| 00:00.42 | ManxPower | kernel_ghost: for readhat style systems "chkconfig zaptel off" and "service zaptel stop" |
| 00:01.10 | kernel_ghost | ManxPower: so that obviously also prevents ztdummy from loading |
| 00:01.11 | kernel_ghost | cool |
| 00:01.23 | kernel_ghost | ManxPower: thankyou very much, you've been a great help |
| 00:01.42 | CGMChris | dont forget /etc/init.d/zaptel (start|stop|restart) |
| 00:01.55 | CGMChris | or /etc/rc.d/init.d/zaptel (start|stop|restart) |
| 00:02.12 | CGMChris | or, pick your favoritate applicable runlevel in /etc/rcX.d/init.d/ |
| 00:02.23 | *** join/#asterisk RB2 (n=RB2@pool-71-255-92-53.nwrknj.east.verizon.net) |
| 00:02.33 | kernel_ghost | lol, chkconfig is always easy |
| 00:02.47 | CGMChris | believe it or not, I can type /etc/init.d faster than chkconfig |
| 00:02.52 | CGMChris | years of practice. |
| 00:03.06 | kernel_ghost | yeah with the help of the tab key |
| 00:03.13 | x86 | <PROTECTED> |
| 00:03.17 | CGMChris | bah, who needs tab key. |
| 00:04.02 | kernel_ghost | i dont care about how standard it is, typing chkconfig servicename off or on is a good guarantee its off or on at boot |
| 00:05.04 | x86 | but it's not portable |
| 00:05.06 | CGMChris | sure... or goto /etc/rc3.d and either start your filenames with K or S. This is also a better way to see what is going to start/stop at subsequent reboots. |
| 00:05.10 | kernel_ghost | lol |
| 00:05.19 | x86 | removing it from /etc/init.d/rc3.d, is, however ;) |
| 00:05.25 | kernel_ghost | ok, i dont need a debate on linux internals, thanks for the answer ManxPower |
| 00:05.45 | x86 | kernel_ghost: sure, your way will work just fine for your purposes |
| 00:05.48 | kernel_ghost | at least with chkconfig its more of a switch, on and off |
| 00:05.55 | kernel_ghost | and if i need it later |
| 00:05.56 | x86 | kernel_ghost: just saying it's not portable, that's all :) |
| 00:05.58 | kernel_ghost | its still there |
| 00:06.07 | kernel_ghost | yep u have me on that point |
| 00:07.58 | CGMChris | I really like the set caller ID functionality...But is that not a security issue? Would it not be trivial to assign caller id as someone ELSES cell phone and crack their 4 digit passcode to get to voicemail? |
| 00:09.26 | CGMChris | Especially with unlimited concurrent calls being offered by some SIP trunk providers on a per minute basis... you could literally crack a voicemail password in an hour (or less). Is that not correct? |
| 00:10.07 | CGMChris | anyone? |
| 00:12.42 | Qwell | ...if they require passwords when calling from that CID |
| 00:13.32 | CGMChris | hmmm, good point. |
| 00:13.43 | hardwire | CGMChris: you'd be busted at around the 12th channel |
| 00:14.00 | CGMChris | hardwire: "Busted" how? |
| 00:14.02 | hardwire | cell phone voicemail systems don't have a super huge capacity from the outside networks |
| 00:14.17 | hardwire | it would seem like it was a DoS |
| 00:14.33 | hardwire | and LEC's can identify the carrier it came from |
| 00:15.08 | hardwire | you could just as easily call the voicemail access number then enter the mailbox number |
| 00:15.12 | hardwire | then crack the pin |
| 00:15.27 | hardwire | you'd be tracked down and skull****ed either way. |
| 00:15.32 | CGMChris | so, say 5 channels... say each channel tries 1 time every 60 seconds... you'd have the password in a day. Even if you slowed it down to take a week, you'd be ok. |
| 00:15.40 | hardwire | going after Palins voicemail? |
| 00:15.50 | hardwire | try the year she graduated high school. |
| 00:16.05 | CGMChris | LOL |
| 00:16.18 | hardwire | CGMChris: I wonder how many times a failed password is allowed. |
| 00:16.32 | hardwire | try it on your own cell. |
| 00:16.47 | CGMChris | hardwire: I wouldnt have the balls to test rate limits any other way. :) |
| 00:16.48 | hardwire | how would you know you cracked it? |
| 00:17.26 | hardwire | btw.. voicemail systems are notoriously bad with consistancy |
| 00:17.45 | hardwire | so you'd have to do sound detection |
| 00:17.48 | hardwire | matching |
| 00:17.48 | CGMChris | hardwire: Would probably be hardcore to implement, however, you could take a sample of the "invalid" passcode message from the provider and pattern match on it... or vice versa |
| 00:18.03 | hardwire | pattern matching sound is hard |
| 00:18.16 | CGMChris | I agree... so is CAPTCHA but i've matched it before |
| 00:18.29 | hardwire | more power to yah. |
| 00:18.40 | CGMChris | pass to function and return a percentage "likeness"... but I agree, hard. the timing has to be correct or you're comparing the wrong bits |
| 00:18.43 | hardwire | I personally think it's less invasive, easier, etc.. to just steal the phone and request a password reset. |
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| 00:18.52 | voxter | it seems like it would be pretty easy to defeat captcha by play the sound file thru TTS |
| 00:18.57 | voxter | play/playing |
| 00:19.01 | *** part/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 00:19.14 | CGMChris | voxter, yes.. that's easier than trying to decode a yahoo captcha, for example. |
| 00:20.18 | CGMChris | voxter: Your code may very well be able to successfully decode your 5 sample yahoo captchas, but as soon as you try it in the wild it will bust and work less than 5% of the time... you'll probably be lucky to get 1% success rate. |
| 00:20.37 | CGMChris | Back to voicemail, It could be equally difficult. |
| 00:21.40 | CGMChris | As long as it took to create a @#$@#$% dialplan, I'm not even going to attempt to crack passwords at this point. |
| 00:22.02 | ManxPower | CGMChris: how long did it take for you to create your dialplan? |
| 00:24.21 | hardwire | wait |
| 00:24.25 | hardwire | a peer can have multiple contexts? |
| 00:24.46 | ManxPower | hardwire: no |
| 00:25.05 | hardwire | ManxPower: got extconfig.txt handy? |
| 00:25.12 | CGMChris | ManxPower: Yesterday from 8am-10pm, 5pm to 9:30pm... then again today 6:30am to 8:15am, 11:00am to 12:00pm, and then 4:45pm to 6:30pm... let me add that up for you. |
| 00:26.13 | ManxPower | CGMChris: Ah, you're only 1% done then. |
| 00:26.20 | CGMChris | If my math is correct, 11 hours. |
| 00:26.26 | ManxPower | hardwire: never looked at it |
| 00:26.41 | CGMChris | I have a fully functional dialplan with queues, voicemail, agents, custom messages for holidays, etc. |
| 00:26.42 | hardwire | ManxPower: it has an example for multiple contexts for a peer. |
| 00:27.10 | ManxPower | wait until your users are using it. |
| 00:27.21 | CGMChris | dialplan rules to match patters to implement gizmo5 or my paid provider.. patters for long distance, lazy long distance (omit the 1), local calling... etc |
| 00:27.41 | ManxPower | *whine* Our old phone system did it this way! *whine* I have echo. *whine* I have to dial 9 *whine* I have to press send. |
| 00:28.14 | CGMChris | I wont force them to dial 9... and its already setup to dial w/o having to press send. |
| 00:28.30 | CGMChris | what else you got? :) |
| 00:28.51 | ManxPower | So how do you not use a timeout when dialing 12135551212 and 2315551212 |
| 00:29.26 | CGMChris | there is a timeout, but it dials by itself. You mean, the users may want it to dial immediately? |
| 00:29.50 | ManxPower | You have much to learn, grasshopper. |
| 00:30.52 | CGMChris | lol. I realize it can be modified to dial immediately, however, for my internal office use it is fine. |
| 00:32.10 | *** part/#asterisk gcbirzan (n=gcbirzan@pida/gcbirzan) |
| 00:32.21 | ManxPower | CGMChris: try calling a disconnected number |
| 00:32.38 | ManxPower | or a number that has been changed |
| 00:32.50 | ManxPower | is off to watch TiVo |
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| 00:41.15 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 00:42.07 | CGMChris | Eh, disconnected numbers seem to ring thru |
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| 00:58.33 | CGMChris | Good night * land. |
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| 01:14.31 | RipeR-81 | good evening.. gents |
| 01:14.57 | RipeR-81 | im having issues with a digium TE121 card |
| 01:15.07 | RipeR-81 | on an E1 connection |
| 01:15.16 | RipeR-81 | the card stays blinking on yellow |
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| 01:16.54 | RipeR-81 | any ideay? |
| 01:16.58 | RipeR-81 | any idea? |
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| 01:26.22 | RipeR-81 | :( |
| 01:28.09 | drmessano | Sorry, I am just here for the whoresdurves |
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| 01:34.20 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:38.08 | RipeR-81 | :( |
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| 01:48.55 | Qwell | jameswf: tested yet? |
| 01:49.00 | Qwell | drmessano: ...you too |
| 01:49.40 | jameswf | Qwell: Yeah i told you earlier i made it my dev whore :) |
| 01:50.00 | jameswf | needed to do dahdi testing as well so it worked out |
| 01:50.23 | Qwell | any issues? |
| 01:50.43 | Qwell | I'm also sitting on 1.6 RPMs now :D |
| 01:50.45 | *** join/#asterisk el_critter (n=el_critt@190.39.203.141) |
| 01:56.46 | NovceGuru | jameswf is having dahdi issues |
| 01:57.47 | drmessano | I had issues with my Dahdi too |
| 01:57.54 | drmessano | <-- Lots of therapy |
| 01:58.10 | drmessano | Im much better now |
| 01:58.13 | drmessano | But even if I wasnt |
| 01:58.17 | drmessano | WHATS IT TO YA, SUCKA? |
| 02:08.13 | *** join/#asterisk Nag (n=Nag@office.influence-reseaux.fr) |
| 02:08.21 | Nag | hi ! |
| 02:08.36 | Nag | someone here isn't sleeping please ? |
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| 02:13.08 | seanbright | ~ask |
| 02:13.09 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:16.25 | Nag | someone have tested cisco 7971G with SIP firmware v 8-4-1SR1S on asterisk please ? |
| 02:16.53 | Swabby | tu problemo asterisk? |
| 02:17.54 | Nag | i have a dialing problem, phone register weel, i can call him, i can call voicemail via direct key, but when i dial by hand i always have the "reorder" message... |
| 02:19.11 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:27.17 | el_critter | Hi, how can I obtain the latest sip.cfg for my polycom? |
| 02:27.36 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 02:31.56 | RipeR-81 | help |
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| 02:37.50 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:38.06 | jaytee | el_critter, from your Polycom vendor? |
| 02:39.11 | jaytee | el_critter, you might also try Polycom's site. They don't always list the very latest release. |
| 02:39.19 | el_critter | jaytee: Can't I download it from polycom's page? |
| 02:40.06 | el_critter | jaytee: Hmmm ok, didn't understand why it was impossible to download. |
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| 02:40.32 | jaytee | you can download the packages but you've gotta have an FTP server setup so you can provision the phone with it to update it. You can't just do the sip.cfg. You need to read their Admin guide for it and the upgrade docs |
| 02:42.44 | jaytee | el_critter, what version of SIP is your Polycom currently running? |
| 02:44.00 | el_critter | jaytee: 2.1.2.0049 |
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| 02:44.06 | jaytee | http://www.polycom.com/common/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf |
| 02:44.31 | jaytee | that's an excellent document for provisioning. |
| 02:45.01 | jaytee | what model Polycom phone? |
| 02:46.16 | el_critter | 330 |
| 02:46.24 | jaytee | http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
| 02:47.32 | el_critter | excellent, thanks a lot, first time playing with IP phones |
| 02:47.37 | jaytee | I see multiple version of the sip firmware packages that come with the sip.ld file (firmware) and the sip.cfg file plus phone.cfg 2.2.2 is the latest in the 2.x series and 3.0.4 in the 3.x series |
| 02:48.00 | msalaque | ~ask |
| 02:48.01 | jbot | it has been said that ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:48.31 | msalaque | ~help |
| 02:48.38 | jaytee | if you're going to have more than 3 Polycoms it makes sense to setup FTP provisioning. it's covered in The Book |
| 02:48.44 | jaytee | ~book |
| 02:48.45 | jbot | extra, extra, read all about it, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:48.52 | el_critter | jaytee: whats the difference between them? |
| 02:49.16 | jaytee | bug fixes, additional features |
| 02:49.47 | jaytee | I'm running mine at 2.2.0 at the moment and testing 2.2.2 |
| 02:51.19 | jaytee | el_critter, you can configure basic stuff by using a browser to go to the IP address of the phone but advanced stuff isn't configurable from there, only through provisioning via FTP or TFTP, FTP preferably. |
| 02:52.36 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 02:53.06 | jaytee | like making it ring for a 1 ring or warble and then go off-hook speakerphone. |
| 02:56.57 | el_critter | jaytee: didn't know that |
| 02:57.26 | jaytee | I had to go to 2.2.0 from 2.1.2 to get that feature in SIP |
| 02:58.52 | [TK]D-Fender | jaytee: Just about every version could do that... |
| 02:59.18 | jaytee | read that white paper and read the section in the book on provisioning and the Polycom SIP Admin guide for whatever version you're going to upgrade to and read the Release Notes for that version |
| 03:00.17 | jaytee | [TK]D-Fender, maybe I'm thinking of the Applications stuff like the weather ticker microbrowser app |
| 03:00.44 | [TK]D-Fender | jaytee: There were MB enhancements for 2.2 list tables, etc |
| 03:00.53 | [TK]D-Fender | jaytee: I think thats one I mentioned to you a while back |
| 03:01.06 | [TK]D-Fender | jaytee: My home 501 and work 320's are on SIP 3.1.0 |
| 03:01.08 | *** join/#asterisk moy (n=moy@189.169.90.224) |
| 03:01.32 | el_critter | jaytee: thanks a lot for your help |
| 03:01.40 | jaytee | el_critter, np |
| 03:01.54 | el_critter | bye |
| 03:02.10 | jaytee | el_critter, once you delve into them it's a little work but they are so powerful and they've got great audio quality and solid |
| 03:02.12 | jaytee | bye |
| 03:02.54 | el_critter | ok, I'll read that tomorrow morning |
| 03:03.02 | jaytee | [TK]D-Fender, what 2.x version of SIP do I need to be at to go to 3.x? can I hop straight from 2.2.0? |
| 03:03.46 | [TK]D-Fender | jaytee: What ver you are running doesn't matter so long as the modeel supports it |
| 03:04.01 | [TK]D-Fender | jaytee: You can jump from an IP 501 stock 1.5.2 cold to 3.1.0 |
| 03:04.42 | jaytee | ok, thought I saw something in the docs that said I had to go in stages. Maybe it was the bootrom stuff |
| 03:05.41 | jaytee | I went to Applebee's earlier. It's amazing what 2 Leinenkugle Summer Wheats will do to an old man's brain for recall :-) |
| 03:06.29 | jaytee | but hey, I had to have something to wash down the 4 mini-bacon cheeseburgers |
| 03:07.23 | jaytee | I don't see 3.1.0 listed on the downloads page. I'd probably have to go through my reseller for that version |
| 03:07.53 | jaytee | latest version I see is 3.0.4 |
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| 04:02.52 | drmessano^ | So.. |
| 04:03.07 | drmessano^ | Now that Asterisk 1.6 supports TCP SIP |
| 04:03.11 | drmessano^ | and UDP SIP |
| 04:03.17 | drmessano^ | 1.8 = IPX SIP? |
| 04:03.24 | Nugget | NetBEUI is where it's at. |
| 04:03.31 | [TK]D-Fender | drmessano^: Not so bold... you nead ARCNET first! |
| 04:04.06 | Nugget | LANtastic VoIP. |
| 04:04.29 | [TK]D-Fender | Nugget: I ran a LANTastic setup a loong time back, Novell 3, etc |
| 04:04.36 | Nugget | too |
| 04:04.47 | [TK]D-Fender | Nugget: at the office we still run Novell 5.0... BLEH |
| 04:04.56 | Nugget | http://macnugget.org/photos/archive-misc/lantastic |
| 04:04.57 | drmessano^ | Netware 3.12 FTW |
| 04:05.03 | Nugget | NetWare 3.12 was the pinnacle |
| 04:05.11 | Nugget | they lost sight of the path with 4.0 |
| 04:05.22 | volrepair | :( |
| 04:05.28 | [TK]D-Fender | I actually felt I had control with 3.12 honestly. I feel useless on 5.0 |
| 04:05.31 | *** join/#asterisk axisys (n=axisys@117.18.230.167) |
| 04:05.40 | volrepair | I had some 3.2 servers |
| 04:05.44 | volrepair | Good times |
| 04:05.44 | Nugget | http://macnugget.org/photos/media/netware312 |
| 04:06.08 | volrepair | niice |
| 04:06.13 | volrepair | I still have mine too lol |
| 04:07.57 | volrepair | I know a guy that lost his job over Netware 3.12 |
| 04:08.25 | volrepair | Steps to fail: |
| 04:08.41 | volrepair | 1. Make company buy large amount of memory |
| 04:08.49 | volrepair | 2. Add the memory |
| 04:09.06 | volrepair | 3. Be a dumbass and not config the OS to see it |
| 04:09.27 | volrepair | 4. Have someone go behind you and not see memory and have you fired for theft |
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| 04:12.48 | Nugget | heh, ouch |
| 04:15.41 | volrepair | So maybe I helped get him fired a little |
| 04:15.48 | volrepair | But that was the clincher |
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| 05:38.46 | alancio | hi, I'm having trouble setting up a digium TDM card, can anybody help me? |
| 05:39.40 | alancio | loading the wctdm modules results in: wctdm: probe of 0000:02:06.0 failed with error -5 |
| 05:43.06 | *** join/#asterisk fubada (n=fubada@adsl-75-28-151-170.dsl.wlfrct.sbcglobal.net) |
| 05:43.08 | fubada | hi |
| 05:43.20 | fubada | where can i find a reliable sip provider |
| 05:43.24 | fubada | for doing sip over iphone |
| 05:43.37 | fubada | any big names? |
| 05:44.39 | alancio | les.net |
| 05:45.33 | *** join/#asterisk MrNaz (n=mrnaz@bb116-14-129-45.singnet.com.sg) |
| 05:45.50 | fubada | canada? |
| 05:45.53 | fubada | im in the usa |
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| 05:47.36 | alancio | http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential |
| 05:48.02 | fubada | HUGE LIST |
| 05:48.05 | fubada | im already there |
| 05:57.19 | volrepair | moron |
| 06:20.59 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 06:21.54 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 06:22.01 | mahlon | alancio: did you provide power to the TDM card? (I know, sounds silly. Commonly overlooked.) |
| 06:22.33 | alancio | mahlon: I just switched the TDM card and its now working |
| 06:22.52 | alancio | the strange thing is that this card was working before with 4 fxo modules, I just changed 2 fxo with 2 fxs |
| 06:23.28 | mahlon | ugh. transient failures suck. |
| 06:24.44 | alancio | I guess maybe one fxs module isn't working |
| 06:43.00 | mahlon | I guess you could start module swapping between card to find the faulty one. |
| 06:43.08 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 06:43.11 | mahlon | However - yuck. |
| 06:45.56 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
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| 06:50.22 | flohack | Hi! I have an asterisk server 1.4.17 which shows very strange behaviour. It works quite well for some time, but suddenly some calls (all SIP) are stuck and I'm unable to establish new ones and the old calls do not disappear. |
| 06:50.54 | flohack | I'm currently at such a point, and have a debugger attached. Could someone please lend me a hand? |
| 06:51.05 | *** join/#asterisk swampwork (n=rew@64.238.252.218) |
| 06:51.50 | *** join/#asterisk AndyML (n=AndyML@c-68-62-170-33.hsd1.al.comcast.net) |
| 06:53.18 | AndyML | had dinner with Mark Spencer... |
| 06:53.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:02.01 | alancio | flohack: I would start by using the latest asterisk version |
| 07:02.30 | alancio | I'm using 1.4.21.2 and don't have that problem |
| 07:04.54 | flohack | Hmm....I don't like the windows way of fixing problem "upgrade or reinstall". More so because the problem is difficult to reproduce. |
| 07:05.52 | alancio | well then have fun with gdb |
| 07:06.15 | flohack | Is there a way to find out which channel is handeled by which thread? |
| 07:07.42 | flohack | As far as I can see from the show channels, the problem could be a stale channel (showing DIALSTATUS=CHANUNAVAIL) |
| 07:07.57 | alancio | there is a "core show threads" command, but I'm not sure its the same thread you want |
| 07:10.03 | flohack | ok, I can probably rule out some of the threads (IAX, mgcp, h323), but I'm still unable to tell which channel belongs to which thread |
| 07:12.18 | alancio | you have to add some flags to the code |
| 07:13.20 | flohack | ok, I'll probably have to be able to reproduce the problem reliably. Thanks for your help! |
| 07:14.11 | alancio | no problem |
| 07:16.49 | flohack | Hmm...one channel is stuck in ast_waitfor_nandfds, which (according to the comments in the source) is supposed to wait on a fd for a limited amount of time. There could be a bug in this function, or the timeout is way too high. |
| 07:25.53 | phpboy | Hey all, how do I set a timeout for a user to select an option and if the timeout is reached it must replay the menu? |
| 07:27.10 | *** join/#asterisk angryuser (n=Miranda@9.214.20.81.dynamic.adsl.abo.nordnet.fr) |
| 07:29.47 | alancio | there is a special "t" extension, meaning timeout |
| 07:29.57 | alancio | read the manual |
| 07:30.43 | alancio | read the dialplan basics and the reference for the applications you are using |
| 07:31.54 | *** join/#asterisk nasls_lsa (n=chatzill@cust-242-192.on1.ontelecoms.gr) |
| 07:33.41 | nasls_lsa | hello, I am trying to make a linksys SPA 3102 to work with a n80 ( nokia ) , that supports VoIP and I got little bit confuseb about settings , and if that I waant is posible, anyone can help ? |
| 07:35.58 | alancio | the n80 supports sip? |
| 07:40.20 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 07:41.53 | nasls_lsa | yeep |
| 07:42.08 | nasls_lsa | and I tested successfully with a voip provider |
| 07:42.37 | *** join/#asterisk pluesch0r (n=pluesch0@91.186.158.42) |
| 07:43.34 | ana_micho | Hi My sip clients connect to my asterisk box and I forward calls to my SIP PSTN Gw...however one of those gateways is H323 can Asterisk translate from Sip to H323 ? |
| 07:47.00 | angryuser | ana_micho : hm, try ;) normally iax can call sip so why not h323 ? |
| 07:47.19 | ana_micho | angryuser, actually it is the other way around |
| 07:47.25 | ana_micho | Sip -> H323 |
| 07:48.01 | angryuser | ana_micho : it does not matter |
| 07:48.52 | angryuser | ana_micho : noone would do one directional compatibility, both ways or nothing at all |
| 07:55.10 | nasls_lsa | see ya all latter.. |
| 07:55.12 | *** part/#asterisk nasls_lsa (n=chatzill@cust-242-192.on1.ontelecoms.gr) |
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| 08:16.54 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 08:16.57 | casix | hello |
| 08:20.27 | casix | for 30 minutes i'he had this error in all my outgoing calls: "Re-invite to non-existing call leg on other UA." where can I find information of this error? |
| 08:22.12 | *** join/#asterisk FabiOne (n=FabiOne@151.13.190.20) |
| 08:24.12 | *** join/#asterisk someoneelse (n=savek@ip-174.emscb.ruhr-uni-bochum.de) |
| 08:28.25 | someoneelse | is there any problem with the asterisk.org server? |
| 08:28.47 | FabiOne | hi all |
| 08:29.02 | someoneelse | path traces end right after level3->apid |
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| 09:18.26 | *** join/#asterisk sluxor (n=expatjob@server10.expatjob.net) |
| 09:19.08 | sluxor | Which version of asterisk has more codec support ? 1.4.22 or 1.6.0.1 |
| 09:19.41 | sluxor | and I do mean video codec? |
| 09:35.26 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
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| 09:38.00 | flohack | I have found a deadlock in asterisk 1.4.17, can someone please lend me a hand in getting all the relevant information in order report a bug (I may not be able to reproduce it)? |
| 09:38.19 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 09:56.28 | ITguru | greets everyone in the room |
| 09:56.44 | flohack | Can someone tell me what the asterisk autoservice if for? |
| 09:58.38 | ITguru | Am I right to say that, first you set up the extensions, then you setup the trunk then you setup am inbound route? |
| 10:11.45 | obruT | anyone has experience with faxing over TDM ? I got asterix box connected to PSTN over E1, I would like to send and receive faxes... I don't need complete mail-to-fax and fax-to-mail solution, just send-this-tiff to fax, and receive-fax-and-store-to-tiff.... |
| 10:16.19 | flohack | I'm pretty certain that I hit the following bug: http://bugs.digium.com/view.php?id=12470 Can someone tell me how I can find out in which release the fix was incorporated? |
| 10:17.43 | ITguru | My trunk is a zaptel device, which has a number of lines coming into it - I've got no alarm lights on it - everything is green. That trunk is configured to call one particular extension. Each time I try, the number is unobtainable - Can someone help be devise a test, so that I know the hardware is actually all working? |
| 10:22.05 | *** join/#asterisk feeds (n=feeds@85-135-237-72.adsl.slovanet.sk) |
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| 10:28.21 | *** part/#asterisk __chris (n=chris@unaffiliated/footy247) |
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| 10:47.40 | *** join/#asterisk blunation (n=blunatio@70.89.8.201) |
| 10:47.57 | blunation | Can anyone here help me with a few things? |
| 10:49.52 | feeds | I sure would but I am a begginer in Asterisk |
| 10:49.53 | feeds | :( |
| 10:50.06 | blunation | well thats good. |
| 10:50.12 | blunation | What have you started with so far? |
| 10:50.24 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 10:50.39 | feeds | I installed it and tryied to start up but errors come up. |
| 10:51.05 | feeds | BTW: im using fedora9 |
| 10:51.09 | blunation | Have you ever tried PBX In a Flash |
| 10:51.17 | feeds | no |
| 10:51.25 | feeds | what' that? |
| 10:52.21 | blunation | Well if your learning you want to start with that to get use to the platform and the gui. |
| 10:52.52 | feeds | what OS are you using? |
| 10:52.55 | blunation | I recommend using a static IP and make sure you have INTEL |
| 10:53.04 | blunation | I have several. |
| 10:53.08 | feeds | me too |
| 10:53.11 | blunation | I use Fedora |
| 10:53.29 | blunation | I use PBX in FLASH on some and haven't had a problem. |
| 10:53.34 | blunation | its an ISO easy to install. |
| 10:53.58 | feeds | you mean: PBX in a Flash is a OS/ |
| 10:54.00 | feeds | ? |
| 10:54.23 | blunation | yes |
| 10:54.45 | feeds | too bad I don't have much Disk Space left |
| 10:55.03 | feeds | PBX in a flash i think is relevant to AsteriskNOW |
| 10:57.15 | blunation | asteriskNOW |
| 10:57.22 | blunation | which one's better? |
| 10:57.47 | feeds | no idea |
| 10:57.58 | feeds | haven't tried anyonbe |
| 10:58.05 | feeds | of them |
| 10:58.18 | feeds | BTW i tried running asterisk under fedora 9 |
| 10:58.42 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:02.00 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:09.33 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 11:09.39 | KermitTheFragger | hi all, is it possible to tell misdn which port to use for outgoing calls ? |
| 11:10.14 | KermitTheFragger | so instead of using port 1,2,3,4 in that order, tell it to use port 4,3,2,1 in that order ? |
| 11:11.01 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-115.moldtelecom.md) |
| 11:11.26 | kotique | hello. I want to match peer on number in INVITE. Is it possible ? |
| 11:12.47 | blunation | Can someone tell which is better IAX or SIP |
| 11:12.58 | blunation | which devices handle IAX? |
| 11:13.15 | blunation | And what are the most cheapest but reliable SIP/IAX Devices? |
| 11:13.27 | blunation | ATA's and Office Units? |
| 11:15.24 | blunation | Seriously can anyone assist me. |
| 11:15.30 | feeds | I think SIP is better than IAX but not shure |
| 11:15.46 | feeds | about the SIP phones, no idea |
| 11:15.58 | blunation | Anyone here either an ITSP? |
| 11:16.08 | blunation | that'll will guide me in the right direction. |
| 11:16.27 | blunation | Feeds.. make sure you get providers that go on per minute. |
| 11:16.39 | blunation | don't do packages ... |
| 11:16.47 | feeds | blunation, thnx |
| 11:17.18 | blunation | well great what a wasted time |
| 11:17.24 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
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| 11:56.07 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 11:56.16 | _Raptor_ | hi folks |
| 11:56.40 | *** join/#asterisk write_erase (n=Olivier@goodgw.m2m-fleet.com) |
| 11:57.03 | write_erase | Hi. Any command to check an extension has defined destinations ? |
| 11:57.08 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 11:58.08 | tzafrir_laptop | dialplan show extension@context |
| 11:58.09 | write_erase | I often have app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) that makes Asterisk crash |
| 11:58.14 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:01.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:02.21 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 12:09.37 | UnixDawg | ok who stole asterisk.org and asterisknow.org |
| 12:09.46 | UnixDawg | they are not responding |
| 12:09.52 | UnixDawg | digium.com is thou |
| 12:09.56 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:09.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:10.52 | UnixDawg | this bites |
| 12:11.03 | coppice | asterisk.org responds for me |
| 12:11.18 | UnixDawg | I get digium.com fine |
| 12:11.31 | UnixDawg | asterisk.org and asterisknow.org time out |
| 12:11.59 | coppice | you've been censored. move to the land of the free |
| 12:12.47 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 12:16.08 | UnixDawg | bee fre with freeswitch |
| 12:16.19 | UnixDawg | lol |
| 12:16.53 | coppice | the US is just the land of the suddenly much cheaper |
| 12:17.59 | *** join/#asterisk stevej678 (n=steve@202.7.249.15.dynamic.rev.aanet.com.au) |
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| 12:23.28 | riddlebox | for what its worth, I watched an Ad in St Louis about Mckain, and it showed his "call Center" which was using GXP2000 phones |
| 12:23.37 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:24.24 | coppice | not GOP2000 phones? |
| 12:24.32 | riddlebox | nope lol |
| 12:24.36 | coppice | or GOP2008 |
| 12:24.57 | riddlebox | you are doing to be doing that all day now |
| 12:25.20 | riddlebox | I am going to assume they were using asterisk |
| 12:25.53 | coppice | maybe the use of GXP2000 phones is why they are failing to get their message across |
| 12:25.55 | riddlebox | the weird part is that we do the phones for all senators here and the government requires the use Avaya Partner systems |
| 12:25.59 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 12:27.16 | riddlebox | this sucks i have to be up early for Charter to come give me digital cable |
| 12:28.29 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 12:29.30 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 12:29.34 | Dr-Linux|home | hi |
| 12:30.09 | riddlebox | well I think I am done playing with the gui, and am going to switch my home system back to regular asterisk |
| 12:31.24 | Dr-Linux|home | what is the best way to upgrade asterisk? |
| 12:32.04 | riddlebox | ./configure make make install |
| 12:34.21 | *** join/#asterisk [netman] (n=netman@241.Red-88-17-243.staticIP.rima-tde.net) |
| 12:38.57 | Dr-Linux|home | riddlebox: i'm running PRI's as well |
| 12:39.13 | petererer | bah, i failed upgrading a 1.2 install to 1.4 :( |
| 12:39.30 | petererer | different machines, fortunately :) |
| 12:39.31 | Dr-Linux|home | ahh |
| 12:39.38 | Dr-Linux|home | petererer: what happend? |
| 12:39.52 | petererer | it... didn't start |
| 12:40.00 | petererer | a few warnings but no actual error message |
| 12:40.30 | petererer | probably need to check the configs over a bit more |
| 12:41.52 | Dr-Linux|home | petererer: is it the same machine and same OS? |
| 12:42.19 | Dr-Linux|home | did you remove modules proir to install new asterisk version? |
| 12:43.51 | petererer | Different machine, different OS... |
| 12:44.07 | petererer | ponders another go :) |
| 12:44.52 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
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| 13:01.24 | cfh | hi, is best use tcp openvpn tunnel to encapsulate IAX trunk or udp tunnel ? |
| 13:01.25 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 13:05.28 | riddlebox | weird how I go back to manual configs and my external sip clients can connect, but when I use the gui, they dont work |
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| 13:26.36 | dandre | hello, |
| 13:27.44 | *** join/#asterisk angryuser (n=Miranda@9.214.20.81.dynamic.adsl.abo.nordnet.fr) |
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| 13:30.05 | jazzmann | hi deos anyone know how to integrate sphinx 4 with asterisk thanks |
| 13:30.17 | dandre | I am trying to have the same hold functionality than on a sip phone but from the manager interface. How can I do? |
| 13:30.17 | dandre | I already have read many docs with transfering to an extension but I don't see how to use it to have hold function |
| 13:31.33 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
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| 13:47.55 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 13:53.24 | CGMChris | Who does everyone use for SIP trunking? I'd like to get as much input as possible in hopes of finding the best rates/plans. |
| 13:55.05 | sdaniels | im with VoicePulse |
| 13:58.12 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 13:59.13 | jaytee | I don't use SIP trunking |
| 13:59.25 | *** join/#asterisk fred-tmft (n=fred-tea@ip70-171-36-194.ga.at.cox.net) |
| 14:00.02 | *** part/#asterisk fred-tmft (n=fred-tea@ip70-171-36-194.ga.at.cox.net) |
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| 14:02.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.20 | Knightfal | Guys I am trying to Pass the DID code into the URL parameter exten => s,8,Queue(queue-myqueue|t|DIDHERE||180) But Im not having much luck can anyone assist :) |
| 14:09.19 | *** join/#asterisk bradleyprice86 (n=bprice@68-113-101-90.static.leds.al.charter.com) |
| 14:09.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:10.04 | bradleyprice86 | Quick question, which file specifies which t1 is being used if you have multiple? |
| 14:14.00 | *** join/#asterisk mytux (n=oli@213.219.165.210.adsl.dyn.edpnet.net) |
| 14:14.11 | mytux | hello |
| 14:14.41 | mytux | is it possible to download the webinterface (that you see on the website of asterisk now) |
| 14:15.01 | mytux | I want to use it on a box I installed myself with asterisk 1.6 |
| 14:15.24 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 14:15.47 | jaytee | bradleyprice86, could you be a little more descriptive about "which t1 is being used"? |
| 14:21.51 | jaytee | bradleyprice86, are you asking which files defines what Zap group the T1 channels of a T1 span are a member of or which file determines how to route the a call out over a T1 channel? |
| 14:24.18 | dandre | I am trying to have the same hold functionality than on a sip phone but from the manager interface. How can I do? I already have read many docs with transfering to a special extension but I don't see how to use it to have hold function |
| 14:25.22 | bradleyprice86 | Ok, We have 2 t1's one is used for incoming and the other for outgoing. |
| 14:25.59 | bradleyprice86 | One of the t1's is currently down. I believe it is group 2. So, I am trying to revert the outbound calls back through group 1 |
| 14:27.33 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-7a93497a9dab1e93) |
| 14:27.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:31.35 | CGMChris | jaytee: If you dont use SIP trunking, does that mean you just use hard lines? |
| 14:36.37 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.93) |
| 14:36.49 | *** join/#asterisk iwws (n=iwws@213.129.44.243) |
| 14:38.19 | jaytee | CGMChris, yes at this time we only use PRI for inbound/outbound. It's costlier than using an ITSP but compared to most ITSPs it's more reliable. |
| 14:40.08 | *** join/#asterisk oilinki (n=oil@ppp-124-120-13-103.revip2.asianet.co.th) |
| 14:44.13 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 14:45.10 | *** join/#asterisk ManxPower (n=manxpowe@195.sub-70-223-176.myvzw.com) |
| 14:47.48 | *** join/#asterisk kpettit (n=keith@99-172-37-26.lightspeed.tblltx.sbcglobal.net) |
| 14:49.12 | kpettit | I'm having a hard time with MeetMe rooms. I'm thinking it's a "timing" issue sense I don't have a zaptel card. Whenever I dial a meet me room, I get "That is not a valid confrenece number" |
| 14:49.58 | kpettit | I've tried doing the meetme's with and without passwords, and I've verified from the CLI that it's being dialed correctl. But it always does the "invalid conference". |
| 14:50.25 | *** join/#asterisk oilinki7 (n=oil@ppp-124-120-9-96.revip2.asianet.co.th) |
| 14:51.16 | *** join/#asterisk jtodd (n=jtodd@ip67-88-206-98.z206-88-67.customer.algx.net) |
| 14:51.28 | kpettit | Any ideas? |
| 14:51.59 | ManxPower | kpettit: is ztdummy loaded? |
| 14:52.23 | kpettit | no |
| 14:52.43 | ManxPower | kpettit: what zaptel compat card do you have? |
| 14:53.02 | CGMChris | kpettit: Are you watching the output of "asterisk -r" when you are testing? |
| 14:53.52 | kpettit | I don't know what you mean by "compat card". But I don't have any digum hardware |
| 14:54.02 | CGMChris | jaytee: I have not had any downtime yet with net2phone. Should I expect frequent outages in the future? |
| 14:54.16 | CGMChris | kpettit: Run asterisk -r and watch its output for important error details when testing. |
| 14:54.20 | kpettit | CGMChris: yes. It looks like the dialplan is correct. |
| 14:54.43 | ManxPower | kpettit: MeetMe requires a zaptel timer and mixer. This can be provided by ztdummy or by a zaptel card. |
| 14:54.50 | kpettit | CGMChris: been doing that. It's going to the conference room, I can see pin is entered correct. Conference just does the "invalid conference" |
| 14:55.34 | kpettit | ManxPower: My rpm of zaptel and zaptel-devel has ztdummy, but it's not loading. I'll try to compile it. Thanks |
| 14:56.44 | adr3nalin3 | Hey guys, I am having a problem forwarding faxes to outside line, the faxes are coming in on a PRI then being forwarded out a 1888 number. When I call the number I get fax tones but the fax doesn't go through, are there any special considerations b/c it is a fax instead of voice? |
| 14:58.28 | ManxPower | adr3nalin3 does it work if you don't "forward the fax"? |
| 14:58.50 | adr3nalin3 | If I call the 888 # direct? yes. |
| 14:59.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:59.19 | adr3nalin3 | One thing I noticed is that it rings twice before asterisk sees the call. |
| 14:59.27 | ManxPower | adr3nalin3: I was trying to determine if Asterisk is causing a problem with faxes or not. Since you don't have a local fax machine installed, we can't really test that. |
| 14:59.50 | ManxPower | adr3nalin3: put your /etc/zaptel.conf on pastebin.ca |
| 15:00.04 | adr3nalin3 | ManxPower: I do have a local fax machine, and I was attempting to use the Linksys PAP2 at first. |
| 15:00.14 | adr3nalin3 | but that seemed to be utterly unreliable. |
| 15:00.37 | ManxPower | adr3nalin3: fax over SIP is not reliable for most people |
| 15:01.01 | ManxPower | and Fax over Voice over IP over Internet works for almost nobody. |
| 15:01.41 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 15:02.14 | adr3nalin3 | ManxPower: I should probably have the telco forward our fax number to the 888 number instead to alleviate the problem. |
| 15:02.34 | adr3nalin3 | take * out of the equation completely |
| 15:02.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:04.19 | *** join/#asterisk jplank (n=GBove@reports.nyigc.net) |
| 15:04.40 | ManxPower | adr3nalin3: Yes. We don't run any fax machines thru Asterisk. Any fax machines get a POTS line direct from the telco |
| 15:05.09 | jplank | I see in the wiki that in order to use option e with the directory command, I have to apply a patch, but that patch is from 2006, and I'm running 1.4, and the patch shows it was committed to the trunk in 2006, is it safe to assume thats just old info? |
| 15:05.17 | adr3nalin3 | ok, thanks ManxPower |
| 15:05.36 | ManxPower | jplank: what does the "e" option do? |
| 15:05.48 | jplank | speaks the extension before transferring |
| 15:06.08 | ManxPower | jplank: what does the "core show application directory" tell you? |
| 15:06.16 | jplank | good poiny |
| 15:06.16 | *** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net) |
| 15:06.31 | ManxPower | jplank: that is the OFFICIAL docs for Asterisk applications |
| 15:06.32 | jplank | its there :) |
| 15:06.47 | jplank | I always forget about that, my first place is always the wiki :( |
| 15:06.58 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:07.22 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 15:07.27 | Rienzilla | Hey everyone |
| 15:07.54 | Rienzilla | Anyone here experience with hooking up snom phones (300's and 360's) with a central directory? |
| 15:08.04 | kpettit | ManxPower: I just compiled and install my own zaptel. Which worked, I now have loaded ztdummy. But when I try to start asterisk I get "Zaptel time test failed to set ZT_TIMERCONFIG to 160" |
| 15:08.13 | ManxPower | jplank: going to the wiki first is like going to a witch doctor for a broken leg. |
| 15:08.13 | kpettit | Haven't seen that error before, any idea what that's about? |
| 15:08.32 | ManxPower | kpettit: I suspect your kernel HZ is not set to 1000 |
| 15:08.41 | ManxPower | it's a kernel compile setting. |
| 15:08.44 | jplank | I know, most of that info is out of date, but it *usually* puts me on the right track |
| 15:09.00 | jplank | I always forget about show application though, I think I need to put a little sticky up on my monitor |
| 15:09.00 | ManxPower | jplank: except when it doesn't. |
| 15:09.18 | kpettit | ManxPower: oh bugger. So is there anything I can do about it? I don't have the option to be able to re-compile kernel |
| 15:09.21 | ManxPower | Yes you do. Some time you will waste a day because you used the Wiki |
| 15:09.42 | ManxPower | kpettit: if you can't recompile the kernel then you cannot change the kernel HZ |
| 15:09.52 | ManxPower | so just give up on MeetMe |
| 15:10.17 | kpettit | ouch. My issue is I'm running on a Xen server. Voice Quality is good, but I guess this zaptel thing isn't going to work |
| 15:10.21 | *** join/#asterisk Jas_Williams (n=Jason@host86-147-180-245.range86-147.btcentralplus.com) |
| 15:10.25 | ManxPower | kpettit: I believe the kernel defaults to 1000Hz or 1024Hz, but vendors change that. |
| 15:10.36 | kpettit | I've looked at the "app_conference" module, but apparently it only works on asterisk 1.2 |
| 15:10.46 | ManxPower | kpettit: no I would not expect zaptel to work in a VM |
| 15:11.30 | kpettit | I was exciteted about app_conference becuase it's supposed to work without zaptel, but dosen't look like it's been update to work with 1.4 series |
| 15:13.17 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279393795.dsl.bell.ca) |
| 15:17.36 | *** join/#asterisk moy (n=moy@nat/ibm/x-532a622737bc2e8e) |
| 15:17.47 | kpettit | does asterisk 1.6 have any different way to do meetme? I know duhdi is used instead of zaptel in 1.6, not sure if that would make a difference for me or not |
| 15:18.28 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 15:19.22 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 15:29.11 | CGMChris | ManxPower: You sure do field alot of 1000Hz questions dont you. |
| 15:31.27 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 15:33.06 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:33.10 | jplank | Isn't the only major difference between zaptel and duhdi the name? |
| 15:33.55 | Zeeek | hey thete hi there ho there |
| 15:34.06 | ManxPower | kpettit: 1.6 is no different in it's MeetMe requirements |
| 15:34.55 | file | 1.6.0 and 1.6.1 is no different, 1.6.2 will have a conferencing solution that will not require Meetme |
| 15:36.26 | Zeeek | we be talkin' asterisk live in 20 minutes or so, so ask your DAHDI if you can come by |
| 15:36.42 | file | erm... not require dahdi |
| 15:37.02 | Zeeek | there seems to be a rant about dahdi on the ml? |
| 15:37.33 | *** join/#asterisk AndyML (n=AndyML@nat/digium/x-e48e4e172d9c6c54) |
| 15:38.13 | ManxPower | Zeeek: so business as usual on the mailing list? |
| 15:38.35 | Zeeek | heh |
| 15:39.05 | Zeeek | ManxPower: did you go to astricon? |
| 15:39.34 | ManxPower | Zeeek: no |
| 15:39.48 | ManxPower | ~manxpower |
| 15:39.49 | jbot | extra, extra, read all about it, manxpower is NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org http://www.fnords.org/skillslist.html |
| 15:40.03 | Zeeek | You remember we met in Madrid, right? (I have so many online personna I'm not sure who I am) |
| 15:40.12 | ManxPower | Yup |
| 15:40.31 | Zeeek | In fact, I was surfing the other day and found a dead link to your site about * |
| 15:40.53 | Zeeek | I had it in a list about asterisk docs |
| 15:41.02 | ManxPower | For some reason I thought we also met in Amsterdam -- my memories of the Amsterdam part of my trip is a little vague. |
| 15:41.11 | Zeeek | s/list/bookmark from 1990's/ |
| 15:41.42 | Zeeek | ManxPower: didn't you do the full tour that year? |
| 15:41.56 | ManxPower | Zeeek: my customer of 10 years got a new IT manager and he's bringing all his own people, so I have like 1/8 the normal number of billable hours. |
| 15:42.05 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:42.15 | Zeeek | ah. More time for "research" |
| 15:42.18 | ManxPower | Zeeek: Stockholm, Antwerp, amsterdam, delft, eindhoven, madrid |
| 15:42.24 | dominic1 | hello guys.... Can anybody tell me where I can download the tar gz of dahdi? |
| 15:42.32 | dominic1 | I can't find it on asterisk.org |
| 15:42.41 | ManxPower | Zeeek: Need revenue first. 8-| |
| 15:42.43 | dominic1 | okay... found it |
| 15:42.53 | dominic1 | on the right menue.... omg |
| 15:43.21 | ManxPower | Zeeek: I'm trying my hand at selling echo cancelers on ebay |
| 15:43.33 | Zeeek | EBAY! EBAY IS THE DEVIL |
| 15:43.44 | Zeeek | and Paypal ain't far behind |
| 15:43.47 | bradleyprice86 | How would I change the group my outgoing calls were using. I have 2 T1's an internal and outgoing. And I am trying to figure out where I would change that. |
| 15:43.55 | ManxPower | Zeeek: I sold my soul to the devel 5 years ago for a pack of smokes and a tank of gas. |
| 15:44.04 | ManxPower | devel? LOL! Devil |
| 15:44.31 | ManxPower | Zeeek: If I'm successful I'll try my own site to sell the stuff |
| 15:44.39 | Zeeek | hey if anyone wants to sell echo cancellers or anything else asterisk-related join us now: http://voipusersconference.org gives the SIP URI (actually it's an IP) |
| 15:44.52 | Zeeek | DEVEL devil |
| 15:44.54 | Zeeek | muhahaha |
| 15:45.17 | ManxPower | Zeeek: my internet connection doesn't really work well with VoIP. 8-| |
| 15:46.09 | ManxPower | bradleyprice86: Um, you change the group= line in /etc/asterisk/zapata.conf |
| 15:46.40 | Zeeek | ManxPower: voip doens't play well on the Internets |
| 15:46.53 | Zeeek | gotta run over to #voip-users-conference |
| 15:47.02 | ManxPower | Zeeek: it does not play well with a high jitter CDMA IP connection. |
| 15:47.54 | Zeeek | ok, you have a rain check... for now |
| 15:48.02 | Zeeek | Later good people |
| 15:48.06 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:49.51 | *** join/#asterisk ManxPower (n=manxpowe@195.sub-70-223-176.myvzw.com) |
| 15:51.22 | riddlebox | ManxPower, what do you think of snom phones? |
| 15:53.16 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 15:56.55 | tzafrir_laptop | dominic1, as usual, under http://downloads.digium.com/pub/telephony/ |
| 15:57.32 | tzafrir_laptop | dahdi-linux and dahdi-tools are separate |
| 15:58.22 | riddlebox | wohoo charter is here to install my dvr, they told me it was to compilcated to install myself haha |
| 15:58.45 | riddlebox | the guy saw my mythtv setup and said I think you could have handled this |
| 16:04.47 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:07.39 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:08.49 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:09.05 | JerJer | has GET DATA on AGI changed? |
| 16:09.47 | JerJer | i am sending: GET DATA theprompt 6000 10 and it won't play theprompt nor wait 6 seconds |
| 16:10.45 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:14.01 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 16:14.58 | *** join/#asterisk luke-jr (n=luke-jr@2002:18fc:16e6:0:20e:a6ff:fec4:4e5d) |
| 16:15.11 | luke-jr | Is it possible that Asterisk would leave out DTMF in Monitor() recordings? |
| 16:15.42 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-074-132.pools.arcor-ip.net) |
| 16:15.58 | shriven | Hello, I'm hoping someone can give me a pointer in figuring out this issue... The beginning of my calls, whether it be to voicemail or a person, as well as the beginning of sentences are getting clipped off a little. How would I go about diagnosing that? I have read about sip clients not sending silence and that causing what seems like similar issues, but none of the clients I'm using have a setting to enable or disable that. |
| 16:16.32 | seanbright | how are you answering the calls? |
| 16:16.58 | shriven | It doesn't seem to matter. |
| 16:17.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:17.03 | seanbright | hmm |
| 16:17.17 | shriven | It happens at the beginning of a voicemail response, when I call a real mobile... another sip client... etc. |
| 16:17.32 | shriven | I have tried adding Wait() to my dialplan |
| 16:17.35 | shriven | but that doesn't help |
| 16:17.42 | luke-jr | :/ |
| 16:18.12 | shriven | For instance when I get a, "user not available, leave a voicemail" whatever from voicemail, sometimes the audio doesn't come in until "available, leave a voicemail" |
| 16:18.20 | shriven | and it sounds slightly garbled at first. |
| 16:18.23 | shriven | then it works fine |
| 16:19.22 | *** join/#asterisk [gnubie] (n=[gnubie]@cm5.omega249.maxonline.com.sg) |
| 16:19.24 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-203-252.bflony.east.verizon.net) |
| 16:19.28 | shriven | I'm on the lan that the asterisk is on, so I think we should be able to safely rule out bandwidth issues.... |
| 16:20.00 | [gnubie] | tzafrir_laptop: building from testing => http://paste.debian.net/18946/ |
| 16:20.47 | luke-jr | assuming Asterisk left DTMF out of the recording, can i obtain the times and digits of it? |
| 16:20.55 | luke-jr | the call is still running |
| 16:22.28 | luke-jr | (if it helps, there is a possibility someone's life is in danger) |
| 16:24.03 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 16:25.31 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.93) |
| 16:26.46 | Rienzilla | shriven: I've had that too |
| 16:26.53 | Rienzilla | in my case the system was too busy |
| 16:27.05 | shriven | : ( no one else is using the asterisk. |
| 16:27.13 | Rienzilla | well not necessarily asterisk |
| 16:27.20 | shriven | ? |
| 16:27.24 | Rienzilla | the cause in my system was a rebuilding raid array that took most i/o |
| 16:27.36 | shriven | ahhh, yeah nothing else is on the computer... load is low |
| 16:27.39 | shriven | thanks though |
| 16:27.42 | Rienzilla | ok |
| 16:28.02 | dandre | I have seen that there is a new Atxfer in asterisk 1.6 manager interface. Is there any plan to have an hold command to? |
| 16:28.30 | shriven | I am wondering if part of it for the voicemail is because the extension is a named extension, and the voicemail system doesn't have anything recorded to say for that |
| 16:28.45 | shriven | but that doesn't really make a lot of sense I guess.... |
| 16:28.54 | shriven | it should just say nothing or w/e it has clearly. |
| 16:30.07 | bradleyprice86 | I am having a problem with asterisk making outbound calls. When I go to asterisk cli and type pri show span 2, it says that it is down. I have called our t1 provider and they test and said the t1 is up. What steps do I need to use to fix this issue? |
| 16:30.24 | *** join/#asterisk naif (n=naif@213-140-6-112.ip.fastwebnet.it) |
| 16:30.27 | naif | i need to test GSM CSD call in international environment. I developed an encrypted mobile client. I am in Italy with Vodafone and want to try calling some GSM number abroad in USA, Germany, Span, UK and another to see if the GSM CSD call reach the other party. In a lot of situation it does not work or arrive as Voice Call. Can someone provide me his GSM number, i will call him with GSM CSD and tell me if he receive a Data Call or not? Answerin |
| 16:34.35 | bradleyprice86 | jaytee: you mentioned I would temporarily need to change my group in zapata.conf. What exactly would I need to change? |
| 16:35.25 | jaytee | bradleyprice86, no I didn't. I askes a couple of questions to get clarification and then waited forever for an answer. |
| 16:35.37 | jaytee | s/askes/asked |
| 16:36.00 | bradleyprice86 | Ok. Sorry, I had to run to the server room because all of our servers crashed at once. |
| 16:36.18 | bradleyprice86 | jaytee: so would it be ok if I explain everything now? |
| 16:36.45 | jaytee | sorry, busy |
| 16:37.04 | bradleyprice86 | Ok. |
| 16:39.54 | gr0mit | am trying to get music on hold working on 1.4.x - i get pbx_extension_helper: No application 'MusicOnHold' for extension |
| 16:40.02 | gr0mit | anyone able to advise me pls? |
| 16:48.12 | *** join/#asterisk steliosk (n=Stelios@athedsl-324329.home.otenet.gr) |
| 17:01.41 | flohack | is russell (svn name) around? |
| 17:01.54 | flohack | russellb: is that you? |
| 17:02.12 | russellb | that is me, but i'm heading out for lunch ... |
| 17:02.24 | russellb | did i break the world? |
| 17:02.43 | flohack | just my telephone system :-) |
| 17:02.59 | Blackvel | hehe |
| 17:02.59 | russellb | i see. |
| 17:03.04 | flohack | I found a deadlock related to http://bugs.digium.com/view.php?id=12470 |
| 17:03.08 | CGMChris | Quick question: ${EXTEN} is the extensian I dialled. Whats the variable for the extenion that I dialed it FROM ? |
| 17:03.23 | Qwell | CGMChris: no such thing |
| 17:03.28 | Qwell | You don't dial from an extension |
| 17:03.38 | Qwell | You dial from a device. |
| 17:03.41 | flohack | russellb: have you got a few minutes to spare after lunch? I've got a debugger attached (with source) |
| 17:03.44 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 17:03.48 | russellb | flohack: k ... assuming you're on the latest version, compile with DEBUG_THREADS, and grab "core show locks" output |
| 17:03.51 | CGMChris | QWell: How do I default VoicemailMain() to the mailbox of the device I am calling from |
| 17:03.55 | russellb | and then attach the output to an issue on bugs.digium.com |
| 17:04.02 | Qwell | CGMChris: You could muck with callerid stuff |
| 17:04.18 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 17:04.30 | CGMChris | exten => *99,1,VoiceMailMain(${CALLERID(num)}@default) <- Should work? |
| 17:04.35 | Qwell | CGMChris: perhaps |
| 17:04.47 | flohack | russellb: I'm on 1.4.17 plus the patch from the bug tracker, but that file is unchanged from that revision on and the bug definitely in there |
| 17:05.11 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 17:05.21 | Qwell | voxter: done any testing yet? :p |
| 17:05.22 | flohack | russellb: It's quite easy: Two threads obviously stuck on the same poll() |
| 17:05.36 | voxter | Qwell: on asterisknow? |
| 17:05.37 | russellb | it is completely normal to be blocking on poll() |
| 17:05.40 | Qwell | mmhmm |
| 17:06.02 | russellb | most of the time that's what you'll see when you look at asterisk threads |
| 17:06.05 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 17:06.08 | russellb | most will be blocking on poll ... |
| 17:06.17 | voxter | Qwell: i just yesterday booted it in a virtual machine. I just spent the last week migrating one of my biggest customers to just a plain old freepbx install, so i will be testing the distro this weekend |
| 17:06.24 | Qwell | ahh |
| 17:07.19 | russellb | anyway, lunch ... |
| 17:07.31 | flohack | russellb: hmmm....I'll have a closer look and come back to you (how long for lunch)? |
| 17:08.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.29 | CGMChris | qwell: sip show peer [my_peer_name] yields "Mailbox : 1000" for my device. Why is this information not passed onto VoicemailMain by default ? |
| 17:09.47 | CGMChris | or rather, how do I pass that mailbox ID on? |
| 17:10.35 | CGMChris | anybody? |
| 17:12.01 | DarKnesS_WolF | CGMChris: u want each extension to listen to it is own mailbox only ? |
| 17:12.07 | DarKnesS_WolF | so u get to enter the password directly ? |
| 17:12.15 | CGMChris | Yes. |
| 17:12.31 | DarKnesS_WolF | CGMChris: the mailbox number is the sanme as the extension ? |
| 17:12.37 | CGMChris | Yes |
| 17:12.46 | ManxPower | CGMChris: Asterisk does not have the mailbox ID accessable in the dialplan |
| 17:13.18 | DarKnesS_WolF | exten => 555,1,VoiceMailMain(${CALLERID(NUM)}@${CONTEXT}) <--- i use this |
| 17:13.21 | CGMChris | ManxPower: The GUI did it by default, however, it was using caller ID per user and passing that to VoicemailMain as a parameter. |
| 17:13.48 | DarKnesS_WolF | so it will use the CALLERID(NUM)@CONTESXT as the mailbox@mailbox contexts |
| 17:13.49 | ManxPower | You will need to use your dialplan to figure out the mailbox ID then pass that to VoicemailMain |
| 17:13.51 | CGMChris | Darkness: That works, but only if the caller ID matches the mailbox number. |
| 17:14.01 | ManxPower | CGMChris: and GUI was doing it like I'm telling you need to do it. |
| 17:14.34 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 17:14.59 | CGMChris | ManxPower: So I need to program a solution... a method of retriving mailbox# based on device? |
| 17:15.00 | ManxPower | too bad you won't be letting users check their voicemail from anywhere other than their extension |
| 17:15.16 | ManxPower | CGMChris: personally I just set the mailbox be the same as the extension and be done with it. |
| 17:15.34 | CGMChris | ManxPower: Incorrect. Its trivial to setup 2 extensions for voicemail. Purpose: People want their voicemail button on their phone to go to THEIR voicemail. |
| 17:16.54 | ManxPower | CGMChris: that is not what you said. You said and I quote "(12:09:47 PM) CGMChris: or rather, how do I pass that mailbox ID on?" If you are automatically passing on the user's extension to Voicemailmain then you'll only be able to access it from your phone. You did not say anything about "two extensions". If you don't give us complete information we can't give you complete answers. |
| 17:17.06 | DarKnesS_WolF | CGMChris: yes in that case use same number for voicemail and extension |
| 17:18.17 | ManxPower | Many people setup their systems allow the user to call their own number and automatically sent to the voicemailmain with the correct mailbox number. Then they have a separate voicemailmain extension |
| 17:18.23 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
| 17:18.26 | CGMChris | Darkness: What if I want to set callerid for that user to a public phone number, but want their mailbox number to be their internal extension? then passing caller id does not work. |
| 17:18.59 | ManxPower | CGMChris: I never advocated passing the callerid as that is not secure. |
| 17:19.07 | DarKnesS_WolF | CGMChris: true |
| 17:19.15 | ManxPower | it also does not work well in the real world. |
| 17:19.37 | ManxPower | and if you are really that desperate then in sip.conf for each peer setvar=mailbox=1234 |
| 17:19.45 | ManxPower | then ${MAILBOX} is accessable in the dialplan. |
| 17:19.52 | CGMChris | PERFECT !@#!@# |
| 17:19.58 | Qwell | ${mailbox}.. |
| 17:20.12 | CGMChris | I am already doing "mailbox=1000" for each peer... so, i just need to make a minor adjustment. PERFECT ! |
| 17:20.17 | ManxPower | I'm not aware of anyone actually doing it that way, but it should work. Of course if you use different voicemail contexts it gets even more complicated. |
| 17:20.35 | CGMChris | The prefered method is... dialing your own extension, then? |
| 17:20.46 | ManxPower | CGMChris: welcome to your first user whine. You'll get many of those. |
| 17:20.55 | ManxPower | CGMChris: there is no preferred method. |
| 17:21.27 | CGMChris | I still havent gotten any user whine... I am thinking of everything I possibly can before deploying my first system. I am still in beta testing over here. |
| 17:21.29 | ManxPower | I find that if people have buttons preprogrammed on their phone they forget what they need to dial when they are not at their phone. |
| 17:21.45 | ManxPower | CGMChris: then just make them enter their mailbox when they call into voicemail main |
| 17:22.03 | CGMChris | ManxPower: I personally want my button to work. :) |
| 17:22.11 | ManxPower | CGMChris: so your machine is not even in production yet? |
| 17:22.31 | CGMChris | Waiting on my boss to decide if we are using net2phone or another SIP trunk provider. |
| 17:22.39 | ManxPower | Remember Asterisk is not really a PBX. Asterisk is a toolkit that allows you to BUILD a PBX |
| 17:22.48 | CGMChris | I have a demo net2phone account that I'm using for testing purpose. |
| 17:22.57 | CGMChris | ManxPower: Understood, that's why I am using it. |
| 17:28.08 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 17:28.36 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:30.31 | CGMChris | ManxPower: Do you know how redundancy would work if one of my internet lines went down? I have 2 lines going thru a multiplexer that merges and load balances. *Should* my SIP provider pickup my new IP via REGISTER and send incoming calls to it... or might they use an IP based solution? |
| 17:31.23 | ManxPower | CGMChris: I would not even guess. |
| 17:31.38 | ManxPower | I use PSTN for Asterisk, not VoIPoInternet |
| 17:31.43 | CGMChris | ah, ok. |
| 17:31.46 | ManxPower | Specifically PRIs |
| 17:32.14 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 17:32.19 | petererer | uses a PRI, along with some SIP providers for incoming calls on 'special' numbers |
| 17:32.26 | naif | i need to test GSM CSD call in international environment. I developed an encrypted mobile client. I am in Italy with Vodafone and want to try calling some GSM number abroad in USA, Germany, Span, UK and another to see if the GSM CSD call reach the other party. In a lot of situation it does not work or arrive as Voice Call. Can someone provide me his GSM number, i will call him with GSM CSD and tell me if he receive a Data Call or not? Answerin |
| 17:32.50 | ManxPower | petererer: the only time we send calls over the internet is when all 4 PRIs are down at all 4 offices. Doesn't happen very often. |
| 17:33.35 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 17:33.55 | *** join/#asterisk Corydon76-dig (i=two@pdpc/supporter/bronze/Corydon76-home) |
| 17:33.55 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 17:34.50 | *** join/#asterisk vonkleist (n=vonkleis@189.155.116.80) |
| 17:36.42 | petererer | heh |
| 17:36.55 | petererer | i tried using enum service with e164.org |
| 17:37.07 | petererer | it added 5 seconds of dns timeouts before always using the pri anyway... |
| 17:39.41 | DarKnesS_WolF | trying to compile dahdi-tools-2.x and getting this error in make config |
| 17:39.42 | DarKnesS_WolF | make: dahdi: Command not found |
| 17:39.52 | DarKnesS_WolF | make: *** [config] Error 127 |
| 17:40.27 | tzafrir_laptop | DarKnesS_WolF, dahdi-tools tarball? |
| 17:41.26 | DarKnesS_WolF | yes |
| 17:41.47 | tzafrir_laptop | what is the output of: make -n config |
| 17:42.36 | ManxPower | DarKnesS_WolF: what distro / what kernel version? |
| 17:42.46 | DarKnesS_WolF | http://pastebin.ca/1224709 |
| 17:42.59 | DarKnesS_WolF | archlinux |
| 17:43.00 | DarKnesS_WolF | Linux Desert 2.6.26-ARCH |
| 17:43.07 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 17:43.24 | DarKnesS_WolF | is there should be a command called dahdi ? |
| 17:45.12 | ManxPower | DarKnesS_WolF: did you compile dahdi-linux before dahdi-tools? |
| 17:45.59 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 17:46.14 | Bladerunner05 | anybody using italian squillo voip account? |
| 17:46.22 | Bladerunner05 | or eutelia ? |
| 17:47.33 | DarKnesS_WolF | ManxPower: sure i did |
| 17:49.09 | ManxPower | DarKnesS_WolF: DAHDI would not even compile on my system -- but I'm using a slightly odd distro that is quite old so it's not really DAHDI's fault. |
| 17:49.38 | DarKnesS_WolF | :) |
| 17:50.09 | DarKnesS_WolF | i don't have any zap hardware but i am writing a howto so i am trying to emulate everything while i am writing but we do still need dahdi for timing source right ? |
| 17:51.35 | DarKnesS_WolF | ManxPower: what distro ur usuing ? |
| 17:51.44 | ManxPower | DarKnesS_WolF: yes. file says 1.6.1 will no longer require zaptel for MeetMe, he did not say anything about IAX2 trunking. |
| 17:51.48 | ManxPower | DarKnesS_WolF: Mandriva 2007 |
| 17:52.21 | DarKnesS_WolF | ManxPower: i love mandriva ;-) |
| 17:52.22 | *** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs) |
| 17:52.34 | ManxPower | Mandriva backported something so DAHDI sees the kernel version, tries to do something the old way, but that fails because the new way was backported. |
| 17:52.37 | DarKnesS_WolF | my 1st distro since mandriva 7.1 |
| 17:52.48 | ManxPower | DarKnesS_WolF: back in 1999 Mandriva was one of the best ones out there. |
| 17:52.50 | DarKnesS_WolF | ah the .config file in kernel source ? |
| 17:53.03 | DarKnesS_WolF | it is still quite good better than ubuntu believe me |
| 17:53.19 | DarKnesS_WolF | i still have some * prduction installation on mdk 2006 |
| 17:53.21 | DarKnesS_WolF | and 2007 |
| 17:53.24 | ManxPower | DarKnesS_WolF: no, like /usr/src/linux/linux.h or something like that |
| 17:53.52 | ManxPower | DarKnesS_WolF: Since I won't have any legacy customers soon I'll be looking at CENTOS |
| 17:54.19 | DarKnesS_WolF | codec_dahdi = sliner? |
| 17:54.39 | ManxPower | I need to call up a friend and have them download the ISO, burn it, and mail it to me. |
| 17:55.59 | DarKnesS_WolF | :-o |
| 17:56.24 | ManxPower | <-- metered, capped internet |
| 17:56.37 | ManxPower | well capped at least |
| 17:58.46 | file | ManxPower: 1.6.1 will not require dahdi timing for IAX2 trunking or internal timing but it will still be required for meetme, 1.6.2 will have an alternate conference application that will not use dahdi |
| 17:59.52 | ManxPower | file: Sorry, I went back and looked and I mis-read your original statemnet |
| 18:01.06 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 18:01.18 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 18:01.26 | DarKnesS_WolF | file perfect ! |
| 18:02.52 | DarKnesS_WolF | file: anyidea about this compiling error for dahdi-tools ? |
| 18:06.09 | seanbright | DarKnesS_WolF: your pastebin expired, could you re-paste? |
| 18:06.53 | gr0mit | hmmm- having prbs getting music on hold runnning on 1.4 |
| 18:07.04 | flohack | naif: have you found someone? I'm from Austria if that would help you. |
| 18:07.25 | gr0mit | app musiconhold does not appear to exist |
| 18:07.30 | naif | flohack: yes please, i would be glad to test connection between Italy and Austria on GSM CSD calls |
| 18:09.05 | gr0mit | how many peeps in his office? |
| 18:09.11 | gr0mit | oops wrong window! |
| 18:14.27 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 18:19.13 | DarKnesS_WolF | seanbright: ok |
| 18:19.53 | DarKnesS_WolF | http://pastebin.ca/1224753 |
| 18:21.39 | tzafrir_laptop | DarKnesS_WolF, nice bug :-) |
| 18:22.51 | tzafrir_laptop | that's a typo in Makefile |
| 18:23.02 | DarKnesS_WolF | tzafrir_laptop: why me :( wy i always find bugs :P |
| 18:23.08 | DarKnesS_WolF | i am a bug hunter :P |
| 18:23.20 | DarKnesS_WolF | or i think i just have a bad luck ;-) |
| 18:23.46 | lmadsen | DarKnesS_WolF: you must just be doing things that are non-standard. I find bugs fairly often (not as often anymore) because I do things that not a lot of people do |
| 18:24.29 | DarKnesS_WolF | lmadsen: how dare u ! asterisk is a bug free app :P |
| 18:24.36 | DarKnesS_WolF | tzafrir_laptop: ok fix ? patch ? |
| 18:24.39 | lmadsen | exactly :) |
| 18:24.52 | DarKnesS_WolF | lmadsen: i did really follow the standard :-s i didn't change anything almost |
| 18:24.57 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 18:25.16 | pigpen | So anyone use the Polycom Spectralink 8020? If so, what do you think? |
| 18:25.34 | DarKnesS_WolF | not me, i am happy with snom quality |
| 18:25.49 | pigpen | heh |
| 18:26.13 | pigpen | I have a snom 360 I need to stick on ebay now that I think about it. |
| 18:26.36 | DarKnesS_WolF | pigpen: why 370 for me works perfect |
| 18:26.39 | DarKnesS_WolF | nice screen ;-) |
| 18:26.57 | DarKnesS_WolF | the mini xml browser so nice with a smile python scipt i did i can read BBC news on the phone ;-) |
| 18:27.18 | pigpen | Just got burned on the early 360's...took it in the shorts for about $5000..... |
| 18:29.22 | DarKnesS_WolF | pigpen: huh ? |
| 18:29.43 | DarKnesS_WolF | tzafrir_laptop: where is that typo ? |
| 18:29.43 | tzafrir_laptop | DarKnesS_WolF, in latest dahdi-tools svn (specifically: r5082) |
| 18:29.53 | DarKnesS_WolF | :) |
| 18:30.12 | DarKnesS_WolF | mmmm ok let me fetch and diff |
| 18:30.16 | tzafrir_laptop | there is a 'ifndef' instead of 'ifneq' |
| 18:31.13 | tzafrir_laptop | duh. It's the same in Zaptel |
| 18:31.30 | DarKnesS_WolF | perfect ;) |
| 18:31.36 | DarKnesS_WolF | works for me after i fixed it |
| 18:31.58 | DarKnesS_WolF | i find the typo u mentioned fixed and works it is the only ifndef in the whole makefile |
| 18:33.02 | DarKnesS_WolF | tzafrir_laptop: good catch :-) |
| 18:33.08 | sdaniels | Will asterisk do SS7? |
| 18:35.14 | *** join/#asterisk Corydon76-dig (i=beige@pdpc/supporter/bronze/Corydon76-home) |
| 18:35.20 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 18:42.55 | flohack | sdaniels: Google 'asterisk SS7', first hit: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 |
| 18:43.05 | tzafrir_laptop | sdaniels, it should . That is: chan_dahdi in 1.6.0 with libss7 |
| 18:44.31 | sdaniels | I have been asked if i can replace an Mpathix voicemail platform with Astrisk, we have about 10k mailboxes, and we have to be able to provice MWI and stuttertone.... |
| 18:45.15 | flohack | russellb: I found that there was some inconsistency between my source code and asterisk, I'm recompiling with thread debug and will try to reproduce |
| 18:45.52 | sdaniels | think it doable? |
| 18:46.36 | russellb | flohack: k! |
| 18:47.20 | flohack | russellb: I just started asterisk with sipp and it crashed it....Program received signal SIGSEGV, Segmentation fault. |
| 18:47.20 | flohack | [Switching to Thread 0xb500ab90 (LWP 11713)] |
| 18:47.20 | flohack | 0x0809e636 in ast_waitfor_nandfds (c=0xb500a158, n=1, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb500a154) at channel.c:2010 |
| 18:47.21 | flohack | 2010 CHECK_BLOCKING(c[x]); |
| 18:47.29 | flohack | ouch |
| 18:48.23 | russellb | you should _not_ enable DO_CRASH |
| 18:48.24 | russellb | heh |
| 18:48.29 | russellb | DO_CRASH makes it ... crash! |
| 18:48.59 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
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| 18:49.05 | rene- | question |
| 18:49.26 | rene- | if i generate a call using originate to say channel Zap/gx/somenumber |
| 18:49.27 | flohack | russellb: I though it would be a good idea to see if something else went wrong before the deadlock |
| 18:50.06 | rene- | and i assign variables to it, is there a way i can i run getvar on /Zap/gx/somenum and get something thos vars back? |
| 18:50.19 | rene- | s/something// |
| 18:50.42 | flohack | russellb: shall I enable DETECT_DEADLOCKS? |
| 18:50.57 | rene- | btw hey all |
| 18:51.22 | russellb | flohack: no :) |
| 18:51.26 | russellb | flohack: just DEBUG_THREADS |
| 18:51.29 | russellb | and DONT_OPTIMIZE |
| 18:51.37 | rene- | sdaniels: what type of technology your phones are |
| 18:51.42 | gr0mit | anyone know what dependencies i need to get music on hold compiled? |
| 18:52.05 | sdaniels | rene-: these are for residential customers on the PSTN |
| 18:52.26 | rene- | hmmm |
| 18:53.31 | rene- | how are u planning to interfase to asterisk? |
| 18:54.10 | sdaniels | rene-: I work for a CLEC we resell ILEC (ATT, etc..) phone lines, we have our own voicemail platform (Mpathix), We pay 20k a month for support of the gear, if I can replace it with Asterisk that would rule... |
| 18:55.54 | sdaniels | rene-: Im just not sure that its possible |
| 18:56.32 | rene- | well |
| 18:56.46 | rene- | asterisk supports voicemail for TDM analog and sip devices |
| 18:57.30 | rene- | basically your phones, will dial the voicemail number |
| 18:57.50 | rene- | and you would route that number to asterisk along with callerid information so u can lookup your voicemail |
| 18:58.23 | rene- | the way asterisk stores voicemail is easily understood so u could migrate it |
| 18:58.36 | AlexTO | Hi, i have a question about autodial, after you put a file .call and make asterisk call automatic you sand the call to and specific context, the call is been place fine but the CDR do no get the called number, just the extension "s" in this case, some one can put me on the right place to find you how can i set on my cdr the destination as the number called? |
| 18:58.42 | rene- | provided you can get out of the Mpathix |
| 18:58.49 | rene- | the info and audio files |
| 18:59.11 | sdaniels | My mail concern is with the SS7 signaling. I would have to be able to send MWI info back to the customers. |
| 18:59.20 | rene- | yeah |
| 18:59.58 | rene- | well SS7 is a bit beyond my experience, but from my understanding, asterisk can be made to run in an SS7 setup, look for the sangoma asterisk gear for a solution |
| 19:00.15 | sdaniels | sangoma? Ill check it out. |
| 19:00.38 | *** join/#asterisk Greek-Boy (n=email@41.222.89.77) |
| 19:01.09 | sdaniels | ahh, this looks like FXO gear |
| 19:02.34 | rene- | no |
| 19:02.37 | rene- | they have ss7 geat |
| 19:02.38 | rene- | gear |
| 19:02.53 | rene- | made from asterisk servers and sangoma e1 boards |
| 19:03.01 | rene- | part of the solution is commercial |
| 19:03.07 | rene- | but a big deal is open source |
| 19:04.01 | rene- | and btw you could |
| 19:04.13 | sdaniels | yeah it does, your right. its probably cheaper than what were using now. |
| 19:04.13 | russellb | Asterisk 1.6 has native ss7 support |
| 19:04.19 | russellb | developed by a guy at Digium. |
| 19:04.30 | russellb | in fact, it's already being used a lot in some very high load environments |
| 19:04.37 | russellb | no need for any commercial anything. |
| 19:04.39 | sdaniels | is that the most current Rev?? |
| 19:05.49 | sdaniels | sorry im on day 3 of Asterisk |
| 19:06.46 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 19:08.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:09.07 | russellb | sdaniels: asterisk 1.6.0 was just released, it's still quite new. |
| 19:09.22 | russellb | sdaniels: asterisk 1.4 has less features, but is more mature. |
| 19:09.29 | russellb | but if you want ss7 support, asterisk 1.6 it is. |
| 19:09.53 | sdaniels | cool, thanks for the info yall... |
| 19:10.13 | *** join/#asterisk kj5t (n=steve@65-120-138-35.dia.static.qwest.net) |
| 19:10.48 | kj5t | Is there someone here that can tell me how I can add to my dialplan to ability that when someone dials my extension it goes to my cell phon? |
| 19:10.51 | kj5t | phone* |
| 19:11.41 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 19:17.28 | jaytee | kj5t, do you mean all the time if you don't answer your desk phone? |
| 19:18.32 | jaytee | kj5t: or just and extension that always dials your cell |
| 19:28.07 | *** part/#asterisk jplank (n=GBove@reports.nyigc.net) |
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| 19:46.29 | cesar_CR | helo guys, I want to use the python lib for AGI pyst, anybody knows if its development is going to continue ?? |
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| 19:50.28 | *** join/#asterisk cj (n=cjac@pdpc/supporter/monthlybronze/cj) |
| 19:50.35 | cj | yay |
| 19:51.04 | cj | okay... so how do I get the telco to send calls to number XXX to my asterisk server? :) |
| 19:51.18 | jaytee | ask them nicely? |
| 19:51.22 | cj | do I need a T1 or BRI or something? |
| 19:51.38 | cj | analog number + analog pci card ? |
| 19:51.54 | jaytee | one of those will do :-) |
| 19:51.55 | cj | I'd prefer to avoid all that expensive stuff and have it all come in over ye olde SIP |
| 19:52.11 | jaytee | SIP is not really "ye olde" |
| 19:52.13 | cj | someone wants to charge me $75/mo for a BRI |
| 19:52.23 | jaytee | you're not in the US |
| 19:52.33 | cj | I am |
| 19:52.46 | jaytee | I didn't think anyone did BRI in the states anymore |
| 19:52.57 | cj | yeah, that's why they want to charge $75, I guess |
| 19:53.00 | *** join/#asterisk RB2 (n=RB2@pool-71-255-92-53.nwrknj.east.verizon.net) |
| 19:53.57 | jaytee | well, you could go with BRI and get a BRI card for the * server or go with POTS analog and get a Digium TDM410 card with one or more FXO modules. |
| 19:54.19 | cj | I've got a TDM410 and a FXS/FXO module |
| 19:54.28 | jaytee | how many concurrent incoming and outgoing calls do you expect to make? |
| 19:54.52 | *** part/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 19:54.58 | cj | only incoming: 1 |
| 19:55.06 | _ShrikE | cj: just get yourself an decent itsp |
| 19:55.28 | riddlebox | is there any asterisk related podcasts? |
| 19:55.28 | jaytee | cj, you mean you have a TDM410 with an FXS AND an FXO, meaning 2 modules on the card, one green, one red. Correct? |
| 19:56.00 | cj | I want to start taking calls for my cell phone # and redirect them to voicemail and forward *that* to my smtp server |
| 19:56.12 | cj | jaytee: yes. one TDM410. One FXS. One FXO |
| 19:56.20 | cj | _ShrikE: itsp? |
| 19:56.50 | cj | Internet Telephony Service Provider? |
| 19:56.52 | jaytee | so you can use that with a pots line which will run you on average about 30 to 50 bucks a month depending on whether you include long distance. |
| 19:57.08 | jaytee | ITSP lets you go SIP over your INTERNET connection |
| 19:57.09 | cj | jaytee: I don't want outgoing calls, just incoming. |
| 19:57.36 | jaytee | cj, then just get an ITSP that features free incoming calls and be done with it :-) |
| 19:57.36 | cj | jaytee: oh? does it handle the routing of incoming calls to my server? |
| 19:58.01 | cj | I've already got a number. Using it is a requirement of my use case. |
| 19:58.09 | jaytee | cj, you need to setup you asterisk server to take the calls coming into your local network. |
| 19:58.35 | cj | jaytee: that part is easy. I need to get those calls to arrive at the server; that's the part that confuses me :) |
| 19:58.49 | jaytee | and that may involve configuring your router for sip/rtp forwarding. |
| 19:58.55 | jaytee | ~sipnat |
| 19:58.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:59.08 | cj | jaytee: nah, I've got a /27 |
| 20:00.26 | cj | and a /48 :) |
| 20:00.44 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 20:00.48 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:00.51 | jaytee | whoop dee fricken doo! what's that got to do with port forwarding IP traffic? |
| 20:01.10 | nny_1 | anyone using the linksys 962 or 942 found a way to have the phones download a directory from ftp etc? |
| 20:01.18 | nny_1 | looking for docs on such a thing |
| 20:01.18 | cj | jaytee: why would I port-forward IP traffic if I could just handle the traffic on the edge? |
| 20:02.24 | jaytee | cj, are you behind a router? |
| 20:02.40 | cj | jaytee: define "behind"? |
| 20:02.43 | cj | and "router" |
| 20:03.09 | cj | it seems that you are referring to a NAT "firewall" as a "router" |
| 20:03.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:03.22 | jaytee | yeah, firewall |
| 20:03.27 | nny_1 | also voicemail.conf: the user name (not the email address) sending the email appears to be the system account under which asterisk is running. (correct me if i am wrong.) any way to later this behavior? |
| 20:03.32 | cj | as I said above, I have a /27. all of those addresses are internet routable |
| 20:03.40 | nny_1 | alter this behavior* |
| 20:04.22 | jaytee | then just have your ISP point to the address of your Asterisk box. without a NAT'd firewall it should be a no-brainer |
| 20:04.28 | jaytee | ~book |
| 20:04.28 | jbot | i heard book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:06.41 | cj | jaytee: okay. I probably have no brain, but how do I tell the telco to route calls to 206-123-4567 to 66.152.65.9:sip ? |
| 20:07.05 | nny_1 | does the telco support sip?! |
| 20:07.12 | cj | nny_1: not afaik. |
| 20:07.13 | nny_1 | needs to hook up with them heh |
| 20:07.22 | nny_1 | than you can't |
| 20:07.25 | nny_1 | however |
| 20:07.34 | jaytee | cj, you'd have to have the number transferred over to the ITSP from your current telco |
| 20:07.43 | nny_1 | you can forward that number to a itsp supplied one |
| 20:07.48 | nny_1 | or that |
| 20:07.56 | cj | awesome. |
| 20:08.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:08.15 | nny_1 | cj iout of curiosity what are you trying to do? |
| 20:08.17 | cj | can you give me an example of an ITSP? |
| 20:08.31 | jaytee | the Number Portability Act was passed a couple years back to allow consumers to keep their phone numbers when moving around and also to give telcos ulcers. |
| 20:08.34 | cj | nny_1: re route calls currently going to my cell phone to my asterisk server |
| 20:08.39 | jaytee | ~itsp |
| 20:08.40 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:08.52 | jaytee | ~itpslist-us |
| 20:09.03 | cj | ~itsplist-us |
| 20:09.04 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:09.08 | jaytee | hehe |
| 20:09.28 | cj | jaytee: okay, assuming that I don't want to pay an ITSP to do the work for me, how do I do it? |
| 20:09.59 | cj | how do I "become" an ITSP? |
| 20:10.05 | jaytee | oh, sure! jbot's smart enough to tell me that someone else just said the same thing but not smart enough to correct my misspelled queue. Damn stupid bots!!! |
| 20:11.20 | cj | I assume I need to get a BRI (at the minimum, probably better a T1) to receive calls directly, eh? |
| 20:11.23 | jaytee | cj, dunno. |
| 20:11.45 | nny_1 | cj on phone one sec |
| 20:11.53 | cj | nny_1: thanks |
| 20:12.34 | jaytee | cj, no. not if you have an internet connection you can recieve calls via SIP but if you need to use that specific number you mentioned earlier you'd need to either forward it to your ITSP or "move" it to your ITSP. |
| 20:13.01 | cj | jaytee: if I want to *be* an ITSP? |
| 20:13.10 | cj | if I want to receive the calls directly? |
| 20:13.19 | jaytee | if you want to be an ITSP you've got to do alot more reading |
| 20:13.31 | cj | jaytee: okay. subjects, please? |
| 20:13.37 | jaytee | ~book |
| 20:13.38 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:13.45 | jaytee | ~telephony101 |
| 20:13.53 | jaytee | ~101 |
| 20:13.54 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 20:14.10 | nny_1 | cj: so you want to forward your cell # to an itsp? |
| 20:14.33 | jaytee | and there's this really cool thing called Google! you may not have met it yet but rest assured, it's your friend!!!! |
| 20:14.36 | nny_1 | first of all if it can't be ported, you will usually still end up paying minutes for the calls |
| 20:14.38 | cj | nny_1: I want to forward the cell # to my server in the telco. you could consider me to be an itsp and your statement would be true. |
| 20:14.57 | nny_1 | your server in the telco? |
| 20:15.19 | nny_1 | is there a number that goes to that server? |
| 20:15.27 | cj | nny_1: nope. how do I get one to do so? :) |
| 20:15.32 | nny_1 | (btw do read those docs) |
| 20:15.42 | cj | okay, I'll go read those before bothering you. thanks. |
| 20:15.54 | nny_1 | where is your server? just a box at your house? does it have ap ublic IP? |
| 20:16.01 | luke-jr | sigh, iConnectHere/DeltaThree are such idiots |
| 20:16.05 | nny_1 | well I can help you figure out what you need to research |
| 20:16.19 | nny_1 | but ultimately the knowledge won't be gleamed from this conversation |
| 20:16.19 | cj | nny_1: http://www.westinbuilding.com/telecom/ |
| 20:16.38 | cj | nny_1: I've got a /27; I can probably get the whole /24 if I ask nicely. |
| 20:16.42 | nny_1 | not needed |
| 20:17.44 | nny_1 | ok you have an asterisk server with a public IP |
| 20:18.07 | cj | well, I have a debian xen machine. spinning up a vm to run asterisk would be simple |
| 20:18.21 | nny_1 | ok well, you'll have a lot more complicted tasks at hand |
| 20:18.26 | cj | right. :) |
| 20:18.27 | nny_1 | like configuring asterisk etc |
| 20:18.49 | nny_1 | but basically somewhere a number needs to terminate from the nationwide telephone network into your box |
| 20:18.51 | *** join/#asterisk rethus (n=rethus@xdsl-81-173-176-47.netcologne.de) |
| 20:18.57 | rethus | hi there, |
| 20:19.01 | cj | I've configured it to take incoming calls before. I was given an analog card by one of my co-workers in '05 and set it up to dump voicemail to disk. it would be simple to send it to my smtp server instead. |
| 20:19.08 | rethus | what is the best way to record a channel? |
| 20:19.16 | nny_1 | rethus: look up chanspy |
| 20:19.44 | rethus | is it a cli command, or an alternate api? |
| 20:19.52 | nny_1 | it's an asterisk application |
| 20:19.58 | nny_1 | so i guess "cli command" is close enough |
| 20:20.09 | cj | nny_1: I could terminate it by having one of the many telcos in the building sell me an analog line and run it to the rack, put the analog digium card in the box, and plug the analog line directly into it |
| 20:20.11 | nny_1 | rethus: it's aprt of your dial statement |
| 20:20.16 | nny_1 | er dial plan* |
| 20:20.18 | rethus | I have found this... but i didn't know whats best for my needshttp://www.voip-info.org/wiki-Asterisk+record+calls |
| 20:20.45 | nny_1 | rethus: http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy |
| 20:20.59 | rethus | a command would be great which i can enter in cli |
| 20:21.10 | rethus | nny_1 thanks i watch on this side now |
| 20:21.21 | nny_1 | cj: depends on the amount of calls you'll get |
| 20:21.27 | nny_1 | cj mind if i ask what the goal is? |
| 20:21.46 | nny_1 | cj no cell service in building? |
| 20:21.54 | cj | nny_1: only a single number. I want to receive the calls currently coming to my cell phone at my asterisk box |
| 20:22.02 | cj | nny_1: the cell service is fine |
| 20:22.17 | nny_1 | cj: hmm then an analog line would be simple enough |
| 20:22.22 | cj | I don't want the number serviced by the cell company, though |
| 20:22.24 | nny_1 | cj: your other option is to use an itsp |
| 20:22.31 | nny_1 | cj: than it will have to be ported |
| 20:22.41 | nny_1 | cj: what is the area code and first 3 digits? |
| 20:22.49 | cj | +1 206 226 |
| 20:22.50 | rethus | is it possible to record withot loosing performance ? i only have an 800MHZ pc |
| 20:22.58 | nny_1 | cj: standby |
| 20:23.01 | cj | nny_1: danke |
| 20:23.02 | nny_1 | rethus: hmm 1 call? |
| 20:23.07 | nny_1 | rethus: or many? |
| 20:23.18 | rethus | i have onne conference with nearly 10 clients |
| 20:23.31 | rethus | (means 10 persons logged into this conference) |
| 20:23.35 | *** join/#asterisk pecanha (n=tyap@189.36.181.134) |
| 20:23.47 | nny_1 | rethus: worth a shot.. although between codec changes and recordings it may be too much for an 800 mhz. there is no right answer to that question |
| 20:23.54 | nny_1 | i say test and see |
| 20:23.54 | rethus | i only need to rec the voice i get from my ekiga-client |
| 20:24.03 | nny_1 | rethus: than it may work |
| 20:24.16 | rethus | what have i to do? |
| 20:24.17 | cj | rethus: if you are okay with recording the aggregate of the calls rather than each call individually, it should be okay |
| 20:24.21 | pecanha | I'm wondering... can I have a dedicated server to host asterisk/freepbx and this server connects to my SIP number. Then when someone calls my number, can I provide automate answer and extensions? I mean... can I have two or more connections to different extensions? |
| 20:24.30 | *** join/#asterisk ManxPower (n=manxpowe@204.sub-75-251-143.myvzw.com) |
| 20:24.37 | rethus | my ekiga comes into conference with 100 |
| 20:24.46 | rethus | so how can i record this line? |
| 20:25.03 | nny_1 | cj: it appears vitelity (one of many providers) has coverage in that area |
| 20:25.05 | nny_1 | cj: http://www.vitelity.com/index.php?p=network&opt=npa&action=searchx&nnpa=206&nxx=226 |
| 20:25.10 | cj | rethus: doesn't ekiga have a "record conversation" feature? |
| 20:25.21 | nny_1 | cj: your next step would be to call them and try to get the number ported to their service |
| 20:25.31 | nny_1 | cj: alternatively, you could just have the local telco do ti as well |
| 20:25.49 | rethus | i have problems with soundcard sharing... cause * use the soundcard an doesn't let me record form other programms |
| 20:25.51 | nny_1 | cj: one costs X per month (telco), the other is X per minute (itsp) |
| 20:26.19 | nny_1 | rethus: recording should not use your soundcard |
| 20:26.20 | rethus | whats with tis: |
| 20:26.21 | rethus | http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 20:26.30 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 20:26.36 | cj | nny_1: I'm looking for the minimum cost. If it would be cheaper for me to terminate the call myself than it would be to have an ITSP do it for me, I'm happy to put in the leg work. |
| 20:26.58 | rethus | but it does. I have Linux... and if i try to record /dev/dsp i get the answer (device busy or someting else) |
| 20:27.09 | cj | I'll check with telco to see what they charge for an incoming-only line |
| 20:27.19 | nny_1 | rethus: just have asterisk dump the conversation internally using chanspy, the soundcard has nothing to do with it |
| 20:27.23 | nny_1 | cj: yeah |
| 20:27.26 | nny_1 | cj: probably better |
| 20:27.36 | ManxPower | rethus: noload => chan_alsa.so and noload => chan_oss.so in /etc/asterisk/modules.conf |
| 20:27.40 | nny_1 | cj: and easier to setup |
| 20:27.56 | nny_1 | rethus: ManxPower ahh sorry misunderstood |
| 20:28.00 | rethus | yeah, this is my intension... therfor i want to use * for recording. But whats about the quality of the recording voice? |
| 20:28.06 | nny_1 | rethus: it |
| 20:28.19 | nny_1 | 's the same as you would hear from the normal code/phone |
| 20:28.20 | nny_1 | codec |
| 20:28.31 | nny_1 | granted the machine can handle it |
| 20:28.42 | rethus | ok |
| 20:28.53 | nny_1 | one sec let me check which codec it saves it in |
| 20:28.54 | rethus | good enough for first test |
| 20:29.33 | rethus | but chanspy only can start in extensions.conf... or can i use it on CLI? |
| 20:29.45 | luke-jr | does Vitelity limit # of channels? |
| 20:29.47 | nny_1 | rethus: easier in extensions.conf |
| 20:29.49 | nny_1 | luke-jr: no |
| 20:30.15 | nny_1 | luke-jr: i have had some dtmf issues here and there, but the verdict is not out as to wether it's asterisk or Vitelity's fault |
| 20:30.24 | nny_1 | rethus: I can give you an example |
| 20:30.39 | rethus | jes please |
| 20:30.54 | luke-jr | hmm, Vitelity considers Kansas City to be ruralâ |
| 20:31.08 | nny_1 | rethus: WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 20:31.26 | nny_1 | rethus: it's not memorex, but it will sound just as good as a voicemail or musiconhold |
| 20:31.38 | nny_1 | luke-jr: hilton head island SC is rural to them too |
| 20:31.45 | rethus | this is good enough |
| 20:31.49 | ManxPower | You would use Monitor to record or one touch recording |
| 20:31.50 | nny_1 | luke-jr: i have yet to find an itsp that is supported in my area |
| 20:31.56 | nny_1 | ManxPower: Monitor is what i meant |
| 20:32.06 | nny_1 | ManxPower: chanspy is not heh |
| 20:32.40 | nny_1 | yeah actually that is in there by default right? |
| 20:32.55 | nny_1 | so just need to enable it in features.conf and at the end of your dial statements |
| 20:33.14 | luke-jr | any idea if they allow you to get a DID without E911? |
| 20:33.19 | ManxPower | Most ITSPs use the same set of carriers and so have the same DID coverage. |
| 20:33.25 | nny_1 | luke-jr: yeah they do |
| 20:33.49 | nny_1 | rethus: basically, anywhere you have a Dial in extensions.conf, you can add switches at the end |
| 20:33.57 | nny_1 | exten => _xxxxxxxxxx,4,Dial(Zap/g1/1${EXTEN},,wW) |
| 20:34.02 | nny_1 | the wW is Monitor |
| 20:34.31 | luke-jr | ponders $5/mo with MyPhoneCompany or $1.50/mo + 1.6c/min with Vitelity |
| 20:34.40 | nny_1 | well its not Monitor it's Automon |
| 20:34.51 | nny_1 | see http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 20:35.03 | luke-jr | any easy way to see if I usually use 218 minutes per month? :þ |
| 20:35.44 | rethus | this is my external.conf: |
| 20:35.45 | rethus | http://pastebin.com/d4c5af534 |
| 20:36.37 | rethus | so i can call |
| 20:36.38 | rethus | exten => callin,n,DIAL(ZAP/g1/1${EXTEN},,wW) at line 4? |
| 20:37.31 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-12-177.phlapa.east.verizon.net) |
| 20:40.35 | nny_1 | rethus: no |
| 20:42.22 | nny_1 | rethus (correct me if i am overcomplicating this channel) you'll have to have the dial command call the extension for meetme |
| 20:43.25 | nny_1 | rethus hmm let me think about that s ec |
| 20:43.26 | rethus | i don't understand |
| 20:43.37 | nny_1 | rethus: I dont think Meetme supports the wW switch |
| 20:44.27 | rethus | shortly i will describe how my system work. |
| 20:44.48 | nny_1 | rethus: 'r' â Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. â requires chan_zap.so |
| 20:45.11 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 20:45.12 | nny_1 | so just add r to your Meetme app line like exten => 400,n,MeetMe(1,Mmpqwr) |
| 20:45.22 | rethus | olk, seems to be the easyest way |
| 20:45.23 | nny_1 | and it should auto record all conference calls |
| 20:45.30 | rethus | only add an r on meetmee !? |
| 20:45.36 | nny_1 | rethus: test it out |
| 20:45.42 | nny_1 | rethus: I haven't done this here yet |
| 20:45.43 | *** join/#asterisk drigo- (n=drigo@145.97.105.1) |
| 20:45.55 | drigo- | hello! |
| 20:46.02 | rethus | where would the soundfiles be reorded? |
| 20:46.26 | drigo- | i am experimenting with the chan_mobile plugin :D |
| 20:46.26 | nny_1 | rethus: i think /car/spool/asterisk/monitor |
| 20:46.29 | nny_1 | var* |
| 20:46.38 | nny_1 | rethus: that is where automon puts them anyways |
| 20:46.59 | nny_1 | rethus: test it and pastebin the console output |
| 20:47.02 | rethus | any suggestion what the filename woould be? |
| 20:47.15 | drigo- | i finally got it to work, but i have not figured out how to put the sms function to good use |
| 20:47.21 | nny_1 | rethus: meetme-conf-rec-${CONFNO}-${UNIQUEID} |
| 20:47.27 | rethus | i first have to reinstall asterisk here on my homepc. |
| 20:47.35 | drigo- | for example i would like to forward a received sms, to an email adress.. |
| 20:48.00 | nny_1 | rethus: heh k |
| 20:48.03 | drigo- | but how to send an email automatically from asterisk's, extensions.conf ? |
| 20:48.23 | rethus | cause the live-server isnÃ't here, and one week befor, i had a hdd-crash |
| 20:49.24 | nny_1 | btw anyone know if musiconhold app supports mixing in adverts during moh? |
| 20:50.59 | jaytee | nny_1, you'd have to mix a custom MOH file that has them already inserted. You can use Audacity to edit most audio formats that * uses. |
| 20:51.18 | nny_1 | jaytee: yeah thats what i figured |
| 20:51.42 | nny_1 | jaytee: replacing a nortel system they had purchased a mixer for it, need to see what it does |
| 20:51.51 | jaytee | unless you're in a queue and then you can use MOH plus announcement messages at preset intervals |
| 20:52.10 | jaytee | nny_1, Meridian or Norstar? |
| 20:52.38 | nny_1 | jaytee: Meridian i believe lemme check |
| 20:53.01 | nny_1 | jaytee: correction norstar |
| 20:53.13 | jaytee | Meridian is their full PBX line, Norstar is their line of key systems |
| 20:53.13 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 20:53.25 | rwaite | i unplugged one of the pots lines going into my tdm400p today and a message popped up saying alert red alert. is there any way to configure some sort of notification action when those alerts appear? |
| 20:54.17 | nny_1 | jaytee: looks like they just got software from Nortel |
| 20:54.34 | linuxer_igor | hi, anybody use gtk console? |
| 20:54.40 | linuxer_igor | how to use this ? |
| 20:55.07 | jaytee | linuxer_igor, gtk console? for writing gtk apps? umm, you're in the wrong channel |
| 20:55.33 | linuxer_igor | pbx_gtkconsole |
| 20:55.46 | jaytee | never used it or looked at it |
| 20:55.54 | linuxer_igor | module load pbx_gtkconsole.so |
| 20:56.01 | linuxer_igor | it's ok |
| 20:56.06 | jaytee | got a link to where you got it? |
| 20:56.07 | linuxer_igor | but how to use ? |
| 20:56.31 | rethus | other question... have anybody installs * on a S100 Settopbox (http://wiki.zenega-user.de/index.php?title=Features) |
| 20:58.14 | *** join/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 20:58.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:00.38 | cj | nny_1: is there a central registration entity (not unlike ARIN) that is responsible for phone number allocation? |
| 21:04.45 | *** join/#asterisk kj5t (n=steve@65-120-138-35.dia.static.qwest.net) |
| 21:05.07 | kj5t | I am sorry, I stepped out. Is there anyone here who can help me with fowarding calls to my asterisk extension to my cell phone? |
| 21:05.29 | cj | kj5t: assume there is. what's your question? |
| 21:08.09 | kj5t | What I want to know is how to go about doing that. something like exten = 127,1,Dial(SIP/Excel/cellphonenumber) but I am not sure |
| 21:08.10 | *** part/#asterisk rethus (n=rethus@xdsl-81-173-176-47.netcologne.de) |
| 21:08.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:09.42 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 21:10.22 | nny_1 | cj you mean http://www.nanpa.com/ ? |
| 21:11.02 | *** join/#asterisk AndyML (n=AndyML@206.166.206.34) |
| 21:11.04 | nny_1 | got to flee.. the day star beckons. bbl |
| 21:11.07 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 21:12.12 | kpettit | wooo hoo, finally got meetme to work. |
| 21:12.26 | kpettit | Asterisk 1.6.1beta meetme works without Zaptel |
| 21:12.35 | kpettit | that was killing me before |
| 21:14.37 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:14.41 | Qwell | O.o |
| 21:14.45 | russellb | o.O |
| 21:15.00 | remont | hi, someone test, integrate asterisk with a nortel meridian? |
| 21:15.48 | DarKnesS_WolF | russellb: what about the eye did u spot redbull with Qwell ;-)? |
| 21:17.28 | *** join/#asterisk mateo_au (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 21:22.08 | ManxPower | cj: there is an authority for each country. In the USA that authority is NANPA/ |
| 21:22.52 | cj | ManxPower: aha. thanks. |
| 21:34.06 | l2trace99 | anyone know of any issue that would cause asterisk to delete vm directories ? |
| 21:35.51 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:36.26 | luke-jr | hey, is SellVoip actually in business again? |
| 21:36.35 | luke-jr | eg, is someone using them successfully? |
| 21:37.38 | magic_hat | hey all. Our asterisk/voip setup has a lot of call-quality fluctuation to it -- several times a day, calls end up failing over to our backup VM system at Teliax. Other times audio quality is sketchy. I'm guessing this is related to bandwidth ups and downs with our ISP, and I'm looking to make a change. What kind of service am I looking for to support up to 12 grandstream IP phones, as well as regular old office internet use? |
| 21:38.11 | magic_hat | t1? synchronous DSL? something else? |
| 21:41.44 | magic_hat | anyone home in here? |
| 21:46.36 | lmadsen | I'm home |
| 21:54.58 | ManxPower | magic_hat: Best of luck with that. |
| 21:56.10 | ManxPower | Many people end up getting a 2nd internet connection just for voice. |
| 21:56.26 | ManxPower | You could get a T-1 with QoS, but expect that to be at least $300/month |
| 21:56.38 | ManxPower | but you still can't QoS once the data leaves your ISP. |
| 22:00.10 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 22:00.27 | DarKnesS_WolF | guys what might be the problem i am trying to solve sine months sipphone nated ------> internet ----> asterisk ----> sip provder the link in all ways are 512/128 kb |
| 22:00.50 | DarKnesS_WolF | 1st call from phone to asterisk works fine after asterisk calling hte provder delay up to 8 secounds |
| 22:00.57 | DarKnesS_WolF | and asterisk eating all the connection |
| 22:01.05 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 22:01.12 | magic_hat | I think solving this problem would probably be worth $300 a month. |
| 22:01.28 | DarKnesS_WolF | i tried that with GSM / g729 /speex all the same results and i don't think it is a bandwidth issue |
| 22:01.41 | cj | so... am I correct in assuming that the smallest amount of POTS address space one can purchase from NANPA is 10,000 - 1 ? |
| 22:02.17 | ManxPower | cj: you are not correct. |
| 22:02.40 | ManxPower | with LNP the smallest block is 1 number |
| 22:03.01 | ManxPower | Telcos deal with NANPA not end users or busineess |
| 22:03.02 | cj | ooh. I can probably afford that... |
| 22:03.31 | cj | well, I'm considering becoming a telco/isp |
| 22:03.46 | ManxPower | NANPA won't even give you a logon on their web site if you are not a carrier. |
| 22:03.56 | russellb | i can haz be next vonage? |
| 22:04.08 | ManxPower | russellb: No! |
| 22:04.12 | russellb | ;( |
| 22:04.49 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 22:04.53 | ManxPower | You might be able to become a non-facilities based telco for $250,000, but I doubt it. |
| 22:05.04 | cj | non-facilities based? |
| 22:05.35 | ManxPower | cj: you have no equipment, switches, or lines, you rent it from the telco |
| 22:05.41 | generalhan | hey all ... can some one point me in the right direction as to what i need to research to find out how i can playback a certain wav file on a channel that is already opened ? |
| 22:05.45 | ManxPower | i.e. basic resale |
| 22:05.59 | cj | ManxPower: is there one that has equipment and switches, but no lines? :) |
| 22:06.07 | generalhan | like i have a call on a SIP channel and i need my employee to be able to hit a few buttons and have it playback a sound clip on that channel |
| 22:06.18 | ManxPower | generalhan: Playback(certain_wav_file). |
| 22:06.26 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 22:06.39 | ManxPower | cj: nobody has lines except the local telco |
| 22:06.45 | generalhan | ManxPower: so if i use that in the internal context it will playback that file on the channel ? |
| 22:06.53 | ManxPower | but if you have your own switch you can just rent the copper |
| 22:07.07 | ManxPower | generalhan: what you are describing is a IVR |
| 22:07.09 | generalhan | i use Playback() all the time ... but never after a channel has been bridged ! |
| 22:07.22 | ManxPower | generalhan: you didn't say two channels were bridged. |
| 22:07.40 | generalhan | ManxPower: im sorry im probably doing a horrible job of explaining ... let me try again |
| 22:07.50 | ManxPower | generalhan: see features.conf for DTMF based call control |
| 22:08.00 | *** join/#asterisk s0lid (n=s0lid@122.53.111.85) |
| 22:08.20 | generalhan | ManxPower: i need to have an employee here call a client. then after they have that client on the phone i need the employee to hit a key combination so that both the client and the employee can hear a certain audio file |
| 22:08.30 | cj | ManxPower: wouldn't a T1 card + asterisk count as a switch? |
| 22:10.32 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:14.13 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:19.48 | ManxPower | generalhan: why not just conference in a 3rd leg of the call and that leg play the file |
| 22:20.03 | ManxPower | cj: Not according to the telcos |
| 22:20.07 | generalhan | hmm, interesting thought ! |
| 22:20.28 | ManxPower | generalhan: then you can change the sound files without making the user change what they dial |
| 22:21.18 | generalhan | so then i could do something like Exten => 1234,1,Playback(sound_to_play) and then have the caller just hit conference and hit 1234 ? |
| 22:21.27 | ManxPower | generalhan: yup |
| 22:21.46 | generalhan | ManxPower: wonderful ... i like that soo much better than messing in features.conf !! thanks for the idea, ill try it now ! |
| 22:21.52 | cj | ManxPower: how would the telcos know the difference? |
| 22:21.57 | ManxPower | if you tun a Hangup as the next priority the 3rd leg should go away automatically |
| 22:22.13 | ManxPower | cj: if you are putting the equipment in their facilities then they will know |
| 22:22.39 | ManxPower | Also until recently Asterisk did not even support SS7, and good luck getting a telco in the USA let you connect via SS7 |
| 22:22.46 | ManxPower | (which would required if you are a real carrier) |
| 22:22.54 | cj | ManxPower: hmmm... the telco doesn't really pay attention to what's in my rack... |
| 22:24.01 | ManxPower | cj: You just go do what makes you happy. |
| 22:24.35 | cj | ManxPower: I'm sorry. I don't mean to insult... it sounds like I've offended you in some way |
| 22:25.02 | ManxPower | cj: nope, just tired of teaching you for free when I have other things I could be doing. |
| 22:25.15 | angryuser | cj: no it's his normal condition ;) |
| 22:25.20 | cj | ManxPower: okay. thank you for the information. |
| 22:25.22 | cj | angryuser: :) |
| 22:25.27 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 22:27.59 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 22:29.57 | generalhan | ManxPower: thanks a lot ... that worked like a charm ! |
| 22:30.24 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-7a93497a9dab1e93) |
| 22:30.56 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 22:31.36 | jaytee | ManxPower, the other nite you and [TK]D-Fender were talking about how a sip account should not be named the same as an extension. I understood your arguement as to why but I have a couple questions if you're not too busy? |
| 22:32.34 | _ShrikE | cj: before you even think about what you are proposing you should find yourself a good telecom attorney. |
| 22:33.35 | cj | _ShrikE: *sigh* always with the lawyers... |
| 22:33.58 | _ShrikE | cj: yep.. the regulation is endless |
| 22:40.49 | De_Mon | lastlog cj |
| 22:40.58 | cj | lastlog de_mon :) |
| 22:42.39 | De_Mon | the things I say don't produce responses like: 06:32PM <_ShrikE> cj: before you even think about what you are proposing you should find yourself a good telecom attorney. |
| 22:43.02 | De_Mon | not in irc anyways |
| 22:43.44 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:43.58 | cj | De_Mon: yeah, sorry for being "interesting" as they say... *sigh* |
| 22:47.34 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
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| 22:53.19 | *** join/#asterisk voxter (n=voxter@mail.metrobridge.com) |
| 22:57.35 | riddlebox | anyone know of any asterisk related podcasts? |
| 22:58.04 | ManxPower | I don't think cj is actually wanting to become a phone company. Sounds like he wants to be a VoIP service provider |
| 22:58.08 | ManxPower | jaytee: ask |
| 22:59.12 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:59.18 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 23:00.36 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 23:01.00 | jaytee | ManxPower, I looked at your sample Polycom configs on your site and was wondering that since the example sip.conf is using MAC addresses for the phones I'm assuming that you have to hardcode that in the dialplan in extensions.conf. Am I correct? |
| 23:01.26 | ManxPower | jaytee: We do hardcode it. If we used a database we might not have to. |
| 23:01.29 | *** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk) |
| 23:01.41 | jaytee | ManxPower, understood. |
| 23:01.55 | cj | ManxPower: I dunno. I should take baby steps. the first thing I want to do is to switch my cell phone to be answered at my server. |
| 23:02.21 | cj | ManxPower: I should wait 'till I get done with that before I start getting "fancy" :) |
| 23:02.31 | cj | oh, and maybe I should learn what I'm doing first :) |
| 23:02.59 | ManxPower | The thing is that things like exten => _XXXX,1,Dial(SIP/$[EXTEN}) does not scale well |
| 23:02.59 | jaytee | ManxPower, so you'd have something like exten => 222,1,Dial(SIP/0004f21abc23) in a context for internal extensions to dial that phone when someone dials 222. |
| 23:03.08 | cj | is mattf here? |
| 23:03.09 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 23:04.01 | jaytee | ManxPower, yeah and it's been on my mind since you guys discussed it the other night so I've been pondering it. I've also looked at streamlining my extensions.conf |
| 23:04.02 | ManxPower | jaytee: standby |
| 23:04.06 | jaytee | yep |
| 23:04.44 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-219-16.phlapa.east.verizon.net) |
| 23:04.57 | cj | is there a driver for getting dial plans from a databse instead of an on-disk file? |
| 23:05.06 | cj | I think I recall hearing something about that at one point... |
| 23:05.27 | _ShrikE | cj: realtime |
| 23:05.39 | cj | _ShrikE: hmm? |
| 23:05.42 | cj | is that the driver name? |
| 23:06.27 | seanbright | it's a core part of asterisk's architecture |
| 23:06.47 | ManxPower | jaytee: What we actually do is something more like exten => 222,1,Set(DIAL_DEST=SIP/0004f21abc23-a) exten => 222,n,Set(CFBL_DEST[1]=SIP/0004f21abc23-b) exten => 222,n,Macro(std-exten-v2) |
| 23:08.39 | ManxPower | see http://www.fnords.org/~eric/macro-std-exten-v2.inc |
| 23:08.55 | jaytee | ah, ok. was gonna ask for that :-) |
| 23:09.32 | ManxPower | jaytee: we COULD set all those variables via an AGI or database, we just don't want the headache of doing it that way. |
| 23:09.34 | jaytee | wow! |
| 23:10.17 | jaytee | that's gonna take me a little while to "walk through" :-) |
| 23:10.42 | ManxPower | jaytee: don't worry too much about the actual macro, it's spagetti code and really ugly. Just pay attention to the concept and the docs near the top |
| 23:11.40 | jaytee | ManxPower, you ain't seen ugly till you've seen my dialplan :-) |
| 23:11.43 | ManxPower | The idea here is: set variables via any method you want, then run the macro. The macro is controlled by the variables you set before running it. |
| 23:11.54 | seanbright | that would look so hot with AEL... ;) |
| 23:12.11 | ManxPower | seanbright: My plan is to re-write it for 1.6 in AEL |
| 23:12.28 | seanbright | ManxPower: you'll be happy with the result |
| 23:12.35 | jaytee | seanbright, got any good links for decent documentation on AEL? what I've found so far is pitiful. |
| 23:13.32 | seanbright | jaytee: murf updated most of the voip-info docs, and there is a sample extensions.ael in the distro |
| 23:13.37 | ManxPower | seanbright: I would have rewritten it for 1.4, but we skipped 1.4 entirely. |
| 23:14.17 | seanbright | yeah i know |
| 23:14.20 | *** part/#asterisk bradleyprice86 (n=bprice@68-113-101-90.static.leds.al.charter.com) |
| 23:14.35 | seanbright | the main win of AEL in IMHO is that it is less noisy |
| 23:14.49 | *** join/#asterisk chazz (n=chazz@nat/digium/x-3687c1af34eb87d4) |
| 23:14.53 | seanbright | too many "in"s in that last statement |
| 23:15.09 | jaytee | ManxPower, when you setup the sip.conf using the MAC address of the phone was it because 1) it's always unique (or should be) and 2) it gives you a way to have a record of the MAC address for each user's phone? |
| 23:15.17 | ManxPower | seanbright: I started that macro before there was a While, For, or Gosub and before functions existed. It's still close to it's heritage. |
| 23:15.24 | seanbright | though with the patch Corydon76-dig introduced today (same =>) that will help with that somewhat |
| 23:15.30 | ManxPower | jaytee: Yes! 8-) |
| 23:15.55 | seanbright | well, committed today i should say. |
| 23:16.12 | jaytee | ManxPower, excellent!! I was thinking of using our network logon ids which is also unique but I like this better. |
| 23:16.18 | seanbright | so you'll be able to do: |
| 23:16.22 | ManxPower | It was the easiest way to uniquely identify the phone+line appearance, we use polycoms so the MAC is very important to know (used for centralized provisioning) and the MAC is printed on the bottom of every phone. |
| 23:16.32 | seanbright | exten => XXXXXXXXXX,1,NoOp(Hey!) |
| 23:16.46 | seanbright | same => 2,NoOp(hot!) |
| 23:17.18 | jaytee | seanbright, thanks for pointing me in that direction. I've looked at the extensions.ael sample but didn't look at voip-info's stuff. Most of the stuff on that site is horrifically out of date. |
| 23:17.53 | seanbright | jaytee: it's one of those things you just have to learn as you go |
| 23:18.09 | jaytee | seanbright, like so much of life :-) |
| 23:18.11 | ManxPower | We have extension numbers that ring on multiple phones so it's important to be able to find the actual SIP device, the users only know the extension. We ask "what are the last 5 letters/numbers on the white sticker on the bottom of the phone that starts with '0004'" |
| 23:18.40 | ManxPower | even the stupidest of users can give us that information |
| 23:18.41 | seanbright | now if i could only convince murf to only allow strings if they are quoted... |
| 23:18.44 | jaytee | should have learned my lesson about the stock market back in 1987 during the junk bond crisis but noooooooooo! |
| 23:19.17 | ManxPower | jaytee: As of now I'm basically unemployed so quit whining! *tease* |
| 23:19.39 | seanbright | ManxPower: i thought #asterisk was your full-time gig? |
| 23:19.40 | jaytee | but screw retirement, I mean who want's to go on leaf-peeping tours and bus trips to factory outlet malls anyways :-) |
| 23:19.54 | ManxPower | seanbright: #asterisk is an unpaid hobby. |
| 23:20.22 | ManxPower | I'll be here a lot less as I get busy trying to generate revenue. |
| 23:20.25 | seanbright | was joking |
| 23:20.31 | ManxPower | Anyone want to buy a tellabs echo canceler 8-) |
| 23:20.33 | codefreeze-lap | seanbright: that'll be AEL3. |
| 23:20.43 | seanbright | codefreeze-lap: don't tease me! |
| 23:21.02 | jaytee | you and [TK] ought to get a monthly stipend from Digium for keeping their platform afloat. |
| 23:21.14 | ManxPower | jaytee: I've thought that for *years*. |
| 23:21.19 | seanbright | you mean "our" platform |
| 23:21.19 | codefreeze-lap | looks around for someone who might be teasing... |
| 23:21.42 | ManxPower | If they paid me I'd even have to be nice(r). |
| 23:21.49 | jaytee | seanbright, I stand corrected. Our platform |
| 23:21.51 | seanbright | ManxPower: we couln't have that. |
| 23:22.34 | jaytee | although my portion of "our" platform would never have got off the ground if I hadn't had ManxPower and [TK |
| 23:23.05 | jaytee | helping me out when I first got going |
| 23:23.08 | seanbright | codefreeze-lap: you'd have to include an option to allow unquoted strings for all the extensions.conf purists out there. |
| 23:23.24 | seanbright | jaytee: they are great resources, indeed. |
| 23:23.38 | seanbright | jaytee: assuming your skin is /just/ thick enough |
| 23:24.06 | jaytee | seanbright, I now completely understand where they're coming from after spending alot of time in here. |
| 23:24.09 | seanbright | ;) |
| 23:24.13 | seanbright | agreed |
| 23:25.32 | jaytee | I like to try and solve my own problems or at least make an attempt. I use google and I do some trial and error with a "control" and an "experiment" just like in a lab. If I'm still stuck I come here and try to have my info ready in a pastebin before I ask so I don't waste anyone's time. |
| 23:26.24 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:26.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:26.29 | jaytee | but I understand how frustrated ManxPower and [TK] can get when people want to be spoon fed stuff and won't listen when asked for more information to help troubleshoot the problem. |
| 23:26.41 | seanbright | it's much more gratifying to solve your own problems, too. |
| 23:26.53 | jaytee | it is |
| 23:26.59 | jaytee | like busting through a wall |
| 23:27.03 | seanbright | indeed. |
| 23:27.10 | jaytee | or reaching the peak of Everest |
| 23:27.14 | seanbright | i read some of the questions people ask and say to myself... |
| 23:27.30 | seanbright | "man... i could do this guys job with a hand tied behind my back." |
| 23:27.42 | seanbright | meanwhile they are probably making twice what i am |
| 23:27.44 | seanbright | heh |
| 23:28.06 | jaytee | really? my response is usuallly more like "WTF?" or *headdesks |
| 23:28.19 | ManxPower | It seems that most new asterisk users know nothing about telecom, almost nothing about networking, and some of them barely know Linux. |
| 23:28.35 | seanbright | <-- knows squat about traditional telephony |
| 23:28.38 | ManxPower | pile on knowing nothing about Asterisk and the learning curve is very steep. |
| 23:29.29 | ManxPower | seanbright: I've been fascinated by traditional telephony since I was a kid, I started doing it professionally at the end of 2001 with Asterisk |
| 23:29.43 | ManxPower | Been doing networking and linux for years. |
| 23:29.47 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:29.54 | jaytee | seanbright, the network engineer I work with has his MCSE. I'm still working on one test that'll give me my MCSA. I have MCP in XP and MCDST already but he can't even figure out how to setup DHCP to force clients into a particular address scope based on Organizationally Unique Identifiers. |
| 23:30.16 | jaytee | and he knows squat about linux yet he makes 20K more than I do |
| 23:30.29 | seanbright | ugh |
| 23:30.36 | ManxPower | jaytee: I'd have trouble even installing a DHCP server on Win32 |
| 23:30.50 | seanbright | well... i am MCSD and my company has me doing analysis services work :-/ |
| 23:31.04 | jaytee | he's been there 1 year. I've been there 6. My boss is an asshole from hell. I'm taking my dCAP exam, finishing my MCSA and getting the hell out. |
| 23:31.47 | jaytee | installing DHCP on Win2K3 servers is a walk in the park. ManxPower, you could learn it in 10 minutes. |
| 23:31.57 | seanbright | if that |
| 23:32.03 | seanbright | add/remove program |
| 23:32.08 | seanbright | install windows components |
| 23:32.09 | seanbright | done |
| 23:32.10 | seanbright | heh |
| 23:33.00 | lmadsen | I think I joined the wrong room |
| 23:33.02 | seanbright | i guess my point is... imagine the average user of the average question in #asterisk |
| 23:33.04 | jaytee | the scope and option settings are pretty much intuitive and if you've setup DHCP in linux then Windows would be a cakewalk but my coworker, "Forrest Gump" can't figure out the OUI part to shove Polycom phones into their own address pool in a specific VLAN. |
| 23:33.17 | seanbright | and now imagine him advising his boss on technical solutions |
| 23:33.31 | seanbright | it gives me chills |
| 23:33.39 | jaytee | lmadsen, stick around! we'd be happy to teach you about Asterisk, the Future of Telephony! |
| 23:33.39 | lmadsen | seanbright: I choose not to think about that :) |
| 23:33.50 | lmadsen | jaytee: I've heard the future is now! |
| 23:34.00 | ManxPower | jaytee: I'd do that in the router with Cisco DHCP Relay, it stuffs info into the DHCP packet as the router routes it so the DHCP server servs out of the correct pool. |
| 23:34.03 | jaytee | duty now, spuds! |
| 23:34.37 | lmadsen | I've been fascinated with computers since grade 2, and telephones since around grade 6 |
| 23:34.39 | jaytee | ManxPower, hmmm. I'll have to run that by him. We're all Cisco switches and routers at our shop |
| 23:34.40 | lmadsen | ran a BBS for a while |
| 23:34.41 | hardwire | anybody here ever been part of an "open" sip hard phone project? |
| 23:34.54 | ManxPower | jaytee: I'm available for hire 8-) |
| 23:34.55 | jaytee | what's a BBS? ....... Just kidding |
| 23:35.19 | hardwire | lmadsen: the future is tomorrow. |
| 23:35.36 | hardwire | or... if you're headed to Alaska "North to the future!" |
| 23:35.40 | jaytee | remembers back when he thought he was the bomb because he had a 2400 baud Hayes Smartmodem on his 286/12mhz clone |
| 23:35.45 | WimpMan | Mine is still running, but I removed the line on new years day. |
| 23:35.49 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:35.50 | seanbright | so our network admin... he has all of our analog and digital lines coming into the server room. from the 50 pair running through the wall, he has attached individual biscuit jacks to each pair |
| 23:35.51 | lmadsen | actually the future is now, because by the time your brain processes now, it is already delayed |
| 23:35.58 | seanbright | i can't even get him to put in a 66 |
| 23:36.02 | hardwire | lmadsen: now |
| 23:36.02 | WimpMan | But I'm still using it as mail server. |
| 23:36.03 | lmadsen | the future is latent! |
| 23:36.03 | hardwire | no now |
| 23:36.05 | hardwire | no now |
| 23:36.07 | hardwire | no now |
| 23:36.07 | hardwire | now |
| 23:36.16 | ManxPower | jaytee: I saved one company %40 off their $10,000/month bill for T-1s, etc after evaluating what they had, what they needed, and how we could use the tarrifs to our advantage. |
| 23:36.25 | hardwire | actually.. the future just passed. |
| 23:36.31 | jaytee | ManxPower, you wouldn't want to live in Indiana most likely and my boss is too much of a dick to hire anyone to work remote. |
| 23:36.49 | seanbright | i'd just miss DST |
| 23:36.49 | hardwire | is mad when the future keeps zipping right by him |
| 23:37.19 | ManxPower | jaytee: I was revering to one-off small projects like setting up DHCP relay on the routers and coordinating testing with the server admin |
| 23:37.26 | lmadsen | off to dinner! |
| 23:37.29 | ManxPower | seanbright: I'd miss DST too. |
| 23:37.31 | jaytee | ManxPower, once I get everyone off our Nortel Option 11 and on Asterisk and can pull the plug on it we're going to look at a SIP trunk solution with a company called Axia. |
| 23:37.38 | ManxPower | those perverts in IN don't believe in DST! |
| 23:38.07 | jaytee | ManxPower, no we adopted DST starting in 2006 |
| 23:38.12 | jaytee | finally |
| 23:38.16 | ManxPower | Cool! |
| 23:38.30 | hardwire | showing a beginner programmer how python works |
| 23:38.34 | seanbright | packs his things |
| 23:38.39 | jaytee | coming from Massachusetts it was quite an adjustment when I first moved here. |
| 23:38.46 | ManxPower | seanbright: winters there suck |
| 23:38.52 | seanbright | yeah, my uncle lives there |
| 23:38.54 | hardwire | wrote a script to interface to asterisk manager interface, then mute my soundcard my sip channel was in use. |
| 23:38.56 | jaytee | summer are horrible too |
| 23:39.02 | hardwire | he's been enjoying muting my volume by calling me |
| 23:39.15 | jaytee | high 80's and low 90's with high humidity and dead air |
| 23:39.17 | ManxPower | jaytee: better than summers in TN or LA |
| 23:39.34 | seanbright | wanders off |
| 23:39.38 | jaytee | ManxPower, that'll I'll give ya but nothing is worse than Biloxi, MS. |
| 23:39.54 | ManxPower | at least in the north it's not usually really hot every day and it usually cools off at night. |
| 23:40.43 | jaytee | yeah, I miss Southern Oregon. I lived there for 5 years. It gets hot but it's so dry, like 19% humidity that you don't feel as uncomfortable. |
| 23:41.04 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net) |
| 23:41.15 | ManxPower | I'd move back to Santa Monica / Venice Beach (southern California) in a moment if I could. |
| 23:42.00 | coil | me too |
| 23:43.06 | jaytee | oh, god!!! I went rollerblading from Santa Monica pier through Venice Beach to Redondo and back in Feb of 98 and loved it. I almost broke my neck slipping on my own drool from all the bikini babes with incredible ta-ta's rollerblading in the opposite direction. |
| 23:43.34 | jaytee | I was down there for a 2 week class at Nortel's site in La Palma. |
| 23:44.14 | WimpMan | Oh, They're real? I.e. not only on TV? |
| 23:44.57 | jaytee | WimpMan, ever see the movie L.A. Story with Steve Martin and Sarah Jessica Parker? |
| 23:45.25 | jaytee | "SanDee, you're breasts feel weird!" "Oh, that's cuz they're real!" |
| 23:45.48 | WimpMan | Yes, that rings a bell. But I don't know wich. I'm bad at remembering films. |
| 23:46.40 | jaytee | a funny movie that poked fun at the L.A. lifestyle |
| 23:47.03 | jaytee | "I'll have a double-decaffienated half-caf with a twist of lemon!" |
| 23:47.57 | WimpMan | That was the one with the canned sweat, wasn't it? |
| 23:48.16 | jaytee | in one scene he has to go to a bank and have a financial portfolio review just to get a reservation at a new trendy restaurant called L'Idiot |
| 23:48.31 | jaytee | I don't remember the canned sweat |
| 23:52.41 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 23:53.36 | alancio | hello, I'm trying to configure a fxs interfase on asterisk, almost there, but can't place calls |
| 23:54.27 | alancio | I have: immediate=no, but as soon as I press a number I get a busy tone |
| 23:54.50 | alancio | I can receive calls to the phone connected to the fxs |
| 23:55.33 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
| 23:56.01 | *** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:56.28 | jaytee | alancio, fxs interface is kinda generic info. Is it a Digium card or an ATA adapter? |
| 23:56.52 | alancio | Digium TDM410 card with a FXS module |
| 23:58.18 | alancio | I think my problem is with the channel definition in zapata.conf |
| 23:58.30 | jaytee | pastebin it |
| 23:58.33 | alancio | I have a single [channels] section for all the channels |
| 23:58.51 | jaytee | or use the book as a guide |
| 23:58.54 | jaytee | ~book |
| 23:58.55 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:59.21 | alancio | ohhh I think I have it |
| 23:59.28 | coil | ~buybook |
| 23:59.29 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 23:59.46 | alancio | first you configure the channels parameters and then you define it with a channel => X line |