please |
| 10:39.34 | Pilko | A-Data - a special language is not required. Perl, Python etc. are ok. My choice is Ruby. nice thing |
| 10:41.01 | *** join/#asterisk friedrich| (n=friedric@e177248248.adsl.alicedsl.de) |
| 10:41.40 | *** join/#asterisk Vec2 (n=Vec@dsl-243-89-196.telkomadsl.co.za) |
| 10:42.17 | zeeesh | killing me ... trying to build new remote machine .. rhel ...using asterisk-1.4.4 ... getting too many errors at CLI .. now getting .. "Can't locate LWP.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi"? |
| 10:48.43 | HarryR | zeeesh: which script's that coming from? |
| 10:49.18 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 10:50.41 | jaike | has anyone here used the TCB400B card? for codec translation? |
| 10:51.10 | A-Data | zeeesh do you have perl installled? |
| 10:51.26 | *** join/#asterisk samarora (i=minesh@203.88.149.166) |
| 10:51.26 | A-Data | LWP is bundled with perl |
| 10:52.26 | A-Data | if you Need to get the LWP.pm here is a downloadable one but take care as i suggest installing it in correct way from perl package http://search.cpan.org/src/GAAS/libwww-perl-5.64/lib/LWP.pm |
| 10:53.04 | *** join/#asterisk ne0- (n=randy@190.80.157.190) |
| 10:53.43 | dreamind | hm, still nobody here who can help me with Dial() and Macros being called through M()? |
| 10:53.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:57.02 | *** part/#asterisk FliTTi (n=chatzill@212.218.65.194) |
| 10:57.15 | *** join/#asterisk FliTTi (n=chatzill@212.218.65.194) |
| 10:57.21 | FliTTi | i have a question: i whant to log if somebody is doing a 3way transfer. is this possible? I have tryied to see it in the cdr, but ic can't. The same if I look in cdr_mysql. Have anyone an idear, how it works? |
| 10:57.58 | *** join/#asterisk Vec (n=Vec@dsl-243-97-122.telkomadsl.co.za) |
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| 11:03.37 | *** part/#asterisk chribac (n=chatzill@p57A8156D.dip0.t-ipconnect.de) |
| 11:09.10 | zeeesh | <HarryR>: perl script |
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| 11:16.33 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 11:16.37 | JoelSolanki | Good Morning all |
| 11:17.05 | JoelSolanki | i want to create gui where i can display realtime calls going of extensions |
| 11:17.20 | JoelSolanki | can anybody tell me where can i get take this data and create gui ? |
| 11:19.18 | A-Data | JoelSolanki asterisk gui have this GUI |
| 11:20.16 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 11:20.31 | *** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 11:21.01 | *** join/#asterisk mihinomenest (i=MEX1@cerebus.clandestineresearch.com) |
| 11:22.45 | JoelSolanki | u mean the new Aserisk gui softwware |
| 11:23.36 | *** part/#asterisk samarora (i=minesh@203.88.149.166) |
| 11:25.50 | *** part/#asterisk FliTTi (n=chatzill@212.218.65.194) |
| 11:28.10 | *** part/#asterisk jaike (n=jaike@125.5.144.90) |
| 11:31.04 | *** join/#asterisk expose (n=nobody@82.139.196.236) |
| 11:31.09 | expose | hi |
| 11:31.42 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 11:31.57 | expose | Does anyone know wether emergeny numbers are working using german VoIP lines? |
| 11:34.54 | *** join/#asterisk CapriCorn^80 (i=Administ@209.8.41.149) |
| 11:40.01 | *** join/#asterisk theBrave (n=jberniol@AMontsouris-753-1-4-91.w90-2.abo.wanadoo.fr) |
| 11:41.22 | theBrave | Hello, i want to learn more about sip, and make some experiments. I have seen that asterisk is availlable for WRT devices, how well could it perform on an Asus WL500GP (arm 266MHz, 32M ram) ? |
| 11:41.41 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-217-59.cablep.bezeqint.net) |
| 11:48.05 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
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| 11:49.52 | *** join/#asterisk xzu (n=otto@brucester.a20.net) |
| 11:50.16 | xzu | hi |
| 11:50.26 | Dovid | hi |
| 11:50.54 | xzu | is it possible to route incoming faxes on my asterisk box, using dtmf? |
| 11:51.01 | Dovid | yes |
| 11:51.09 | Dovid | well what kind of DTMF ? |
| 11:51.17 | Dovid | in an IVR ? |
| 11:51.19 | xzu | nice! so not based on DID but DTMF |
| 11:51.20 | xzu | nope |
| 11:51.35 | Dovid | where is the DTMF coming from ? |
| 11:51.42 | xzu | i'd like my clients to be able to fax to a number, append a pound and a code |
| 11:51.50 | xzu | and route the faxes based on that code |
| 11:51.54 | Dovid | ah ok |
| 11:51.57 | xzu | they would use a normal analogue fax |
| 11:52.05 | Dovid | ast4erisk vanialla or trixbox? |
| 11:52.09 | Dovid | asterisk* |
| 11:52.20 | Dovid | oh analouge fax modem and route it to asterisk ? |
| 11:52.31 | xzu | this would save us having to reserve huge blocks of numbers *and* solve an identification issue |
| 11:52.45 | Dovid | what kind of lines will you be using ? |
| 11:52.49 | xzu | is trixbox open? |
| 11:52.49 | Dovid | POTS ? |
| 11:52.55 | xzu | uhm.. isdn 30 |
| 11:53.07 | Dovid | then asterisk should be able to do it |
| 11:53.08 | xzu | sorry, i'm a total telco n00b |
| 11:53.15 | Dovid | we all were at one point |
| 11:53.27 | xzu | ok, any pointers to links/howto's/whatnot? |
| 11:53.28 | Dovid | i would not reccomend using trixbos for something like this |
| 11:53.49 | Dovid | i dont know of any faxing support for the latest version of 1.4.X |
| 11:54.06 | Dovid | well first - are using regular fax machines ? or u want fax to a file ? |
| 11:54.50 | xzu | regular |
| 11:55.10 | Dovid | are the fax machines going to be in a different location or near the server ? |
| 11:55.18 | Dovid | because faxing + IP + issues |
| 11:55.24 | Dovid | = issues* |
| 11:55.35 | xzu | all over the place |
| 11:55.46 | Dovid | faxing over IP isn't that good |
| 11:55.54 | Dovid | asterisk 1.4.X supports T.38 pass through |
| 11:56.13 | xzu | ok, I prefer to have an isdn card in the box and have isdn 30 hooked up to it |
| 11:56.21 | Dovid | you can try using ATA's with ur ISDN line but there is no garuntee |
| 11:56.24 | HarryR | and OpenPBX supports T.38 origination & termination iirc |
| 11:56.52 | Dovid | brb |
| 11:56.56 | HarryR | same goes for Yate :) |
| 11:57.38 | xzu | ok, i was thinking of using hylaxfax for the solution but asterisk seems to have a much more active userbase? |
| 11:58.08 | HarryR | well, most people just use hylafax for fax termination unless their platform supports it already |
| 11:58.34 | xzu | what would you guys recommend? (have digital phone lines, have server and isdn card, need dtmf based routing to email addresses) |
| 11:59.17 | HarryR | Asterisk and sandsp or yate & sandsp |
| 11:59.23 | A-Data | does any one know VOIP termination that can give me usa phone numbers so that calls from this usa number route to my ASterisk |
| 11:59.28 | xzu | actually, a spool directory would good enough |
| 12:06.22 | A-Data | does any one know VOIP termination that can give me usa phone numbers so that calls from this usa number route to my ASterisk |
| 12:06.28 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 12:08.06 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 12:08.20 | HarryR | A-Data: google for USA pstn numbers |
| 12:08.46 | HarryR | or even better "USA incoming numbers" |
| 12:09.02 | HarryR | ah: http://www.voiptalk.org/products/Phone+the+USA+Numbers |
| 12:10.11 | A-Data | ty HarryR |
| 12:12.47 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.59 | xzu | HarryR: so sandsp for the analogue fax end, and yate or asterisk for taking care of the received data |
| 12:14.10 | xzu | any pref between yate or asterisk? |
| 12:14.42 | HarryR | YATE, but asterisk is easier to use |
| 12:15.03 | xzu | YATE is written in c++, so harder to hack for sysadmins |
| 12:15.08 | Dovid | SpanDSP doesnt work on 1.4.X :( |
| 12:15.15 | xzu | or is asterisk c++ to? |
| 12:15.19 | xzu | Dovid: woof |
| 12:15.20 | Dovid | cant seem to find a coder to have it work on 1.4.x |
| 12:15.43 | Dovid | xzu: issue is that digium is stoping to support 1.2.X in a few months |
| 12:15.51 | HarryR | xzu: nah wrong :) YATE is written extremely cleanly which makes it easier to hack for sysadmins |
| 12:16.04 | HarryR | xzu: Asterisk is a horrible maze of C which is a nightmare |
| 12:16.25 | xzu | HarryR: ok, check |
| 12:16.35 | HarryR | OpenPBX then? |
| 12:16.58 | HarryR | uh.. callweaver* |
| 12:17.43 | xzu | the thing is, i just need the fax to number#$SOMEPIN to be routed to a dedicated number and I'll be fine. that's all this setup will be used for |
| 12:18.18 | xzu | moving our office pbx to asterisk is a long term project, we don't have enough voice/fax traffic to make that a priority |
| 12:18.33 | HarryR | ah |
| 12:18.34 | creativx | anyone here what SIP call-info header x-lite needs in order to autoanswer? |
| 12:21.54 | [TK]D-Fender | creativx: Consider the strong likelyhood that being the free version, the functionality you are hoping for does not exist at all, or has been removed. |
| 12:23.28 | *** join/#asterisk Vec2 (n=Vec@dsl-243-97-122.telkomadsl.co.za) |
| 12:24.01 | HarryR | [TK]D-Fender: there shouldn't be any header to make a phone auto-answer |
| 12:24.12 | HarryR | if there is, it should be considered a security hole |
| 12:24.57 | Vec2 | HarryR : there is, so you can use Grandstream phones as pagers. |
| 12:25.00 | [TK]D-Fender | HarryR: And you share this for every other product out there? And considering calls are AUTH'D and coming from a server that determines such priveldges.....? |
| 12:25.06 | creativx | i use it for autoanswering our ip10s |
| 12:25.24 | creativx | but yes [TK]D-Fender, i will investigate if the pro version has support for it |
| 12:25.39 | HarryR | [TK]D-Fender: /me shrugs |
| 12:25.47 | HarryR | bah |
| 12:25.49 | creativx | security holes are fun anyways |
| 12:26.16 | HarryR | not really security, but just stuff which could lead to one |
| 12:26.34 | creativx | well |
| 12:27.36 | [TK]D-Fender | HarryR: Next you'll lobby for gov't to ban ball-point pens because they could be used to write down confidential notes for illegal dissemination. |
| 12:27.46 | [TK]D-Fender | HarryR: you TERRIST |
| 12:28.17 | HarryR | not at all, it's like having a big sign outside your house saying if you're in or out |
| 12:28.30 | creativx | presence is in these days HarryR |
| 12:28.35 | HarryR | :\ |
| 12:28.51 | HarryR | it's just me being anally retentive then |
| 12:28.59 | HarryR | I want to go back to 1995 :( |
| 12:29.40 | *** join/#asterisk matsk (n=mk@194.68.102.112) |
| 12:29.44 | creativx | pffff HarryR |
| 12:29.53 | creativx | i bet google streetmaps have taken your picture already |
| 12:30.14 | HarryR | I am on loads of tourists videos |
| 12:30.44 | HarryR | they always seem to like filming the financial district in london going "wow.. its so busy" |
| 12:31.00 | creativx | canary wharf? |
| 12:31.02 | creativx | or city? |
| 12:31.16 | HarryR | city |
| 12:31.25 | creativx | mkay |
| 12:31.28 | creativx | only been to wharf |
| 12:31.32 | creativx | didnt seem to busy there |
| 12:31.33 | HarryR | there are always loads of people on london bridge filming random stuff |
| 12:31.47 | HarryR | in the rush hour it's majorly hectic |
| 12:32.29 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 12:33.36 | *** join/#asterisk vgster (n=vgster@h147170.navonline.net) |
| 12:34.27 | vgster | has anyone had any luck with aastra 53i's and distinctive rings |
| 12:37.18 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:39.57 | walhala | hi all |
| 12:44.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:46.49 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 12:47.37 | VJFROMGT | I used to be with freepbx and want to go pure asterisk now, does asterisk have a reporting module like freepbx? |
| 12:47.57 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:53.15 | s0ck | when i try to transfer a call to call park, i hit #7, before i can get to the next number, the caller gets a high pitched whine sent to them and the call appears to d/c |
| 12:53.24 | s0ck | verbose/debug reveals nothing at all |
| 12:53.55 | s0ck | blind xfer works fine... |
| 12:57.16 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:00.50 | creativx | [TK]D-Fender: by looking at the counterpath forums it does not seem like xlite likes answer-after |
| 13:00.58 | creativx | or the call-info header |
| 13:01.22 | VJFROMGT | does asterisk now add alot of extra to the conf files like trixbox do:? |
| 13:01.40 | *** join/#asterisk mirco (n=mirco@87.139.74.16) |
| 13:02.14 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 13:02.43 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:04.34 | creativx | VJFROMGT: only if you do make samples |
| 13:06.25 | [TK]D-Fender | vgster: No, * does not have a reporting module. * is one PIECE of the overall Trixbox system |
| 13:07.45 | VJFROMGT | creativx, i dont understand, what i am looking for is a bit of gui on asterisk but i must be able to hand code files also, with trixbox, original file gets overwrited once u use gui |
| 13:08.13 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 13:08.41 | [TK]D-Fender | VJFROMGT: its all or nothing with pretty much every GUI out there. If you don't like it, write your own. |
| 13:09.22 | VJFROMGT | hmm |
| 13:09.33 | VJFROMGT | guess i will have to try pure asterisk then |
| 13:10.29 | *** part/#asterisk exoxe (i=exoxe@ip70-171-16-60.ga.at.cox.net) |
| 13:10.38 | [TK]D-Fender | VJFROMGT: What are you here for if not that? Its not Raw Cat science.... |
| 13:10.49 | VJFROMGT | haha |
| 13:10.49 | creativx | pure asterisk is the shit |
| 13:11.03 | VJFROMGT | well i have been using trixbox for a while but want to migrate |
| 13:11.15 | vgster | ? trixbox? |
| 13:11.17 | VJFROMGT | but kinda scared of the non-gui |
| 13:11.26 | VJFROMGT | trixbox=freepbx |
| 13:12.10 | vgster | i dont understand the comment - vgster: No, * does not have a reporting module. * is one PIECE of the overall Trixbox system |
| 13:12.14 | creativx | i was scared of the .conf files to begin with |
| 13:12.25 | vgster | i has adked about distinctive rings |
| 13:12.35 | VJFROMGT | creat. u migrate from trix to pure? |
| 13:12.41 | creativx | i think he hit the wrong nick vgster.. |
| 13:12.48 | vgster | ok |
| 13:12.50 | creativx | VJFROMGT: nope, went straight to pure from the beginning |
| 13:12.51 | [TK]D-Fender | vgster: What don't you understand? There is no web viewing tool to visualize CDR, queue logs, etc. |
| 13:12.54 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 13:13.05 | vgster | i dont want to visualize cdrs and logs |
| 13:13.07 | [TK]D-Fender | vgster: not that is a PART of Asterisk. They are all THIRD PARTY APPS. |
| 13:13.27 | vgster | all i want to know is if anyone has setup a 53i with distinctive rings from asterisk |
| 13:13.29 | VJFROMGT | guys, vg did not ask abotu cdr, i did |
| 13:13.33 | vgster | ffs |
| 13:13.41 | [TK]D-Fender | vgster: Well stop using generic words like "reporting", since my answer covered 99% of what the common usage of that wording implies |
| 13:13.42 | creativx | heheh |
| 13:13.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.48 | creativx | [TK]D-Fender ---> its friday |
| 13:13.50 | vgster | i have used the work reporting |
| 13:13.54 | vgster | havent |
| 13:13.55 | [TK]D-Fender | Whatever.... silly auto-complete fails again |
| 13:14.07 | creativx | sure blame the computer |
| 13:14.08 | creativx | :P |
| 13:14.12 | [TK]D-Fender | creativx: Thank ^&%$##ing God |
| 13:14.17 | creativx | haha indeed |
| 13:14.28 | creativx | my job would suck if I couldnt blame the computers from time to time |
| 13:14.34 | [TK]D-Fender | creativx: There ;) |
| 13:14.38 | creativx | hehe |
| 13:14.43 | creativx | i will forward your blame to the computer dept. |
| 13:14.46 | VJFROMGT | is centos the prefered OS for asterisk? |
| 13:15.20 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 13:16.16 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:16.27 | [TK]D-Fender | VJFROMGT: It is used very often. And "preferred" depends on who you ask |
| 13:16.39 | vgster | so anyone distinctive rings, 53i's? or is this the cdr reporting and visualisation channel :D |
| 13:16.51 | vgster | ive been using centos more since i dumped suse |
| 13:17.31 | VJFROMGT | tk,, i pc has limited resources, what OS would you recommend? |
| 13:18.52 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:18.59 | Qwell | VJFROMGT: Linux |
| 13:22.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:26.55 | *** join/#asterisk mazpe (n=email@adsl-074-173-243-244.sip.bct.bellsouth.net) |
| 13:28.54 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 13:31.06 | *** join/#asterisk theBrave (n=jberniol@AMontsouris-753-1-13-117.w90-2.abo.wanadoo.fr) |
| 13:35.12 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:35.44 | *** join/#asterisk bob-b (n=bob@Jade.NetSurf.Net) |
| 13:38.01 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 13:38.25 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:39.27 | bob-b | Where should I start looking if I am hearing static on the SIP phones connected to Asterisk with inbound FXO lines on a TDM2400p card? |
| 13:40.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:43.39 | NOT_guru | are you sure the line itself is clean? |
| 13:43.56 | NOT_guru | do you have an analog phone you could plug individual lines into |
| 13:44.16 | NOT_guru | at least thats what I would do |
| 13:44.50 | *** join/#asterisk Gringo_ (n=N3TW4LK3@34.124-247-81.adsl-dyn.isp.belgacom.be) |
| 13:45.13 | Gringo_ | is it impossible to use asterisk-addons with 1.4? |
| 13:45.25 | Gringo_ | i just cannot seem to get it to compile |
| 13:45.48 | _DAW | Gringo_: I'm not having any problems. |
| 13:46.20 | Gringo_ | _DAW: what distro? |
| 13:46.40 | Gringo_ | the module I need is res_mysql |
| 13:46.42 | file | Gringo_: if you gave a pastebin URL to the output of what trying to compile it gives you... someone might be able to help you |
| 13:46.50 | _DAW | CentOS 4.4 :: Asterisk 1.4.4 |
| 13:47.05 | Gringo_ | file: okay, just a moment |
| 13:48.40 | Gringo_ | http://pastebin.com/929604 |
| 13:49.08 | file | that is an ebuild |
| 13:49.26 | Gringo_ | yes, for gentoo |
| 13:49.31 | file | and you can't use 1.2 addons with 1.4 |
| 13:49.33 | Gringo_ | do you want me to try with just the svn |
| 13:49.53 | Gringo_ | oh, res_mysql is for 1.2 only? |
| 13:50.08 | file | no, there's a 1.4 addons |
| 13:50.23 | file | you use asterisk-addons 1.4 with asterisk 1.4, and asterisk-addons 1.2 with asterisk 1.2 |
| 13:50.45 | Gringo_ | aaaah, I understand now where i've made my mistake |
| 13:50.49 | Gringo_ | damnit, that was stupid |
| 13:50.58 | Gringo_ | i had to use a gentoo overlay to get asterisk 1.4 to work |
| 13:51.08 | *** join/#asterisk redlob (i=dbolderm@xs3.xs4all.nl) |
| 13:51.10 | Gringo_ | but didn't use that same overlay for the addons |
| 13:51.19 | Gringo_ | haha, sorry to waste your time :) |
| 13:51.24 | Gringo_ | tnx, file |
| 13:51.26 | Gringo_ | ! |
| 13:53.13 | [TK]D-Fender | Genpooooooooooooooooooooooooooooooooo! |
| 13:53.38 | Gringo_ | trolling, are we? ;) |
| 13:55.58 | Gringo_ | it's actually very easy to bash gentoo :) i'm doing a complete install as we speak |
| 13:56.08 | Gringo_ | it'll be done by the day after tomorrow :) |
| 13:56.32 | [TK]D-Fender | PATIENCE?!?!?!? yeah yeah, how long will THAT TAKE?! |
| 13:56.34 | Gringo_ | openoffice alone takes 5 hours to compiile |
| 13:57.59 | blitzrage | jeebuz |
| 14:00.48 | tzanger | I sent it to blitzrage but you guys may as well have the fun too |
| 14:01.00 | tzanger | http://www.youtube.com/watch?v=kYvZnTFpip0 |
| 14:01.06 | dreamind | finally it works, but not the way I wanted :( |
| 14:01.09 | dreamind | anyhow, bye |
| 14:01.14 | tzanger | Gringo_: yeah and I bet you see that 3% speed increase too |
| 14:01.23 | *** join/#asterisk foxtrot- (n=lfc@c90696a5.static.spo.virtua.com.br) |
| 14:01.53 | foxtrot- | Hey, does anybody know how to reload my zaptel configuration, i have added more channels, but its not listing and nor using it |
| 14:02.02 | Gringo_ | tzanger: no :) you can't tell the difference |
| 14:02.20 | Gringo_ | however, the installation is the only annoying bit |
| 14:02.50 | Gringo_ | because it's quite acceptable afterwards, most of the programs you need to install afterwards are compiled within 5 mins |
| 14:05.22 | ManxPower | foxtrot-: you must stop and start asterisk or unload chan_zap.so and load chan_zap.so |
| 14:05.24 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:05.36 | tzafrir_laptop | foxtrot-, 'reload' or 'reload chan_zap.so' will reload zapata.conf (and users.conf's zaptel config) |
| 14:05.52 | tzafrir_laptop | ztcfg will reload /etc/zaptel.conf changes... |
| 14:06.28 | tzafrir_laptop | several changes, such as signaling changes, adding / removing channels will not apply on reload |
| 14:06.43 | *** join/#asterisk perf3kt (i=perf3kt@149.166.33.155) |
| 14:06.44 | ManxPower | tzanger: reload will NOT add/remove channels |
| 14:06.54 | ManxPower | oh, you said that already |
| 14:07.00 | tzanger | ManxPower: indeed I did :-p |
| 14:07.04 | tzafrir_laptop | foxtrot-, also: if you have just analog channels, use: zap restart |
| 14:07.08 | ManxPower | Where is my coffee??? |
| 14:07.14 | tzanger | ManxPower: IN MUH BELLY |
| 14:07.31 | perf3kt | can I use asterisknw to get asterisk and linux loaded onto a machien and just use the cli to configure the files |
| 14:07.34 | tzafrir_laptop | for digital channels with spans I'm not sure exactly what it does... |
| 14:07.38 | ManxPower | We are getting our first significant rain in almsot 4 months. |
| 14:07.39 | perf3kt | is that acceptabel to the cli users? |
| 14:08.03 | *** part/#asterisk Gringo_ (n=N3TW4LK3@34.124-247-81.adsl-dyn.isp.belgacom.be) |
| 14:08.05 | ManxPower | perf3kt: perhaps you could ask on #asterisknow |
| 14:08.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:08.39 | *** join/#asterisk Gringo_ (n=N3TW4LK3@34.124-247-81.adsl-dyn.isp.belgacom.be) |
| 14:08.41 | perf3kt | manxpower: well no I ask because I'll be looking for support here, but I konw you all hate *now |
| 14:09.17 | tzafrir_laptop | ManxPower, the qiestion about CLI users is a good question for here... |
| 14:09.21 | ManxPower | perf3kt: Most of us don't even know how *Now sets up it's config files. |
| 14:09.24 | perf3kt | manxpower: but for the sake of simplicity just to get all the packages and * installed I was aking if it woudl be okay to just the *now cd iso |
| 14:09.41 | ManxPower | So, I guess if you rewrote all the config files from scratch we could help you |
| 14:09.48 | tzafrir_laptop | one thing I find a bit "surprising" about asterisknow is the editing of zapata.conf / zaptel.conf |
| 14:09.49 | tzafrir_laptop | by the zaptel init.d script |
| 14:09.53 | tzanger | ManxPower: it's been dry here too |
| 14:10.04 | perf3kt | manxpower: but its the same program, if I delete the configs and start from scratch, editing them myself |
| 14:10.12 | tzafrir_laptop | Also: this is a matter of using or not using the asterisk GUI |
| 14:10.15 | ManxPower | perf3kt: yes. |
| 14:10.29 | tzafrir_laptop | currently asterisknow runs asterisk as root. I'm not really sure what it takes to change that |
| 14:10.32 | ManxPower | tzanger: what is suprizing? |
| 14:11.08 | tzafrir_laptop | ManxPower, did you ask me? |
| 14:11.11 | perf3kt | manxpower: cool, i mean I have centOS currently on a machine, but I'm not a wiz with linux and dont' know what packages are required for 1.4 |
| 14:11.35 | ManxPower | (09:09:42) tzafrir_laptop: one thing I find a bit "surprising" about asterisknow is the editing of zapata.conf / zaptel.conf |
| 14:11.39 | ManxPower | I asked what did you find suprizing |
| 14:11.58 | tzafrir_laptop | perf3kt, essentially the same that are needed for 1.2 . There is someadded functionality |
| 14:12.15 | tzafrir_laptop | so you can get iksemel if you want jabber support |
| 14:12.37 | ManxPower | perf3kt: Unfortunately if you expect to use asterisk effectively you are going to learn linux. |
| 14:12.58 | ManxPower | You are also going to learn networking, telecom, and protocols |
| 14:13.27 | tzafrir_laptop | ManxPower, the init script edits zaptel.conf / zapata.conf for you and oerrides yur choices. |
| 14:13.27 | tzafrir_laptop | You can't set an analog channel to be loopstart, for instance |
| 14:13.34 | [TK]D-Fender | perf3kt: you are looking for the "EVERYTHING" button. Just do it. |
| 14:13.56 | ManxPower | tzafrir_laptop: Ah, OK. |
| 14:14.44 | tzafrir_laptop | but I guess that the real question is: does it run as root |
| 14:15.57 | perf3kt | tk: what? |
| 14:16.16 | *** part/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 14:18.08 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-152-135.ny325.east.verizon.net) |
| 14:19.11 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 14:20.29 | [TK]D-Fender | perf3kt: In your CentOS install, you were wondering what packages, etc to install. At the bottom of the list is an option for EVERYTHING. |
| 14:20.44 | [TK]D-Fender | perf3kt: Pick it and you'll be gauranteed to ahve everything you need for *. |
| 14:20.55 | ManxPower | tzafrir_laptop: Well it is called AsteriskNOW and not AsteriskGreat |
| 14:22.01 | [TK]D-Fender | ManxPower: Sign my petition to rename it "FunctionalLATER!" |
| 14:22.16 | tzafrir_laptop | yes, recent asterisknow (from beta 5.5) still runs as root |
| 14:22.32 | ManxPower | I'm Looking for Asterisk Mr. Right, not Asterisk Right Now |
| 14:23.54 | s0ck | [TK]D-Fender: any ideas why one of my sip trunks disconnects after n seconds |
| 14:24.00 | s0ck | where n is anything up to 24 seconds |
| 14:24.06 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 14:24.13 | s0ck | since i added qualify, it's more like 5 seconds |
| 14:24.29 | coppice | NOW is in capitals. it must stand for something |
| 14:24.38 | coppice | Not Often Working? :-\ |
| 14:24.55 | tzafrir_laptop | Actually it does work |
| 14:24.55 | foxtrot- | hey guys, im using unicall instead of zaptel, and i already have 60 channels going, i was wondering what are the steps to create 30 more channels and make asterisk recognize them, can anybody help me? |
| 14:25.03 | tzafrir_laptop | did you get to look at the asterisk gui? |
| 14:25.12 | tzafrir_laptop | it is nice, and it work |
| 14:25.16 | tzafrir_laptop | works |
| 14:25.28 | *** part/#asterisk redlob (i=dbolderm@xs3.xs4all.nl) |
| 14:26.10 | tzafrir_laptop | if you let it just show configuration and not write configuration, it can be a very handy tool sometimes |
| 14:26.32 | mmlj4 | unicall? |
| 14:26.58 | mmlj4 | hey ManxPower |
| 14:27.10 | foxtrot- | unicall is the zaptel module for r2mfc e1circuits |
| 14:27.16 | tzafrir_laptop | mmlj4, something a guy on hongkong does |
| 14:27.23 | mmlj4 | ah. |
| 14:27.25 | foxtrot- | i have already changed unicall.conf and reload chan unicall.so |
| 14:27.57 | foxtrot- | then i reloaded zaptel.conf by using ztcfg |
| 14:28.38 | foxtrot- | is this correct? can you help me? |
| 14:29.05 | perf3kt | tk: thanks, actually I selected minimal, I only had cd 1... |
| 14:29.20 | tzafrir_laptop | seems like you need to refresh zaptel first, and only then userspace (chan_unicall) could be aware of the changes |
| 14:29.36 | tzafrir_laptop | hence ztcfg first, reload chan_unicall.so later |
| 14:30.03 | s0ck | disconnects bang on 6 seconds every time lol |
| 14:30.18 | foxtrot- | im sorry, thats what i did...first i ztcfg and then reload ...unical.so |
| 14:31.11 | tzafrir_laptop | also, IIRC chan_unicall.so behaves the same as chan_zap: not adding channels on reload |
| 14:31.41 | tzafrir_laptop | maybe unload chan_unicall.so and re-load it as ManxPower suggested earlier for chan_zap.so |
| 14:32.28 | foxtrot- | can you be more specific?? |
| 14:32.30 | tzafrir_laptop | or fully restart asterisk |
| 14:32.56 | foxtrot- | fully restart would do great, but i have a workstation here with 150 people using it |
| 14:32.58 | foxtrot- | hehehe |
| 14:33.54 | tzafrir_laptop | what verion of asterisk is it? |
| 14:34.11 | tzafrir_laptop | unload chan_unicall.so |
| 14:34.16 | tzafrir_laptop | load chan_unicall.so |
| 14:34.17 | foxtrot- | asterisk 1.0.9 |
| 14:34.49 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 14:35.04 | foxtrot- | unload & load == reload, right? |
| 14:35.40 | tzafrir_laptop | wow. chan_unicall works with it? nice |
| 14:36.19 | foxtrot- | why surprised?:) |
| 14:36.39 | *** join/#asterisk `Sean (i=Un1x@CPE000c258d147c-CM000a73a94167.cpe.net.cable.rogers.com) |
| 14:37.52 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 14:38.11 | foxtrot- | do asterisk accept reload cmd? |
| 14:38.46 | *** join/#asterisk ghento (n=ghento@CPE001124d2c50e-CM0011e6c416f1.cpe.net.cable.rogers.com) |
| 14:39.11 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:41.11 | foxtrot- | tzafrir after doing reload chan unicall.so , do i still need to reload chan_zap? |
| 14:47.31 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:50.02 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f201ba001471e699) |
| 14:51.13 | tzafrir_laptop | for the others here: reload != unload + load , AFAIK |
| 14:52.40 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 14:53.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:54.05 | *** join/#asterisk angryuser (n=Miranda@df01t2-195-36-137-125.d4.club-internet.fr) |
| 14:55.20 | *** join/#asterisk fonnerk (n=fonnerk@dsl093-012-082.cle1.dsl.speakeasy.net) |
| 14:56.04 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 14:57.18 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:58.43 | fonnerk | I just bought a digium tdm card so that I could set up an asterisk system, however I am having trouble getting the card to inialize properly... Is this a good place to ask for help? |
| 14:59.26 | blitzrage | if you bought a card from Digium, you can call their support for help |
| 14:59.33 | blitzrage | that's your best bet |
| 14:59.33 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:59.42 | fonnerk | excellent... thanks |
| 14:59.44 | blitzrage | all hardware comes with an hour of setup support |
| 14:59.58 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 15:00.07 | fonnerk | did I have to buy it directly from digium? |
| 15:00.10 | asteriskguy | hello all |
| 15:00.22 | asteriskguy | is there an irc chat like this for Dundi? |
| 15:00.41 | asteriskguy | or a better question, has anyone here ever used Dundi before? |
| 15:01.05 | UCFmethod | _/list isnt working at the moment, so I don't know about a room... |
| 15:01.13 | tzafrir_laptop | asteriskguy, #asterisk is the channel for dundi, I guess |
| 15:01.40 | asteriskguy | that's cool, thanks both you guys |
| 15:01.50 | tzafrir_laptop | I don't know if there is actually any other program that implements it (well, maybe callweaver) |
| 15:01.59 | asteriskguy | has anyone here ever worked with Dundi? |
| 15:01.59 | blitzrage | there used to be a #dundi, but no one is there anymore |
| 15:02.18 | *** join/#asterisk [jwb] (n=me@schizophrenia.paravolve.net) |
| 15:02.22 | blitzrage | asteriskguy: just ask your question -- people generally aren't going to say, "ya, I used it", because they know you'll jump all over them |
| 15:02.36 | blitzrage | and DUNDi isn't that hard really |
| 15:03.29 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:03.47 | blitzrage | two sources of documentation: |
| 15:03.49 | blitzrage | ~book |
| 15:03.50 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:03.51 | blitzrage | http://leifmadsen.com/papers/dundi-intro.pdf |
| 15:04.07 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:04.15 | blitzrage | both cover DUNDi |
| 15:04.25 | anonymouz666 | anyone in here already used the pap2 to pass through FAX ? |
| 15:04.45 | anonymouz666 | I am using g711 sometime works, sometimes does not work. |
| 15:05.24 | anonymouz666 | what is impressive I have some grandstream running on LAN here and works all the time using 711 |
| 15:05.48 | UCFmethod | would anyone like to share a bit of advice... I am going to upgrade from 1.2 to 1.4 this morning/afternoon... anything drastically different, and pointers or hints anyone would love to share |
| 15:07.22 | coppice | If you haven't worked that out, I think it might be a bit premature to upgrade today :-\ |
| 15:07.39 | blitzrage | UCFmethod: I hope that's on a test system |
| 15:07.52 | blitzrage | if you're doing it to a production system... yer just stupid |
| 15:09.57 | UCFmethod | of course on a test server.... come on now.... I meant.... oh this app Meetme() is called ConfRoom() now or something along those lines that people have noticed and worked around. I read through the UPGRADE.txt but nothing jumped out at me |
| 15:10.19 | blitzrage | UCFmethod: you'd be surprised at the kind of people who come in here... :) |
| 15:10.32 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 15:10.39 | coppice | dismayed might be more accurate |
| 15:10.59 | brea | Is it possible to use DIDs over analog and a TDM400P? |
| 15:11.10 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:11.58 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:14.02 | VJFROMGT | what is the difference between a voip gateway and a voip ata? |
| 15:14.22 | coppice | about 100% added to the price |
| 15:14.22 | rob0 | ~ata |
| 15:14.23 | jbot | i guess ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 15:14.53 | asteriskguy | thanks blitzrage |
| 15:14.59 | waKKu | folks.. someone there using/know this equip: Grandstream HT488 ? - http://www.myphonecall.co.uk/pt/voip/telephoneadapters/grandstream/grandstream_handytone_488.aspx |
| 15:15.35 | waKKu | i'm thinking to use it instead of TDM400P .. what do u think ? |
| 15:15.37 | rob0 | Hmm, isn't any ATA by definition a VoIP device? |
| 15:16.09 | angryuser | i am searching a nice headset for snom 360 |
| 15:16.15 | UCFmethod | side note, how longs has "make menuselect" been here |
| 15:16.21 | UCFmethod | f'ing sweet |
| 15:16.41 | *** join/#asterisk n00dle (n=ccraft@officewall.springsips.com) |
| 15:16.41 | rob0 | waKKu: people here who seem to know consistently say that Grandstreams are junk. (I'm just relaying this, never had one.) |
| 15:16.59 | n00dle | rob0, I use GS... no probs. |
| 15:17.05 | waKKu | hehe.. |
| 15:17.13 | waKKu | just in time n00dle ;) |
| 15:17.24 | *** join/#asterisk jtoy_ (n=jtoy@mail.backchannelmedia.com) |
| 15:17.30 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:17.33 | coppice | if you assumed that everything someone here says is junk really is junk, you wouldn't use anything |
| 15:17.33 | waKKu | n00dle do u know GS HT 488 ? http://www.myphonecall.co.uk/pt/voip/telephoneadapters/grandstream/grandstream_handytone_488.aspx |
| 15:17.40 | n00dle | Sure their firmware seems quirky, but they're quite usable and I drive all the features I can from *. |
| 15:17.54 | n00dle | no, I only have BT101 and GXP2000 |
| 15:18.06 | rob0 | I've also heard they're cheaply made. |
| 15:18.34 | rob0 | I avoided GS devices on the advice from this channel. |
| 15:19.01 | n00dle | They're a tad light, and you have to make sure that your power connector stays tight, other than that... |
| 15:19.13 | rob0 | I have a TDM400 at one site, Sipura ATA's here. |
| 15:19.27 | Teccy | i'm having an odd issue with a TDM400 w/2FXO |
| 15:19.34 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:19.45 | n00dle | So I have a question... anyone using meetme and/or SLA? |
| 15:20.21 | UCFmethod | n00dle: I am sure most people use meetme... whats up? |
| 15:20.45 | rob0 | The only issue I have with Sipuras is an 8-10 second delay between dialing and the dial() application being executed in *. |
| 15:20.53 | Teccy | it's connected to a toshiba strata CT PBX. If i ring it from a PBX phone, * shows 'Starting simple switch on Zap1-1' and follows the correct dialplan, however, the PBX phone continues to ring and the asterisk connection is never made |
| 15:20.59 | Teccy | any thoughts? |
| 15:21.19 | russellb | n00dle: i'm about to commit a patch for your issue ... give it a try and let me know if it does what you need |
| 15:21.35 | n00dle | Well, the SLA code uses meetme to do its thing, but I'm missing an important functionality if I use it... can't transfer a call to voicemail. |
| 15:21.51 | Teccy | also, if i try calling out through the zaptel device from a sip phone, the destination phone never rings, and i only hear silence on the line |
| 15:22.05 | Teccy | but according to * the connection has been made |
| 15:22.41 | russellb | n00dle: why can't you do that? |
| 15:22.49 | *** join/#asterisk gerwinin (n=gerwinin@ip5457b30e.direct-adsl.nl) |
| 15:23.10 | russellb | n00dle: if you create an extension in the context the SIP phones use that goes to voicemail, the SIP phone's transfer button should let you do it |
| 15:23.22 | gerwinin | I would like to invite the asterisk project for our event who should I contact ? |
| 15:23.33 | russellb | gerwinin: what is the event? |
| 15:23.59 | russellb | gerwinin: in general, contact the marketing department at Digium |
| 15:24.09 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 15:24.10 | santibiotico | hi |
| 15:24.15 | gerwinin | It is an event in the netherlands for opensource projects |
| 15:24.18 | santibiotico | any help with AGI+php? |
| 15:24.38 | santibiotico | i'm trying to run my first php script using agi |
| 15:24.42 | blitzrage | santibiotico: best bet is to ask a specific question... |
| 15:24.51 | n00dle | russellb, Nope... doesn't seem to do it... hang on, I'll try again. |
| 15:24.53 | russellb | gerwinin: contact marketing and tell them to send russellb :-D |
| 15:25.07 | blitzrage | and blitzrage :D |
| 15:25.08 | *** join/#asterisk bintut (n=bintut@cm14.gamma177.maxonline.com.sg) |
| 15:25.12 | gerwinin | russelb okay |
| 15:25.30 | gerwinin | santibiotico what is your problem ? |
| 15:25.34 | rob0 | Send a chartered A300 or MD11 to KHSV ... Digium will see to it that it gets filled. |
| 15:25.40 | bintut | anyone here running debian etch i386? how can i install fxoload? |
| 15:26.01 | gerwinin | santibiotico : I worked quite a lot with php_agi :) |
| 15:26.02 | santibiotico | the 1st thing i do is to create $stdin, $stdout and $stdlog |
| 15:26.09 | santibiotico | then |
| 15:26.16 | gerwinin | ok |
| 15:26.25 | blitzrage | santibiotico: did you check out the PHP/AGI section in the TFoT book? (that's where I started... :)) |
| 15:26.25 | santibiotico | when i try to do sth with stdin it just crashes |
| 15:26.25 | angryuser | i am searching a wireless snom 360 headset, with the possibility to answer the call with the headset button, where can i find one? |
| 15:26.30 | *** join/#asterisk gigot (n=gigot@mea77-2-82-239-228-128.fbx.proxad.net) |
| 15:26.44 | santibiotico | blitzrage: nops |
| 15:26.57 | gerwinin | santibiotico , with which message does it crash ? |
| 15:27.00 | blitzrage | santibiotico: give that a read.... it might help |
| 15:27.01 | waKKu | angryuser did u already see hs810 motorola ? |
| 15:27.13 | santibiotico | blitzrage: i've started with some wiki from voip-info |
| 15:27.20 | blitzrage | ~book |
| 15:27.21 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:27.30 | blitzrage | jbot is responding very fast today |
| 15:27.31 | santibiotico | gerwinin: with no error message |
| 15:27.34 | russellb | that book is great, the guys that wrote it are such a pain, though |
| 15:27.45 | blitzrage | ya... I heard that Leif guy is an ASSHOLE |
| 15:27.46 | angryuser | waKKu: no will see thx |
| 15:27.54 | n00dle | russellb, Hmm... I hit transfer, get MOH on caller and dialtone on the set, dial the "xfer to voicemail" for the mbox I want to drop the call in to... then hang up... |
| 15:28.35 | blitzrage | santibiotico: this is how I start mine.... |
| 15:28.37 | blitzrage | / Be sure STDIN, STDOUT and STDERR are defined |
| 15:28.37 | blitzrage | $stdin = fopen('php://stdin', 'r'); |
| 15:28.37 | blitzrage | $stdout = fopen('php://stdout', 'w'); |
| 15:28.37 | blitzrage | $stderr = fopen('php://stderr', 'w'); |
| 15:28.41 | russellb | n00dle: you're probably not supposed to hang up ... most phones have you press a button after dialing the extension for the transfer |
| 15:28.56 | santibiotico | blitzrage: they are |
| 15:29.37 | n00dle | I've tried.... maybe I'll have to do an ether sniff for SIP. |
| 15:31.51 | bintut | how can i install fxoload? |
| 15:32.18 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 15:34.03 | angryuser | waKKu: it is useless for snom 360 i need a headset to plug in to it, or you have some wayt to connect a bluetooth headset to snom's jack? |
| 15:34.31 | tzafrir_laptop | bintut, fxload? |
| 15:34.32 | tzafrir_laptop | on which distro? |
| 15:34.39 | n00dle | russellb, Ok, this time I actually read the manual on attended and blind transfers (SO different from traditional systems) and the caller remains on MOH while the CLI says it's playing the VM unavailable message... Hmmm. |
| 15:35.09 | the_5th_wheel | is it poosible to setup sms sending ove the isdn modem, connected to either an bri-ftc or a normal bri? |
| 15:35.59 | gerwinin | the_5th_wheel: yes if you figure out your dtmf settings correctly |
| 15:36.01 | russellb | n00dle: stop finding bugs |
| 15:36.07 | blitzrage | heh |
| 15:36.49 | the_5th_wheel | gerwinin: explain further please, im a noob to asterisk |
| 15:36.56 | russellb | blitzrage: it's quite out of hand |
| 15:37.03 | russellb | blitzrage: actually, i'd like to just hire you. |
| 15:37.13 | russellb | but I don't get to do that (yet) :) |
| 15:37.14 | gerwinin | the_5th_wheel: sms over isdn and analog is dtmf based |
| 15:37.42 | n00dle | russellb, well... I could just abandon SLA entirely and re-code the entire dialplan, and then retrain our entire staff. |
| 15:37.52 | n00dle | ;) |
| 15:37.55 | russellb | n00dle: that's one option, yes. |
| 15:37.58 | gerwinin | the 5th_wheel as long as you find a way that asterisk is not making those dtmfs to short or to long you can receive and send the dtmfs correctly |
| 15:37.59 | russellb | and then i don't have to do anything |
| 15:38.09 | russellb | n00dle: but seriously, let me test that here |
| 15:38.18 | gerwinin | the_5th_wheel: which country are you living in ? |
| 15:38.53 | n00dle | russellb, that was attended transfer... blind transfer got the caller stuck in MOH while the transfer-er got the voicemail box. |
| 15:39.09 | russellb | the transferer got the voicemail box? that's bizarre :) |
| 15:39.12 | gerwinin | the_5th_wheel: phone--------------> asterisk--------------> analog --------------> sms server |
| 15:39.14 | the_5th_wheel | so i would just send some dtmf tones over audio line? i would have thought that one would done that digitally |
| 15:39.26 | n00dle | russellb, That's what I thought. |
| 15:39.31 | the_5th_wheel | gerwinin: south africa |
| 15:39.50 | gerwinin | the_5th_wheel: you want to sms from a voip phone ? |
| 15:39.55 | the_5th_wheel | when you say sms server you mean smscenter? |
| 15:40.01 | gerwinin | yes |
| 15:40.05 | russellb | n00dle: well, if you just give me time, i'll make sure to fix all of your bugs with using SLA. You have been the most helpful and responsive user of it yet, by far. |
| 15:40.16 | the_5th_wheel | gerwinin: i want to be able to send smsses from my bri |
| 15:40.27 | the_5th_wheel | so from probably the asterisk server |
| 15:40.38 | n00dle | russellb, Glad to help. :) The boss is getting impatient, but I think I can manage that. |
| 15:41.04 | russellb | n00dle: tell him that free != done yesterday :-p |
| 15:41.09 | gerwinin | the_5th_wheel: than you need to make a number on which the bri is accessable |
| 15:41.44 | gerwinin | the_5th_wheel: and than you send the dtmf to this number |
| 15:42.04 | gerwinin | the_5th_wheel: Are you doing this for a specific application ? |
| 15:42.04 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 15:43.43 | *** part/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 15:44.35 | the_5th_wheel | gerwinin: i will be getting an ftc from a local company, and i will be using this for all our communications (its a community otchestra) |
| 15:45.32 | n00dle | russellb, SHE knows... :) ...but she has many other projects going that distract her from the phones not being entirely done yet. |
| 15:45.52 | russellb | n00dle: ha, my bad on the poor gender assumption. |
| 15:47.03 | n00dle | Happens all the time. |
| 15:48.27 | *** join/#asterisk essenzolo (n=xtc@unaffiliated/essenza) |
| 15:48.48 | essenzolo | are anyone for support me ? |
| 15:48.49 | gerwinin | the_5th_wheel: what is an ftc ? |
| 15:49.02 | the_5th_wheel | gerwinin: a premicell |
| 15:49.21 | essenzolo | are any application for count the time and the cost for my call center? |
| 15:50.01 | gerwinin | the_5th_wheel okay |
| 15:50.13 | gerwinin | essenzolo: yes |
| 15:50.32 | essenzolo | oh good where i can find it? |
| 15:51.13 | gerwinin | essenzolo: http://trac.asterisk2billing.org/cgi-bin/trac.cgi |
| 15:51.13 | essenzolo | gerwinin: where i can find it? |
| 15:51.22 | essenzolo | tnx !!! |
| 15:51.28 | gerwinin | the_5th_wheel: I will help you |
| 15:51.41 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:52.25 | the_5th_wheel | I dont have the Hardware as yet, but i should have it somewhere soon. |
| 15:52.34 | the_5th_wheel | but aslong as it is doable im happy |
| 15:52.40 | gerwinin | the_5th_wheel: okay let me know if you have it :) |
| 15:53.09 | *** join/#asterisk perlmonke (n=perlmonk@hubert.perlmonkee.com) |
| 15:53.16 | gerwinin | essenzolo: take into account that it is a beta so you probarly need to add something , most important for callcenters is peering although |
| 15:53.46 | brea | Is it possible to use DIDs over analog and a TDM400P? Like some sort of analog trunking? |
| 15:53.54 | gerwinin | brea: yes |
| 15:54.06 | perlmonke | Inbound routing of multiple DIDs from a single SIP registration... I've been told there is a "much" easier way to do this other than parsing the "To" header "manually" in the dial plan... can anyone point me in the right direction? |
| 15:54.36 | brea | gerwinin: Well then... how would I go about doing this? |
| 15:55.06 | gerwinin | the_5th_wheel: sent me a mail on gerwin@vanderkruis.net so we can stay in touch and I can help you further out |
| 15:55.15 | *** join/#asterisk blackbyte01_ (n=blackbyt@89.119.146.121) |
| 15:55.16 | perlmonke | brea: distinctive ring, etc. |
| 15:55.35 | gerwinin | brea: see perlmonke :) |
| 15:55.43 | blackbyte01_ | hi! |
| 15:56.04 | blackbyte01_ | can i explain my question? |
| 15:56.04 | Teccy | i'm having an odd issue with a TDM400 w/2FXO |
| 15:56.10 | Teccy | it's connected to a toshiba strata CT PBX. If i ring it from a PBX phone, * shows 'Starting simple switch on Zap1-1' and follows the correct dialplan, however, the PBX phone continues to ring and the asterisk connection is never made |
| 15:56.19 | Teccy | any thoughts anyone? |
| 15:56.20 | gerwinin | blackbyte: go ahead |
| 15:56.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:56.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:56.36 | blackbyte01_ | thks, i own an Internet Point in Rome |
| 15:56.41 | russellb | Teccy: it's supposed to keep ringing until asterisk answers the line |
| 15:56.48 | blackbyte01_ | and i'd like to start a VOIP business |
| 15:56.51 | russellb | which may be done by an application, or explicitly using the Answer() application |
| 15:57.02 | russellb | Teccy: and if that doesn't help, you'll have to talk to support@digium.com |
| 15:57.03 | Teccy | russellb: the dialplan does have an Answer |
| 15:57.08 | gerwinin | Blackbye: okay |
| 15:57.21 | blackbyte01_ | i installed asterisk |
| 15:57.27 | blackbyte01_ | and it works well |
| 15:57.32 | gerwinin | Blackbyte: okay |
| 15:57.35 | blackbyte01_ | i have 10 terminals |
| 15:57.47 | blackbyte01_ | but i need a GUI to calculate the time |
| 15:57.56 | blackbyte01_ | and my charges |
| 15:58.19 | gerwinin | blackbyte : use a2billing or make something yourself with php-agi :) |
| 15:58.30 | russellb | n00dle: i was able to recreate the attended transfer weirdness ... i'll work on it today |
| 15:58.51 | gerwinin | Blackbyte: I think a2billing is not available in italian but I am not sure |
| 15:59.08 | blackbyte01_ | gerwinin: That's not a problem |
| 15:59.10 | n00dle | russellb, Thanks. :) |
| 15:59.20 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 15:59.25 | blackbyte01_ | gerwinin: Thanks! |
| 15:59.33 | gerwinin | blackbyte: :) |
| 15:59.54 | gerwinin | Blackbyte: take good care of your peering , the better your peering the lower your rate :) |
| 16:00.57 | bintut | tzafrir_laptop: i already managed to install it.. thanks.. :) |
| 16:01.17 | russellb | n00dle: i know exactly why this is happening ... yikes ... i'm going to have to ponder this one for a bit |
| 16:01.52 | *** join/#asterisk seele_ (n=seele@200.30.85.186) |
| 16:02.03 | russellb | n00dle: but for now, can I ask you a bit about expected behavior? Like, let's say there are 3 sets on that line, and one phone transfers ... are all parties transferred? |
| 16:02.10 | n00dle | Ok... ponder away. I only discovered this one yesterday evening. |
| 16:02.17 | n00dle | I thought about that this morning... |
| 16:02.19 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:02.44 | syzygyBSD | how can I customize the voicemail email sent to pagers? |
| 16:02.45 | russellb | n00dle: like, would all of the other parties all be talking to Voicemail, together? |
| 16:02.56 | russellb | syzygyBSD: voicemail.conf |
| 16:02.59 | seele_ | please help how can i test or configure the mail server on asterisk to voicemail ?? |
| 16:03.00 | syzygyBSD | I see how to change the normal one, but the pager email looks different |
| 16:03.32 | n00dle | ...but I would expect that if there were a feature-code like transfer (say *4 for group) that would transfer everyone else and *2 would just transfer the calling trunk. Otherwise, a "regular" transfer button thingy would only do the trunk. |
| 16:04.07 | russellb | i'm not even sure i can make the transfer button work the way this is |
| 16:04.14 | russellb | this really was not intended to support transfer at all. |
| 16:04.29 | russellb | just basic shared lines ... |
| 16:04.48 | gerwinin | russelb: I was working on a sip phone a while with a csr chipset they had some weird behavious as well in the sip headers |
| 16:05.56 | russellb | anyway, i'll think about it for a while ... |
| 16:05.58 | gerwinin | russelb: fixed that on the phone than it had some audio problem but that had to do with electrical design of the phone so that went back to my collegue |
| 16:05.59 | syzygyBSD | russellb: thanks, it wasn't on voip-info though, why I asked at all, didn't realize there was better documentation elsewhere |
| 16:06.23 | russellb | syzygyBSD: no problem, i don't know if it is supported for sure, i just know that if it was, it would be in voicemail.conf.sample |
| 16:06.35 | russellb | and on that note, it's lunch time. |
| 16:08.18 | seele_ | what default email server use asterisk??? |
| 16:08.31 | seele_ | sendmail, exim, postfix or qmail ??? |
| 16:08.31 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 16:08.36 | gerwinin | I think it is sendmail but coorect me if I am wrong |
| 16:08.44 | Nugget | asterisk has nothing to do with your mail server. |
| 16:08.47 | syzygyBSD | well, not a server, just a program, and I am pretty sure it is sendmail |
| 16:08.51 | seele_ | and how can I change it or configure it?? |
| 16:08.54 | *** part/#asterisk jtoy_ (n=jtoy@mail.backchannelmedia.com) |
| 16:09.06 | syzygyBSD | for voicemail... it is an option in voicemail.conf |
| 16:09.15 | syzygyBSD | http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 16:09.25 | Nugget | yes, but asterisk doesn't INCLUDE a mail server. It uses whatever you have. |
| 16:09.43 | n00dle | russellb, Ok,... 99.9% of the time I expect that there would only be one party on the inside of the call and one party on the trunk that would be transferring. Group transfer is just a weird thing. |
| 16:09.52 | denon | voicemail.conf has a directive like: mailcmd=/usr/sbin/sendmail -t |
| 16:10.17 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:10.38 | Nugget | which works with sendmail, postfix, qmail, and exim. :) |
| 16:10.52 | seele_ | denon, ok thanks |
| 16:10.54 | denon | nod |
| 16:11.02 | denon | thats got nothing to do really with sendmail daemon |
| 16:11.16 | denon | but you could change it to something else if you wanted, I guess |
| 16:11.39 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 16:17.37 | tzafrir_laptop | or even with queue-less mailers such as nullmailer and ssmtp, if you don't care loosing a mail message or two... |
| 16:22.07 | *** part/#asterisk Gringo_ (n=N3TW4LK3@34.124-247-81.adsl-dyn.isp.belgacom.be) |
| 16:22.49 | LeBowlingAlley | Is there a difference between when a parked call times out and rings back to the original extension AND what is referred to as an "orphaned call"? |
| 16:22.51 | seele_ | where can I configure the default voicemail message text ??? |
| 16:23.55 | zeeesh | Can't locate LWP.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi ? |
| 16:27.18 | UCFmethod | seele_: voicemail.conf |
| 16:27.31 | file | zeeesh: start cpan and install LWP |
| 16:27.47 | UCFmethod | seele_: emailbody= blah blah |
| 16:27.52 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 16:31.13 | zeeesh | <file>: i tried through cpan sometime and geeting msg"Can't locate object method "data" via package "CPAN::Modulelist" (perhaps you forgot to load "CPAN::Modulelist"?) at (eval 20) line 1. |
| 16:31.14 | zeeesh | <PROTECTED> |
| 16:31.26 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 16:31.33 | zeeesh | Can't locate object method "data" via package "CPAN::Modulelist" (perhaps you forgot to load "CPAN::Modulelist"?) at (eval 20) line 1. |
| 16:31.33 | zeeesh | <PROTECTED> |
| 16:33.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:34.49 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 16:34.59 | _VoiceMeUp_COM | did anyone implement a good test script for pris ? |
| 16:35.08 | _VoiceMeUp_COM | coz this one falls once every 4 days |
| 16:35.27 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 16:35.43 | _VoiceMeUp_COM | wel asterisk zap gives dead locks.. so i was thinking of assignin the BTN number as test numbers. that push to a script.. and remote box should get a certain response from script |
| 16:35.53 | seele_ | UCFmethod, ok thanks |
| 16:38.06 | _VoiceMeUp_COM | Ring requested on channel 0/2 already in use on span 1. Hanging up owner. |
| 16:38.14 | _VoiceMeUp_COM | this is waht happends.. i think its a bad hangup |
| 16:38.31 | _VoiceMeUp_COM | EX: asterisk thinkgs it hung up.. but zap never got the command |
| 16:38.41 | Corydon76-work | or glare |
| 16:38.44 | _VoiceMeUp_COM | so on next call it assigns same port.. ex 0/2 and zap complains |
| 16:38.49 | _VoiceMeUp_COM | do i make sense ? |
| 16:39.23 | Corydon76-work | More likely, you requested something from remote and remote decided not to let you have it |
| 16:39.43 | _VoiceMeUp_COM | heu |
| 16:40.30 | _VoiceMeUp_COM | channel.c: Avoiding initial deadlock for |
| 16:40.41 | _VoiceMeUp_COM | hmm so when that happends a show channels will crap otu |
| 16:40.42 | _VoiceMeUp_COM | out |
| 16:40.47 | _VoiceMeUp_COM | and needs a killall -9 |
| 16:42.42 | perlmonke | Inbound routing of multiple DIDs from a single SIP registration... I've been told there is a "much" easier way to do this other than parsing the "To" header "manually" in the dial plan... can anyone point me in the right direction? |
| 16:43.13 | _VoiceMeUp_COM | heu |
| 16:46.28 | MRH2 | hi - does h264 (passtrhough) only work with asterisk 1.4 |
| 16:46.55 | perlmonke | I wish I knew why people clam up so much when I ask this question =( |
| 16:47.16 | MRH2 | did someone say something? lol |
| 16:53.38 | MRH2 | for me multiple stuff comes in as destination extension = did |
| 16:55.29 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:58.36 | irule | I have exten = 1,1,Macro(options) and in [macro-options] I have exten = 1,... well, once I am in the macro, 1 is used from [default] instead of the one in the macro, am I missing something? |
| 17:02.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:05.54 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:07.52 | [TK]D-Fender | perlmonke: the To: header should basically be effectively mapped to an exten. So 1 auth, multiple tartget extens. |
| 17:10.19 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 17:12.13 | seele_ | someone uses phonebridge ??? |
| 17:12.52 | [TK]D-Fender | irule: [macrTDMoE = ass |
| 17:12.57 | [TK]D-Fender | TDMoE = ass |
| 17:13.05 | *** join/#asterisk mrichmanM (n=richmanm@c-24-20-124-243.hsd1.mn.comcast.net) |
| 17:13.06 | *** join/#asterisk luxvero (n=atomic@189.3.87.60) |
| 17:13.09 | [TK]D-Fender | irule: Go read how macro's MERGE with the context that CALLS them. |
| 17:13.28 | [TK]D-Fender | irule: And DON'T make IVR's in Macros |
| 17:14.23 | luxvero | you can use MACRO_CONTEXT to go back to where you called the macro from |
| 17:15.26 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 17:16.08 | Luxvero | how does asterisk calculates the estimated hold time in QUEUEs? |
| 17:16.09 | *** join/#asterisk bbryant (i=brett@nat/digium/x-03bc32561b17731e) |
| 17:17.48 | irule | [TK]D-Fender oh great thanks :) |
| 17:19.30 | Luxvero | doesnt anyone uses report-holdtime? |
| 17:20.18 | *** join/#asterisk FinboySlick (n=Miranda@207.134.8.202) |
| 17:22.29 | perlmonke | [TK]D-Fender: I wish the first half of what you said was true - I don't understand what you are trying to convey with the second half. |
| 17:22.33 | [TK]D-Fender | Luxvero: rare... for one having a system give you hope while you sit in line for hours only to time out to VM is disenheartening :) |
| 17:22.34 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 17:23.06 | perlmonke | sipdebug shows that the "To" header contains things like: |
| 17:23.08 | perlmonke | To: <sip:15039728913@198.65.166.131>;tag=as46780ea4 |
| 17:23.17 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 17:23.49 | perlmonke | but when I create an exten for 15039728913 - it is ignored. |
| 17:23.49 | FinboySlick | I need a few pointers with my zapata.conf. Three fxs lines: 1,2,3, two contexts telco_in, telco_out. I need channels 1 and 2 to pick up in context telco_in. I need a group for dialing out on lines 2 and 3. |
| 17:24.02 | Luxvero | perlmonke, you must CUT the HEADER and then use GOTO |
| 17:24.10 | perlmonke | example? |
| 17:24.11 | Luxvero | wait a sec |
| 17:24.28 | FinboySlick | My problem is that since channel 2 is defined in both contexts, all incoming calls fall into the last context. |
| 17:24.52 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 17:25.16 | Luxvero | perlmonke, right here: http://www.aussievoip.com/wiki/How+to+get+the+DID+of+a+SIP+trunk |
| 17:25.34 | [TK]D-Fender | perlmonke: pastebin the incoming call at verbose 10 SIP debug enabled. |
| 17:25.37 | [TK]D-Fender | ~pb |
| 17:25.39 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:26.04 | Jon335 | I currently have a SPA-3000. As the echo is unbearable, I need to find another ATA or PCI card(s) that will support Asterisk with no echo. Any recommendations? I need a FXO and a FXS port. |
| 17:26.25 | [TK]D-Fender | FinboySlick: Contexts have NOTHING to do with what zap devices you use to dial. |
| 17:26.55 | FinboySlick | [TK]D-Fender: I know. |
| 17:27.03 | [TK]D-Fender | Jon335: I presume you have gone through several firmware revisions and tweaked the echo settings a lot at this point? |
| 17:27.24 | Jon335 | [TK]D-Fender, yes, I've tried everything |
| 17:28.00 | [TK]D-Fender | Jon335: Well guaranteed quality will cost you. Newer zaptel + a non-ec card might get the job done but is riskier |
| 17:28.04 | FinboySlick | [TK]D-Fender: It turns out that my problem was due to me defining the telco_in before the telco_out context. Since the channel was defined in both, it fell into the telco_out context. |
| 17:28.14 | *** join/#asterisk awk (n=awk@kia.inet-corp.com) |
| 17:28.16 | Luxvero | D-fender, it may be disheartening for sure, hehe.. But my callers would wait like 5-20 minutes, I was hoping asterisk could be set to calculate 10 x queuedCalers / agentsLogged |
| 17:28.16 | FinboySlick | [TK]D-Fender: Now I probably have the inverse problem. |
| 17:28.29 | [TK]D-Fender | FinboySlick: If you set one then the other it gets overriden. But if its solved, congrats.... |
| 17:28.44 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:28.51 | [TK]D-Fender | Luxvero: I honestly have NO idea of the math used for it.... |
| 17:29.06 | FinboySlick | [TK]D-Fender: Can I have a channel in two different contexts though? |
| 17:29.07 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 17:29.31 | FinboySlick | [TK]D-Fender: I don't want it overridden, I want it in both :P |
| 17:30.20 | perlmonke | Luxvero: thanks... thats what I was doing already, but some people told me there was a "better way"... I'm giving up on finding it. |
| 17:30.38 | [TK]D-Fender | FinboySlick: you have misunderstood something. the context is where it sends INCOMING calls into. |
| 17:31.04 | [TK]D-Fender | FinboySlick: an incoming call can't GO to 2 places off a single channel. |
| 17:31.20 | n00dle | Anyone use sipsak to send text to sip phones? |
| 17:31.38 | FinboySlick | [TK]D-Fender: Then I don't need to define channels for my telco_out context? |
| 17:32.06 | [TK]D-Fender | FinboySlick: context is where calls FROM that channel go. |
| 17:32.33 | [TK]D-Fender | FinboySlick: You can shove Dial(Zap/3/1234567) anywhere you FELL LIKE in your dialplan. |
| 17:32.37 | [TK]D-Fender | FEEL* |
| 17:32.58 | FinboySlick | I want to dial a group so that it takes whichever line is free. |
| 17:33.10 | FinboySlick | How do I define a group without a 'channel' statement? |
| 17:33.19 | [TK]D-Fender | FinboySlick: then you should define "group=x" where X is 1,2,3, etc.... |
| 17:33.30 | [TK]D-Fender | FinboySlick: Channel is NOT a grouping. |
| 17:34.00 | [TK]D-Fender | FinboySlick: the channel statement in zapata says "take these parameters I have set and APPLY them to these ports I'm specifying" |
| 17:34.01 | FinboySlick | [TK]D-Fender: Aaaah, so this is what I misunderstood. |
| 17:34.28 | Trevor_b | anyone here tried the Plantronics "S11 System"? |
| 17:35.49 | [TK]D-Fender | Trevor_b: Looks like an RJ9 >2.5mm AMP + cheap-o headset |
| 17:35.55 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:36.08 | [TK]D-Fender | Trevor_b: Non-voicetube mic though... thats good. |
| 17:36.14 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:36.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:36.21 | Trevor_b | [TK]D-Fender: dont like the voicetubes? |
| 17:36.32 | coppice | S11 System - System is added to the name merely to double the price |
| 17:36.35 | [TK]D-Fender | Trevor_b: Jittery annoying POS |
| 17:37.02 | Trevor_b | coppice: includes an AMP where the S11 normally is just the $16 dollar headset |
| 17:37.04 | [TK]D-Fender | coppice: Its have the price of the M12 amp and comes with the headset... so I guess they just named it as such to throw you off ;) |
| 17:37.35 | Trevor_b | Yeah wondering how decent or bad the amp is. Plantronics every make a SHITTY amp? |
| 17:37.49 | Trevor_b | Short of power issues or bat's dying etc. |
| 17:37.54 | LeddyHM | I think my internet tubes are full |
| 17:37.57 | LeddyHM | I can't make a call |
| 17:38.14 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 17:38.37 | FinboySlick | [TK]D-Fender: Thanks a gigantic bunch, btw. That was really dumb of me and I've had this working out of luck for months too. |
| 17:38.50 | [TK]D-Fender | LeddyHM: You have to refill the bellows otherwise the smoke becomes congested. |
| 17:41.16 | coppice | [TK]D-Fender: well the S11 looks like the 50 cent headsets the chinese export because they wouldn't want to use them themselves. |
| 17:41.42 | *** part/#asterisk Luxvero (n=atomic@189.3.87.60) |
| 17:41.55 | coppice | plantronics have always been a crazy price, but it seems you used to get a bit more for the money |
| 17:42.11 | *** join/#asterisk qdk (n=qdk@182.Red-83-39-38.dynamicIP.rima-tde.net) |
| 17:43.37 | denon | you know, Ive always been pretty happy with GN Netcom |
| 17:43.43 | Trevor_b | coppice: depends on where you buy too, direct retail from them is nuts, but some of their stuff has really nice pricing from resellers or partners. |
| 17:44.07 | Trevor_b | seems they have a S12 system as well, wonder if these are m11 and m12 amps repackaged into a "system" |
| 17:44.08 | Jon335 | Can anyone recommend a X100P? (or are they all the same) |
| 17:44.17 | Trevor_b | Jon335: x100p.com |
| 17:44.19 | denon | Jon335: I can recommend not to get one :) |
| 17:44.42 | Jon335 | denon, the are that bad, aren't they |
| 17:44.54 | denon | well, they're kind of a hack |
| 17:44.58 | coppice | denon: operators have never liked GN Netcom. they're usually a lot bulkier. hello voice is better liked |
| 17:45.00 | denon | you'll be much happier with a real tdm card |
| 17:45.15 | denon | coppice: spose .. though I've worn a gn netcom for many years |
| 17:45.22 | Jon335 | denon, how much does one go for? |
| 17:45.35 | denon | Jon335: depends how many channels you want and stuff. . im not in sales :) |
| 17:50.51 | [TK]D-Fender | Jon335: http://www.telephonydepot.com/product_p/105-050-100-a.htm |
| 17:56.04 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:57.24 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 17:57.54 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:58.08 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 18:02.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:02.23 | *** join/#asterisk logyati (n=suporte@201.29.73.49) |
| 18:02.26 | logyati | hi guys |
| 18:03.05 | *** join/#asterisk juuva (i=juuva@peili.org) |
| 18:03.44 | logyati | i have a tdm400p, should i download asterisk 1.4 or 1.2? |
| 18:04.09 | russellb | either one, but i would recommend 1.4 |
| 18:04.32 | logyati | the difference is only more features? |
| 18:04.36 | logyati | and bugfixies? |
| 18:04.39 | denon | you get to be a part of the future :) |
| 18:05.37 | logyati | so, if 1.4 is better, why i see two links at download page? i thought that 1.2 should be at "old releases" section |
| 18:05.47 | logyati | this i cant understand |
| 18:06.13 | denon | logyati: are you familiar with linux kernels? 2.4 and 2.6? |
| 18:06.17 | logyati | yes |
| 18:06.24 | denon | kind of the same idea, why both of those are still around |
| 18:06.25 | *** join/#asterisk angom (n=angom@red-corp-201.143.81.252.telnor.net) |
| 18:06.33 | logyati | but all new distros come with 2.6 right? |
| 18:06.38 | denon | they're starting to |
| 18:06.44 | logyati | hmmm |
| 18:06.51 | denon | as many new asterisk users are going to 1.4 |
| 18:06.51 | logyati | so, 1.4 is recommendend |
| 18:07.06 | denon | and eventually everyone else will be too .. but sometimes it's a hard jump when you have a bunch of stuff on 1.2, and 1.4 has changed a few things |
| 18:07.13 | logyati | is there big changes 1.2 to 1.4? to configure and use |
| 18:07.31 | denon | depends how complex your stuff is .. odds are, you'll be time and effort ahead if you start on 1.4 |
| 18:07.41 | logyati | im learning it from o'reilly book |
| 18:07.50 | logyati | im starting from zero |
| 18:07.52 | denon | but, if you buy now, I'll give you both 1.2 and 1.4 for the same price, so you can see which one you prefer |
| 18:07.53 | denon | :) |
| 18:08.07 | logyati | lol |
| 18:08.09 | logyati | :D |
| 18:08.23 | denon | offer void where prohibited, must be 18 years or older, yada yada |
| 18:08.31 | *** join/#asterisk mutilator (n=WebChat@the.drinkproject.com) |
| 18:09.09 | logyati | im asking all this cos im learning from the book, and i want to know maybe the book is too old to 1.4 and im gonna learn things that wont work |
| 18:09.24 | [TK]D-Fender | logyati: And if you don't like your free download version of Asterisk, we'll give you DOUBLE your money back... |
| 18:09.27 | denon | well, feel free to hit the wiki for more details on stuff that's not in the book |
| 18:09.37 | logyati | hahaha |
| 18:09.45 | denon | it's really worth learning aggressively, not just what the book has to say |
| 18:09.50 | denon | get involved, dont just read :) |
| 18:10.13 | [TK]D-Fender | logyati: There are indeed some differences between 1.2 & 1.4 but mostly small stuff from a beginners POV. And a new book is due out in about a month or so. |
| 18:10.34 | logyati | im not worried with things i cant find inside the book, but with differences between what could be written in the book based on 1.2, and with 1.4 it wont work |
| 18:10.45 | [TK]D-Fender | logyati: indeed, get off your ass, download, install, get a soft-phone (or 2) and get to it! |
| 18:10.55 | logyati | hehe |
| 18:10.58 | [TK]D-Fender | logyati: Stop worrying and get off your ass! |
| 18:11.10 | [TK]D-Fender | logyati: When you hit a bump we'll be here. |
| 18:11.14 | mvanbaak | I dont like typing while standing |
| 18:11.22 | mvanbaak | so I'll sit on my ass and type ok ;) |
| 18:11.22 | logyati | hehehe |
| 18:11.58 | Dr-Linux | anybody is using agent callback login? |
| 18:12.10 | UCFmethod | logyati: the basics will be the same, but alot has changed from the book til 1.4 |
| 18:12.25 | [TK]D-Fender | Dr-Linux: No, we stopped. All of us. just for YOU. |
| 18:13.10 | [TK]D-Fender | logyati: Go DL 1.4, follow the installation instructions on asterisk.org, THEN start with the book. |
| 18:13.48 | Dr-Linux | [TK]D-Fender: Thanks! |
| 18:13.50 | logyati | im trying to find this version of the book (one month ago) that d-fender said... |
| 18:13.56 | [TK]D-Fender | I really shoud write book myself.. |
| 18:14.07 | UCFmethod | logyati: i can put the pdf someplace if you want? |
| 18:14.20 | logyati | yes, please, cos my is from 2005 |
| 18:14.24 | [TK]D-Fender | "Intelligence for Dummies! (A self-help book by IDG)" |
| 18:14.30 | UCFmethod | so is the pdf ;-) |
| 18:14.38 | [TK]D-Fender | ~boot |
| 18:14.40 | jbot | boot is probably what you get when you act like a DalNet user, or #debian-boot |
| 18:14.41 | [TK]D-Fender | ~book |
| 18:14.41 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:14.49 | [TK]D-Fender | There , go DL |
| 18:14.56 | logyati | ty |
| 18:15.02 | Dr-Linux | since i know very rare peoples use agent callback login, so i should ask before asking my odd question |
| 18:15.04 | [TK]D-Fender | logyati: that IS the only book out. a NEW one is COMING. |
| 18:15.20 | [TK]D-Fender | Dr-Linux: Never ask to ask, you should know better, just get to it! |
| 18:15.36 | [TK]D-Fender | some people sure do take a long time to make their point(less) |
| 18:16.34 | Dr-Linux | well, the queue "retry" doesn't work for callback agent |
| 18:16.54 | Dr-Linux | also when an agent get a call, i can see on the CLI: |
| 18:16.55 | Dr-Linux | <PROTECTED> |
| 18:16.55 | Dr-Linux | <PROTECTED> |
| 18:17.01 | [TK]D-Fender | Dr-Linux: pastebin your configs & applicable dialplan |
| 18:17.12 | Dr-Linux | so Jason is an agent and his extension is 4510 |
| 18:17.19 | [TK]D-Fender | Dr-Linux: and of course verbose 10 CLI output. |
| 18:17.32 | Dr-Linux | it's set to 100 |
| 18:17.51 | [TK]D-Fender | Dr-Linux: ..... get moving... |
| 18:19.02 | n00dle | I was curious yesterday about the verbosity levels and spelunked in the code... |
| 18:19.10 | n00dle | I found nothing over a 4. |
| 18:19.13 | russellb | n00dle: Transfer just can't be supported right now, at least for 1.4. It's going to be a more significant development effort that could only be targeted for 1.6. |
| 18:19.27 | Dr-Linux | [TK]D-Fender: but if i add simple member in queues.conf that works just fine, but in agent case it shows double at CLI, and it hangs up after 20 sec which time is set in the extensions.conf |
| 18:19.34 | JerJer | Asterisk 1.2 honors DNS TTL now, right ? |
| 18:19.36 | n00dle | Ok... thanks, russellb. Now I know my plan for the immediate future. |
| 18:19.59 | [TK]D-Fender | Dr-Linux: Stop with the useless description and PASTEBIN YOUR CONFIGS AND CLI OUTPUT. |
| 18:21.06 | russellb | n00dle: you're welcome ... |
| 18:21.29 | [TK]D-Fender | russellb: Why kind of "Transfer" is this you're looking at? |
| 18:21.53 | russellb | JerJer: no ... you can enable the DNS manager in dnsmgr.conf to have lookups refreshed, but only some of the code supports it (chan_sip, mainly) |
| 18:21.58 | Dr-Linux | [TK]D-Fender: ok, i gonna paste bin CLI, bcoz my agents.conf for an application |
| 18:22.04 | russellb | [TK]D-Fender: transfers + shared line appearances ... |
| 18:22.09 | n00dle | d-fender: We were looking at the SLA code... it's, er... "complicated" |
| 18:22.12 | JerJer | russellb: lovely |
| 18:22.13 | [TK]D-Fender | russellb: z0mg |
| 18:22.26 | russellb | I can make it work, but it's not a "fix" really |
| 18:22.39 | russellb | it'll take some effort, for sure, to the point i consider it a new feature ... |
| 18:22.46 | JerJer | so 1.4 fully supports dns ttl ? |
| 18:22.54 | [TK]D-Fender | n00dle: SLA = Sorta Like Advertised ;) |
| 18:23.00 | denon | haha |
| 18:23.06 | russellb | ~lart [TK]D-Fender |
| 18:23.06 | jbot | wallops [TK]D-Fender with a main rotation server that needs rehubbing. It won't take long |
| 18:23.09 | [TK]D-Fender | :O |
| 18:23.14 | denon | I think verizon uses that as their SLA definition |
| 18:23.19 | n00dle | Laughing or crying I dunno.... |
| 18:23.33 | [TK]D-Fender | denon: Revizon Math 2.0! |
| 18:23.39 | russellb | it provides basic shared lines. that's all that was ever advertised |
| 18:23.46 | russellb | stop giving me such a hard time people :-p |
| 18:24.20 | russellb | it seems what a lot of people want is something more like a "shared extension", which is already possible IMO ... |
| 18:24.26 | russellb | well, for the most part |
| 18:24.38 | n00dle | russellb, No, no! No hard time, but there were a few hidden gotchas that my boss expected for functions. |
| 18:24.45 | russellb | i understand |
| 18:24.54 | anonymouz666 | JerJer! |
| 18:24.58 | [TK]D-Fender | russellb: Of course its possible, tons of phones support it and other PBX's... its just a question of if, then, how, then WHEN for * :) |
| 18:24.58 | russellb | this is the part where I find out what peopel really want, and I eventually go code it |
| 18:24.58 | n00dle | ...and I didn't know if it worked or not until I tried it. :) |
| 18:25.01 | Dr-Linux | [TK]D-Fender: go here: http://phpfi.com/241976 |
| 18:25.15 | *** join/#asterisk charasky (n=charasky@mx.rima.org) |
| 18:25.23 | denon | russellb: so the trick is to stub a bunch of feature names into the code, then wait for people to say "how come it doesnt .. " |
| 18:25.53 | anonymouz666 | JerJer is running E61 with fully DNS TTL! |
| 18:26.09 | [TK]D-Fender | russellb: SLA for PHONES is the big goal. Improved queues where native SIP transfers don't break agents, etc would be next. SLA for "lines" isn't really needed, thats what parking is for. |
| 18:26.48 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 18:26.49 | n00dle | ...and I haven't even tried parking, so I have some stuff to learn. |
| 18:27.03 | denon | parking is your friend |
| 18:27.09 | [TK]D-Fender | Dr-Linux: Oh God, this crap AGAIN!?!?! Got SIP response 482 "Loop Detected" back from 127.0.0.1 |
| 18:27.12 | russellb | well, i coded what people told me was needed |
| 18:27.21 | russellb | some people like it, and then a lot more say it's not enough |
| 18:27.27 | russellb | that's fine, it's what i need to know, i guess ... |
| 18:27.38 | denon | nod |
| 18:27.49 | russellb | SLA for phones is possible in asterisk today for the most part |
| 18:27.52 | [TK]D-Fender | russellb: supprt group sessions open up tomorrow night at happy-hour :) |
| 18:28.14 | russellb | i'm going ot call them "shared extensions" ... |
| 18:28.14 | Dr-Linux | [TK]D-Fender: yeah, bcoz that's from application i mean the callback agent |
| 18:28.42 | russellb | exten => 123,1,Dial(SIP/123_1&SIP123_2) ... exten => 123,hint,SIP/123_1&SIP/123_2 |
| 18:29.11 | russellb | the only thing missing in that picture is to make SIP/123_2 show that the shared extension is in use, when SIP/123_1 makes a call ... |
| 18:29.21 | russellb | am I on the right track here? |
| 18:30.01 | [TK]D-Fender | russellb: No, thats still just a presence hack w/o the ability to steal held calls and answer incoming / ring along-with |
| 18:30.20 | russellb | ah, interesting point. |
| 18:30.32 | [TK]D-Fender | russellb: useful for other things sure, but not with that title associated with :) |
| 18:31.26 | russellb | i need a freaking spec document from somebody, heh |
| 18:31.41 | [TK]D-Fender | russellb: You know what WOULD be useful ..... having an internal parser allow pattern-matched hints. |
| 18:31.57 | russellb | yep, i think there may even be a patch for that in the tracker, i don't know |
| 18:32.06 | russellb | i lose track. |
| 18:32.56 | [TK]D-Fender | russellb: I excel at coming up with ideas that are already there or in the works :) "Those who fail to understand Unix are doomed to REINVENT IT. (poorly)" |
| 18:33.22 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 18:33.23 | [TK]D-Fender | russellb: Naughty! |
| 18:33.37 | russellb | i think this whole SLA thing was a failure for someone to tell me what people actually wanted |
| 18:33.51 | denon | focus groups :) |
| 18:33.58 | russellb | i wrote somethign which is what I was told people wanted ... but it was too basic |
| 18:34.15 | russellb | for most people, anyway |
| 18:34.19 | lee_is_me | AMI Question: Can you use Redirect() on an existing meetme member? |
| 18:34.23 | russellb | now i get to go write the complicated version |
| 18:34.37 | russellb | denon: yeah, i'll get my focus group team right on that |
| 18:34.40 | russellb | wait ... |
| 18:34.54 | Qwell[] | russellb: I think your focus group has ADD |
| 18:34.57 | Qwell[] | ... |
| 18:35.10 | russellb | it's a requirement to work in swdev at digium. |
| 18:35.23 | denon | redbull-induced add |
| 18:35.51 | Qwell[] | russellb: I'm gonna write a petition to Danny, to provide redbull to engineering. |
| 18:36.31 | russellb | Qwell[]: can I be #2 to sign it? |
| 18:36.35 | Qwell[] | You may |
| 18:36.53 | russellb | screw the coffee bar ... i want free redbull |
| 18:37.47 | russellb | Well if someone would like to write a document which explains what features people want out of shared extensions or whatever, i'll code it |
| 18:38.39 | denon | or ask someone which pbx's implementation they're happy with, then mirror their featureset? |
| 18:38.44 | russellb | or say, go look at system X, like that! |
| 18:38.49 | russellb | yeah |
| 18:38.50 | [TK]D-Fender | russellb: I'd pay for your redbull to work a round-table meeting with you to iron out some of the real world stuff. "People" are never clear on things and can't describe their way out of a paper-bag. ESPECIALLY HERE |
| 18:39.23 | russellb | [TK]D-Fender: well, i'd be happy for that kind of feedback, even without redbull. |
| 18:39.51 | russellb | in theory, we would be getting this information from marketing ... |
| 18:39.59 | [TK]D-Fender | russellb: I've got the dev meetme linked on my home server, we'll work something out :) |
| 18:40.12 | lee_is_me | Ah, so you can actually use Redirect() on an existing Queue member. Sorry, I was being lazy... |
| 18:40.23 | russellb | [TK]D-Fender: alright, cool. |
| 18:40.27 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:40.30 | [TK]D-Fender | russellb: Marketing is a contrivance by those with their heads in the clouds, not hands in the dirt. |
| 18:40.43 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:40.49 | russellb | heh |
| 18:41.01 | denon | russellb: Ive never seen a marketing dept able to give concrete featuresets .. tech sales and such, perhaps |
| 18:41.09 | denon | or implementation folks |
| 18:41.29 | *** join/#asterisk jtoy_ (n=jtoy@mail.backchannelmedia.com) |
| 18:41.36 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 18:41.36 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-173-56-41.socal.res.rr.com) |
| 18:41.48 | jtoy_ | what is the way I should use if I want to have sip users dynamically? |
| 18:41.56 | CoolGuy21 | hey guys |
| 18:42.04 | jtoy_ | I normally use sip.conf, but thats too cumbersome |
| 18:42.07 | CoolGuy21 | anyone here know what format the mp3 for moh should be? |
| 18:42.45 | russellb | CoolGuy21: 8kHz mono |
| 18:43.01 | Qwell[] | personally I would say not to use mp3 for moh... |
| 18:43.04 | [TK]D-Fender | Dr-Linux: does this look SANE to you?! - Registered SIP '4510' at 192.168.0.254 port 5060 expires 120 |
| 18:43.05 | [TK]D-Fender | <PROTECTED> |
| 18:43.05 | Qwell[] | I'd use ulaw or something |
| 18:43.10 | russellb | Qwell[]: i almost said ulaw, heh |
| 18:43.36 | russellb | [TK]D-Fender: man, you're harsh |
| 18:43.50 | denon | Qwell: agreed .. course im usually lazy and use mp3 anyway .. setting up some overpowered xeon running at like 1% cpu all day |
| 18:43.56 | [TK]D-Fender | russellb: I brushed my teeth before biting his head off! |
| 18:43.59 | [TK]D-Fender | *sheesh* |
| 18:44.53 | russellb | alright, well this has been fun...ish ;) ... i'm off to fix more bugs ... |
| 18:45.14 | denon | audios senior russell |
| 18:45.14 | [TK]D-Fender | russellb: Poeple with names like Dr-Linux , MrTelephone , voipMasta (specific examples) really should show a hint of a Clue you know :) |
| 18:45.23 | russellb | lol |
| 18:45.25 | russellb | nice. |
| 18:45.29 | denon | hah |
| 18:45.35 | russellb | <PROTECTED> |
| 18:45.40 | Dr-Linux | huh |
| 18:45.43 | jtoy_ | If I am connecting to postgresql, should I use odbc driver or native driver? |
| 18:45.52 | denon | russellb: this from the guy who doesnt even understand what we want for SLA! |
| 18:45.57 | russellb | jtoy_: the odbc driver is better maintained |
| 18:46.06 | russellb | denon: meanie head |
| 18:46.15 | denon | hehe kidding .. I have no need for sla in general |
| 18:46.25 | denon | but I'm sure you're doing an outstand job! |
| 18:46.37 | Dr-Linux | [TK]D-Fender: do you mean guy like file must have show some editors? |
| 18:46.37 | tzafrir_laptop | but is the postgresql odbc driver well-maintained? ;-) |
| 18:46.41 | russellb | heh, it does what it was intended to do well |
| 18:46.49 | russellb | :) |
| 18:47.02 | denon | nod |
| 18:47.04 | russellb | but you're just not allowed to try to make it do more |
| 18:47.24 | jtoy_ | hmm, so postgresql or odbc? |
| 18:47.40 | russellb | jtoy_: odbc |
| 18:48.13 | [TK]D-Fender | Dr-Linux: No, that means he has paper-work to do :) |
| 18:48.15 | jtoy_ | ok, thanks |
| 18:48.56 | mocker | Woo, odbc just crashed my asterisk. :( |
| 18:51.07 | [TK]D-Fender | Dr-Linux: And this : - Executing VoiceMail("Local/4510@users-99e6,2", "u4510@default") in new stack |
| 18:51.47 | [TK]D-Fender | Dr-Linux: What are you doing using dialplan to call your agents that falls to VM? This completely defeates the idea of being in a queue to get a HUMAN BEING |
| 18:52.33 | Qwell[] | ~hdlc |
| 18:52.40 | jbot | rumour has it, hdlc is High-level Data Link Control |
| 18:52.56 | Dr-Linux | thanks |
| 18:53.04 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:53.11 | [TK]D-Fender | Dr-Linux: All part of the service... bill is in the mail :0 |
| 18:53.34 | [TK]D-Fender | Qwell[]: Do you know who maintains jbot? |
| 18:53.43 | Qwell[] | Tim Riker |
| 18:53.45 | Qwell[] | ~timriker |
| 18:53.46 | jbot | it has been said that timriker is my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 18:53.46 | *** part/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:54.21 | [TK]D-Fender | Qwell[]: spot-on. thanks |
| 18:54.56 | [TK]D-Fender | Qwell[]: Actually... thats more the Coder right? I basically want a dump of everything jbot has been trainied for. |
| 18:55.12 | *** join/#asterisk structure (n=struct@76.97.84.4) |
| 18:55.13 | Qwell[] | no, he runs the instance of the bot |
| 18:55.46 | Hymie | PEOPLE OF ASTERISK LAND!!!!!! Any idea where to snag newer polycom roms? |
| 18:55.50 | *** join/#asterisk jpablo (n=jpablo@200.94.130.197) |
| 18:56.01 | Qwell[] | Hymie: from your reseller |
| 18:56.13 | Hymie | my resller told me that I should go outside, find a donkey, and suck that donkey's balls |
| 18:56.24 | Hymie | so.. I would prefer another method |
| 18:56.33 | Qwell[] | Then call Polycom, tell them that one of their resellers are violating their agreement, and that you'd like it terminated. |
| 18:56.45 | coppice | the attitude to updates in the VoIP business is an absolute disgrace |
| 18:57.02 | Hymie | Qwell[]: my reseller is some dude, probably not 'official' |
| 18:57.06 | Hymie | anyhow |
| 18:57.08 | Hymie | I know there be some site |
| 18:57.12 | Hymie | but, not sure where I found it |
| 18:57.14 | pipwerk | so, find a decent reseller |
| 18:57.15 | Qwell[] | Then call Polycom, and tell them that they have an unauthorized reseller. |
| 18:57.18 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 18:57.27 | Qwell[] | then ask for a legit reseller who can provide the firmware |
| 18:57.35 | Hymie | Qwell[]: I want polycom to provide it |
| 18:57.38 | Hymie | and I demand it now |
| 18:57.43 | Hymie | polycom! I demand it now! |
| 18:58.00 | Hymie | I, who demand, ecome more demanding! |
| 18:58.14 | Hymie | as a demander, I demand that my demanding is taken seriously |
| 18:58.19 | Hymie | or I will demand more! |
| 18:58.26 | *** mode/#asterisk [+b %Hymie!*@*] by Qwell[] |
| 18:58.27 | Qwell[] | shh |
| 18:58.31 | *** mode/#asterisk [-b %Hymie!*@*] by Qwell[] |
| 18:58.36 | [TK]D-Fender | Hymie: .... STFU KTHXBAI |
| 18:58.46 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:58.48 | [TK]D-Fender | :O |
| 18:58.50 | Hymie | [TK]D-Fender: I comprehend the STFU part |
| 18:58.54 | file | Qwell[]: I'd like to know where you got the notion... |
| 18:59.01 | Qwell[] | notion? |
| 18:59.03 | Hymie | [TK]D-Fender: the rest is too .... unknown for me to parse |
| 18:59.05 | file | yes! |
| 18:59.08 | Qwell[] | to?! |
| 18:59.14 | file | rockin' the boat |
| 18:59.17 | Qwell[] | oic |
| 18:59.21 | *** part/#asterisk juuva (i=juuva@peili.org) |
| 19:00.54 | Hymie | Qwell[]: also.. hi, and sorry for the bleeding ears |
| 19:07.47 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 19:09.25 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 19:10.39 | Taadow | Has anyone ever experienced an issue where a peer attempts to register w/ a softphone (Eyebeam) and while doing so causes the pbx to stop responding for all other requests? ie, no one else can register/call. |
| 19:11.19 | Taadow | Only odd occurance I can recognize log wise is doing a 'sip show channels' and seeing a bunch of SUBSCRIBE messages associated w/ the offender (person trying to register w/ soft phone that causes issue). |
| 19:12.37 | seele_ | any application to lock the outgoing calls when i go out of my office for example?? |
| 19:13.22 | [TK]D-Fender | seele_: its your dilaplan, go shove some code in there to see if it should be considered disabled before actually dialing anything |
| 19:15.17 | seele_ | [TK]D-Fender, yes but the dialplan change the extension every time I need some command like *123 to enable/disable the line ... with password, like a queue but for outgoing calls |
| 19:15.46 | mazpe | is there a requirement for asterisk recording? |
| 19:15.58 | gerwinin | Taadow: maybe you problem lays somewhere else |
| 19:16.14 | gerwinin | Taadow : are you using eyebeam with video ? |
| 19:16.15 | [TK]D-Fender | mazpe: A working install of * + storage space & some CPU |
| 19:16.37 | [TK]D-Fender | seele_: Reword that. Its all jumbled up |
| 19:16.44 | gerwinin | Asterisk runs fine on a via c7 |
| 19:16.49 | Taadow | gerwinin: No video. I've been working with voip for over two years now and this is the first issue I cannot troubleshoot. It's quite elusive. |
| 19:16.53 | mazpe | I mean more in format.. what format is required for the recordings? |
| 19:17.25 | [TK]D-Fender | mazpe: Most of the time you get to choose the format. |
| 19:17.38 | mazpe | i have a .wav 256kpbs 16bits 1mono 16khz in PCM ... doesnt seem to play. |
| 19:17.45 | [TK]D-Fender | mazpe: And you can also CONVERT it yourself to whatever you want relatively easily. |
| 19:17.53 | Taadow | I think I might have to pay Digium to have one of their support staff assist w/ this. |
| 19:17.55 | Qwell[] | mazpe: 8khz |
| 19:18.03 | gerwinin | Taadow: Can you pastbin the sip register message ? |
| 19:18.09 | mazpe | Qwell: i see |
| 19:18.10 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:18.14 | [TK]D-Fender | mazpe: ah to PLAY a recorded file... |
| 19:18.21 | gerwinin | Taadow: we can try to look at it together |
| 19:18.30 | [TK]D-Fender | mazpe: For wav : 8khz mono indeed |
| 19:18.48 | gerwinin | Taadow can you pastbin the sip register message from the eyebeam client ? |
| 19:18.54 | mazpe | hmm.. anyone knows a quick remedy to downgrade it? |
| 19:18.57 | Taadow | gerwinin: Yes, give me a moment to grab it. |
| 19:19.18 | logyati | [tk]d-fender im at "configuring sip client" section of chapter 4. I configured SIP.CONF. at this point, i must have openser already installed and working, right? |
| 19:19.44 | logyati | [tk]d-fender page 70, if you want to know |
| 19:20.17 | logyati | [tk]d-fender im asking it cos you said that asterisk is not a sip proxy, but the book doesnt mention that i should have one installed |
| 19:21.04 | mazpe | 8k 16bits mono should be fine? |
| 19:21.14 | [TK]D-Fender | logyati: Do you refer to your motorcycle operators guide when trying to figure out how to make a bag of microwave popcorn often? |
| 19:21.49 | [TK]D-Fender | logyati: PUT DOWN THE CRACK PIPE (c) JerJer |
| 19:22.21 | [TK]D-Fender | logyati: Forget about SER. You are learning ASTERISK. |
| 19:23.02 | logyati | k ehehe |
| 19:23.33 | seele_ | ok, I'm in my office with my phone, I can make outgoing calls, but I need that when I require, be able to block these calls |
| 19:24.03 | seele_ | with a command like *(something) |
| 19:24.16 | *** join/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 19:24.27 | [TK]D-Fender | seele_: its your dialplan, you should know where to insert the disabling code. |
| 19:25.03 | seele_ | and how can I make a disable code? |
| 19:26.31 | _omer | anyone knows about Sending Fax (using SIP..without any hardware) through Asterisk ? |
| 19:26.44 | structure | OK so I have an Asterisk v1.4.2 connected to a 3Com via 1 PRI. Is there anything special in wanpipe I need to configure to send callerid name? phone is working |
| 19:27.25 | structure | I've verified Asterisk is sending callerid name/# via IAX and SIP perfectly. |
| 19:27.45 | seele_ | _omer, no SIP ... but IAXmodem works fine |
| 19:27.55 | _omer | IAXmodem ? |
| 19:28.09 | _omer | u mean IAX ? |
| 19:28.14 | structure | https://sourceforge.net/projects/iaxmodem |
| 19:28.17 | gerwinin | structure: you probarly need to adapt your country settings |
| 19:28.21 | seele_ | _omer, http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem |
| 19:29.36 | structure | gerwinin, In the PRI configuration - ensure both sides match? |
| 19:30.23 | gerwinin | structure: yes are you using a card from digium or sangoma ? |
| 19:30.29 | structure | Sangoma |
| 19:30.47 | gerwinin | Okay and your connection with e1 ? |
| 19:30.51 | structure | So I'm using wanpipe1.conf through ..4.conf, 1 for each port |
| 19:31.00 | structure | crossover t1 |
| 19:31.22 | gerwinin | structure : I know that tool because I had to make some changes to it to make it work with an ericson gateway once |
| 19:31.39 | gerwinin | structure which pabx or gateway are you connecting with ? |
| 19:31.51 | gerwinin | structure: which country are you in ? |
| 19:32.01 | structure | I'm in the US. |
| 19:32.17 | structure | It's Asterisk 1.4.2 -> 3Com. Does that answer your question? |
| 19:32.46 | gerwinin | structure: is this a pabx or gateway ? |
| 19:33.44 | gerwinin | structure : so you have 23 channels and one control channel |
| 19:34.08 | [TK]D-Fender | seele_: Do a GotoIf before dialing if a "disable" value has been set. |
| 19:34.24 | CoolGuy21 | what format should the asterisk MOH mp3 files be? |
| 19:34.45 | structure | gerwinin, I believe it is pabx. It's for one company's use internally. Yes 23 channels and 1 data. |
| 19:35.27 | gerwinin | structure: for number recognition you need to have the correct country settings in both your sangoma card and in asterisk |
| 19:35.41 | [TK]D-Fender | CoolGuy21: 128kbit non-vbr, no ID3 tags |
| 19:36.37 | structure | gerwinin, number is working well, it's name that is not. Can name be affected by mismatched country settings? |
| 19:37.00 | *** join/#asterisk Skarmeth (n=Skarmeth@201009036240.user.veloxzone.com.br) |
| 19:37.15 | Skarmeth | hi all |
| 19:37.19 | gerwinin | structure: well the name is mostly not send by the pabx |
| 19:37.33 | gerwinin | structure: the number is sent by dtmf |
| 19:37.44 | gerwinin | strcuture: the number is working |
| 19:38.35 | structure | gerwinin, Yes the number is working, name is not. |
| 19:39.10 | Skarmeth | I was searching Digium SVN repo for chan_cellphone and chan_bluetooth for testing Asterisk + Bluetooth mobiles but I can't found it on trunk... |
| 19:39.28 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 19:39.40 | gerwinin | structure: I understood that, but the name is not send by the pabx , mostly the name is or in the phone , and with voipphones in the sip messages |
| 19:40.00 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:40.07 | gerwinin | structure: for example sip---> asterisk ------> sip it will give the name |
| 19:40.50 | gerwinin | structure: analogue --------> asterisk -------------> sip it will give only the number unless the name is in extensions.conf |
| 19:41.10 | structure | Hmmm ok |
| 19:42.22 | *** join/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 19:42.51 | CunningPike | [TK]D-Fender: Have you ever used OpenSER/SER as a SIP registrar in front of Asterisk? |
| 19:43.02 | *** join/#asterisk xjagox (n=xjagox@190.8.158.12) |
| 19:43.03 | *** part/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 19:43.37 | xjagox | wenas |
| 19:47.13 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 19:47.38 | [TK]D-Fender | CunningPike: nope |
| 19:47.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 19:48.15 | CunningPike | [TK]D-Fender: We're approaching 400 registrations and I'm wondering about doing that to take the registration load off our Asterisk server |
| 19:48.45 | mazpe | whats a good pc sip phone? |
| 19:48.48 | _VoiceMeUp_COM | dhmmm |
| 19:48.54 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 19:48.55 | _VoiceMeUp_COM | that G flag in dial is messing with me |
| 19:49.03 | [TK]D-Fender | ~softphone |
| 19:49.03 | jbot | something that should be drug out into the street and shot |
| 19:49.08 | mazpe | or softphone i guess is the right term |
| 19:49.14 | mazpe | nod |
| 19:49.15 | [TK]D-Fender | ~softphones^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:49.30 | [TK]D-Fender | mazpe: on the better end is eyeBeam & Ekiga |
| 19:49.51 | _VoiceMeUp_COM | G(getdtmf^s^1)) then in [getdtmf] i wait,read,gotoif on result.. but it send both channels there.. not sure on why or how to make it kill the other one |
| 19:50.09 | _VoiceMeUp_COM | says in docs.. sneds caller to priority and celled party to prio+1 |
| 19:50.21 | _VoiceMeUp_COM | that whould be +101 to make things easier i think |
| 19:50.24 | mazpe | [TK]D-Fender... free? i'm looking for somehting like IDEFISK, for testing |
| 19:50.27 | mazpe | but sip |
| 19:50.36 | [TK]D-Fender | mazpe: idefisk then |
| 19:50.40 | logyati | [tk]d-fender can you see a pastebin of my extensions.conf and say why asterisk doesnt answer the call from pstn? my zapata.conf is ok, with context=incoming |
| 19:50.43 | mazpe | idefisk does sip? |
| 19:50.56 | [TK]D-Fender | mazpe: yes |
| 19:50.56 | mocker | Anyone every done any type of community asterisk server for a local aug or anything? |
| 19:50.58 | logyati | [tk]d-fender everything seems to be fine to me, i cant find where im wrong |
| 19:51.01 | [TK]D-Fender | logyati: go for it |
| 19:51.06 | logyati | [tk]d-fender http://www.pastebin.ca/568804 |
| 19:51.07 | mazpe | [TK]D-Fender interesting :_ |
| 19:51.45 | _omer | anyone who have used AsterFax? |
| 19:51.46 | [TK]D-Fender | logyati: pastebin CLI output of the call attempt as well as your zaptel & zapata config so I can prove that it looks right. also include "zap show channels" |
| 19:53.05 | logyati | [tk]d-fender zapata.conf http://www.pastebin.ca/568816 |
| 19:53.08 | CunningPike | What is the general wisdom around the number of SIP registrations a single Asterisk server can support? |
| 19:53.28 | [TK]D-Fender | logyati: And atr a bare minimum its guaranteed that "autofallthrough=yes" will kill the call right after "Answer" is issued if even |
| 19:54.06 | logyati | [tk]d-fender zaptel.conf http://www.pastebin.ca/568819 |
| 19:55.16 | logyati | [tk]d-fender when i put only exten => s,1,Answer() / exten =>s,2,Echo() everything works fine |
| 19:56.04 | logyati | ok |
| 19:56.08 | logyati | i removed the line you told me |
| 19:56.12 | logyati | now its working ty |
| 19:56.21 | [TK]D-Fender | logyati: you need to learn how to make a proper IVR. |
| 19:58.18 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:59.39 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:04.04 | *** join/#asterisk Nuitari (n=nuitari@melchior.nuitari.net) |
| 20:04.12 | Nuitari | Hi |
| 20:04.21 | Nuitari | Is there a way to get all setted devstate with func_devstate? |
| 20:06.15 | *** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com) |
| 20:08.33 | Zion800 | Hey, I think ChanIsAvail is broken in Asterisk 1.4. Can someone teach me how to use bugs.digium.com to search for bugs on this issue? Every time I try using bug.digium.com, and I do a search, it comes up with a bunch of seemingly unrelated bugs... |
| 20:09.25 | [TK]D-Fender | Zion800: You have described precisely NOTHING about this "problem" you seem to think you have. Congratulations, you are the 100th to do so today. |
| 20:09.39 | [TK]D-Fender | Zion800: You can pickup your prize at the door :) |
| 20:09.42 | Nuitari | ~prize |
| 20:10.23 | Zion800 | Well, before wanting to bother you guys with the problem, I'd rather do a little research.. :-) |
| 20:10.57 | Zion800 | If its already a noted bug, then I wouldn't think its something I'm doing wrong |
| 20:10.59 | [TK]D-Fender | Zion800: show us the problem, don't just sit there telling us you found one. |
| 20:11.06 | Zion800 | Alright...fine :-) |
| 20:13.12 | Zion800 | So, I upgraded from Asterisk 1.2, and in Asterisk 1.2, ChanIsAvail would priority jump if the channel was unavailable |
| 20:13.24 | Zion800 | Well, in 1.4, they added the 'j' option for this. However, it doesn't jump. |
| 20:13.37 | [TK]D-Fender | Zion800: stop talking and start PASTEBINING. |
| 20:14.35 | Zion800 | ok |
| 20:15.17 | Qwell[] | Priority jumping has been completely removed in trunk, and will not be available in 1.6. |
| 20:15.24 | Qwell[] | I strongly recommend NOT using it at this point. |
| 20:16.17 | Zion800 | hmm...ok...well, I'm using 1.4.4, so it *should* still work |
| 20:16.35 | Zion800 | http://pastebin.ca/568859 |
| 20:16.39 | Qwell[] | Yes, but I still wouldn't recommend using it at all. |
| 20:16.53 | [TK]D-Fender | Qwell[]: We |
| 20:17.00 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 20:17.05 | [TK]D-Fender | Qwell[]: We'll worry about 1.6 when it comes out... in 2020 :) |
| 20:17.46 | Zion800 | [TK]D-Fender: does the pastebin have enough info to work with? |
| 20:18.44 | Zion800 | keep in mind, this worked in 1.2 (without the 'j' option of course) |
| 20:19.23 | [TK]D-Fender | Zion800: pastebin "show dialplan" |
| 20:20.11 | Zion800 | just the section this extension is under? |
| 20:20.39 | [TK]D-Fender | Zion800: yes |
| 20:21.02 | _VoiceMeUp_COM | oh well so i got my PRI checker up and running .. basically i cron a script.. that makes a call out file and makes boxes talk via pri's and if a string is not received on tester box.. then it sends an ssh coomand to restart the pbx via killall -9 name |
| 20:21.14 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 20:21.20 | _VoiceMeUp_COM | so i guess its all good.. |
| 20:22.48 | *** part/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 20:22.57 | Zion800 | ok....its in the same pastebin |
| 20:22.58 | Zion800 | http://pastebin.ca/568859 |
| 20:23.12 | *** join/#asterisk Kumba_ (n=kumba@208.177.233.66) |
| 20:23.14 | Corydon76-work | Zion800: there is no n+101 for the chanisavail |
| 20:23.24 | _VoiceMeUp_COM | sipsock_read: We could NOT get the channel lock for S |
| 20:23.25 | _VoiceMeUp_COM | hmmm |
| 20:23.31 | [TK]D-Fender | Corydon76-home: Sure used to be. |
| 20:23.34 | _VoiceMeUp_COM | nasty |
| 20:23.43 | Zion800 | Corydon76-home: It says in "show application chanisavail" |
| 20:23.48 | _VoiceMeUp_COM | <PROTECTED> |
| 20:23.53 | Kumba_ | Anyone used the VTech 8100's? Do the handsets just act as one sip registration or are the handsets all individual registrations? |
| 20:23.59 | Corydon76-work | Zion800: in other words, there is no 102 priority. There is a 105 priority, though |
| 20:24.04 | _VoiceMeUp_COM | theres a trixbox imlpementation that will break asterisk and core dump it |
| 20:24.21 | Qwell[] | ... |
| 20:24.24 | Zion800 | ah...so how do I fix that? |
| 20:24.24 | _VoiceMeUp_COM | Can you guys ENFORCE trixbox to use UAgent as Trixbox Asterisk Mod |
| 20:24.27 | Qwell[] | there's a trixbox implementation that DOESN'T? |
| 20:24.32 | Qwell[] | </troll> |
| 20:24.37 | _VoiceMeUp_COM | well i mean why would asterisk crash |
| 20:24.45 | _VoiceMeUp_COM | that not good |
| 20:24.45 | Qwell[] | Don't they patch it? |
| 20:24.51 | _VoiceMeUp_COM | yeah they break it |
| 20:24.55 | Corydon76-work | Zion800: 1(start),ChanIsAvail |
| 20:25.03 | _VoiceMeUp_COM | but cant you Enforce the fact they still use Asterisk as the UA |
| 20:25.05 | Kumba_ | Aren't patches in Trixbox kind of like Patches in RedHat? They break everything... |
| 20:25.08 | Corydon76-work | Zion800: start+101,DoWhatever |
| 20:25.12 | _VoiceMeUp_COM | at least i could BLOCK any Trixbox form connecting to us |
| 20:25.26 | _VoiceMeUp_COM | and call us to activate one we knwo what version they use |
| 20:25.33 | Zion800 | Corydon76-work: ahh...gotcha..lemme try that |
| 20:25.34 | _VoiceMeUp_COM | i saw 0.007 pre 1 |
| 20:25.37 | _VoiceMeUp_COM | out ther elol |
| 20:26.12 | [TK]D-Fender | Zion800: its not in there, and you can't add to an old pstebin # |
| 20:27.15 | [TK]D-Fender | ok, time's up... gottaq go, back later-ish |
| 20:27.50 | *** part/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 20:27.51 | Kumba_ | If a PAP2 supports Sip 2.0, that's the V.2.0 one correct? |
| 20:28.59 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:29.51 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 20:29.53 | xo8ox | hello |
| 20:29.55 | Zion800 | Corydon76-work: Well, I guess the priority jumping is set right now, but it still doesn't jump. I think it's because the ${AVAILSTATUS} is 0 (unknown), so it just continues through the dialplan. Before, it used to know the channel status. |
| 20:30.37 | xo8ox | guys I added an exten in extension.conf, then added the added the voicemail in voicemail.conf and also created the sip.conf user but when I try to check the voicemail it says mailbox incorect |
| 20:31.01 | xo8ox | what else am I missing to create this new exten with voicemail ? |
| 20:31.27 | Corydon76-work | Zion800: that's the problem with binary conditions... you can't take into account multiple conditions |
| 20:32.11 | Zion800 | Corydon76-work: So any ideas on a way to fix it? Maybe another app that would what I need it to do? |
| 20:32.24 | Zion800 | that would do* |
| 20:32.35 | Corydon76-work | Zion800: Why don't you do what Qwell already suggested that you do? |
| 20:33.03 | Corydon76-work | Get rid of the jumping and do a conditional immediately after the ChanIsAvail to evaluate what to do next |
| 20:34.04 | Corydon76-work | ${AVAILSTATUS} contains the value of the device state |
| 20:34.08 | Zion800 | Ok...but I would be testing it against the ${AVAILSTATUS} variable, which is 0 AST_DEVICE_UNKNOWN. |
| 20:34.13 | Zion800 | ya... |
| 20:34.17 | _omer | sdf |
| 20:34.18 | mazpe | interesting.. why a sip extension will connect with xlite and it wont register with idefisk? |
| 20:34.19 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
| 20:34.58 | Corydon76-work | mazpe: might I suggest that you talk to the people who develop idefisk? |
| 20:35.13 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 20:35.38 | mazpe | Corydon76-work: i was just thinking maybe a configuration on my part? |
| 20:35.51 | Corydon76-work | Possibly, but they would know better |
| 20:36.06 | Corydon76-work | This is #asterisk, not #idefisk |
| 20:37.58 | Taadow | heheh |
| 20:38.28 | xo8ox | guys to setup an extension with voicemail I have to modify 3 files right ? extension.conf, voicemail.conf, and sip.conf .. right |
| 20:38.37 | Nuitari | and reload them yes |
| 20:38.44 | xo8ox | reload all ? |
| 20:38.47 | xo8ox | or just extensions |
| 20:38.54 | Nuitari | extensions is the dialplan reload |
| 20:39.06 | Nuitari | sip.conf is sip reload |
| 20:39.06 | xo8ox | so then how do I reload them all |
| 20:39.12 | xo8ox | aha i c |
| 20:39.16 | xo8ox | and for voicemail ? |
| 20:39.28 | Nuitari | reload app_voicemail.so, I think |
| 20:39.38 | xo8ox | in the asterisk cli ? |
| 20:40.20 | xo8ox | do I need to manually create the voicemail directories i the var/spool/asterisk/voicemail/ ?? |
| 20:40.28 | xo8ox | or it will create them |
| 20:40.42 | Nuitari | it will, if * has the proper permissions |
| 20:40.46 | Zion800 | Corydon76-work: I think there is still a problem with ChanIsAvail. No matter what, the ${AVAILSTATUS} is 0. I just redid my dialplan to get rid of priority jumping, as you and qwell recommended... |
| 20:40.47 | Nuitari | and yes |
| 20:42.15 | Nuitari | Zion800: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
| 20:43.17 | Zion800 | Nuitari: ? I think there is a bug in ChanIsAvail... |
| 20:44.07 | Zion800 | And I understand it is not useful in all cases, however, given that it worked perfectly for my use in Asterisk 1.2, and now it doesnt, leaves me feeling very suspicious that it is a bug |
| 20:51.33 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 20:53.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:55.57 | mocker | DUNDi question, is there a way to tell asterisk that an extension does *not* live on a server? |
| 20:55.59 | Kumba_ | Yay! Wont my Amp 25-pin crimper for $95... now I can charge all the other suckers to crimp custom cable length... |
| 20:56.14 | mocker | I have 64XX on one server, and 6442 on a different server |
| 20:56.21 | Kumba_ | err 25-pair |
| 20:56.41 | mocker | dundi sees 64XX first I'm guessing and routes the call their instead of to the more exact server.. |
| 20:58.05 | *** part/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 20:59.55 | Kumba_ | Anyone having issues with the new Polycom IP320/330's hanging after being on for a day or so? (i'm using PoE) |
| 21:01.23 | bkruse | Kumba_: I believe I have some in the lab on POE and havent had a problem |
| 21:01.27 | bkruse | no error messages? just froze? |
| 21:01.33 | bkruse | does it stay registered? or stop responding network wise |
| 21:02.08 | Kumba_ | Dial key stops responding |
| 21:02.14 | Kumba_ | can still navigate the menu |
| 21:02.18 | Kumba_ | asterisk says it's no longer there |
| 21:02.27 | Kumba_ | cant dial numbers on the keypad |
| 21:02.34 | Kumba_ | Like it's just generally locking up |
| 21:04.30 | Kumba_ | Kind of like a Grandstream :) |
| 21:05.13 | _DAW | I have 4 330's on 1.2.18 that have been up for a few weeks or so. No problems. |
| 21:05.57 | Kumba_ | Hmm... |
| 21:06.30 | Kumba_ | Just have one phone... guess i'll try plugging the other 2 in and see what happens... funny thing is, after I plugged the one phone in, the Sipura started acting up... |
| 21:10.34 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:11.34 | *** join/#asterisk steepcreek (n=steepcre@adsl-068-209-010-062.sip.asm.bellsouth.net) |
| 21:11.37 | structure | gerwinin, are you still here? |
| 21:12.55 | gerwinin | structure : yes |
| 21:13.25 | structure | I have more information on my issue if you have a moment. |
| 21:14.42 | structure | You say "the name is mostly not send by the pabx." If Asterisk connects over a digital PRI, wouldn't it pass the name over the data channel? |
| 21:14.48 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 21:15.02 | structure | Thanks for your help btw. |
| 21:18.10 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 21:19.43 | gerwinin | structure : let me check this |
| 21:23.03 | gerwinin | structure: it seems not |
| 21:23.38 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 21:23.39 | gerwinin | Structure : it is or a data channel or voice channel |
| 21:23.40 | mxmasster | hi all |
| 21:23.42 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:24.13 | mxmasster | i am looking for an example - i need to implement some decision logic based on times. i.e. if between 8am-5pm m-f send the call to a different location |
| 21:24.28 | mxmasster | can someone point me to an example |
| 21:25.03 | bkruse | gotoiftime() |
| 21:25.05 | bkruse | ;] |
| 21:25.08 | structure | gerwinin, They are data channels, as designated by signalling=pri_net in the wanpipe configs |
| 21:25.08 | bkruse | fixed! |
| 21:25.57 | gerwinin | structure: I know but what I am checking is in the pri spec how callerid is defined |
| 21:26.22 | irule | [Jun 15 14:36:11] NOTICE[17481]: chan_iax2.c:5636 update_registry: Restricting registration for peer '200' to 60 seconds (requested 300) how can I make it longer? I see this message very often every hour |
| 21:26.36 | Teccy | i'm having some trouble with a TDM400P w/2 FXOs. If i ring the card (i've tried with it both on an analogue PBX port and a plain PSTN line) asterisk answers and follows the dialplan, but the card never actually answers the line |
| 21:26.43 | Teccy | log from asterisk at: http://pastebin.com/929953 |
| 21:27.07 | Teccy | i've checked it's all using UK signalling, but still no luck. any thoughts? |
| 21:29.03 | gerwinin | structure: the d-channel can contain this info , checked here , but with me the card is showing the number and not the name as well on the analogue phone |
| 21:29.21 | gerwinin | structure: on the voip phone I have the correct number althoguh |
| 21:29.45 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:30.49 | gerwinin | teccy: I am not so into digium cards but it seems there is some problem with the connection factor |
| 21:31.22 | gerwinin | teccy: with analogue you can only connect a certain amount of things to the line |
| 21:32.05 | Teccy | gerwinin: it is the only device on the line |
| 21:32.30 | gerwinin | Teccy: let me check the logs for a moment |
| 21:35.14 | gerwinin | teccy can you set callprogress=no and busydetect=no |
| 21:36.11 | gerwinin | Teccy: can you set callprogress=no and busydetect=no |
| 21:38.24 | gerwinin | Structure: I am pretty sure now that the software from sangoma will not suppot this |
| 21:40.00 | structure | gerwinin, Really? I spoke to them and they claimed that no settings in their software had anything to do with callerid information |
| 21:40.22 | structure | gerwinin, So maybe that was their version of 'We don't support it'.. |
| 21:40.35 | gerwinin | structure: hehehe did you speak with the south african bloke :) |
| 21:40.49 | structure | gerwinin, heh no, he was Indian |
| 21:41.04 | gerwinin | structure: haha it seems that almost all companies have that nowadays |
| 21:41.07 | JerJer | has anyone else noticed the SPA-942s blink their backlight on every single SIP registration attempt ? |
| 21:41.19 | tzafrir_home | Teccy, it has answered and played a file. So what is hte problem? |
| 21:41.26 | [hC] | JerJer: yes. it drives me bonkers. |
| 21:41.31 | [hC] | JerJer: im not sure how to make it stop. |
| 21:41.32 | tzafrir_home | A wait(10) is a bit too much |
| 21:41.40 | gerwinin | structure: in a way he is right |
| 21:42.20 | JerJer | [hC]: we need to figure out WTF is up with that |
| 21:42.39 | structure | gerwinin, Why do you think it is the Sangoma software that will not allow it? |
| 21:43.05 | tzafrir_home | Teccy, how do you know that the card does not answer the line? |
| 21:43.19 | structure | JerJer, happens to me often when using speakerphone and the person speaking is loud.. I try not to look at it :) |
| 21:43.23 | [hC] | JerJer: yes, absolutely. it would be handy if it was controllable, to come on at certain times like cisco... but yeah, thats extremely frustrating seeing it blink. people ask me all the time why it does that. |
| 21:43.47 | gerwinin | structure: it is not that it doesn't allow it but it is more that it is a message that is not translated |
| 21:44.15 | gerwinin | tzafrir_home: I had this issue before and fixed it by setting the things I told him to set |
| 21:44.50 | tzafrir_home | gerwinin, maybe it answers, but only after 10 seconds? |
| 21:45.11 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 21:45.15 | gerwinin | tzafrir_home: that is what I thought could be an issue as well |
| 21:45.42 | tzafrir_home | he has an explicit Wait(1) before the Answer |
| 21:45.58 | tzafrir_home | Wait(10), that is |
| 21:46.05 | JerJer | structure: no its very much tied to the sip registration process |
| 21:46.25 | JerJer | i can set my sip reg to 30 seconds and every 30 seconds the backlight blinks |
| 21:46.30 | JerJer | but only on v1.4 |
| 21:46.33 | JerJer | not on v1.2 |
| 21:46.40 | JerJer | or ser/openser |
| 21:46.50 | structure | gerwinin, Interestingly enough, I just found that on incoming calls from the 3Com, if I insert a Wait(1) before logging the name it shows up correctly. |
| 21:47.03 | structure | gerwinin, If I do not have a Wait(1) and attempt to log the name, it is blank. |
| 21:47.09 | gerwinin | structure: :) |
| 21:47.19 | gerwinin | structure: than it seems to be solved |
| 21:47.46 | structure | gerwinin, So perhaps we need to have the 3Com perform a similar wait before ringing their extention. Nice :D |
| 21:47.59 | structure | (sp) |
| 21:48.18 | JerJer | structure: if you are talking about Caller*iD Name, that is totally expected |
| 21:48.51 | structure | JerJer, yes I am. So the sending of name is delayed a bit then? |
| 21:48.52 | gerwinin | structure: that seems pretty interesting |
| 21:49.09 | JerJer | structure: yes |
| 21:49.37 | JerJer | in fact I run with a wait,1 as the first priority on everything inbound |
| 21:50.23 | anonymouz666 | JerJer: looking at SIP register messages, there is nothing different from 1.4 to 1.2? |
| 21:50.43 | JerJer | i haven't dove that deep yet |
| 21:51.45 | JerJer | s/dove/dug |
| 21:52.40 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:54.46 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 21:54.52 | *** join/#asterisk bbryant (i=brett@nat/digium/x-fc2314ca80723780) |
| 21:58.50 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 21:59.10 | *** part/#asterisk joetester (n=joeteste@216.191.34.13) |
| 22:00.09 | mxmasster | when you specific an extension i.e. exten => 1, the next config item is the priority |
| 22:00.17 | mxmasster | what is the behavior of "n" as a priority |
| 22:02.57 | *** part/#asterisk angom (n=angom@red-corp-201.143.81.252.telnor.net) |
| 22:03.31 | _charly_ | mxmasster: n just adds +1 to the previous priority, but you must have at least priority 1 in your extension |
| 22:04.57 | gerwinin | teccy/structure: does it work now ? |
| 22:05.54 | structure | Well there's a little more to it, but alas I must disconnect I'm heading out soon. Thanks for the help and have a good weekend! |
| 22:06.18 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 22:07.02 | gerwinin | structure: okay, if I am not online you can reach met at gerwin@vanderkruis.net |
| 22:07.30 | gerwinin | structure: it would be interesting to know how it went further |
| 22:12.18 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 22:13.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:14.36 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:20.02 | mxmasster | i found an application but don't remember how to do it - i want to keep track of how many calls i send to a destination and when the count is reached send the calls somewhere else |
| 22:21.55 | *** join/#asterisk chronomex (n=duncan@c-24-19-6-204.hsd1.mn.comcast.net) |
| 22:23.03 | *** join/#asterisk Metfan2007 (n=metfan@189.136.82.104) |
| 22:23.28 | *** join/#asterisk snuff-away (n=snuffy@C-61-68-189-13.bur.connect.net.au) |
| 22:24.22 | Metfan2007 | Hi! I have a problem with Asterisk and NAT, I have ALL the ports in Asterisk side forwarded to the server, The phone in the other side can register ok, but no media pass!!! Maximum retries exceeded on transmission 21a9d69572c4fbeb6a3e5ab1e9255428@189.136.82.104 |
| 22:24.31 | Metfan2007 | <PROTECTED> |
| 22:24.35 | Metfan2007 | Any idea? |
| 22:24.41 | mmlj4 | Metfan2007: lemme guess, you're using SIP to connect, right? |
| 22:24.48 | Metfan2007 | yea, correct |
| 22:25.08 | mmlj4 | SIP no workie with NAT, unless you jump through hoops |
| 22:25.24 | Metfan2007 | I'm using an Aastra 480i |
| 22:25.28 | mmlj4 | check the wiki, maybe it has suggestions |
| 22:26.13 | mmlj4 | SIP only /sets up/ calls, and it's ignorant of NAT after that |
| 22:26.37 | Metfan2007 | what about Sip express router? does it helps?ยก??? |
| 22:26.48 | mmlj4 | again, check the wiki |
| 22:27.25 | Metfan2007 | I chekced and rechecked and googleit and read.... but I really don't understand all... |
| 22:27.51 | mmlj4 | ok... SIP is SESSION INITIATION or whatever PROTOCOL |
| 22:27.56 | Metfan2007 | yeap |
| 22:28.11 | mmlj4 | the media streams after that are independent of SIP |
| 22:28.14 | xkev | also referred to by some as shitty implementation protocol |
| 22:28.22 | Metfan2007 | HEHEHEHE |
| 22:28.29 | mmlj4 | there's no defined route for the media to traverse, if you're behind NAT |
| 22:28.35 | Metfan2007 | RTP right? |
| 22:29.04 | mmlj4 | hence, the error you got |
| 22:29.30 | mmlj4 | the quickest fix is for you to run asterisk on your edge device |
| 22:29.41 | mmlj4 | SER might help, but I've never messed with it |
| 22:30.17 | *** join/#asterisk perf3ktion (n=perf3kt@adsl-68-73-150-167.dsl.ipltin.ameritech.net) |
| 22:30.36 | mmlj4 | hrm... anyone know if arp munging might help with the NAT problem? turning it into a layer 2 issue instead? |
| 22:30.39 | blitzrage | SIP works fine behind NAT, as long as your NAT device doesn't suck |
| 22:30.45 | blitzrage | if you're using a Sonicwall, give up now and run |
| 22:32.37 | Metfan2007 | no, actually I'm using a 2WIRE |
| 22:33.46 | putnopvut | /disconnect |
| 22:33.52 | putnopvut | d'oh! |
| 22:33.54 | blitzrage | weak |
| 22:33.55 | Metfan2007 | do you know any IAX harphone that I can offer? |
| 22:33.55 | blitzrage | :) |
| 22:33.59 | blitzrage | nope |
| 22:34.00 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-f201ba001471e699) |
| 22:34.03 | blitzrage | they don't exist |
| 22:34.10 | Metfan2007 | what a... |
| 22:34.26 | blitzrage | it's because IAX isn't a real RFC yet (still draft) |
| 22:34.35 | Metfan2007 | so what's the solution in this kind of problem, I think this is common |
| 22:34.45 | blitzrage | get a better NAT device |
| 22:34.51 | blitzrage | it'll work fine behind IPtables |
| 22:34.55 | blitzrage | iptables* |
| 22:37.12 | mxmasster | how do i count how many calls set to an extension so i can limit the number to a value? |
| 22:37.31 | blitzrage | call-limit in sip.conf, or use the GROUP() and GROUP_COUNT() dialplan functions |
| 22:37.52 | mxmasster | blitzrage: thank you |
| 22:39.13 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 22:39.33 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.19 and 1.4.5 (June 15, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 22:39.38 | blitzrage | w00t! |
| 22:39.45 | russellb | heh |
| 22:40.02 | russellb | now to write some sort of announcement ... |
| 22:42.10 | blitzrage | russellb: what is the file that I would keep if I wanted to keep the same menuselect options again? |
| 22:42.56 | mxmasster | blitzrage: the docs on the wiki are overly confusing, i want to send up to 23 calls to a destination (simple via a dial command), calls 24+ should go to another destination (via a different dial command) - is there a simple example for this? |
| 22:43.59 | Corydon76-work | blitzrage: menuselect.makeopts |
| 22:44.21 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 22:45.07 | blitzrage | when the call comes in, you do Set(GROUP()=username). Then to check the number of channels in use, do something like: GotoIf($[${GROUP_COUNT(username)} >= 24]?too_many_calls) |
| 22:45.13 | blitzrage | Corydon76-work: thx! |
| 22:47.33 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 22:50.21 | Taadow | Any "Digium Staff" available to help resolve an elusive issue with our company Asterisk pbx? We will, of course, pay for the service. |
| 22:51.04 | Qwell[] | Taadow: You can call Digium sales, and see about some support contract stuff |
| 22:51.41 | Taadow | I called but it put me to voicemail, no queue. :( Is it possible to get per issue support or does it need to be a yearly subscription? |
| 22:51.56 | *** part/#asterisk xjagox (n=xjagox@190.8.158.12) |
| 22:52.50 | Qwell[] | Taadow: I don't know, sorry.. You might try calling on Monday, it's nearly 6pm here |
| 22:54.12 | Taadow | Qwell: Cool, thanks. I'll try that. |
| 23:03.19 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 23:08.43 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 23:09.14 | *** join/#asterisk Lann (i=Dewayne@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 23:09.40 | Lann | hey hey |
| 23:10.29 | Taadow | Qwell: Do you have a quick moment to check a log of 'sip debug ip <host>' which illustrates sip messsages occuring when a certain peer attempts to place a call which brings down the pbx for all other staff? I'm hoping you may recognize something with a quick glance. |
| 23:11.43 | Lann | i just thought of a crazy ambitious project idea...i'd like to hear thoughts on this...a voip based MUD |
| 23:13.05 | Lann | all controls being either keypad or voice based |
| 23:13.18 | mazpe | trade wars? |
| 23:13.25 | Taadow | I used to love that game. |
| 23:13.35 | mazpe | trade wars was the bomb! |
| 23:13.36 | Lann | yeah ok say like it takes place in modern day |
| 23:13.36 | blitzrage | Barren Realms Elite! |
| 23:13.41 | Taadow | mazpe: If you liked trade wars you may get a kick out of the game I wrote. :D www.northworld.ca |
| 23:13.46 | Taadow | bre kicked ass too. |
| 23:13.50 | blitzrage | fuck ya |
| 23:13.57 | tzanger | blitzrage: what else bad happened today? |
| 23:13.59 | Lann | but use like asterisk maybe combined with a mud server |
| 23:14.04 | Lann | and lots of voicea cting |
| 23:14.19 | tzanger | I got a contract to make a userspace app that bridges t1/e1 channels and ztdummy channels, heh |
| 23:14.33 | tzanger | barren realms elite |
| 23:14.34 | tzanger | oh lord |
| 23:14.38 | Lann | whats that? |
| 23:14.44 | blitzrage | tzanger: today was pretty good considering |
| 23:14.45 | Lann | gemstone III ftw |
| 23:14.47 | Taadow | exitilus |
| 23:15.13 | blitzrage | Legend Of the Red Dragon! |
| 23:15.18 | tzanger | yep LORD |
| 23:15.18 | Lann | haha aol muds! |
| 23:15.23 | tzanger | how many of those did I keygen, heh |
| 23:15.24 | blitzrage | BBS |
| 23:15.29 | blitzrage | tzanger: all of them |
| 23:15.36 | Lann | would it be a bad idea though, a mud over voip? |
| 23:15.47 | Lann | i think it could potentially be fun if done well |
| 23:16.01 | Lann | you could play it in traffic if you minimize the necessary keypresses |
| 23:16.23 | tzanger | hahah |
| 23:16.25 | Lann | like maybe one key toggle between talk and commands |
| 23:16.26 | Taadow | http://www.northworld.ca/downloads/nworld10.zip - Run in DOS or Win CMD shell, but remember... in Win use Alt-Enter for full (text) screen. Much nicer. :D |
| 23:16.53 | Lann | can asterisk create ...basically rooms |
| 23:17.08 | Lann | conference call rooms |
| 23:17.46 | Lann | could you imagine how fun it could be |
| 23:17.52 | Lann | you could have like a prison for people that act up |
| 23:18.00 | Lann | and a moderator could mess with people there |
| 23:18.05 | mxmasster | Jun 15 23:14:30 WARNING[8128] pbx.c: Requested contexts didn't get merged |
| 23:18.08 | Lann | that's what i'd do all day |
| 23:18.13 | mxmasster | how do i find out what this error is referring to |
| 23:19.42 | JunK-Y | mxmasster: cause you prolly ahve 2 contexts with the same name |
| 23:20.27 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:20.33 | mxmasster | JunK-Y: yeah, but what file? |
| 23:22.06 | *** join/#asterisk jtoy_ (n=jtoy@c-24-60-178-47.hsd1.ma.comcast.net) |
| 23:23.08 | JunK-Y | extensions.conf |
| 23:23.32 | JunK-Y | pbx is related to dialplan. |
| 23:24.31 | Lann | actually i guess i could use asterisk and some script to connect an asterisk user directly to a mud server that is for this purpose |
| 23:24.44 | Lann | but can asterisk connect two users voices? |
| 23:25.08 | Lann | i never really tried anything like that |
| 23:25.24 | Lann | could i somehow identify each user of the phone system and connect users arbitrarily |
| 23:25.41 | mxmasster | JunK-Y: yeah - except i definately do not have duplicate contexts in extension.conf |
| 23:28.11 | Lann | sub question, can asterisk say more than one thing at a time to a user? |
| 23:28.20 | Lann | or play more than one audio clip at once |
| 23:29.08 | JunK-Y | mxmasster: and in sip.conf or iax.conf ? |
| 23:30.51 | mxmasster | no - can i raise a debug so it will tell me the specific duplicate |
| 23:31.40 | JunK-Y | yes, in logger.conf, turn debug on for console and do set debug 4 |
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| 23:48.41 | Lann | anyone? |
| 23:48.57 | Lann | i mean i guess it can do background and foreground sound clips right? can it do more than 2? |
| 23:51.13 | Vorondil | Why would you want to do more than one? |
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