| 00:00.04 | slowshutt | already connect? |
| 00:00.11 | slowshutt | past your extension.conf |
| 00:00.28 | asdx | slowshutt: kiax |
| 00:01.18 | slowshutt | should look something line this exten => _X.,1,Dial(IAX/diego/S{EXTEN}) |
| 00:02.14 | asdx | exten => _1XXXXXXXXXX,1,DIAL(IAX2/diego@teliax/$\{EXTEN},20) |
| 00:02.25 | obnauticus | On what band does CDMA run, does anyone know? |
| 00:02.53 | slowshutt | why the \? |
| 00:03.55 | asdx | slowshutt: teliax told me to use that |
| 00:04.10 | fujin | o_0 |
| 00:04.12 | fujin | that's odd. |
| 00:04.22 | slowshutt | 800 and 1900mhz obnauticus |
| 00:04.49 | slowshutt | doesn't look right to me |
| 00:04.59 | slowshutt | take it out and test dialing |
| 00:05.26 | asdx | [Nov 15 18:03:17] NOTICE[20290]: chan_iax2.c:5258 register_verify: Peer 'diego' is not dynamic (from 190.52.147.91) |
| 00:05.29 | asdx | when i try to connect |
| 00:05.46 | fujin | insecure=host |
| 00:07.20 | slowshutt | did they say anything about having g729 licences if you want to connec to to them? |
| 00:07.46 | asdx | i'm trying to connect from my localhost (my computer) with kiax to the asterisk server and i get that in the server |
| 00:07.50 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 00:08.01 | asdx | i'm confused, should i connect the pbx to the ITSP first? |
| 00:08.23 | asdx | maybe i should go to read the book... |
| 00:08.53 | slowshutt | sorry i forgot add bindport=4569 to general in iax.conf |
| 00:09.16 | slowshutt | and bindaddr=0.0.0.0 |
| 00:09.40 | asdx | slowshutt: when i connect my client to *, asterisk will connect to the ITSP right? |
| 00:09.54 | asdx | with that user |
| 00:09.54 | slowshutt | yes |
| 00:10.28 | *** part/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 00:10.32 | slowshutt | you can add qualify=yes in diago context to force it to check the connection |
| 00:14.35 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:15.41 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 00:17.14 | asdx | slowshutt: ok |
| 00:19.08 | slowshutt | any luck asdx |
| 00:19.09 | slowshutt | ? |
| 00:19.33 | asdx | [Nov 15 18:17:20] NOTICE[9321]: chan_iax2.c:5258 register_verify: Peer 'diego' is not dynamic (from 190.52.147.91) |
| 00:20.22 | slowshutt | change type =frined to type=peer |
| 00:21.18 | slowshutt | did they tel you to put the register string in you general under iax.conf? |
| 00:21.50 | JT | slowshutt: why change to peer? |
| 00:21.56 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 00:22.15 | slowshutt | add the register command and egnore last statement |
| 00:22.51 | slowshutt | JT got iax and sip confused |
| 00:22.56 | asdx | slowshutt: yes they told me to add the register string |
| 00:23.09 | slowshutt | add it and test |
| 00:23.24 | slowshutt | remember to do iax reload before you test |
| 00:23.58 | JT | slowshutt: either way, friend is usually what you want, sip or iax |
| 00:25.06 | Mackes | I worked some more with Polycoms today. |
| 00:25.16 | Mackes | They are a decent phone |
| 00:25.24 | Mackes | Im not sure they are the end all |
| 00:25.28 | Mackes | but very good |
| 00:25.44 | asdx | still same message |
| 00:25.53 | asdx | is not dynamic |
| 00:26.35 | asdx | maybe i should change something in my teliax account configuration? |
| 00:27.50 | asdx | i'll go to read |
| 00:28.13 | JT | asdx: is it set as dynamic from teliax side? |
| 00:28.21 | JT | asdx: are you accessing it from a static ip? |
| 00:28.26 | JT | with no nat |
| 00:28.35 | asdx | JT: i'm behind a router |
| 00:28.45 | asdx | JT: with nat |
| 00:29.33 | asdx | under "features" |
| 00:29.39 | asdx | in the teliax account |
| 00:29.42 | slowshutt | make sure port 4569 is open |
| 00:29.58 | asdx | i see the radiobuttom is check to SIP |
| 00:30.05 | asdx | slowshutt: ok |
| 00:30.17 | asdx | it lets me update to IAX |
| 00:30.23 | asdx | or SIP+IAX |
| 00:31.09 | asdx | damn it |
| 00:31.24 | asdx | Apr/02/2002 16:28:25 Target IP(192.168.0.1), Target Port(5678) Packet Dropped |
| 00:31.28 | asdx | in the router |
| 00:31.52 | asdx | what a gay router |
| 00:32.45 | BBHoss | maybe increase the qualify time? |
| 00:33.14 | BBHoss | aren't you trying to do iax2? |
| 00:33.24 | asdx | BBHoss: yes |
| 00:33.24 | JT | erm |
| 00:33.39 | JT | maybe don't use static config |
| 00:33.42 | BBHoss | then why would it be hitting 5678? |
| 00:33.47 | BBHoss | should be 4569 |
| 00:34.02 | JT | because that's how tcp/ip usually works |
| 00:34.08 | JT | different source and dest ips |
| 00:34.11 | JT | s/ips/ports/ |
| 00:34.20 | asdx | oops |
| 00:34.35 | asdx | i confused myself, it's 4569 actually |
| 00:35.04 | JT | asdx: set it to dynamic at the provider. |
| 00:35.15 | asdx | JT: ok |
| 00:35.36 | JT | registration with static config is pointless |
| 00:36.38 | JT | static hostname config, that is |
| 00:39.34 | Zuchmir | which is the best format to use for FXO |
| 00:39.53 | *** join/#asterisk De_Mon (i=de_mon@fl-71-52-101-157.dhcp.embarqhsd.net) |
| 00:40.18 | De_Mon | Dang, I really need to add you guys to my autojoin list |
| 00:40.21 | slowshutt | asdx: should look something like this http://pastebin.com/m6fca014a |
| 00:40.24 | De_Mon | I just lost 3 hours of my life! |
| 00:40.55 | asdx | slowshutt: ok |
| 00:40.56 | asdx | slowshutt: thanks |
| 00:41.25 | asdx | <PROTECTED> |
| 00:41.29 | asdx | (914) 214-4052 voip-co4.teliax.com iax Off |
| 00:41.31 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 00:43.48 | slowshutt | iax2 show peers do you have anything online? |
| 00:44.29 | asdx | 1 iax2 peers [0 online, 0 offline, 1 unmonitored] |
| 00:44.35 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 00:46.08 | asdx | i connected many times already but using host=dynamic |
| 00:46.15 | asdx | without a ITSP |
| 00:48.12 | slowshutt | the only diference is that you are now connecting to a static address |
| 00:48.20 | asdx | [Nov 15 18:46:20] NOTICE[22418]: chan_iax2.c:5933 update_registry: Restricting registration for peer 'teliax' to 60 seconds (requested 300) |
| 00:48.33 | slowshutt | did you put back the register string? |
| 00:48.43 | asdx | yes |
| 00:48.57 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 00:50.11 | riddlebox | hrmm we are taking out an asterisk server as a trade in for a new Partner system, the salesman said that Asterisk sucks, because they couldnt transfer and see when people were on the phone, I asked him what phones they have and he said all are Grandstream budget tone phones lol |
| 00:56.59 | GreggB | riddlebox: Well, when you buy crappy stuff, then you get crappy "features". |
| 00:57.35 | riddlebox | thats what I told him, whoever sold that system to them didnt do a good job |
| 00:57.57 | riddlebox | the good news is he is giving me the server and the phones to play around with |
| 00:59.01 | GreggB | Heh - I'd eBay the phones and pickup a few SPA942's, or if you need BLF (and need to keep costs down) a few GXP-2000s. |
| 00:59.22 | riddlebox | I have 3 GXP-2000's already in my house |
| 00:59.24 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 00:59.33 | GreggB | :-) |
| 00:59.52 | riddlebox | I would like to get a cisco phone and a polycom phone |
| 01:00.03 | riddlebox | GreggB, what do you think of the Aastra phones? |
| 01:00.15 | GreggB | Aastra? Never heard of it |
| 01:00.34 | coppice | sheltered life? :-) |
| 01:00.44 | GreggB | apparently |
| 01:01.24 | GreggB | I've been hooked on these great little SPA942s, and just one GXP2000 |
| 01:01.48 | fujin | spa942's are fun. |
| 01:01.56 | fujin | never bothered with BLF though |
| 01:01.59 | fujin | we've got 50~ of em here. |
| 01:02.01 | coppice | the SPA942 looks like it was spray painted by a 4 year old in someone's garage |
| 01:02.09 | GreggB | Yea - seen the latest firmware? It removes the license requirement for the extra two lines. |
| 01:02.11 | fujin | sif, they're very clean looking |
| 01:02.14 | fujin | GreggB: yep :) |
| 01:02.24 | fujin | coppice: maybe azn knockoff ones. |
| 01:02.36 | fujin | they're exactly the same colour as the Cisco ones, I think.. |
| 01:02.43 | fujin | not as flashy as polycom though |
| 01:02.50 | coppice | the look like a knockoff of a cisco |
| 01:02.51 | GreggB | coppice: huh? maybe yours, but mine look like they were painted by a teen |
| 01:02.58 | slowshutt | Unable to support trunking on user 'bar' without zaptel timing |
| 01:03.08 | slowshutt | what does the above mead |
| 01:03.11 | slowshutt | what does the above mean? |
| 01:03.12 | coppice | GreggB: you must have version 2 |
| 01:03.20 | GreggB | coppice: they're not all that terrible, for what you paying for at least...and I cant complain too much about the software/hardware either. |
| 01:03.47 | GreggB | coppice: Ver2? They have a version 2 already? |
| 01:04.18 | coppice | well V1 looks like it was painted by a 4 year old. maybe you have low expectations of teens |
| 01:05.05 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:05.11 | GreggB | coppice: mine looks like...well....any of these: http://images.google.com/images?q=Linksys+spa942 |
| 01:05.27 | coppice | the polycoms look much better built. they are expensive here, but seem to be cheap in the US |
| 01:05.53 | GreggB | fujin: I dont know if the SPAs support BLF yet |
| 01:06.04 | GreggB | fujin: havent tried it at least... |
| 01:06.31 | fujin | GreggB: I know that the SPA962 with the SPA932 addon supports blf |
| 01:06.41 | fujin | coppice: ip330 |
| 01:06.47 | fujin | 's are comparable in price to the spa942 here |
| 01:06.52 | fujin | shame I didn' tknow that when I purchased 50 spa942's :p |
| 01:06.57 | GreggB | coppice: Well, our Polycom costs around $750, but is a conference phone... cant say I'm too fond of it at the moment though, as the audio pulses about every 2-3 secs |
| 01:07.23 | coppice | the simple polycoms are something like $80 |
| 01:08.07 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 01:09.50 | riddlebox | coppice, which phone is like $80 for polycom? |
| 01:10.03 | coppice | the 320 |
| 01:11.27 | riddlebox | coppice, what store do you use? |
| 01:12.14 | coppice | I don't. they are expensive here, but I was pointed at US sites offering them at that price |
| 01:14.35 | Mackes | Cisco phones are very good |
| 01:14.45 | Mackes | Aastra phones are great |
| 01:14.51 | Mackes | What would you like to know? |
| 01:15.25 | riddlebox | Coppice, those 320's do look nice |
| 01:15.59 | riddlebox | Mackes, I was just wondering what people thought of the aastra phones, they are not really talked about here |
| 01:17.37 | asdx | i think i already connected |
| 01:18.52 | asdx | i have register in [global] and host=dynamic |
| 01:19.12 | Mackes | I have deployed about 300 of them in my org |
| 01:19.16 | Mackes | what would you like to know? |
| 01:19.40 | Mackes | We have 480i's and 57i |
| 01:20.08 | asdx | Mackes: how do i make calls to normal telephones |
| 01:21.11 | J4k3 | asdx: using an ITSP is the easiest way |
| 01:21.22 | J4k3 | ~itsp |
| 01:21.25 | jbot | hmm... itsp is an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 01:21.54 | asdx | yeah i already have a teliax account |
| 01:24.57 | Mackes | How? |
| 01:25.19 | Mackes | You will need an account with a SIP provider that allows calls to the outside world |
| 01:25.25 | GreggB | riddlebox: do you use Polycom phones? Have you ever experienced an audible pulsing (volume lowers, and the restores to normal) every 2-3 seconds? |
| 01:25.55 | asdx | Mackes: teliax does sip and iax2 i think |
| 01:26.08 | GreggB | asdx: I use teliax with a couple of my customers, for both inbound and outbound calls over the PSTN - what issues are you running into? |
| 01:26.24 | Mackes | I have had that happen with a conference room polycom 4000 I have |
| 01:26.38 | asdx | GreggB: i don't know how to configure * yet |
| 01:26.57 | GreggB | Mackes: oh? was it * related, or something the IP4000 itself? |
| 01:27.02 | asdx | they tell me to add: host=voip-co4.teliax.com |
| 01:27.04 | Mackes | I use a company named http://vitelity.net/ |
| 01:27.04 | GreggB | ~book |
| 01:27.05 | jbot | 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 |
| 01:27.10 | asdx | they tell me to add: host=voip-co4.teliax.com |
| 01:27.15 | asdx | but when i connect |
| 01:27.15 | GreggB | asdx: ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 01:27.18 | Mackes | Yep, Read the book |
| 01:27.18 | asdx | yeah |
| 01:27.25 | Mackes | its is the bible of Asterisk |
| 01:27.33 | asdx | ok |
| 01:27.36 | Mackes | The authors chat online ofter |
| 01:27.52 | Mackes | and they are in the Toronto Asteirsk USer group |
| 01:27.58 | asdx | Mackes: mark spencer? |
| 01:28.04 | Mackes | no.... |
| 01:28.20 | Mackes | The Authors of the Orilly book |
| 01:28.26 | Mackes | Oreilly |
| 01:28.30 | asdx | k |
| 01:28.47 | GreggB | asdx: The teliax folks have a great support site which gives you working config snippets for *, but you'll likely still need a few other additions to your dialplan and sip.conf before you can truly use it. |
| 01:29.07 | asdx | GreggB: yeah |
| 01:30.04 | Mackes | Check out http://vitelity.net/ |
| 01:30.17 | Mackes | very cheap and great service |
| 01:30.19 | Mackes | no issues |
| 01:30.23 | Mackes | IAX and Sip |
| 01:30.31 | Mackes | I use them for my home phones |
| 01:30.50 | GreggB | Mackes: did that pulsing occur because of something in the * config, or was it something with the phone itself? I'm guessing the phone, as the pulsing is heard even at the initial dialtone |
| 01:31.02 | Mackes | Nope, not Asterisk |
| 01:31.30 | asdx | i'm just confused, when i set host=dynamic i can connect to my * pbx without problems, when i set host=voip-co4.teliax.com which the teliax examples tells me to use, i get: Peer 'teliax' is not dynamic (from 190.52.147.91) |
| 01:32.05 | Mackes | I have 100's of extensions (aastra 480's) with no issue |
| 01:32.07 | asdx | i guess i need to understand first what that exactly means |
| 01:32.08 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 01:32.16 | UserReg_CL | (alguien habla espaņol?) |
| 01:32.17 | Mackes | My Polycom phones act wierd |
| 01:32.35 | asdx | UserReg_CL: si yo porque? pero creo que aca no van a permitir que se hable espanol |
| 01:32.38 | J4k3 | Mackes: shhh, don't wake [TK]D-Fender |
| 01:32.40 | Mackes | I have Cisco 7960s with no issue |
| 01:32.46 | BBHoss | asdx: pastebin your configs (minus the passwords of course) |
| 01:32.51 | Mackes | and 2 Snoms with no issues |
| 01:32.59 | Mackes | <PROTECTED> |
| 01:33.06 | J4k3 | Mackes: woah, what brand wifi phone? |
| 01:33.11 | Mackes | 3 Polycoms that drive me crazy |
| 01:33.16 | Mackes | Lots |
| 01:33.22 | J4k3 | well |
| 01:33.24 | UserReg_CL | asdx: puedo preguntarte ? |
| 01:33.29 | GreggB | Mackes: Same here, I have a cocktail of SPA942s and LoudHush phones (~50), and this pulsing is only heard on the IP-4000. Did you ever resolve the pulsing issue? |
| 01:33.30 | J4k3 | I had the ut starcom, it was beyond worthless. |
| 01:33.33 | J4k3 | the f1000g |
| 01:33.34 | asdx | UserReg_CL: si, claro... pero soy un noob |
| 01:33.42 | J4k3 | did some googling, found that was pretty universal |
| 01:33.44 | Mackes | Zultys --- 500 of them |
| 01:33.47 | Mackes | WIP 2 |
| 01:33.48 | J4k3 | now I have a pocketpc pda w/ wifi |
| 01:33.59 | Zuchmir | ztmonitor 4 -vv shows me when the phone rings, but asterisk does not pick up, any clues? |
| 01:34.04 | Mackes | and I have several "Test" units. |
| 01:34.06 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 01:34.12 | Mackes | I bought one of everything |
| 01:34.12 | asdx | UserReg_CL: http://pastebin.com/m11fd7cea |
| 01:34.55 | Mackes | http://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones |
| 01:35.18 | UserReg_CL | asdx: solo quiero algo simple.. permitir cierto tipo de llamadas y el resto bloquearlas.. |
| 01:35.19 | Mackes | The Starcom is a weak phone |
| 01:35.29 | Mackes | The Hitachi is FANTASTIC |
| 01:35.46 | UserReg_CL | asdx: permitir todas las llamadas que empiecen con 09 y el resto de las llamadas bloquearlas... |
| 01:35.47 | GreggB | WIP2 = $300? |
| 01:35.49 | Mackes | and the Zultys is AWSOME, but looks like a mid 80's cell phone |
| 01:36.00 | J4k3 | I'd carry a moto brick if it worked :) |
| 01:36.02 | Mackes | We get them for $230 ish |
| 01:36.04 | UserReg_CL | asdx: se que interviene el archivo extensions.conf |
| 01:36.11 | asdx | BBHoss: http://pastebin.com/m11fd7cea |
| 01:36.11 | Mackes | Zultys work |
| 01:36.16 | Mackes | really well |
| 01:36.19 | Mackes | http://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones |
| 01:36.24 | Mackes | check that link |
| 01:36.40 | J4k3 | hmm, I'm going to have to try them, then. |
| 01:36.47 | GreggB | Mackes: oy... Well, we defaulted to Polycom as they've always made a great conference room phone. I just wish I could get rid of that pulsing |
| 01:36.50 | asdx | UserReg_CL: soy muy nuevo con asterisk, todavia no llegue a eso, pero supongo que debe haber algun tipo de ACL que puedas mirar, mira en el libro de asterisk |
| 01:36.53 | asdx | ~book |
| 01:36.54 | jbot | 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 |
| 01:37.30 | UserReg_CL | asdx: quisiera ver un ejemplo pero no encuentro como los que quiero.. |
| 01:37.59 | Mackes | Yeah... I was just chating with a guy who installed Asterisk on his IPhone |
| 01:38.05 | Mackes | Its availible for download |
| 01:38.07 | UserReg_CL | helpme please... need permit all call begin with 09 and block all others... helpme please |
| 01:38.15 | Mackes | Check out the Hitachi |
| 01:38.19 | Mackes | http://www.voipsupply.com/product_info.php?products_id=2996 |
| 01:38.27 | BBHoss | asdx: have you read the instructions that teliax provides? |
| 01:39.20 | GreggB | Mackes: * on the iPhone? Now that's a bit crazy |
| 01:39.33 | Mackes | yep, it is.. but he did |
| 01:39.38 | asdx | BBHoss: yes, that conf file is from the teliax samples |
| 01:40.08 | Mackes | http://www.mgamble.ca/oss/iphone_asterisk/ |
| 01:40.12 | UserReg_CL | helpme please... need config about extensions.conf for block any calls |
| 01:40.22 | Zuchmir | which is the prefered sound-file format for asterisk? |
| 01:40.33 | Mackes | what do you want to block? |
| 01:40.54 | UserReg_CL | (sorry by english :) |
| 01:40.59 | GreggB | asdx: On one of my systems, when I added the teliax config, I took the default/template sip.conf, and just added the teliax stuff - it worked out of the box after I added the appropriate entries into my dialplan. |
| 01:41.04 | Mackes | yep, english |
| 01:41.17 | BBHoss | UserReg_CL: use this in extensions.conf: _09.,1,Dial(SIP/${EXTEN}@sipprovider,,r) |
| 01:42.31 | GreggB | Mackes: weird...now what would be even more weird is if someone wrote a program for actually using the iPhone w/ * via SIP or IAX2...as...oh...a phone... :-) |
| 01:42.32 | Mackes | That will dial your sip provider when ever you dial 09, and a number... leaving the 09 on the number that is sent to your sip provider? |
| 01:42.43 | Mackes | IS that what you want? |
| 01:43.36 | UserReg_CL | need : allow all call begin with 09**** and all other call block !!! |
| 01:43.56 | UserReg_CL | ~book |
| 01:43.57 | jbot | 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 |
| 01:44.23 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 01:44.35 | Mackes | Ahhhh... Yes, well the the line BBHoss provided will do that.... if is the only line in your extentions config |
| 01:45.05 | BBHoss | UserReg_CL: the string a posted will dial 09 and ANYTHING after it, if you want 09 + 4 digits from 0-9 then replace the _09. with _09XXXX |
| 01:45.06 | Mackes | You dont really block calls in your extentions.conf, you write rules that allow calls. |
| 01:45.32 | BBHoss | asdx: what is the error you are getting with that config? |
| 01:45.39 | Mackes | So the line that BBHoss provided will allow calls to your provider when 09 is dialed |
| 01:45.55 | [TK]D-Fender | ~wifisip |
| 01:45.56 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 01:45.56 | Mackes | if it is the only line, then that is all your users will be able to do |
| 01:45.57 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 01:46.08 | Mackes | They do not suck man |
| 01:46.14 | Mackes | Many are VERY good |
| 01:46.17 | BBHoss | lets not start this shit again |
| 01:46.28 | Mackes | you are right |
| 01:46.28 | asdx | i told my customer i had to research... he told me "ok, research it, i will give you the time" |
| 01:46.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:46.35 | asdx | and now is he saying to drop everything.... |
| 01:46.37 | Mackes | lets agree to disagree |
| 01:46.41 | asdx | what a fucking idiot |
| 01:46.47 | asdx | :( |
| 01:46.57 | [TK]D-Fender | Mackes, they do. Poor call handling, low CPU that often doesn't allow for standard encrptions, crappy range and battery life. Many don't support even BASIC calling features like transfer, etc. |
| 01:47.01 | J4k3 | asdx: clue first, money later. |
| 01:47.22 | J4k3 | [TK]D-Fender: really, give it the fuck up. |
| 01:47.27 | J4k3 | goddamn. |
| 01:47.29 | asdx | J4k3: i really wanted to do this to learn :/ |
| 01:47.35 | Mackes | and what phones have you both used? |
| 01:47.37 | J4k3 | asdx: learn on your own time/dollar. |
| 01:47.43 | [TK]D-Fender | Mackes, Then there are those with very poor AP browsers, the Hitachi has a very weak screen (hi-res, but hard to read). |
| 01:47.49 | JT | Mackes: let's say again |
| 01:47.54 | asdx | J4k3: i don't have money |
| 01:47.56 | JT | Mackes: it's not so much the phones |
| 01:47.58 | [TK]D-Fender | J4k3, unload chan_bile.so ;) |
| 01:48.00 | JT | Mackes: it's the technology |
| 01:48.08 | BBHoss | heh |
| 01:48.11 | Mackes | ok |
| 01:48.29 | Mackes | but your wrong. We use 300 phones... every day, |
| 01:48.39 | Mackes | alll over 30 locations |
| 01:48.48 | BBHoss | there are numerous hospitals using WiFi phones |
| 01:48.51 | Mackes | using over 300 Access Points |
| 01:48.55 | Mackes | right |
| 01:48.57 | JT | Mackes: your spelling is wrong, it's "you're" |
| 01:49.05 | Mackes | yes |
| 01:49.06 | Mackes | ok |
| 01:49.07 | Mackes | thatnk |
| 01:49.10 | Mackes | thanks |
| 01:49.11 | asdx | is there a free ITSP that i can use for testing? |
| 01:49.15 | J4k3 | Mackes: you'll see you found a bunch of dumbfucks on IRC |
| 01:49.24 | JT | asdx: not for the pstn |
| 01:49.26 | J4k3 | I stay in this channel purely for the entertainment factor these days. |
| 01:49.27 | BBHoss | it may not be the BEST way to do wireless, but they work |
| 01:49.31 | [TK]D-Fender | Mackes, It may be workable, and in some cases a bit more necessary to do that in your specific application, but in the greater scheme of things, typically a bad idea. |
| 01:49.33 | BBHoss | asdx: yeah there is |
| 01:49.40 | Mackes | yeah... I dont know why I bother |
| 01:49.42 | BBHoss | asdx: VoipJet gives 25 cents free |
| 01:49.46 | UserReg_CL | BBHoss: |
| 01:49.50 | [TK]D-Fender | Mackes, I think he have spoken on this. |
| 01:49.50 | Mackes | but if you are interested, check out these phones |
| 01:49.52 | JT | J4k3: it's not about being dumbfucks |
| 01:49.55 | Mackes | http://www.voipsupply.com/product_info.php?products_id=354 |
| 01:49.56 | J4k3 | because, quite simply, you can't ask for someone more full of shit than [TK]D-Fender... |
| 01:50.05 | Mackes | yep |
| 01:50.06 | Mackes | we have |
| 01:50.09 | JT | the problem is, if we roll out wifi phones and people have crappy consumer APs |
| 01:50.15 | JT | then they complain that it works shit |
| 01:50.15 | Mackes | I am happy to let it go |
| 01:50.16 | BBHoss | you have both made yourselves very clear over the past 2 weeks :) |
| 01:50.21 | [TK]D-Fender | J4k3, You seem to spead the most personally targeted bile, and I seem to be the star of that show... |
| 01:50.22 | Mackes | yes |
| 01:50.27 | Mackes | done and done |
| 01:50.30 | BBHoss | JT: thats why you roll out DECT :) |
| 01:50.37 | J4k3 | [TK]D-Fender: excuse me, I guess polycom doesn't sell a wireless handset. |
| 01:50.45 | J4k3 | [TK]D-Spambot |
| 01:50.45 | Mackes | lol |
| 01:50.48 | BBHoss | technically |
| 01:50.51 | BBHoss | they do :) |
| 01:50.52 | JT | BBHoss: no. |
| 01:50.57 | Mackes | really? |
| 01:51.19 | BBHoss | they bought Kirk Telecom or w/e it was |
| 01:51.19 | [TK]D-Fender | J4k3, You can get over the fact that I don't hold the same opinion as you do for the gear you both use and recommend. I don't attack you personally for it so please calm down. The personally attacks are getting a little irritating. |
| 01:51.39 | J4k3 | [TK]D-Fender: I don't recommend equipment... you otoh, spread fud. |
| 01:51.39 | JT | yeah J4k3 needs to learn how to keep it professional |
| 01:51.52 | J4k3 | yeah, stay out of the way of the spambot. |
| 01:51.56 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 01:52.00 | J4k3 | thats professional, at least on #asterisk standards apparently. |
| 01:52.04 | JT | no, stop making personal attacks |
| 01:52.05 | JT | it's simple |
| 01:52.11 | J4k3 | JT: stay out of it. |
| 01:52.17 | JT | J4k3: stay off channel then |
| 01:52.21 | J4k3 | JT: thats the simplest way for you. |
| 01:52.28 | [TK]D-Fender | J4k3, you do regularly recommend your GS favourites, etc, and make scathing personal attacks on users here. Time for an attitude re-adjustment. |
| 01:52.34 | J4k3 | [TK]D-Fender: no, I don't |
| 01:52.47 | [TK]D-Fender | J4k3, no? |
| 01:52.48 | J4k3 | see, I say quite simply that you don't *have* to buy polycom's equipment to get working results. |
| 01:52.49 | [TK]D-Fender | <J4k3> [TK]D-Spambot <----- |
| 01:52.52 | JT | J4k3: you are the one who makes personal attacks, from what i've seen, [TK]D-Fender sticks to the issues |
| 01:52.53 | BBHoss | i haven't seen any personal attacks from anyone |
| 01:53.04 | [TK]D-Fender | <J4k3> because, quite simply, you can't ask for someone more full of shit than [TK]D-Fender... <-------------- |
| 01:53.14 | JT | BBHoss: not looking carefully? |
| 01:53.17 | BBHoss | i have seen overly biased opinions though, but that is human nature :) |
| 01:53.22 | J4k3 | I call you a spambot because all you do is toot polycom's horn, and talk shit about anyone else's gear. You're being pointed out as a spammer. |
| 01:53.25 | [TK]D-Fender | <J4k3> [TK]D-Fender: really, give it the fuck up. <--------- |
| 01:53.40 | JT | J4k3: polycom is not the only thing that [TK]D-Fender recommends |
| 01:53.55 | BBHoss | yeah i mean except now :) |
| 01:54.02 | [TK]D-Fender | JT, that's OK, he wants to see me as one-dimensional... |
| 01:54.06 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 01:54.10 | J4k3 | JT: since when? |
| 01:54.17 | Mackes | What ATA do you all recomend? |
| 01:54.20 | JT | J4k3: since ever |
| 01:54.32 | J4k3 | JT: really, go find your own argument. |
| 01:54.42 | RypPn | Mackes: I like the pap2t |
| 01:54.49 | BBHoss | Mackes: the PAP2s are alright |
| 01:54.49 | [TK]D-Fender | J4k3, I recommend Linksys & Aastra depending on implementations and geography, and plenty moer. |
| 01:54.52 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:55.01 | *** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 01:55.01 | J4k3 | [TK]D-Fender is at least entertaining... JT wants an argument that isn't even *for* him. Theres some professional IRCing for you. |
| 01:55.19 | JT | J4k3: i don't want an argument, i just want the emotional personal attacks to stop. |
| 01:55.37 | Mackes | Are those the Linksys Model? |
| 01:55.46 | BBHoss | Mackes: yes |
| 01:56.01 | Mackes | Yeah, those are nice |
| 01:56.01 | J4k3 | [TK]D-Fender: you lack any useful output, all you do is spread fud. I'm here to inform the people that might actually get confused with your bullshit that you *are* actually spreading nonsense. |
| 01:56.01 | RypPn | yeah linksys/cisco depending on which badge you look under first |
| 01:56.09 | [TK]D-Fender | J4k3, Nice to see you fanning flames for "entertainment". You are forfeiting any chance of not coming off as a hypocrite. |
| 01:56.21 | BBHoss | Mackes: if you can find a PAP2 V2 in stores, you can unlock it and get it to work with * without having to pay the unlocked price |
| 01:56.25 | UserReg_CL | mmm |
| 01:56.25 | Mackes | Does Cisco sell the PAP _septura as well? |
| 01:56.34 | J4k3 | [TK]D-Fender: hypocrite? I'm not the one spreading fud, sir. |
| 01:56.45 | BBHoss | why don't yall move to #flood :) |
| 01:56.52 | Mackes | How much should they go for... I'm paying about 80 |
| 01:57.05 | RypPn | expensive |
| 01:57.20 | [TK]D-Fender | J4k3, FUD? Hardly when so many cases validate the warning I hand out. |
| 01:57.23 | BBHoss | you're not in the US correct? |
| 01:57.34 | Mackes | I am |
| 01:57.38 | Mackes | Buffalo NY |
| 01:57.42 | BBHoss | ok |
| 01:57.49 | J4k3 | [TK]D-Fender: "so many cases"... you were a dumbfuck who bought a cheap AP, and you insist that everyone else is as dumb as you. CONGRATS. |
| 01:57.53 | [TK]D-Fender | J4k3, We all know the problems that follow these items, and not user-config type things either. |
| 01:57.58 | J4k3 | thank you, move the fuck on. |
| 01:57.58 | BBHoss | just go to walmart and get the PAP2 V2 or go to Staples or something, you can then unlock them |
| 01:58.06 | Mackes | I need the FXO model |
| 01:58.09 | [TK]D-Fender | J4k3, Who said anything about a "cheap AP"? I didn't |
| 01:58.13 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e07e3e7b36674aec) |
| 01:58.16 | JT | J4k3: enough with the personal attacks |
| 01:58.16 | BBHoss | oh |
| 01:58.27 | UserReg_CL | exten => _2.,2,Dial(IAX2/locales/${EXTEN}) |
| 01:58.27 | Mackes | Are they less expensive at Walmart? |
| 01:58.27 | JT | J4k3: it's fine to disagree about what to buy |
| 01:58.31 | Mackes | D |
| 01:58.31 | JT | just stick to the issues |
| 01:58.31 | RypPn | Mackes: http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=3508 in USD is x2, so about 60bucks |
| 01:58.35 | [TK]D-Fender | J4k3, and I'm not swearing like a hurt little child lashing out. |
| 01:58.35 | Mackes | Do they carry them? |
| 01:58.56 | BBHoss | yeah they are subsidized by vonage, if they have them |
| 01:59.05 | BBHoss | last time i checked they did |
| 01:59.05 | J4k3 | [TK]D-Fender: read the shit you type. "most". "Most" in your experience is most likely what you implement. Nobody wants to read your fud, provide something useful instead. |
| 01:59.09 | Mackes | Yeah, but I need them in the US, and with an FXO port |
| 01:59.09 | J4k3 | ~wisip |
| 01:59.13 | BBHoss | you have to make sure you get the v2 though |
| 01:59.26 | BBHoss | but they only proivide FXS |
| 01:59.27 | UserReg_CL | exten => _2.,2,Dial(IAX2/locales/${EXTEN}) now need block all others calls.. ŋhow? |
| 01:59.27 | BBHoss | you will need something else |
| 01:59.38 | J4k3 | ~wifisip |
| 01:59.38 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 01:59.42 | RypPn | like a spa3102 |
| 01:59.46 | J4k3 | [TK]D-Fender's favorite bot 'factoid' |
| 01:59.50 | RypPn | but they blow for echo |
| 02:00.00 | Mackes | You dont need to block just remove the exten that lets them call |
| 02:00.04 | J4k3 | when, quite simply, its not true |
| 02:00.04 | [TK]D-Fender | J4k3, My AP reference is for Wifi phones that don't have the CPU to support higher and very standard encryptions that the *AP* itself does and should be using. Use of those phones would FORCE you to use a weak encryption thereby comprimising your network. |
| 02:00.09 | BBHoss | i personally like to keep my FXO ports in the pc |
| 02:00.17 | Mackes | no blocks, just allows in Asterisk |
| 02:00.20 | BBHoss | loads better quality |
| 02:00.25 | [TK]D-Fender | J4k3, No, its not. Actually, it'd be ~book <-- |
| 02:00.31 | Mackes | I use ATA's at our retail locations for 911 |
| 02:00.35 | J4k3 | when you have to cover your ass by saying 'most' (with no evidence of 'most' even being the case) you're not adding anything useful to the conversation. |
| 02:00.41 | UserReg_CL | :( |
| 02:00.45 | Mackes | All phones, and one ATA, no Asterisk on site |
| 02:01.02 | BBHoss | ok that is fine |
| 02:01.02 | Mackes | and our HQ acts as the itsp |
| 02:01.04 | J4k3 | you could say, add something useful that says 'not all phones are good, and you need decent APs and network otherwise to expect decent performance' or whatever may be the case to make you comfortable. |
| 02:01.07 | [TK]D-Fender | J4k3, You can stop putting words in my mouth now. Your accuracy leaves far too much to be desired |
| 02:01.22 | BBHoss | the sipura 3102 (linksys) works well enough |
| 02:01.42 | BBHoss | provides 1 fxo and 1 fxs |
| 02:01.47 | BBHoss | http://www.voipsupply.com/product_info.php?products_id=1646&searchid=29033&gclid=CPWRpeii4I8CFQUsPAodEBkH8g |
| 02:01.51 | J4k3 | [TK]D-Fender: hint: this conversation is text and is well logged. Would you like me to paste the line above where you had to cover your ass with 'most'? |
| 02:01.51 | RypPn | I'd disagree in my limited experience of the spa3102 |
| 02:01.56 | [TK]D-Fender | J4k3, And again on the AP, it isn't saying you need good AP, I'm saying that many phone won't LET you use AP's to their proper capacity. the fault lies with the PHONE, not the AP. |
| 02:02.09 | RypPn | horrible echo and mobiles never converge |
| 02:02.09 | J4k3 | [TK]D-Fender: but not always, or even most. |
| 02:02.32 | J4k3 | [TK]D-Fender: so the statement, or calling the bot to spew the statement out, is entirely incorrect. |
| 02:02.46 | JT | RypPn: mobiles never converge? |
| 02:02.54 | J4k3 | [TK]D-Fender: and since it seems like daily you're getting checked on it, by more people than just myself, you'd move paste it |
| 02:02.58 | J4k3 | er past |
| 02:03.03 | Mackes | Yeah.. I have found some echo with the sipura 3102 as well.. sometimes, not always. |
| 02:03.17 | Mackes | Is there a more pro unit anyone recomends? |
| 02:03.21 | Zuchmir | is "fxsks=4" the correct setting for an FXO? |
| 02:03.24 | JT | does the SPA-3102 have any HWEC at all? |
| 02:03.29 | RypPn | thats right, it can take up to 30 seconds for a lanline pstn call to converge, mobiles just refused to work at all |
| 02:03.29 | JT | Zuchmir: yes |
| 02:03.33 | [TK]D-Fender | J4k3, No, the bot's statement and my samples do not all apply to every phone simultaneously. if there are 10 common bad things out there, it doesn't mean every bad product out there shares ALL 10 of them. Its like "one or the other" largely across the board. |
| 02:04.10 | Zuchmir | tk-fender: any ideas asterisk is not answering the call? |
| 02:04.33 | J4k3 | [TK]D-Fender: but the statement is pure fud, and you calling it regularly basically makes you the fat kid holding the butter knife. |
| 02:04.40 | [TK]D-Fender | Zuchmir, give me a moment to read up on your issue, I've been too focussed on this conversation. |
| 02:04.48 | J4k3 | keep spreading, [TK]D-Fender. |
| 02:05.03 | [TK]D-Fender | J4k3, Again taggin personal insults to the end of every statement. Grow up. |
| 02:05.20 | J4k3 | [TK]D-Fender: and yet you're still guilty of them, entirely. |
| 02:06.33 | [TK]D-Fender | J4k3, I have just validated the comment that the bot was triggered to say. Feel free to counter the clear explanations I have just given. I have explained it and given precise examples. it therefor not *FUD*, but the abbreviated summary without the 10 paragraph explanation. |
| 02:06.56 | [TK]D-Fender | J4k3, Should we make all bot comments 10 paragraphs long? |
| 02:07.18 | J4k3 | no, the bot shouldn't have the statement at all since its pretty much entirely inaccurate |
| 02:07.25 | J4k3 | it was accurate, say, 2 years ago. |
| 02:07.29 | RypPn | define lame |
| 02:07.29 | *** join/#asterisk johann8384 (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 02:07.33 | [TK]D-Fender | J4k3, I've backed up what I've said. Your turn now... |
| 02:07.38 | J4k3 | RypPn: [TK]D-Fender |
| 02:07.48 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 02:08.08 | *** join/#asterisk mdance (n=mdance@S01060050bafb9c77.ed.shawcable.net) |
| 02:08.35 | J4k3 | [TK]D-Fender: take a step back and come to the realization you're not the end-all-be-all expert to ip pbx related compatability, and shut the fuck up. |
| 02:08.48 | J4k3 | (I'm not either, so don't even try to go there..) |
| 02:09.01 | mdance | why would my incoming calls get disconnected as soon as they get connected? like i hear 1 second of my IVR then the call gets cut |
| 02:09.12 | UserReg_CL | helpme ... need example configuration extensions.conf and sip.conf |
| 02:09.25 | J4k3 | and if you want a more professional response than that, either gain the respect to get it or pay for it. |
| 02:09.37 | [TK]D-Fender | J4k3, I see.... you nothing of real value to say and just want a fight. And not even one where you can land a validated blow. You are remarkably immature and your temper betrays you so much more. |
| 02:09.56 | J4k3 | [TK]D-Fender: your bot is still incorrect. |
| 02:10.00 | BBHoss | UserReg_CL: what exactly are you trying to do, block people from dialing calls other than 09, or block incoming calls other than 09 |
| 02:10.05 | J4k3 | and you're still the fool that requests the data. |
| 02:10.39 | [TK]D-Fender | Zuchmir, could you please pastebin your zapata.conf and extensions.conf. From your channel description (which looked correct) you would appear to have a TDM400/800/2400 with 4 FXO ports worth of modules. Correct? |
| 02:11.23 | Mackes | UserReg - helpme ... need example configuration extensions.conf and sip.conf ? |
| 02:11.27 | [TK]D-Fender | J4k3, I have asked you repeatedly to backup your claim that it is incorrect. I have not discarded your opinion. You have simply refused to validate it. |
| 02:11.37 | Mackes | UserRege check out: |
| 02:11.41 | UserReg_CL | Mackes: yes |
| 02:11.44 | J4k3 | [TK]D-Fender: is english your second language? |
| 02:11.49 | [TK]D-Fender | J4k3, No, it isn't. |
| 02:12.01 | BBHoss | UserReg_CL: would you answer my question? |
| 02:12.07 | J4k3 | [TK]D-Fender: then you really have no excuse. |
| 02:12.13 | J4k3 | ~wifisip |
| 02:12.14 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 02:12.17 | J4k3 | read it. |
| 02:12.31 | J4k3 | while its obviously a dumbass's opinion |
| 02:12.48 | [TK]D-Fender | J4k3, and you say its inaccurate. I explained the basis of its statement. Now tell me where all the models are that have none of the issues I've listed very specifically for you. |
| 02:12.54 | J4k3 | this channel is *supposed* to be professional, and some bullshit fud like that shouldn't be acceptable. |
| 02:12.59 | *** join/#asterisk johann8384 (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 02:13.09 | Mackes | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 02:13.13 | UserReg_CL | BBHoss: need what for all users only call to number _09. and for other number dial block... |
| 02:13.37 | *** join/#asterisk johann8384 (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 02:13.37 | BBHoss | J4k3: even if he does emit fud, what you and he are doing right now is **highly** unprofessional |
| 02:13.48 | BBHoss | carry it to pms if you want to drag it out |
| 02:14.07 | J4k3 | BBHoss: don't care, this channel doesn't deserve it if his activity is encouraged. |
| 02:14.14 | [TK]D-Fender | PMS... yes, that would be an strangely appropriate analogy :) |
| 02:14.17 | BBHoss | its not |
| 02:14.17 | J4k3 | the professional wasn't here long before my first /join |
| 02:14.43 | J4k3 | BBHoss: then address him and I, not just me. |
| 02:14.55 | BBHoss | im not saying its you, or anyone, just everyone stfu and quit airing dirty laundry |
| 02:15.27 | [TK]D-Fender | BBHoss, for the record I have fully validated that "bot message" with a complete and coherant list of specific points. it is not FUD. They may only be "less applicable", and even that has not been suitably challenged. |
| 02:15.28 | BBHoss | J4k3: what do you really expect from a bunch of telecom geeks anyways :) |
| 02:15.48 | J4k3 | BBHoss: I'm trying to keep the channel from *discouraging* people from using IP-based voice. His bot basically say "you're fucked" and thats it. |
| 02:16.09 | J4k3 | BBHoss: If the task of #asterisk is to encourage people not to use asterisk, then it really doesn't belong on freenode. |
| 02:16.31 | J4k3 | BBHoss: telecom geeks my ass. This is straight up gamesnet level bullshit. |
| 02:16.36 | BBHoss | can anyone win here? |
| 02:16.40 | [TK]D-Fender | J4k3, I gave specifics. Counter them in kind and show me examples that might help me CHANGE my opinion. So far you are merely confrontational and not likely to get any positive response with that attitude. |
| 02:17.07 | BBHoss | I do wish the bot responses were worded a bit more professionally |
| 02:17.09 | [TK]D-Fender | BBHoss, Vegas has better odds. |
| 02:17.09 | J4k3 | [TK]D-Fender: your opinion isn't relevent |
| 02:17.30 | UserReg_CL | mmm |
| 02:17.39 | J4k3 | [TK]D-Fender: see, quite simply, if #asterisk is a professional channel, you venting your *opinions* via the channel bot is inappropriate. |
| 02:17.53 | [TK]D-Fender | J4k3, So now who thinks they are the "know-it-a-all" of voip? As I said, "hypocrite". |
| 02:18.03 | J4k3 | BBHoss: you'll find more professional responses on urbandictionary. |
| 02:18.03 | BBHoss | UserReg_CL: the dial string i gave you before ONLY allows 09 |
| 02:18.19 | J4k3 | [TK]D-Fender: I know enough about voip to know your statements are incorrect. |
| 02:18.21 | [TK]D-Fender | J4k3, and my opinions are shared by a great many other peoplle who've been in here and in telecom far longer than my 4+ years |
| 02:18.30 | Zuchmir | <PROTECTED> |
| 02:18.31 | [TK]D-Fender | J4k3, And have nothing to show us for it. |
| 02:18.41 | J4k3 | [TK]D-Fender: your bitch list concerning wifi voip phones are simply all results of you being too stupid to read the specs list, in about 80% of all cases. |
| 02:18.58 | [TK]D-Fender | J4k3, Your comments are unvalidated. THAT would be FUD. |
| 02:18.58 | J4k3 | I'll agree, I'd say about 20% of the wifi-capable-voip-gear I've touched has been garbage. |
| 02:18.59 | Zuchmir | [TK]D-Fender: TDM01B TDM400P |
| 02:19.04 | J4k3 | and unusable. |
| 02:19.12 | J4k3 | beyond that, it all pretty much met manufacturer's specifications |
| 02:19.18 | [TK]D-Fender | Zuchmir, For 5 FXo total across 2 cards? |
| 02:19.19 | BBHoss | just create a context and make my dialstring the ONLY thing in that context, then assign your phones to that context |
| 02:20.09 | J4k3 | this is the 00's... you can *always* find the owners manual before you buy the product. |
| 02:20.21 | [TK]D-Fender | Zuchmir, "immediate=no" <- should be NO. |
| 02:20.26 | J4k3 | if the owners manual says you can't do something you need to do, and you still buy it, its your bad for not researching it. |
| 02:21.01 | Zuchmir | [TK]D-Fender: 1 FXO on channel 4 |
| 02:21.24 | asdx | BBHoss: cool |
| 02:21.30 | asdx | BBHoss: do they have IAX? |
| 02:21.42 | asdx | BBHoss: VoipJet |
| 02:21.49 | Zuchmir | [TK]D-Fender: doesn't "immediate=no" turn off auto answer on 1st ring |
| 02:21.54 | BBHoss | i have a solution for you two: [TK]D-Fender, write out your list of 10 things that suck about wifi, J4k3: refute all of them |
| 02:22.07 | BBHoss | asdx: yeah |
| 02:22.08 | [TK]D-Fender | J4k3, and does just thrusting newbies at "any old product" that is amongst the 20% of "total crap" out there blindly doing them a favour? When it leaves a sour taste on them for * and voip? And their boss'? When it costs projects because they can't know EVERYTHING walking in the door? Isn't that the point of our giving advise which is what they ASKED for in the first place? |
| 02:22.17 | asdx | BBHoss: whoaaa!! thanks :D |
| 02:22.21 | BBHoss | asdx: https://www.voipjet.com/join.php |
| 02:22.23 | J4k3 | [TK]D-Fender: suggest things that work, then. |
| 02:22.45 | J4k3 | I have. my personal best wifisip experience has been with a wm6 capable phone. |
| 02:22.58 | [TK]D-Fender | Zuchmir, No, immediate=yes" is only used with FXS channels (analog phones) for use sas a "bat-phone" that automatically enters a starting point in the dialplan the moment it is picked up without waiting for DTMF. |
| 02:23.11 | [TK]D-Fender | Zuchmir, like an "emergency only" phone. |
| 02:23.25 | [TK]D-Fender | J4k3, funny... I DO <- |
| 02:23.52 | J4k3 | lacks a few features, but alas, its most functional. You had a guy here saying he's got 300 handsets or somesuch working across 30 locations without problems. You say it all sucks. |
| 02:23.57 | [TK]D-Fender | J4k3, Polycom, Linksys and Aastra all work quite well as SIP hardphones. |
| 02:24.02 | Zuchmir | [TK]D-Fender: ok "immediate=no" didn't help any |
| 02:24.04 | J4k3 | [TK]D-Fender: people aren't going to be nailed down |
| 02:24.16 | [TK]D-Fender | J4k3, and did you see my SPECIFIC response to him? |
| 02:24.20 | J4k3 | [TK]D-Fender: cordless phones have been a pretty standard consumer item for 20 years now |
| 02:24.24 | J4k3 | cellphones for 15 |
| 02:24.39 | J4k3 | [TK]D-Fender: yeah, you called the bot. |
| 02:24.49 | J4k3 | I have backscroll too. |
| 02:24.56 | asdx | BBHoss: thanks a lot :-) |
| 02:25.13 | BBHoss | asdx: np |
| 02:25.15 | [TK]D-Fender | J4k3, no, I'm talking to my specific response to HIM that he said AFTER |
| 02:26.03 | J4k3 | ok... lets see |
| 02:26.04 | [TK]D-Fender | <[TK]D-Fender> Mackes, It may be workable, and in some cases a bit more necessary to do that in your specific application, but in the greater scheme of things, typically a bad idea. <---- |
| 02:26.09 | BBHoss | asdx: you can even call some paraguay numbers, like Asuncion and the Proper |
| 02:26.09 | [TK]D-Fender | J4k3, there. |
| 02:26.11 | J4k3 | people were talking, you chimed in with your idiotic bot call |
| 02:26.18 | BBHoss | asdx: though they are like 10 cents a minute |
| 02:26.25 | J4k3 | and your personalized response was.... |
| 02:26.27 | [TK]D-Fender | J4k3, so in summary "TYPICALLY a bad idea", but it validates HIS usage |
| 02:26.28 | J4k3 | 19:45 < [TK]D-Fender> ^^^^^^^^^^^^^^^^^^^^ |
| 02:26.32 | J4k3 | unless thats a macro. |
| 02:26.51 | J4k3 | then you cried about a lot of shit that doesn't matter to people who *need* a wifi phone |
| 02:26.51 | [TK]D-Fender | J4k3, BELOW that... you know the part where you stopped actually listening... |
| 02:27.20 | J4k3 | didn't matter |
| 02:27.29 | [TK]D-Fender | J4k3, So see above and open up a bit... |
| 02:27.33 | J4k3 | you did what you did, and backslid. |
| 02:27.43 | J4k3 | take the shit off the bot, or at least stop calling it |
| 02:27.48 | Zuchmir | [TK]D-Fender: how can i see if * is listening to channel 4 |
| 02:27.50 | J4k3 | thats all I'm asking |
| 02:28.00 | J4k3 | I'm asking YOU to be professional, or quit trying to act like you know someething. |
| 02:28.00 | Mackes | wow guys. |
| 02:28.08 | Mackes | I left and came back |
| 02:28.13 | [TK]D-Fender | J4k3, I made a general warning statement (later validated), and am open to other viewpoints. Thats considerably more than you've shown us here lately. |
| 02:28.15 | Mackes | I'm sorry I startted this |
| 02:28.17 | BBHoss | still going... |
| 02:28.35 | [TK]D-Fender | Zuchmir, pastebin "zap show channels" and see if it matches what zaptel & zapata say |
| 02:28.39 | BBHoss | it happens occasionally, i think they should get married |
| 02:28.49 | Mackes | I think we can all agree that we are lucky that we all buy different products |
| 02:28.49 | BBHoss | :) |
| 02:28.56 | Mackes | It allows for choice |
| 02:29.01 | J4k3 | Mackes: yeah, I'm glad [TK]D-Fender buys what sucks. ;) |
| 02:29.03 | [TK]D-Fender | Mackes, BBHoss : Energizer Bunny will be suing us for patent infringement shortly ;) |
| 02:29.09 | Mackes | and without it, Asterisk would have died a long time ago |
| 02:29.12 | BBHoss | heh |
| 02:29.16 | [TK]D-Fender | J4k3, See, you just keep at it. |
| 02:29.24 | Zuchmir | [TK]D-Fender: No such command 'zap show' (type 'help' for help) |
| 02:29.26 | J4k3 | [TK]D-Fender: you still exist. |
| 02:29.44 | J4k3 | [TK]D-Fender: until you say "I am wrong and I'll fix it" I plan to stay in your ass. |
| 02:29.53 | [TK]D-Fender | J4k3, Ok, clearly its as was said long ago "some people you just can't reach". |
| 02:30.03 | Mackes | I have blocked J4k3 from my scroll list. |
| 02:30.10 | Mackes | Its neat, I dont even see it |
| 02:30.18 | Zuchmir | [TK]D-Fender: how do i load that module? |
| 02:30.34 | J4k3 | mackes thinks we care whats on his screen. glee |
| 02:30.37 | J4k3 | take your drama to efnet. |
| 02:30.48 | [TK]D-Fender | Zuchmir, Ok, its looking like zaptel isn't even loaded. lets try manually. try "module load chan_zap.so" |
| 02:31.41 | Zuchmir | [TK]D-Fender: Signalling must be specified before any channels are. |
| 02:31.51 | [TK]D-Fender | J4k3, I think you've missed the point. You ARE the drama. We're all kind of spectators. You are not really interacting or learning with us. You are a one sided conversation, which as far as drama goes is merely a "monologue". |
| 02:32.07 | [TK]D-Fender | Zuchmir, It is before your channel declaration |
| 02:32.27 | [TK]D-Fender | Zuchmir, line 18 sets fxs_ks, line 21 sets your channel |
| 02:32.44 | [TK]D-Fender | Zuchmir, that you have "zap" commands from CLI says Zaptel isn't loaded. |
| 02:32.50 | [TK]D-Fender | Zuchmir, So lets focus there. |
| 02:33.06 | J4k3 | [TK]D-Fender: how about this. I'll /part, and you can keep spreading lies and bullshit to this channel |
| 02:33.35 | J4k3 | because, quite simply, your existance here proves theres no professionals here. |
| 02:33.36 | [TK]D-Fender | J4k3, you haven't and reasonable discredited a single solid poitn I provided. |
| 02:33.39 | J4k3 | toodles. |
| 02:33.41 | *** part/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 02:33.50 | [TK]D-Fender | *sigh* |
| 02:33.59 | BBHoss | lets end it there |
| 02:34.00 | [TK]D-Fender | I really don't know how I can actually keep doing this... |
| 02:34.31 | [TK]D-Fender | BBHoss, Oh, he's apparently gone. I won't keep rambling on about it now. My points are made and I can stand by them. |
| 02:34.32 | [TK]D-Fender | :) |
| 02:34.35 | Zuchmir | [TK]D-Fender: "[Nov 16 13:31:01] ERROR[16146] chan_zap.c: Signalling must be specified before any channels are." |
| 02:34.54 | [TK]D-Fender | Zuchmir, hmm |
| 02:35.13 | JT | never seen j4k3 freak out so much |
| 02:35.16 | [TK]D-Fender | Zuchmir, pastebin your zapata again please (new). |
| 02:35.46 | Zuchmir | [TK]D-Fender: I didn't change it |
| 02:36.15 | [TK]D-Fender | JT, basically I completely "called him out" on his scathing remarks. I do naot take an assault on my character lightly. I've been wrong plenty of times and accept correction graciously. He however has nothing positive to contribut in exchange. |
| 02:36.54 | [TK]D-Fender | Zuchmir, fix the "immediate", stop *. Do "ztcfg -vvvv" and if that works fine, restart *. |
| 02:37.42 | fujin | j4k3 is always raging about Polycom. |
| 02:38.25 | *** join/#asterisk BeeBuu (n=chatzill@218.13.66.237) |
| 02:38.51 | *** join/#asterisk J_5 (n=J_5@cpe-71-72-210-44.cinci.res.rr.com) |
| 02:40.43 | [TK]D-Fender | fujin, I'm not so sure... just seems to be ME. You can see from the comment he made as soon as Polycom was even mentioned and I wasn't around... |
| 02:40.53 | fujin | heh yeah |
| 02:40.59 | fujin | well, you do kind of push them a little :) |
| 02:41.01 | fujin | but so does JT |
| 02:41.07 | fujin | he's just ragehappy man, I dunno |
| 02:41.28 | [TK]D-Fender | fujin, Oh, I make no claim to the contrary, and I am proud of what it is I promote. |
| 02:41.36 | fujin | indeed |
| 02:41.59 | [TK]D-Fender | fujin, but the fact is the opinion IS shared by what, 99% of their users here? |
| 02:42.11 | Zuchmir | [TK]D-Fender: http://pastebin.ca/776484 |
| 02:42.14 | fujin | generally, yeah |
| 02:42.16 | fujin | he's like the oddity |
| 02:42.19 | fujin | says that GS is the best |
| 02:42.24 | fujin | and polycom "aren't that great" |
| 02:42.40 | fujin | and the widely purported solution of "ditching your current phones, adn buying polycom" is crap |
| 02:42.41 | fujin | or something. |
| 02:42.42 | [TK]D-Fender | fujin, Frankly the only negative point you hear here is that the provisioning can be a PITA learning curve, which we accept whole-heartedly. |
| 02:42.46 | fujin | i had a big argument with thim the other day |
| 02:42.56 | Mackes | YESSSSS!!!! |
| 02:42.58 | [TK]D-Fender | fujin, I'm not the least bit surprised. |
| 02:43.06 | Mackes | fujin.. man you rock |
| 02:43.24 | fujin | eh, why? |
| 02:44.03 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:44.37 | BeeBuu | how can i get which channel is available in a group? |
| 02:45.03 | Mackes | because I was apart of a huge argument last night on this very list where I said that Polycoms are good but not the best, and they are hard to provision |
| 02:45.12 | Mackes | I didnt have alot of support |
| 02:45.35 | Mackes | We roll out Aastra Wired phones and Zultys Wifi |
| 02:45.43 | fujin | In my limited experience with them they'v ebeen very good, although, I prefer the XML-based configuraiton of the spa942. |
| 02:45.44 | Mackes | and they are both "ok" |
| 02:45.47 | [TK]D-Fender | Mackes, support should come from your reseller first and there are many of us here who specialise in them. |
| 02:46.04 | [TK]D-Fender | Mackes, And your opinion on thier usage might change once you get to know them better. |
| 02:46.04 | fujin | Indeed. Most resellers are shit though |
| 02:46.14 | [TK]D-Fender | Mackes, Which models do you have? |
| 02:46.15 | Mackes | no, I ment my statement wasnt well supported in this group |
| 02:46.20 | [TK]D-Fender | fujin, Agreed... |
| 02:46.23 | Mackes | not as far as documentation |
| 02:46.37 | Mackes | Ploycoms are a very good product |
| 02:46.42 | Mackes | they have lots of docs |
| 02:46.46 | [TK]D-Fender | Mackes, Actually... the admin guide really DOES say it all.. you do have to be in the right mindset to understand it however... |
| 02:46.46 | Mackes | and lots on the web |
| 02:46.52 | Mackes | yes, it dos |
| 02:46.55 | Mackes | does |
| 02:47.02 | [TK]D-Fender | Mackes, I will say that it isn't the most intuitive thing in the world, but it IS "complete" |
| 02:47.21 | [TK]D-Fender | Mackes, So is there something specific I can help you with? :) |
| 02:47.26 | Mackes | however, if you have ever worked with a Snom TFTP config file or and Aastra |
| 02:47.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:47.39 | obnauticus | [TK]D-Fender: Is this out of date? Asterisk SVN-branch-1.4-r71230 |
| 02:47.40 | Mackes | it makes Polycom XML look like a puzzle |
| 02:47.42 | Zuchmir | [TK]D-Fender: maybe my build of * isbroken (see http://bugs.digium.com/view.php?id=7496 ) |
| 02:48.02 | [TK]D-Fender | Mackes, I have, and I agree. 100% more complex.... and in exchange, 300% more powerful. I can live with that trade-off ;) |
| 02:48.03 | Mackes | anyways |
| 02:48.21 | [TK]D-Fender | obnauticus, I am not sure how to read exactly how old that is... |
| 02:48.22 | Mackes | maybe.. I dont think they are all that more powerful personally |
| 02:48.28 | obnauticus | uhh |
| 02:48.38 | [TK]D-Fender | Mackes, 3 specific things then, ok? |
| 02:48.41 | *** join/#asterisk saftsack (n=saftsack@pD9E0741C.dip.t-dialin.net) |
| 02:48.50 | Mackes | Snow I can create an OpenVPN tunnel right from the phone--- no Nat limitations ever |
| 02:48.54 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:48.57 | MrTelephone | hey guys |
| 02:49.02 | [TK]D-Fender | Mackes, "Join/Split" <- the most awesome dynamic call handling feaure I've seen on any phone, and only Polycom seems to have it,. |
| 02:49.34 | [TK]D-Fender | Mackes, Following that : You can hang up on a 3-way and leave the other parties CONENCTED (again rare to see anywhere). |
| 02:49.44 | Mackes | Snom has a live logging system (not just snmp) right in the web interface of the phone... every step of the sip connection |
| 02:49.49 | [TK]D-Fender | Mackes, Able to shuffle a nearly obscene number of calls. |
| 02:50.02 | Mackes | I can do that with Asterisk, I dont need it in the call? |
| 02:50.08 | Mackes | in the phone |
| 02:50.11 | Mackes | excuse me |
| 02:50.32 | [TK]D-Fender | Mackes, how would you do it in asterisk? |
| 02:51.19 | [TK]D-Fender | Mackes, and I wasn't saying all the user features were "300% more powerful", I was referring to the actual configurability of it. |
| 02:51.29 | [TK]D-Fender | Mackes, by way of provisioning |
| 02:51.38 | Mackes | ok.... everyone likes what they like |
| 02:51.44 | Mackes | thats cool |
| 02:51.47 | [TK]D-Fender | Mackes, true |
| 02:51.50 | Mackes | I respect it |
| 02:51.51 | MrTelephone | are you guys still arguing about equipment |
| 02:52.04 | Mackes | Polycoms are VERY well built |
| 02:52.04 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 02:52.08 | Mackes | I like that |
| 02:52.09 | MrTelephone | feels like i jumped into a delorion |
| 02:52.12 | MrTelephone | to yesterday |
| 02:52.13 | Mackes | Aastra are not |
| 02:52.16 | Mackes | Snom are not |
| 02:52.20 | BBHoss | MrTelephone: heh |
| 02:52.20 | Mackes | Cisco are ok |
| 02:52.30 | Mackes | Polycom are tanks |
| 02:52.49 | Mackes | hey, its what this room is about |
| 02:52.55 | Mackes | IP phones, asterisk |
| 02:52.57 | [TK]D-Fender | Mackes, To Aastra I conceded that their soft-keys are FRIGGEN AWESOME. State-based, capable of in-line DTMF (AWESOME!), the best Presence detail (polycom doesn't indicate RINGING for example, Aastra DOES), and so MANY of the, |
| 02:53.05 | [TK]D-Fender | Mackes, is that not a glowing review? |
| 02:53.07 | Zuchmir | [TK]D-Fender: how do i know if i have librpi compiled into *? |
| 02:53.16 | Mackes | Thank you. |
| 02:53.31 | Mackes | Polycoms feel like a great phone, not like a toy |
| 02:53.32 | [TK]D-Fender | Zuchmir, you don't even have ZAP, and from what you mentioned, not a PRI, so you shouldn;'t need it. |
| 02:53.54 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 02:54.01 | Mackes | No line monitoring in Polycom? |
| 02:54.04 | [TK]D-Fender | Mackes, I have gripes on every phone out there (sounding like coppice now ;)) |
| 02:54.22 | Zuchmir | [TK]D-Fender: any cles why zap can't load? |
| 02:54.24 | MrTelephone | you don't want to sound like him.. he's kind of an arsehole |
| 02:54.26 | Mackes | me too |
| 02:54.37 | [TK]D-Fender | Mackes, There is, but its either "Idle", "not conencted", or "on call". "ringing" looks like "on call". |
| 02:54.47 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 02:54.51 | [TK]D-Fender | Mackes, Which means you might not know to do a directed pickup for example. |
| 02:54.59 | Mackes | now, I know I asked this last night, but do any of you like the Cisco 7961 or 7941s? |
| 02:55.10 | Mackes | Ahhhh |
| 02:55.15 | [TK]D-Fender | MrTelephone, Yeah, he can be a bit ascerbic at times, but he is not unreasonable :) |
| 02:55.31 | [TK]D-Fender | MrTelephone, He can be quite cool and you CAN learn a lot from him without having to struggle. |
| 02:55.40 | [TK]D-Fender | MrTelephone, I can WORK with that. |
| 02:56.13 | Mackes | We use Aastra 57i 's with 2 Side cars, monitoring 120 lines. ring, in call, available... and they love them |
| 02:56.17 | [TK]D-Fender | Mackes, On cisco : great physical phone, but I think they only just (if even) added Presence support at all. |
| 02:56.18 | MrTelephone | he just reminds me of someone I know who doesn't like his job, hates his life, single, and drinks all the time |
| 02:56.30 | *** join/#asterisk rfxr (n=enigma@adsl-67-126-192-10.dsl.chic01.pacbell.net) |
| 02:56.34 | Mackes | Yeah, I dont think they do |
| 02:56.50 | [TK]D-Fender | Mackes, I heard you can't get presence on ALL of them however and have heard of some glorious crashed when they get flooded with HINT updates due to paging :) |
| 02:57.21 | [TK]D-Fender | Mackes, well the lack of presence, the reduced call handling capabilities, and increased cost over Polycom makes it harder to recommend. |
| 02:57.27 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 02:57.32 | [TK]D-Fender | Mackes, Its a mixed bag you know... |
| 02:58.05 | [TK]D-Fender | Mackes, TBH, Linksys has little "bad" about them. There is such a thing as being too "middle of the road" to really stand out, and thats their "flaw" ;) |
| 02:58.30 | Zuchmir | [TK]D-Fender: I added "load => chan_zap.so" in the modules.conf, this didn't either help |
| 02:58.36 | [TK]D-Fender | Mackes, IMO Linksys is more stable than Aastra, the call handling a little more natural, but they have their quirks too. |
| 02:58.46 | Mackes | ahahh.. |
| 02:58.47 | Mackes | Ok |
| 02:58.48 | [TK]D-Fender | Zuchmir, what did you get from "ztcfg -vvvv"? |
| 02:58.56 | fujin | I haven't noticed any quirkiness, but they linksys are very reasonable for the cost |
| 02:59.10 | [TK]D-Fender | Mackes, For typicaly use I'd probably prefer a Linksys over an Aastra. |
| 02:59.16 | Mackes | Really? |
| 02:59.26 | Mackes | Do they do TFTP provisioning? |
| 02:59.46 | fujin | yep. |
| 02:59.50 | Mackes | Hmmmmm |
| 02:59.51 | fujin | I provision all 50 on mine with tftp |
| 02:59.53 | fujin | firmware/config. |
| 03:00.06 | [TK]D-Fender | fujin, sample : they act like an ATA with an analog phone bolted on top. You get reorder tomes when a call terminates so you can't on-hook dial if a call terminate without literally going on-hook. |
| 03:00.08 | fujin | you can also do 4 configs per device, like a global one, then a user one, then like a group one or whatever. |
| 03:00.09 | Mackes | That is one phone I have not had the plesure of using. |
| 03:00.09 | MrTelephone | im trying out http |
| 03:00.12 | MrTelephone | http provisioning |
| 03:00.22 | Zuchmir | [TK]D-Fender: http://pastebin.ca/776507 |
| 03:00.27 | fujin | Poe, ethernet out to PC, reasonable |
| 03:00.37 | fujin | [TK]D-Fender: ah, indeed. I've not seen those. only had experience with the 9x2 series. |
| 03:00.57 | MrTelephone | i love that vlan id on the phones |
| 03:01.17 | fujin | yeah, vlans are handy here. |
| 03:01.21 | [TK]D-Fender | fujin, Oh, they are, which is why in certaim places you'll see me recommend them over Polycom. But in North America.... well you know what Polycom pricing is like here... seriously, would YOU pick somthing else if not for a highly specialized purpose? |
| 03:01.23 | fujin | have a phone vlan, anc CDP picks it up on bootup |
| 03:01.34 | Mackes | Which Linksys? |
| 03:01.42 | MrTelephone | how do you get vlan from dhcp if you need the vlan to get dhcp in the first place? |
| 03:01.43 | fujin | 942, one 962 for the receptionist |
| 03:01.48 | fujin | MrTelephone: it's CDP |
| 03:01.52 | [TK]D-Fender | Mackes, They're all about the same. its a question of PoE, and # of lines. |
| 03:02.05 | MrTelephone | what broadcasts cdp vlan? the cisco switch? |
| 03:02.08 | Mackes | tell me about the Polycom 320 |
| 03:02.09 | J_5 | what ports do I need to open up to allow someone to connect to my asterisk server from out of my network? |
| 03:02.13 | fujin | MrTelephone: the switch is configured as a 'voice port', the Linksys can speak CDP, and can work out which vlan to hop on |
| 03:02.16 | fujin | yeah, the cisco switch. |
| 03:02.17 | [TK]D-Fender | Zuchmir, ok, I'm not sure why zaptel isn't loading... |
| 03:02.24 | fujin | we run 3560 POE here. |
| 03:02.30 | [TK]D-Fender | J_5, what protocol? |
| 03:02.32 | MrTelephone | I have a sw224p switch and there is no option for "Voice" |
| 03:02.36 | J_5 | SIP |
| 03:02.37 | fujin | ;( |
| 03:02.38 | [TK]D-Fender | Mackes, What would you like to know? |
| 03:02.51 | MrTelephone | I just setup vlans 20 and 30 making 20 pcs and 30 for phones |
| 03:03.38 | MrTelephone | can you put a hostname in the http server and have multiple A records for it? |
| 03:03.50 | fujin | that'd be dome |
| 03:03.54 | fujin | dns SRV records ar the correct way to do that |
| 03:04.05 | MrTelephone | not for sip, for config download |
| 03:04.08 | fujin | oh |
| 03:04.13 | Mackes | well, We have deployed alot of Aastra 480s to our car washs |
| 03:04.19 | fujin | MrTelephone: for wha tphone? |
| 03:04.22 | Zuchmir | [TK]D-Fender: http://pastebin.ca/776508 |
| 03:04.22 | MrTelephone | polycom 501 |
| 03:04.23 | Mackes | but they are distroying them |
| 03:04.27 | fujin | sorry don't know ;) |
| 03:04.36 | Zuchmir | [TK]D-Fender: ok, thanks for your time! |
| 03:04.44 | MrTelephone | i programt he phones through the web so if it can't find a config server it uses the web settings |
| 03:04.53 | MrTelephone | cisco junkie.. |
| 03:04.57 | [TK]D-Fender | Zuchmir, we'll get this figured out... |
| 03:05.01 | Mackes | so I would like to deploy a basic phone that is hardcore |
| 03:05.27 | [TK]D-Fender | Mackes, if its a "high phone rape" area I'd recommend using an ATA + analog phone. |
| 03:05.46 | [TK]D-Fender | Mackes, back in the day I put an IP 301 in our reception are for exactly that reason. it was CHEAPER :) |
| 03:05.52 | Mackes | Na, to many parts |
| 03:06.17 | JT | eh |
| 03:06.22 | Mackes | are the 301's strong... could they be dropped of a counter |
| 03:06.25 | JT | you can get REALLY solid analogue handsets |
| 03:06.41 | MrTelephone | you could drop them from the new york trade center |
| 03:06.49 | [TK]D-Fender | Mackes, well if you want minimal parts, that'll mean PoE, and as far as inexpensive phones go with PoE, the only cheaper option is a GXP and well... the 320 is only like 10$ more and I won't go on on that :) |
| 03:06.49 | Mackes | hmmmm |
| 03:06.54 | Mackes | not a good example |
| 03:07.09 | JT | Cisco 12SP+ series is REALLY solid, unlike the current stuff |
| 03:07.16 | [TK]D-Fender | Mackes, the 320/330 killed off the 301, and pretty mcuh do the same for the 430 and seriously threaten the 501. |
| 03:07.19 | JT | but H.323 support in asterisk is poor |
| 03:07.21 | Qwell | JT: oh god yes |
| 03:07.32 | Mackes | no I mean using ATA's would make my installs more complex. |
| 03:07.36 | Qwell | I ran over mine with a steamroller, and the metal shattered |
| 03:07.44 | Qwell | JT: they run Skinny, and they're awesome |
| 03:07.53 | JT | Qwell: some run H.323 |
| 03:08.02 | [TK]D-Fender | Mackes, If you can't live with the extra wiring, so be it. next up IMO would be the IP 320 |
| 03:08.04 | JT | Qwell: did it work after the steamroller? |
| 03:08.05 | Qwell | you can't flash them to skinny? |
| 03:08.07 | fujin | those 330's look nice. |
| 03:08.13 | JT | Qwell: don't think i'd want to |
| 03:08.17 | JT | properietry protocol |
| 03:08.25 | JT | also limited support |
| 03:08.25 | Qwell | skinny rocks |
| 03:08.34 | fujin | Isn't skinny the CCM proto? |
| 03:08.35 | Qwell | support...on the 12sp? |
| 03:08.41 | Qwell | it's been EOL for years, heh |
| 03:08.42 | [TK]D-Fender | fujin, yes |
| 03:08.52 | Qwell | hell, mine isn't even Cisco branded |
| 03:09.02 | Qwell | erm, maybe that's the 30 VIP |
| 03:09.14 | Qwell | yeah, Selsius Phone |
| 03:09.24 | Qwell | (hence the SP) |
| 03:10.14 | JT | no |
| 03:10.22 | JT | support from stuff that isn't CCM |
| 03:10.29 | JT | H.323 is widely deployed |
| 03:10.29 | Qwell | like asterisk? |
| 03:10.32 | JT | yes |
| 03:10.37 | Qwell | mine works with asterisk |
| 03:11.21 | JT | sure |
| 03:11.28 | JT | it's still rudimentary support |
| 03:11.37 | Qwell | still better than h323 |
| 03:11.40 | obnauticus | Ya, JT, what does CCM do anyway. |
| 03:11.49 | MrTelephone | it routes calls |
| 03:11.53 | JT | H.323 is pretty good |
| 03:12.01 | JT | obnauticus: what asterisk does, sort of |
| 03:12.01 | BBHoss | i didnt know cisco made phones as ugly as that :) |
| 03:12.01 | obnauticus | I was told it was better but anyone that is saying anything but switches from Cisco is good is probably biase. |
| 03:12.02 | obnauticus | d |
| 03:12.09 | Qwell | BBHoss: heh, it's ugly as hell |
| 03:12.10 | MrTelephone | I like sip |
| 03:12.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:12.25 | MrTelephone | why is h323 better than sip? |
| 03:12.26 | BBHoss | im sure its built tough though |
| 03:12.30 | Qwell | MrTelephone: it isn't |
| 03:12.36 | JT | MrTelephone: it's a more efficient protocol |
| 03:12.46 | JT | SIP just gets more complex all the time |
| 03:12.51 | JT | and there are so many implementations |
| 03:12.54 | JT | it's ridiculous |
| 03:13.03 | MrTelephone | i agree with that |
| 03:13.04 | obnauticus | IAX isn't as widely supported though |
| 03:13.12 | BBHoss | obnauticus: did you try another provider? |
| 03:13.26 | JT | using http like strings is retarded |
| 03:13.34 | JT | there's a reason carriers use H.323 and not SIP |
| 03:13.34 | obnauticus | BBHoss, no but it WAS working on and off for me, but now it just doesn't work. |
| 03:14.04 | Qwell | JT: because their stuff already talks Q.931 |
| 03:14.17 | Qwell | that's all h323 really is |
| 03:14.23 | JT | and SIP not talking Q.931 is a big NEGATIVE point |
| 03:14.27 | JT | which is great |
| 03:14.43 | [TK]D-Fender | Mackes, So, anything else you'd like to know about the IP 320? |
| 03:15.05 | MrTelephone | mgcp or h323? |
| 03:15.11 | obnauticus | BBHoss, you know what I should just try rebootign the boxen |
| 03:15.11 | obnauticus | lol. |
| 03:15.14 | BBHoss | network taking a dump |
| 03:15.28 | [TK]D-Fender | obnauticus, NAT issues? |
| 03:15.33 | obnauticus | No. |
| 03:15.40 | obnauticus | Otherwise it would say it didn't respond to the critical packet |
| 03:15.43 | *** join/#asterisk Utah_Dav1 (n=David@c-67-172-225-211.hsd1.ut.comcast.net) |
| 03:15.44 | obnauticus | or give me an error for that matter. |
| 03:15.47 | JT | SIP cause codes suck |
| 03:15.56 | [TK]D-Fender | obnauticus, Actually, thats a great way to GET that problem... |
| 03:16.29 | obnauticus | Well [TK]D-Fender, if it were an configuration issue of the NAT, then it wouldn't have worked in the first place.... |
| 03:16.32 | obnauticus | at all... |
| 03:16.36 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 03:16.59 | [TK]D-Fender | obnauticus, actually, if a UDP tunnel closes down on you it can happen after a successful call... |
| 03:17.02 | obnauticus | But it could be the fact that PfSense is being a little poopier than they describe it. |
| 03:17.04 | MrTelephone | single ring on a handset form an ata device is usually a sign of registration isn't it? |
| 03:17.12 | [TK]D-Fender | obnauticus, please describe the complete call path. |
| 03:17.16 | BBHoss | sorry |
| 03:17.19 | obnauticus | Okay. |
| 03:17.19 | BBHoss | obnauticus: did you try another provider? |
| 03:17.33 | obnauticus | No, I didn't I will when I pay for one which is like tomorrow. |
| 03:17.39 | [TK]D-Fender | obnauticus, Oh wait.. I think we DID look at this already last night or so, didn't we? |
| 03:17.42 | obnauticus | Until then I want to try and see if i can fix this : |
| 03:17.46 | obnauticus | ya. |
| 03:17.59 | obnauticus | I haven't tried rebooting the actual box |
| 03:18.01 | [TK]D-Fender | obnauticus, for sanity's sake can you pastebin your sip.conf |
| 03:18.08 | obnauticus | k |
| 03:18.11 | obnauticus | hold on |
| 03:20.17 | obnauticus | http://pastebin.ca/776522 |
| 03:20.48 | [TK]D-Fender | obnauticus, bindaddr=10.0.0.109 ; IP address to bind to (0.0.0.0 binds to all) <- i generally advise 0.0.0.0. |
| 03:20.58 | obnauticus | K. |
| 03:21.10 | [TK]D-Fender | obnauticus, ;pedantic=yes ; Enable checking of tags in headers, <- this is typically a very good thing to UNcomment |
| 03:21.25 | obnauticus | k |
| 03:22.07 | [TK]D-Fender | obnauticus, localnet=10.0.0.0/16 ; All RFC 1918 addresses are local networks <- I know you were told to /16 but I've always used full dotted... your call if this is accurate... |
| 03:22.30 | obnauticus | Lemme make sure :) |
| 03:22.48 | *** join/#asterisk johann8384-home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 03:22.57 | obnauticus | Yup, subnet on my LAN if is 10.0.0.1/16 |
| 03:23.09 | [TK]D-Fender | obnauticus, do make sure the mask is right. Also which is your provider [section]? |
| 03:23.24 | [TK]D-Fender | AH! |
| 03:23.30 | obnauticus | AH? |
| 03:23.32 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 03:23.52 | [TK]D-Fender | obnauticus, for your ITSP [devicesections] please set "nat=no" <---- and "canreinvite=no" |
| 03:23.59 | Zuchmir | [TK]D-Fender: ok, found it, "signalling" was misspelled in the book!!!! |
| 03:24.12 | [TK]D-Fender | Zuchmir, EEK! I missed it.... |
| 03:24.24 | Zuchmir | [TK]D-Fender: ZTDUMMY/1 (source: HRtimer) 1 UNCONFIGUR 0 0 0 |
| 03:24.25 | obnauticus | so [TK]D-Fender, canreinvite=no if nat=no |
| 03:24.27 | [TK]D-Fender | Zuchmir, missing an l! |
| 03:24.30 | obnauticus | and canreinvite=yes if nat=yes |
| 03:24.42 | Zuchmir | [TK]D-Fender: I had to grep through the src code to find that typo!! |
| 03:24.56 | [TK]D-Fender | obnauticus, No, canreinvite=no is a good thing regardless. |
| 03:24.56 | obnauticus | What about in the [508] section, [TK]D-Fender? |
| 03:25.19 | [TK]D-Fender | obnauticus, but your ITSP shouls definately be "nat=no", because you MUST use the IP they provide in their SIP header. |
| 03:25.45 | obnauticus | I also have nat=yes in the 508 section though. |
| 03:25.57 | [TK]D-Fender | obnauticus, what is 508? |
| 03:26.03 | obnauticus | [508] ;IPKall |
| 03:26.10 | obnauticus | Line 664 |
| 03:26.22 | [TK]D-Fender | obnauticus, oh, definitely fix that |
| 03:26.22 | obnauticus | Or acutally 670... |
| 03:26.26 | obnauticus | K. |
| 03:26.44 | [TK]D-Fender | obnauticus, you should only have nat=yes for PHONES you have behind other remote nats |
| 03:26.58 | obnauticus | Still is not working. |
| 03:27.02 | [TK]D-Fender | obnauticus, And NOTHING should be allowed to reinvite |
| 03:27.15 | Zuchmir | ok, thanks for that!!!!, B"H now it works!!!!!!! |
| 03:27.16 | [TK]D-Fender | obnauticus, pastebin V2 please. |
| 03:27.21 | obnauticus | affermative. |
| 03:27.30 | [TK]D-Fender | Zuchmir, very glad to hear... |
| 03:27.43 | obnauticus | Thanks btw [TK]D-Fender. |
| 03:29.02 | [TK]D-Fender | obnauticus, np..... so... ready yet? :) |
| 03:29.08 | obnauticus | http://pastebin.ca/776539 |
| 03:30.19 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 03:30.31 | [TK]D-Fender | obnauticus, you oopsed : nat=no ; Global NAT settings (Affects all peers and users) <-- that SHOULD have been left as "yes" the individual accoutns outside your NAT should be left as NO except for phones that are themselves behind ANOTHER NAT |
| 03:30.46 | obnauticus | oh ok |
| 03:30.46 | obnauticus | lol. |
| 03:30.58 | obnauticus | Changed global nat=yes |
| 03:31.05 | [TK]D-Fender | obnauticus, thats jsut about the only one I *didn't* tell you to change :) |
| 03:31.26 | [TK]D-Fender | obnauticus, Oh, and for sanity's sake again... permanently remove all that commented out junk, ok? :) |
| 03:31.26 | obnauticus | Fail. |
| 03:31.42 | obnauticus | LOL |
| 03:31.43 | obnauticus | ya i know. |
| 03:31.54 | [TK]D-Fender | obnauticus, and repastebin.... I get dizzy when I spend too much parsing stuff I don't need to care about |
| 03:31.56 | obnauticus | I hate that in apache too, and just about everything else that has text based configuration. |
| 03:35.29 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 03:35.30 | obnauticus | lol 3.03kb to 1.01 kb |
| 03:35.43 | [TK]D-Fender | Diet Crap! Half the crap! |
| 03:36.23 | [TK]D-Fender | obnauticus, if it makes you feel any better its probably taking up the same number of clusters on your drive regardless ;) |
| 03:36.38 | obnauticus | I get dizzy too :) |
| 03:36.39 | obnauticus | http://pastebin.ca/776557 |
| 03:36.56 | [TK]D-Fender | dear God... so much better.... |
| 03:37.32 | [TK]D-Fender | obnauticus, now add canreinvite=no to every section |
| 03:37.47 | obnauticus | Affermative. |
| 03:37.52 | [TK]D-Fender | obnauticus, and set codecs in your [general] as well |
| 03:38.23 | obnauticus | disallow=all \r\n allow=ulaw \r\n allow=g711 |
| 03:38.24 | obnauticus | is that good? |
| 03:38.33 | [TK]D-Fender | obnauticus, 3rd = bad |
| 03:38.39 | obnauticus | probably. |
| 03:38.40 | obnauticus | lol |
| 03:38.51 | [TK]D-Fender | obnauticus, ulaw = g711u, alaw=g711a |
| 03:38.59 | obnauticus | oh i see |
| 03:39.04 | obnauticus | Why can't they just say |
| 03:39.06 | obnauticus | g711u |
| 03:39.07 | [TK]D-Fender | obnauticus, shh! |
| 03:39.23 | obnauticus | k |
| 03:39.25 | obnauticus | it's done :/ |
| 03:39.39 | [TK]D-Fender | obnauticus, it is technically correct. they unabbreviate the U/A |
| 03:39.49 | obnauticus | mmmk |
| 03:39.53 | [TK]D-Fender | obnauticus, final version and a drum-roll please... |
| 03:40.25 | obnauticus | I have 5$ epic fail. |
| 03:40.48 | obnauticus | http://pastebin.ca/776570 <-- final |
| 03:41.21 | obnauticus | Epic fail. |
| 03:41.40 | [TK]D-Fender | obnauticus, Ok, but now your * setup looks 100% clean. |
| 03:41.45 | obnauticus | <PROTECTED> |
| 03:41.50 | [TK]D-Fender | obnauticus, now you can focus on pfsense issues |
| 03:41.54 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 03:41.58 | obnauticus | Or.. |
| 03:42.03 | obnauticus | ipkall. |
| 03:42.07 | BBHoss | is you * behind the firewall |
| 03:42.10 | obnauticus | Word. |
| 03:42.15 | [TK]D-Fender | obnauticus, what are you forwarding to *? |
| 03:42.27 | [TK]D-Fender | BBHoss, I've got this one :) |
| 03:43.05 | obnauticus | 4569 (IAX) 5060 (SIP RTP) 10000-20000 (SIP shits) |
| 03:43.26 | [TK]D-Fender | obnauticus, ummm.. what protocols for those? |
| 03:43.36 | obnauticus | all TCP/UDP |
| 03:43.44 | obnauticus | I got lazy so i just used TCP/UDP |
| 03:43.44 | obnauticus | lol. |
| 03:43.49 | [TK]D-Fender | obnauticus, Ok, that SHOULD do it then. |
| 03:43.56 | obnauticus | ya...SHOULD. |
| 03:43.58 | MrTelephone | chan_sip.c pkt_retrans errors.. that has nothing to do with rtp right? |
| 03:44.06 | [TK]D-Fender | obnauticus, Could be your fps is being assy or doing some sort of SIP transform. |
| 03:44.27 | obnauticus | Ya. |
| 03:44.31 | obnauticus | I could try rebooting the * box though. |
| 03:44.36 | [TK]D-Fender | obnauticus, try to remove that from the equation |
| 03:44.38 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 03:44.41 | obnauticus | Nuts. |
| 03:44.54 | [TK]D-Fender | obnauticus, IIRC you have NO iptables rules filtering there, right? |
| 03:45.00 | obnauticus | Nope. |
| 03:45.02 | obnauticus | not up here |
| 03:45.13 | obnauticus | I got snort running, but it's generating no alerts. |
| 03:45.19 | obnauticus | I'll look in firewall logs. |
| 03:45.22 | obnauticus | see if anything's happening |
| 03:45.38 | MrTelephone | just give up computers |
| 03:45.38 | obnauticus | Some stuff is comming n. |
| 03:45.41 | MrTelephone | give it up |
| 03:46.04 | MrTelephone | i wish I was the one that didn't know anything.. sitting at a desk with barely a clue on how to turn it on |
| 03:46.26 | obnauticus | * Added *!*n=test@S010600195b946fa3.ls.shawcable.net to ignore list |
| 03:46.31 | BBHoss | take a prozac and calm down |
| 03:46.41 | [TK]D-Fender | obnauticus, ? |
| 03:46.48 | obnauticus | i see stuff |
| 03:46.49 | MrTelephone | hey thats me |
| 03:47.02 | obnauticus | sorry... |
| 03:47.05 | obnauticus | lol hold on |
| 03:47.07 | obnauticus | i was looking |
| 03:47.14 | obnauticus | i was going to be more specific...`stuff' was kinda a retarded thing to say |
| 03:47.15 | obnauticus | lol. |
| 03:47.29 | BBHoss | obnauticus: in the firewall rules, it might be helpful to turn on logging for each rule, too see what passes |
| 03:47.35 | MrTelephone | * Added obnauticus to ignore list |
| 03:47.58 | [TK]D-Fender | obnauticus, or remove ALL filters and see if ti works and add back till it breaks :) |
| 03:48.10 | obnauticus | LOL |
| 03:48.28 | obnauticus | k brb in 3 months. |
| 03:48.28 | obnauticus | lol. |
| 03:48.45 | MrTelephone | thank god |
| 03:48.59 | BBHoss | also a wierd quirk with pfsense is that if you add a rule the wrong way in NAT Forwarding, and then change it in NAT Forwarding, it doesnt change it in Firewall Rules |
| 03:49.25 | obnauticus | I figured that out. |
| 03:49.32 | BBHoss | heh fun eh? |
| 03:49.37 | obnauticus | If I mess up in NAT and have it automatically add a firewall rule... |
| 03:49.41 | obnauticus | i just go into rules |
| 03:49.43 | obnauticus | and delete it |
| 03:49.45 | obnauticus | then nat and delete it |
| 03:50.01 | obnauticus | They already have logging enabled actually |
| 03:50.16 | BBHoss | are the packets getting passed? |
| 03:50.21 | obnauticus | Lemme see |
| 03:50.38 | obnauticus | Ya, but the ones im looking at right now |
| 03:50.46 | obnauticus | Are from the 28th? |
| 03:50.47 | obnauticus | wtf |
| 03:50.56 | BBHoss | just resolve the dns of ipkall, then search the logs for their IP address |
| 03:51.05 | obnauticus | ya that's what I tried to do |
| 03:51.15 | obnauticus | hold on i'll restart asterisk to see if it generates any logs. |
| 03:52.26 | BBHoss | FAIL:http://bp2.blogger.com/_TVfEjAyW4Fk/RzumZQusGkI/AAAAAAAAENE/ZmHKqT0jdoA/s1600-h/1.jpg |
| 03:52.37 | obnauticus | LOL |
| 03:53.06 | MrTelephone | 33 years to pay off 5000$ at 21% with minimum payments of 100 |
| 03:53.13 | MrTelephone | did you see that commercial? |
| 03:53.22 | obnauticus | It's erm |
| 03:53.24 | obnauticus | not generating any logs |
| 03:53.33 | BBHoss | try clearing them |
| 03:53.36 | obnauticus | i did |
| 03:53.36 | obnauticus | lol. |
| 03:53.42 | obnauticus | The latest logs were on the 28th |
| 03:54.40 | obnauticus | Hold on :| |
| 03:55.30 | obnauticus | I'm going to log anything going to voiper.ipkall.com (66.54.140.46) |
| 03:57.45 | obnauticus | k i got it. |
| 03:59.12 | obnauticus | http://pastebin.ca/776586 |
| 03:59.18 | BBHoss | hey can u use rafb.net |
| 03:59.21 | obnauticus | k |
| 03:59.44 | obnauticus | http://rafb.net/p/ZNUIzd81.html |
| 03:59.48 | *** part/#asterisk Utah_Dave (n=David@c-67-172-225-211.hsd1.ut.comcast.net) |
| 03:59.49 | obnauticus | As you can see, nothing is even trying to be sent out |
| 04:00.25 | BBHoss | odd |
| 04:00.51 | obnauticus | 5. 742221 rule 1026/0(match): pass in on rl0: 66.54.140.46.5060 > 10.0.0.109.5060: SIP, length: 893 |
| 04:00.53 | obnauticus | That works. |
| 04:00.59 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:01.01 | obnauticus | That's when I get ringed. |
| 04:01.13 | BBHoss | but no audio out :( |
| 04:01.16 | obnauticus | Nope. |
| 04:01.21 | obnauticus | My * box isn't even TRYING to send out |
| 04:01.25 | obnauticus | otherwise I would see it :| |
| 04:01.42 | obnauticus | im gonna tshark on my * box. |
| 04:02.09 | BBHoss | you want to try my server just for shits sake? |
| 04:02.13 | obnauticus | My Outbound ones should be exactly the same length (172) right? |
| 04:02.16 | obnauticus | Ya, sure actually. |
| 04:02.22 | BBHoss | hang on |
| 04:02.37 | BBHoss | SIP correct? |
| 04:06.11 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 04:09.57 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:24.50 | *** join/#asterisk mihinomenest (i=IBi3@66.255.220.17) |
| 04:29.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:29.35 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 04:34.46 | *** join/#asterisk mtaht4 (n=m@158-109-62-200.enitel.net.ni) |
| 04:34.56 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 04:37.51 | mosty | does the CHANNEL(audionativeformat) function show the codec being used for the current channel? or is it the highest priority codec? |
| 04:38.02 | JayTee52 | my boss has finally agreed to purchase a new server for Asterisk. We're getting a Dell PowerEdge with 4GB of RAM and 2 large 15K RPM SATA drives that we'll mirror. If I install Asterisk 1.4 on it, match the hostname and IP address of the original Asterisk server which runs version 1.2 can I just copy all my configs for my dialplan and have it work or will I need to do alot of tweaking? |
| 04:38.42 | [TK]D-Fender | JayTee52, depends how 1.2 compliant your configs were |
| 04:39.02 | [TK]D-Fender | JayTee52, And last I checked there was no such thing as a 15K RPM SATA drive.... |
| 04:39.19 | JayTee52 | well, we used Asterisk TFOT 1st edition as a reference for our dialplan. |
| 04:39.30 | JayTee52 | SAS drives? |
| 04:39.48 | [TK]D-Fender | JayTee52, Doesn't say much. They use some deprecated stuff in there. There are almost guaranteed issues to arise. |
| 04:40.38 | *** part/#asterisk mtaht4 (n=m@158-109-62-200.enitel.net.ni) |
| 04:41.03 | JayTee52 | is there a compiled list of what's been deprecated available or is it scattered all over the web in bits and pieces? |
| 04:41.47 | mosty | JayTee52, you should get warnings when you load the dialplan |
| 04:43.05 | JayTee52 | yeah, I'm getting better at pulling info from the console to help me debug issues. Right now my system loads without any errors. |
| 04:43.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-26e1e0ed0acbf259) |
| 04:45.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a8d08752a9fc1dc3) |
| 04:45.12 | JayTee52 | I'd still like to find documentation that covers what's changed between the two versions |
| 04:45.13 | mosty | that won't catch things where an option has changed behaviour, without being deprecated |
| 04:47.20 | JayTee52 | I was looking forward to buying the O'Reilly Asterisk Cookbook which was supposed to come out on August 1st but so far it's been a no show and O'Reilly's site contains no info on when if ever it will finally ship. |
| 04:48.31 | [TK]D-Fender | JayTee52, read the Rev 2 of the book, and the entire Docs folder |
| 04:48.57 | Zuchmir | how do i let it ring twice so that i can get the callerID? |
| 04:49.25 | [TK]D-Fender | Zuchmir, "usercallerid=yes" , "callerid=asreceived" |
| 04:49.37 | [TK]D-Fender | Zuchmir, add to zapata.conf |
| 04:49.54 | [TK]D-Fender | zuch before your channel declaration. You will have to restart * to take effect |
| 04:50.05 | JayTee52 | [TK]D-Fender, if my X-Chat didn't have autocomplete your screen name would be really annoying to chat with :-) Thanks. I have the 2nd edition in PDF format and I'll definitely check out the Docs folder when I install 1.4 on the new server. |
| 04:50.43 | [TK]D-Fender | JayTee52, copy & paste FTW. And complaining about my nick is pretty low for the help I give :) |
| 04:51.25 | Zuchmir | is "dialplan reload" enough |
| 04:51.58 | JayTee52 | [TK]D-Fender, ok. that was low. I apologize for kidding you. It's actually a cool looking nick cuz it reminds me of my Strat |
| 04:52.00 | [TK]D-Fender | Zuchmir, No |
| 04:52.08 | [TK]D-Fender | Zuchmir, You need to reload zaptel |
| 04:52.23 | [TK]D-Fender | Zuchmir, just do "restart now" if you don't have calls in progress |
| 04:52.46 | [TK]D-Fender | JayTee52, I play Dean & Ibanez personally.... |
| 04:53.39 | JayTee52 | I have 3 Ibanez guitars, 1 acoustic/electric, 1 dreadnought acoustic and 1 classical with a standard neck instead of wide neck. Love them the best. |
| 04:53.57 | [TK]D-Fender | JayTee52, the last sounds cool... up my alley.. |
| 04:54.04 | Zuchmir | ok, thanks! |
| 04:54.28 | [TK]D-Fender | JayTee52, I'm looking for a slim cutaway accoustic & classical next. Then eventually a neck-through electric |
| 04:55.10 | JayTee52 | This is the acoustic electric I've got http://www.ibanez.com/acoustic/guitar.aspx?m=AEF30EVV |
| 04:55.15 | [TK]D-Fender | JayTee52, I dislike the bulky dreadnought look personally and would prefer something a bit more agile with higer fre access |
| 04:55.16 | JayTee52 | I named it Lily |
| 04:55.37 | JayTee52 | the dreadnought I have has some really nice bottom to it. |
| 04:55.48 | [TK]D-Fender | JayTee52, I think I've tried that one at my local shop..... |
| 04:55.54 | MrTelephone | is there a way to goto a context of asterisk and make it forget where it came froM? |
| 04:56.06 | [TK]D-Fender | JayTee52, 1/2 dreadnought body depth? |
| 04:56.11 | MrTelephone | becauase In my dialplan there is sometimes 3 or 4 hops and the dialplan doesn't work properly |
| 04:56.19 | [TK]D-Fender | MrTelephone, look at ResetCDR |
| 04:56.21 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 04:56.38 | JayTee52 | this one has great range though for an acoustic electric and the tuners are awesome. It'll stay in tune for days even with some temperature shifts. |
| 04:56.46 | MrTelephone | if i phone office 1 from office 1 the extensions won't work |
| 04:56.47 | [TK]D-Fender | MrTelephone, if CDR is what you were referring to. Otherwise you're making no sense at all... |
| 04:56.55 | MrTelephone | heh |
| 04:57.00 | JayTee52 | [TK]D-Fender, it's like 3/4 depth of the dread |
| 04:57.09 | [TK]D-Fender | JayTee52, maybe manageable... |
| 04:57.21 | [TK]D-Fender | JayTee52, I've got a wish list I ought to get around to... |
| 04:57.36 | JayTee52 | I had an Epiphone that was 1/2 dread depth but it had next to no bass to it. |
| 04:57.49 | MrTelephone | if i dial out from the office it goes from [office-phones] -> [office-outgoing] -> [incoming-office] which includes office-phones.. |
| 04:57.49 | [TK]D-Fender | JayTee52, I want... THIS : http://www.youtube.com/watch?v=yIu7UA-823Y |
| 04:57.57 | `Sean | ~callweaver |
| 04:57.58 | jbot | somebody said callweaver was something that started off as a fork of Asterisk (b the name of openpbx), but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in Asterisk. see http://callweaver.org/ for more info, or join #callweaver |
| 04:57.58 | `Sean | :p |
| 04:58.25 | JayTee52 | I'm trying to sell my old Ovation Celebrity but so far haven't had any local takers. It's in prime shape. I'm leary of putting it on Ebay though. |
| 04:59.17 | Zuchmir | which variable shows me the calledid? |
| 04:59.26 | MrTelephone | $CALLERID(num) |
| 04:59.37 | MrTelephone | ${CALLERID(num)} |
| 04:59.38 | [TK]D-Fender | Zuchmir, "show function CALLERID" |
| 05:00.11 | JayTee52 | [TK]D-Fender, sounds like my Ibanez a/e classical with the narrow fretboard. That one is 1/2 dread depth |
| 05:00.39 | JayTee52 | I love Ibanez's guitars. |
| 05:00.40 | [TK]D-Fender | JayTee52, neato... I will have to go shoopiing sometime early next year. |
| 05:00.55 | JayTee52 | this is a great video, he's got nice style |
| 05:01.01 | [TK]D-Fender | JayTee52, I'm somewhat partial to them. Thin fast necks and a lot of 24 frets |
| 05:01.05 | MrTelephone | guitar hero |
| 05:01.14 | JayTee52 | I paid like $349 on sale at Guitar Center |
| 05:01.21 | [TK]D-Fender | JayTee52, one of my influences I wish I could say I can play more of... |
| 05:02.00 | [TK]D-Fender | JayTee52, my mainstays are Eric Johnson, Neil Zaza, Joe Satriani, and every 80's hair metal band you can think of :) |
| 05:02.38 | JayTee52 | I had a Yamaha classical with the classical neck but I'd get annoyed going between a standard neck on the Fender strat or Ibanez acoustic and the wide on the Yamaha. When I saw the Ibanez with the standard neck I jumped on it. |
| 05:03.17 | JayTee52 | Zaza's good, love Satriani. Clapton of course and I like Mark Knoppfler |
| 05:03.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:03.41 | [TK]D-Fender | <PROTECTED> |
| 05:03.54 | [TK]D-Fender | JayTee52, You know what Eric Clapton & bad coffee have in common? |
| 05:04.08 | [TK]D-Fender | JayTee52, They both suck without Cream :p |
| 05:04.37 | JayTee52 | hahahaha, I didn't want to say it but the punchline was such an obvious giveaway |
| 05:04.53 | [TK]D-Fender | And Dire Straits is wait too anti-climactic for me :) |
| 05:05.10 | JayTee52 | ever hear of a blues player named Otis Taylor? |
| 05:05.15 | MrTelephone | asterisk guitar |
| 05:05.25 | [TK]D-Fender | JayTee52, I think I've heard the name, but thats all I can so of it. |
| 05:05.37 | [TK]D-Fender | s/so/say/ |
| 05:06.04 | JayTee52 | if you've seen the movie Shooter with Mark Wahlberg it's the song at the end he does. Nasty Letter. great acoustic with a nice bluesy feel to it. |
| 05:06.45 | [TK]D-Fender | JayTee52, lookup Antoine Dufour and Andy Mckee on youtube then :) |
| 05:06.58 | [TK]D-Fender | JayTee52, AD is big on my accoustic list |
| 05:07.10 | JayTee52 | so if you had your choice of a server for Asterisk would you choose A) 3Ghz Dual Core or 2.4 Ghz Quad Core? |
| 05:07.16 | MrTelephone | anyone have a gui for making polycom config files? |
| 05:07.22 | MrTelephone | or a gui that parses them? |
| 05:07.38 | Zuchmir | thanks all! |
| 05:08.47 | JayTee52 | I've seen Antoine before!!! his style is so cool. I love it when you "break the rules" and create with such an unorthodox playing style. |
| 05:09.07 | JayTee52 | Spiritual Groove is awesome |
| 05:09.21 | [TK]D-Fender | JayTee52, definite fave. Makes me wish I knew more about music... |
| 05:10.12 | JayTee52 | you mean music theory? |
| 05:10.35 | [TK]D-Fender | JayTee52, yup, I know remarkably litte for how well I play :) |
| 05:11.16 | [TK]D-Fender | JayTee52, AD really blows my mind... |
| 05:11.32 | JayTee52 | this guy studied at Berkeley College of Music in Boston when I lived there and I took lessons from him. He now teaches there and plays with lots of big name bands. http://www.joestump.com/news.htm |
| 05:12.03 | stubert | can anyone help me with a 491 error question? |
| 05:13.03 | JayTee52 | the main thing he taught me was not to let the "theory" get in the way of the music. Close your eyes more when you play and just feel and listen. I don't consider myself a "real" musician but the guys that are I think with them it's more of a Zen thing. Be one with the instrument. |
| 05:13.24 | [TK]D-Fender | JayTee52, one of my head office coworkers used to jam with Alan Holdsworth and was a pretty good session guitarist himself. |
| 05:14.15 | [TK]D-Fender | JayTee52, Oh, I've got THAT alright, but I'm largely restricted to majors, minors (harmonic & natural), and a blues-d up pentatonic. |
| 05:14.15 | JayTee52 | I pickup something from everyone I've met who plays and even just watching a YouTube like Antoines and then just playing around with it. |
| 05:15.48 | JayTee52 | I kept trying to get the intro to Fuel's Hemmorhage right until I found out it was supposed to be drop d tuning but it was a half step off because he was using a chromatic tuner that with a weak battery. :-) |
| 05:16.09 | obnauticus | [TK]D-Fender BBHoss and I are trying to fix it, and it's apperentally a `unique |
| 05:16.12 | obnauticus | ' issue |
| 05:16.12 | obnauticus | :/ |
| 05:16.19 | obnauticus | (ie. karemtard) |
| 05:17.04 | JayTee52 | damn, it's wicked late. I've gotta catch some zzzz's. It's been a busy week. Got Asterisk talking to Exchange 2007 Unified Messaging through sipX this week so I'm feeling pretty good. |
| 05:17.23 | [TK]D-Fender | JayTee52, TCP intermediary .... *shudder* |
| 05:18.01 | JayTee52 | [TK]D-Fender, blame MS for adopting SIP over TCP. I wish Asterisk gave you the option and there's talk that 1.6 will have it. |
| 05:19.02 | JayTee52 | I make my living supporting Microsoft products. Windows Server 2003 and XP but my heart is in Linux. |
| 05:19.44 | MrTelephone | good thing microsoft frigs up all the time or you'd be out of a job |
| 05:20.38 | JayTee52 | MrTelephone, it's definitely job security :-) |
| 05:23.14 | JayTee52 | well, I'm off for the night. Thanks for the tips [TK]D-Fender and it was great chatting with you. |
| 05:23.17 | JayTee52 | Nite everyone |
| 05:23.23 | [TK]D-Fender | ditto |
| 05:23.31 | *** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 05:23.33 | rfxr | In the docs, zttool is mentioned but in zaptel 1.4.6 it doesn't exist. Was it replaced? |
| 05:26.46 | rfxr | nvm, failed dep libnewt |
| 05:31.44 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 05:37.58 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 05:38.28 | AJaymn | Anyone use VoicePulse? |
| 05:39.42 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:39.46 | joelsolanki | Hi all |
| 05:40.06 | joelsolanki | i m getting below error in asterisk 1.2.0 |
| 05:40.07 | joelsolanki | Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 05:40.28 | joelsolanki | is this code http://bugs.digium.com/view.php?id=5539&nbn=19 already implemented in asterisk 1.2.0 ? |
| 05:40.34 | joelsolanki | or i need to include it ? |
| 05:40.41 | Corydon76-home | 1.2.0 is REALLY old code |
| 05:41.03 | Corydon76-home | beyond unsupported |
| 05:41.21 | Corydon76-home | if you're going to be running 1.2, you should be running the latest bugfix version |
| 05:41.31 | joelsolanki | yes i understand. but we are using on one of the server and right now we can upgrade it :( |
| 05:42.01 | joelsolanki | ok upgrading to latest version will solve this dropping extra fram issue ? |
| 05:43.00 | joelsolanki | ? |
| 05:44.32 | joelsolanki | Corydon76-home: ?? |
| 05:45.33 | mosty | joelsolanki, possibly. at the very least it will get you access to more help |
| 05:48.30 | joelsolanki | hmm ok. |
| 05:51.19 | *** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34) |
| 05:53.44 | MrTelephone | is 1.4 anygood? |
| 05:54.54 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 05:59.29 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.119) |
| 06:03.32 | DataCompBoy | Hi all! Can someone help with ideas... Mine AGI script won't get DTMF on outgoing SIP calls :( |
| 06:03.54 | DataCompBoy | If I have originate call -- DTMF are not handled. If call incoming -- DTMF correctly handled. |
| 06:05.35 | [TK]D-Fender | DataCompBoy, look what you're dialing out using and make sure you're setting your dtmf mode in there. |
| 06:08.50 | *** join/#asterisk datacompboy (n=datacomp@213.187.250.34) |
| 06:09.00 | datacompboy | sorry, was disconnected |
| 06:09.02 | datacompboy | [TK]D-Fender: AMI Packet: Action: Originate, Channel: Local/1@ps_out, Application: AGI, Data: agi://server:port/ps_call/outgoing, Async: yes, Variable: number=00700100100100 |
| 06:09.10 | *** part/#asterisk atari (n=atari@213.144.146.89) |
| 06:09.22 | datacompboy | [TK]D-Fender: on zaptel DTMF handled fine. on call to voip gate -- dtmf fine |
| 06:09.27 | datacompboy | on direct call -- problem |
| 06:10.20 | [TK]D-Fender | datacompboy, pastebin a normal call out your peer and your configs |
| 06:10.56 | datacompboy | [TK]D-Fender: doing. mo |
| 06:16.48 | datacompboy | [TK]D-Fender: http://pastebin.com/d2719b03f here |
| 06:17.58 | [TK]D-Fender | datacompboy, and the rest please, and fast, I'm not here long... |
| 06:18.30 | datacompboy | [TK]D-Fender what rest? |
| 06:19.01 | [TK]D-Fender | datacompboy, configs and CLI output from a SUCCESSFUL call out your peer |
| 06:19.31 | datacompboy | [TK]D-Fender: sip config? |
| 06:19.44 | [TK]D-Fender | datacompboy, yes |
| 06:22.11 | datacompboy | [TK]D-Fender: here is successfull DTMF on dial-in: http://pastebin.com/d1b2c8c94 |
| 06:22.56 | [TK]D-Fender | datacompboy, No, show me a successful call OUT. |
| 06:23.22 | [TK]D-Fender | datacompboy, and pastebin your configs. You are wasting time. |
| 06:25.21 | datacompboy | [TK]D-Fender: http://pastebin.com/d279f3951 -- sip.conf |
| 06:26.11 | [TK]D-Fender | datacompboy, [vf-out] you did NOT set your dtmfmode for this peer. |
| 06:26.22 | [TK]D-Fender | datacompboy, go find out which one they use and set it. |
| 06:27.49 | datacompboy | [TK]D-Fender: http://pastebin.com/d4a4d3b19 -- here is successful dial-out (via voipfone gateway) |
| 06:28.55 | datacompboy | vf-out working fine! the problem only on plain SIP output dial, to 120000, 120001. all calls via @vf-out handle DTMF correctly |
| 06:31.14 | [TK]D-Fender | Sorry, too late for tonight. |
| 06:31.19 | *** join/#asterisk johann8384-home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 06:32.07 | datacompboy | pff:( ok. all bye |
| 06:33.07 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:37.47 | *** join/#asterisk olinux (n=olinux@72.54.254.97) |
| 06:38.20 | olinux | any switchvox users? |
| 06:46.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:50.14 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 06:52.56 | *** join/#asterisk BeeBuu (n=chatzill@218.13.66.237) |
| 06:53.03 | BeeBuu | hi,all |
| 06:53.28 | BeeBuu | hi,all |
| 06:53.59 | BeeBuu | anyone know is AstDB all running in memory? |
| 06:54.39 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 06:54.54 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 06:58.28 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:00.03 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:06.11 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 07:07.16 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:09.18 | BeeBuu | no one here now? |
| 07:17.22 | Strom_M | BeeBuu: astdb is stored on the drive |
| 07:18.08 | BeeBuu | hi,Strom_M |
| 07:18.24 | BeeBuu | Strom_M:it's nice to see you again. |
| 07:18.50 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:19.11 | BeeBuu | Strom_M: would you tell me how to pickup a line but does not dial? |
| 07:20.58 | Strom_M | that question makes no sense. |
| 07:21.59 | BeeBuu | i want check is there avaiaible line to use , and hold that line for use. |
| 07:22.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:23.37 | Strom_M | what do you mean "line" |
| 07:24.06 | BeeBuu | channel |
| 07:25.21 | Strom_M | why would you want to pick it up but not use it? |
| 07:25.56 | BeeBuu | make sure the channel can't be call |
| 07:26.07 | BeeBuu | ed |
| 07:28.03 | Strom_M | uh, ok? |
| 07:28.07 | Strom_M | perhaps I should ask it a different way: |
| 07:28.12 | Strom_M | what problem are you trying to solve? |
| 07:28.54 | *** join/#asterisk quelo (n=andros@host174-134-dynamic.181-80-r.retail.telecomitalia.it) |
| 07:29.13 | quelo | Hi I have a problem... |
| 07:29.29 | Strom_M | quelo: I'm sorry. I can't solve that problem. |
| 07:29.30 | quelo | I use debian etch |
| 07:29.46 | quelo | :) |
| 07:29.47 | BeeBuu | i want to deside answer the callin or not,if all channel busy,does not answer callin |
| 07:29.59 | Strom_M | BeeBuu: you're making no sense |
| 07:30.39 | *** part/#asterisk Raky-2 (n=John@220.157.75.66) |
| 07:30.47 | BeeBuu | user A call in, if no channel can be call out,hangUP user A |
| 07:31.03 | quelo | I've installed asterisk-prompt-it and configured sip.conf and iax.conf with language=it but te prompt is in english |
| 07:31.27 | Strom_M | BeeBuu: uh....but....give me a SCENARIO |
| 07:31.32 | quelo | I've also restarted asterisk |
| 07:34.23 | BeeBuu | Strom_M: example: user A call in ,the system check is there a avaliable channel can be dial out,if true,let the user A dial out with the avaliable channel,if not, hangup User A |
| 07:34.57 | *** join/#asterisk michael-i (n=michael-@Lf075.l.pppool.de) |
| 07:34.57 | BeeBuu | can you get me? |
| 07:35.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-88ac8fe76340c9aa) |
| 07:35.33 | Strom_M | BeeBuu: yeah, but that doesn't mean you need to hold a channel open |
| 07:35.37 | *** join/#asterisk olinux (i=olinux@ip68-107-11-147.sd.sd.cox.net) |
| 07:36.12 | BeeBuu | make sure the channel only can use by user A |
| 07:36.52 | BeeBuu | if at the same time, someone call in ,the channel is inavaliable |
| 07:37.11 | quelo | I have this problem also in trixbox I've followed this guide http://mirror.tomato.it/ftp/pub/asterisk/suoni_ita/index.html |
| 07:37.18 | Strom_M | the problem is you've not got enough channels. |
| 07:37.27 | Strom_M | solution: traffic engineering |
| 07:37.51 | BeeBuu | yes,i know,but always only few channels to use. |
| 07:38.09 | Strom_M | so you're going to kludge it more? wrong solution. |
| 07:38.35 | BeeBuu | so i want to let the user know,no avaliable channel at this time. |
| 07:39.04 | BeeBuu | user can retry later. |
| 07:40.55 | olinux | where can i buy voip minutes? |
| 07:43.30 | joelsolanki | olinux: alot of places |
| 07:43.40 | olinux | thanks! |
| 07:45.44 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de) |
| 07:48.41 | yang | If I do call forwarding to 3 numbers when one number rings, in which file is that defined ? extensions.conf ? |
| 07:53.25 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:57.03 | quelo | anyone can help me to solve the problem with asterisk-prompt-it ? |
| 08:06.49 | olinux | quelo describe the problem |
| 08:09.09 | hmmhesays | yang, yes |
| 08:09.40 | hmmhesays | Dial(Tech/num1&Tech/num2&tech/num3) |
| 08:12.14 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 08:13.24 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:22.00 | yang | thanks hmmhesays |
| 08:22.06 | olinux | no options for an asterisk voip provider? |
| 08:22.39 | kaldemar | ~itsp |
| 08:22.40 | jbot | well, itsp is an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 08:23.27 | olinux | how does that help? |
| 08:23.33 | kaldemar | what was it again... "they all suck. some suck less. teliax seems to suck less than most." |
| 08:24.00 | olinux | heh |
| 08:25.02 | olinux | my system is down until tech support arrives in the morn, i hope i can hook up an alternate and at least provide outgoing calls before that |
| 08:27.37 | *** join/#asterisk clickonce (n=get@unaffiliated/clickonce) |
| 08:27.42 | clickonce | Hello phoners! :) |
| 08:28.03 | clickonce | Does Asterisk and Linksys SPA962 work together? |
| 08:28.53 | *** join/#asterisk olinux (i=olinux@ip68-107-11-147.sd.sd.cox.net) |
| 08:29.41 | yang | hmmhesays: exten => 2967,1,Dial(SIP/Tech/2967&Tech/2931) or without the SIP string ? |
| 08:32.21 | kaldemar | Dial(SIP/2967&SIP/2931), the Tech was supposed to be replaced with SIP, Zap, IAX2 or whatever channel type you're using. |
| 08:33.10 | yang | ok, thanks |
| 08:36.48 | l0verb0y | hey has anyone ever used sipp? |
| 08:51.40 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 08:53.05 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:57.13 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:16.26 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
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| 09:25.04 | *** join/#asterisk cypherdelic (n=cypher@p5B27DFAC.dip.t-dialin.net) |
| 09:29.52 | badcfe | i observe a weird sip behaviour in asterisk; when the INVITE came via a proxy (RR) the response go "To:" the proxy itself. |
| 09:29.56 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 09:30.20 | badcfe | anyone would insepct a trace of this? |
| 09:30.44 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 09:30.51 | jozu | hi to all |
| 09:31.22 | badcfe | hi, would you like a cup of warm morning sip-trace? 8-) |
| 09:31.47 | badcfe | its about incoming sip dialog via a proxy |
| 09:32.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:32.28 | jozu | i have one extension (300) the first line is (s,1,Answer) |
| 09:32.36 | jozu | its possible To save before the number of the caller |
| 09:32.48 | jozu | ? |
| 09:32.56 | *** join/#asterisk funkyhippy (n=notahipp@87.127.49.162) |
| 09:33.08 | juliux | hi all, can somebody give me an advice where to search why this call was not transfered right? http://ubuntuusers.de/paste/18203/ |
| 09:33.40 | jozu | something like s,1,cdr_save_here_to_calllater |
| 09:34.37 | funkyhippy | hi all, can anyone answer a question about Asterisk Business Edition? |
| 09:35.08 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:40.57 | funkyhippy | ok guess thats a no then.. |
| 09:41.00 | funkyhippy | quit |
| 09:55.43 | *** join/#asterisk vgster (n=vgster@217.78.148.66) |
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| 10:11.12 | *** join/#asterisk oej (n=olle@62.97.243.70) |
| 10:11.39 | juliux | what does "Didn't get a frame from channel" means? |
| 10:13.25 | *** join/#asterisk macTijn (i=martijn@rachel.insecure.nl) |
| 10:14.21 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:16.34 | *** join/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 10:16.37 | Haris | Hello guys |
| 10:22.36 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
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| 11:40.14 | *** join/#asterisk Psychobilly (n=Fuzz@online2.ioa.forthnet.gr) |
| 11:42.22 | Psychobilly | can i use regular exprresions inside an app like GotoIf() , for example will somehtin like this work : GotoIF(${EXTEN}=_1XXXX?foo,10) |
| 11:44.12 | ai-a | so number starting with 1 ? whats the "?foo" ? |
| 11:44.22 | ai-a | oh i see. |
| 11:45.45 | ai-a | you can use ${EXTEN:0:1} and & |
| 11:46.25 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:46.36 | *** join/#asterisk snotsnot (n=snot@polar.es6.egwn.net) |
| 11:46.50 | Psychobilly | its not exactly what im looking for ai-a |
| 11:47.25 | snotsnot | hi there |
| 11:47.31 | ai-a | you want to find exten that starts with 1 and is 5 characters long ? |
| 11:47.39 | ai-a | if so, goto foo, else goto 10 |
| 11:48.10 | Psychobilly | this was just an expample, my question is if i can use regulas expressions |
| 11:48.24 | ai-a | no then :) |
| 11:49.04 | ai-a | you can use regexp. |
| 11:49.12 | Psychobilly | hm |
| 11:49.19 | ai-a | http://www.voip-info.org/wiki/index.php?page=Asterisk+Expressions |
| 11:49.50 | snotsnot | i'm using php agi script and i got problem while the user hangup, i want to execute a function so that exiting my script properly |
| 11:49.52 | ai-a | something like "1[0-9]{4}" |
| 11:49.58 | snotsnot | i use if (function_exists('pcntl_signal')) { |
| 11:49.58 | snotsnot | //pcntl_signal(SIGHUP, SIG_IGN); |
| 11:49.58 | snotsnot | pcntl_signal(SIGHUP, "my_callback"); |
| 11:49.58 | snotsnot | } |
| 11:50.05 | *** part/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 11:50.07 | snotsnot | to call mycallback |
| 11:50.16 | snotsnot | but it seems the script don't catch the sig |
| 11:50.19 | snotsnot | any idÃĐes? |
| 11:50.53 | snotsnot | ideas (sorry for my english ) |
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| 12:14.18 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:24.35 | yang | is there a way to change status from unmonitored to pingable ? |
| 12:24.41 | yang | status lines |
| 12:26.35 | yang | i have 2 asterisk versions ne displays status nicely |
| 12:26.41 | yang | ne/one |
| 12:32.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:33.33 | Psychobilly | yang you mean the qualify=yes option? |
| 12:38.03 | *** part/#asterisk clickonce (n=get@unaffiliated/clickonce) |
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| 12:41.06 | *** join/#asterisk Tourinho (n=tourinho@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 12:41.32 | Tourinho | good day, does anyone knows a benchmark por asterisk and g729? |
| 12:45.15 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 12:48.32 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 12:50.33 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:50.45 | davidcsi | can i get any help on compiling asterisk-oh323? both openh323 and pwlib compiled fine, but asterisk-oh323-0.7.3 won't compile... it returns http://pastebin.ca/777001 |
| 12:51.20 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 12:51.23 | davidcsi | anyone? |
| 12:54.54 | ai-a | asteriskaudio.cxx:35:19: error: oh323.h: No such file or directory |
| 12:55.09 | ai-a | davidcsi: where is oh323.h file for including ? |
| 12:56.24 | davidcsi | its /usr/src/asterisk-oh323-0.6.7/asterisk-driver/oh323.h |
| 12:56.56 | *** join/#asterisk bantu (n=Miranda@vpnwww01.rz.uni-karlsruhe.de) |
| 12:58.08 | ai-a | and thats in the include path ? |
| 12:59.33 | ai-a | mine is /usr/src/asterisk-addons-1.4/asterisk-ooh323c/ooh323c/src/ooh323.h |
| 12:59.40 | davidcsi | include paths are, as per instructions |
| 12:59.40 | davidcsi | PWLIBDIR=/usr/src/pwlib_v1_9_2 |
| 12:59.40 | davidcsi | OPENH323DIR=/usr/src/openh323_v1_17_3 |
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| 13:10.26 | yang | Psychobilly: ah in the SIP section? |
| 13:10.39 | Psychobilly | yes |
| 13:12.04 | yang | Psychobilly: hehe, we have this VLINES hardware/software....I wonder where does that gets applies ...there was really no qualify=yes in the sip sections as i can see on the console |
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| 13:17.10 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:17.39 | ai-a | yang: what about your users.conf ? |
| 13:19.36 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:19.36 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:20.06 | ai-a | users.conf is a nice idea... but adds another layer of hassle. I would have prefered a console command asteriskuser that adds / removes a user from the system (manipulating the sip,iax,...exts..) |
| 13:20.27 | yang | ai-a: the whole asterisk config is managed by the vlines software |
| 13:20.41 | ai-a | and ? |
| 13:20.46 | yang | i can see the config files, but i dont know if i can manually edit them |
| 13:21.02 | ai-a | you have to ask Mr VLines. |
| 13:21.08 | yang | hehe |
| 13:21.12 | ai-a | if its re-written from their db, then no. |
| 13:21.42 | ai-a | but they must have some 'base' files for building from. |
| 13:22.07 | ai-a | ask then for some layout info... or request additions that they dontr suply.. or just host it yourself, asterisk install isnt hard. |
| 13:22.29 | ai-a | what do vlines thats better than bascily asterisk ? |
| 13:22.44 | yang | well its just that the company allready uses vlines hardware |
| 13:23.03 | yang | we manage a separate voip on an asterisk boy too |
| 13:23.05 | yang | box |
| 13:23.27 | ai-a | your wanting to modify the vline asterisk box ? |
| 13:24.52 | yang | ok i added qualify=yes to the sip account, but i still cannot see the status it says unmonitored (even that the number is active and shows an ip address) |
| 13:25.31 | *** join/#asterisk jameswf-home (n=that@ip72-204-228-104.ph.ph.cox.net) |
| 13:25.39 | yang | and the peers are show wrongly too |
| 13:25.47 | yang | bad luck |
| 13:27.11 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:27.18 | yang | hi [TK]D-Fender |
| 13:27.47 | *** join/#asterisk dexteruk (n=charper@78.90.15.216) |
| 13:28.35 | yang | ai-a: i think that the software is buggy on that vlines |
| 13:29.16 | ai-a | the vline is a front end system to asterisk, its asterisk your connecting to... get on the box, and pastebin some information of the two systems and how your trying to connect them. |
| 13:29.20 | yang | or we applied some weird settings to disable the status somewhere it could be possible |
| 13:31.03 | yang | http://openpaste.org/en/3926/ |
| 13:31.17 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:31.48 | dexteruk | trying asterisk 1.4.13 using realtime i have added iax user to the iax_buddies database and when i do iax2 show peers it is not shown and rtcachefriends is set to yes |
| 13:32.37 | yang | ai-a: the offline peers show worng |
| 13:34.11 | dexteruk | does anyone have any clues |
| 13:34.28 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:35.03 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:35.36 | yang | ok i fixed it (reloaded:) |
| 13:36.50 | dexteruk | trying asterisk 1.4.13 using realtime i have added iax user to the iax_buddies database and when i do iax2 show peers it is not shown and rtcachefriends=yes |
| 13:38.36 | file | dexteruk: it's not loaded and cached until it is used |
| 13:38.59 | file | do an iax2 show peer <peer name> load and it will be loaded from realtime and cached |
| 13:40.38 | jameswf-home | I am guessing if you ask most people they would not use win98 but they will use trixbox PRO with the 2.6.9 kernel which is ummmm equal to windows98 idiots... |
| 13:42.21 | anonymouz666 | I am using Win98 Second Edition |
| 13:42.31 | jameswf-home | << has a php script that loads show peers to xml and displays with xslt :) |
| 13:43.17 | jameswf-home | my house is windows free... expelled that demon a while ago |
| 13:43.35 | coppice | bit draughty, isn't it? |
| 13:44.22 | jameswf-home | no walls no fences |
| 13:44.37 | coppice | any rooms? |
| 13:44.37 | jameswf-home | verry interesting to shower :) |
| 13:45.21 | jameswf-home | we are all mimes the rooms are virtual :) |
| 13:46.11 | coppice | and you all dress in vmwear? |
| 13:46.43 | jameswf-home | you know since i swiched to linux I havent had to pirate(software) |
| 13:47.14 | jameswf-home | I sat in the corner of a round room |
| 13:47.16 | coppice | this is the main drawback. you can pirate movie to compensate a little |
| 13:48.21 | coppice | there'll still be a something something AA after you |
| 13:48.22 | jameswf-home | my latest thing is ebooks. |
| 13:48.54 | jameswf-home | is there one for books? |
| 13:49.52 | jameswf-home | I use google mostly so the aa's can go after the people who cant manage their servers |
| 13:51.53 | jameswf-home | technicaly I am streaming the music from them and anything residual on my system is cache |
| 13:56.37 | _x86_ | morning |
| 13:56.48 | jameswf-home | it is |
| 13:57.35 | _x86_ | indoubedebly |
| 13:58.11 | jameswf-home | this working from home thing is odd I dont know whaqt to do with out my hour commute |
| 13:59.55 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 14:00.08 | coppice | working at home when we first married was good :-) |
| 14:00.21 | ZenBSDi | any java-agi coders up in here? |
| 14:00.41 | *** join/#asterisk Psychobilly (n=Fuzz@online2.ioa.forthnet.gr) |
| 14:01.39 | Psychobilly | in a macro i want to use only the first 3 digits from an ${ARG} how can i do this? |
| 14:02.07 | [TK]D-Fender | Psychobilly: ${ARG:3} |
| 14:02.26 | [TK]D-Fender | Psychobilly: Time to re-read the section on how to use variable all over again... |
| 14:02.35 | Psychobilly | yes i quess so |
| 14:02.37 | [TK]D-Fender | Psychobilly: ${ARG:0:3} rather |
| 14:02.51 | Psychobilly | im a bit comfuzed :) |
| 14:02.58 | Psychobilly | ok thanks [TK]D-Fender |
| 14:03.53 | Psychobilly | ah it worked perfectly! thanks a lot :) |
| 14:04.34 | Psychobilly | i was using totaly wrong syntax: ${${ARG}:3} |
| 14:04.35 | ZenBSDi | [TK]D-Fender, time to get a paypal account and start charging for your knowledge =p .. |
| 14:04.36 | Psychobilly | :S |
| 14:04.57 | ZenBSDi | you asterisk freak you .. |
| 14:04.57 | Psychobilly | hehe |
| 14:05.22 | [TK]D-Fender | ZenBSDi: I have a paypal account, and I am an * consultant :) |
| 14:05.30 | *** part/#asterisk ming_zym (n=ming_zym@124.14.235.104) |
| 14:05.35 | [TK]D-Fender | ZenBSDi: Donations welcome ;) |
| 14:05.41 | ZenBSDi | how much work you get doing this? |
| 14:05.45 | jameswf-home | I sometimes wonder if i should get offended when people are supprised I am right, It's like you had doubt? screw you! |
| 14:06.12 | [TK]D-Fender | ZenBSDi: Depends on the job. I job cost most stuf, and have a more standard wage for the occasional general stuff |
| 14:06.51 | [TK]D-Fender | jameswf : I think that Dale Carnegie could have used your creative input.... |
| 14:06.52 | jameswf-home | for side work I charge $60/hr 2 hr min |
| 14:07.34 | ZenBSDi | I'm trying to get into the consulting game myself.. got my LPI certification and I've mastered mysql replication and learning about the clustering. I'm learning this * stuff cause I figure the businesses would want to save some change on a pbx. |
| 14:07.39 | jameswf-home | usualy that scares em off for a week then they are like ok |
| 14:07.51 | *** join/#asterisk michael-i (n=michael-@Ld685.l.pppool.de) |
| 14:08.16 | ZenBSDi | $60 an hour doesn't sound bad at all.. I hear 100-120 an hour usually |
| 14:08.29 | jameswf-home | people are often cheap untill they realize they can't be |
| 14:08.41 | ZenBSDi | lol amen to that |
| 14:09.08 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 14:09.33 | jameswf-home | well I figure $120 for 30 minutes (hense the minimum) isnt bad.... gets the wife some spending money and gets her out of the house to leave me alone |
| 14:09.52 | ZenBSDi | bwa hahaha |
| 14:10.06 | ZenBSDi | "leave me woman.. the pr0n3th c@ll3th" =p |
| 14:10.15 | jameswf-home | I went and fixed someones server time and got jeff duhnum tickets :) |
| 14:10.36 | ZenBSDi | no clue who he is |
| 14:10.37 | ZenBSDi | =p |
| 14:10.53 | jameswf-home | the vantriliquist with the purple guy and the old man |
| 14:11.45 | jameswf-home | yeah that one |
| 14:11.48 | jameswf-home | :) |
| 14:11.51 | ZenBSDi | ok.. THAT was worth it |
| 14:11.57 | ZenBSDi | that man is funny |
| 14:12.20 | jameswf-home | the dead terrorisr lmao |
| 14:12.37 | ZenBSDi | so you're a do it all or do you specialize? |
| 14:13.08 | jameswf-home | I am a linux guy who does asterisk plus various asterisk projects |
| 14:13.49 | ZenBSDi | do any agi coding? or just macro's/dialplans? |
| 14:14.37 | jameswf-home | i did an agi script that burns CD's dial ext xxx for ubuntu, xxx for fedora etc.. |
| 14:15.08 | ZenBSDi | I want to figure out how to do credit card processing by phone =p |
| 14:15.09 | jameswf-home | not something i do on a regular basis, usualy only for fun |
| 14:15.31 | jameswf-home | php and a paypal gateway would likely work |
| 14:15.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.09 | ZenBSDi | I probably should give up on using java for this.. |
| 14:17.18 | Titanous | ZenBSDi, use perl |
| 14:18.00 | ZenBSDi | hell no! I was a linux admin for a webhosting company from 2001 - 2003 and thats all I used.. you can't pay me enough to go back to using that chaotic scripting language =p |
| 14:18.53 | jameswf-home | I like to mix perl and php it makes both groups of users nervous |
| 14:19.06 | Titanous | I do a little php also |
| 14:19.45 | jameswf-home | you could jumr on the ruby/on/rails band wagon |
| 14:19.48 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:20.05 | ZenBSDi | I'm a java/oracle guy myself.. having written quite a few jdbc applications ... but figuring out how use the asterisk java interface is a bit painful ... |
| 14:21.31 | jameswf-home | only coooffffeeeeee |
| 14:26.06 | lirakis | can i "Playback" a busy signal? |
| 14:26.36 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:26.50 | lirakis | .. i want to respond differently to different ISDN cause codes.. and on busy 17, i just want to play back the normal busy signal.. |
| 14:29.36 | jameswf-home | busy/reorder/etc are not sound files |
| 14:30.34 | ai-a | lirakis: just send back the busy response |
| 14:33.46 | lirakis | ai-a: can you clarify "send back" ? |
| 14:34.31 | ai-a | are you writing isdn code or in asterisk call flow ? |
| 14:34.59 | lirakis | ai-a: if my Dial() has already finished.. and afterwards I am checking ${HANGUPSTATUS} .. how would i indicate busy to the caller? ... ooo duh i forgot the Busy() command |
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| 14:39.27 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:45.32 | zeeesh | can i send vm at any email address in the shape of wave file or gsm file ? |
| 14:46.44 | jameswf-home | umm can it be done? yes, can you do it? i dunno |
| 14:48.37 | jameswf-home | linux+asterisk can acomplish almost anything with the right person in control |
| 14:49.46 | jameswf-home | I dont think it can wipe your butt but it probably could control a bodait (or however those french goats spell it) |
| 14:50.06 | rob0 | I would like to teach it to feed cats. |
| 14:50.59 | jameswf-home | all you have to do is get an automatic cat feeder, figure out how to set up an rs2r2 connection and you will be golden |
| 14:51.00 | rob0 | I have a * server in Alabama, and 4 cats. But I'm never there. |
| 14:51.16 | jameswf-home | *rs232 |
| 14:52.12 | jameswf-home | we were discussing at work interfacing a soda machine with asterisk, press 1 for dr pepper etc |
| 14:53.08 | *** join/#asterisk mocker (n=ksexton@198.247.173.227) |
| 14:59.23 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:00.03 | Zeeek | ahhhhhhhh |
| 15:00.22 | Zeeek | it's rare I can do it in the middle of the working day. |
| 15:00.37 | mocker | Well, sounds like I'm going to a class on Aspect Unified IP this month. |
| 15:03.19 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:07.35 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 15:13.51 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 15:14.58 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 15:17.35 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 15:17.45 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:18.17 | *** join/#asterisk saftsack (n=saftsack@pD9E05E04.dip.t-dialin.net) |
| 15:21.13 | *** join/#asterisk HarryR`Work (n=harryr@77.240.56.17) |
| 15:23.30 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:23.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:31.20 | ZenBSDi | man this sucks.. http://pastebin.com/m705b47b3 |
| 15:35.05 | viperdudeuk | hi anyone know how to stop sipsak waiting for a response? |
| 15:35.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 15:36.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:36.53 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 15:37.21 | [T]ank | is it normal for a system with a sangoma 4 port t1 card to have very high software interrupts? |
| 15:37.31 | [T]ank | like 45% |
| 15:37.50 | [T]ank | hardware interrupts are at 0 |
| 15:38.24 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:38.46 | HarryR`Work | [T]ank, for high resolution timers? |
| 15:39.02 | [T]ank | not sure what that meand |
| 15:39.04 | [T]ank | means |
| 15:39.43 | HarryR`Work | computers run on smoke, when the smoke comes out they stop working |
| 15:39.56 | HarryR`Work | they help keep the smoke inside the computer |
| 15:40.35 | [T]ank | i dont think we are speaking the same language. I have no idea what you are trying to tell me |
| 15:44.58 | jameswf-home | We had a couple computers loose smoke and keep going, just didnt have an aux fan |
| 15:48.26 | Zeeek | awright |
| 15:49.36 | blitzrage | oh wait... maybe I didn't :) |
| 15:49.39 | blitzrage | Zeeek: ! |
| 15:49.43 | Zeeek | yo blitzrage |
| 15:49.52 | blitzrage | oy |
| 15:49.53 | Zeeek | can I buy you a beer? |
| 15:49.58 | blitzrage | of course you can |
| 15:50.06 | Zeeek | it's waiting for you at the table |
| 15:50.11 | blitzrage | http://www.idialabottle.ca/ |
| 15:50.21 | blitzrage | now you have to :D |
| 15:50.21 | Zeeek | http://voipUsersConference.org |
| 15:51.00 | Zeeek | I can do better than that at LCBO |
| 15:51.29 | blitzrage | I need breakfast |
| 15:51.30 | blitzrage | back in a bit |
| 15:51.39 | Zeeek | O. K. |
| 15:51.39 | *** join/#asterisk kingsob (n=derek@TOROON01-1177844188.sdsl.bell.ca) |
| 15:51.57 | Zeeek | It's 16h51 here, I'm wide awake, too wide awake |
| 15:52.16 | Zeeek | transit strike, but I walk to work anyway, so no biggie |
| 15:52.38 | jameswf-home | Where i live no transit lol |
| 15:52.56 | Zeeek | where is that, transylvania? |
| 15:53.30 | jameswf-home | no we dont have a name either its a rural suburb, unincorperated... |
| 15:53.33 | codefreeze | jameswf-home: what? no cars, horses, bicycles, nothing? |
| 15:53.37 | blitzrage | ewww... suburbs |
| 15:53.47 | Zeeek | I lived in a place with no name once. Didn't have a horse, though |
| 15:53.49 | blitzrage | Toronto's transit system is pretty decent |
| 15:53.51 | jameswf-home | everyone out here drives big suv's |
| 15:53.58 | jameswf-home | a bunch of snooty people lol |
| 15:54.05 | blitzrage | sounds about right |
| 15:54.39 | Zeeek | don need no stinkin' transit in that case |
| 15:54.39 | jameswf-home | I really am too poor to live where i do but screw em |
| 15:54.56 | Zeeek | AMazon S3 is great for backups by the way. Highly recommended, and I don't get a commission |
| 15:55.10 | Zeeek | Backup tot he cloud |
| 15:55.24 | Zeeek | or to the cloud, even |
| 15:55.30 | jameswf-home | bachupps are for wimps, real men restore by hand :) |
| 15:55.45 | Zeeek | you need a backup in order to restore |
| 15:55.58 | jameswf-home | no photographic memory |
| 15:56.06 | jameswf-home | bit for bit from a console |
| 15:56.07 | jameswf-home | :) |
| 15:56.10 | Zeeek | oh, retype the code? Sure, why not? |
| 15:57.02 | Zeeek | We have two onsite bu, one S3 and a 250g drive offsite. In case of nuclear attack, the pen drive will be all we have |
| 15:57.28 | JT | it is an EMP shielded pen drive? |
| 15:57.47 | Zeeek | of course, ya think I'm clueless? err, what is EMP? |
| 15:58.01 | Zeeek | elector-motive pecker? |
| 15:58.05 | JT | Electro-Magnetic Pulse |
| 15:58.13 | Zeeek | exciting stuff |
| 15:58.20 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 15:58.24 | JT | nuclear blasts also generate EMP as well as obvious physical effects |
| 15:58.38 | JT | which destroys electronics |
| 15:59.00 | jameswf-home | https://www.ironkey.com/ <<< bad arse thumbdrive |
| 15:59.17 | Zeeek | prolly won't mind no dialplan iof there's a nuke attack here, but I like to be prepared |
| 15:59.38 | jameswf-home | yeah if you get nuked backups are the least of your worries |
| 15:59.39 | *** join/#asterisk debiano778 (n=nana@213-140-19-123.fastres.net) |
| 15:59.43 | debiano778 | hi all |
| 16:00.13 | debiano778 | anyone can help me with this error in zaptel.con ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 16:00.19 | JT | depends how close the nuke is |
| 16:00.25 | Zeeek | the backups are mainly for: fire, theft, drive corruption. |
| 16:00.47 | JT | backups allow easier data theft ;) |
| 16:01.03 | Zeeek | the local backup is on two direces on two different boxen. Then an offsite weekly. Then the S3 cloud. Then the pendrive |
| 16:01.14 | [TK]D-Fender | My backups are more for idiots who delete stuff without thinking. And its happening at an increasing rate |
| 16:01.21 | JT | [TK]D-Fender: so true |
| 16:01.23 | debiano778 | Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? |
| 16:01.25 | Zeeek | that's why we password protect them. The password is "passcord" |
| 16:01.39 | JT | debiano778: using the wrong signalling are you? |
| 16:01.41 | HarryR`Work | [TK]D-Fender, LVM/ZFS snapshots :D |
| 16:01.47 | Zeeek | [TK]D-Fender never happens to us. Except sometimes |
| 16:01.55 | HarryR`Work | and there are various self-service backup schemes using rsync |
| 16:02.05 | *** join/#asterisk johndbritton (n=john@cpe-74-70-255-158.nycap.res.rr.com) |
| 16:02.16 | debiano778 | i set fxo signal in my zaptel.conf |
| 16:02.19 | johndbritton | could someone help me debug a one-way audio problem im having? |
| 16:02.33 | Zeeek | johndbritton turn off "transmit silence=no" |
| 16:02.36 | debiano778 | cause i have 2 FXO (red) module in the card |
| 16:02.37 | JT | debiano778: is it an fxo port? |
| 16:02.39 | [TK]D-Fender | johndbritton: Describe the situation fully and we'll see |
| 16:02.54 | JT | debiano778: wrong signalling. use fxs signalling. |
| 16:02.55 | Zeeek | no we won't but we might be able to help |
| 16:03.02 | JT | debiano778: exactly as the error message says |
| 16:03.16 | johndbritton | Zeeek: what does that mean... put "transmit silence=no" in the sip.conf for the device with the problem? |
| 16:03.19 | *** join/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 16:03.47 | debiano778 | JT: need to repost the error? |
| 16:03.59 | Zeeek | johndbritton go ahead and describe your hole situation |
| 16:04.04 | JT | debiano778: NO, as i've already said, you are using the WRONG SIGNALLING |
| 16:04.12 | johndbritton | here is the situation: I've got asterisk running on a machine with a pulic ip and no firewall |
| 16:04.20 | *** join/#asterisk Darthclue (n=chatzill@zeus.nisd.net) |
| 16:04.25 | Zeeek | so far, so good! |
| 16:04.28 | debiano778 | JT: can i ask some question in private? |
| 16:04.34 | JT | debiano778: what for? |
| 16:04.42 | jameswf-home | to save face |
| 16:04.45 | debiano778 | for configuring my card |
| 16:04.48 | johndbritton | ive got a few softphones configured that work fine |
| 16:04.59 | JT | debiano778: why can't that be done on channel? |
| 16:05.01 | johndbritton | all used from homes across the country... behind nat |
| 16:05.09 | debiano778 | ok np i ask here |
| 16:05.28 | jameswf-home | anyone use moxphone |
| 16:05.31 | Nuitari | I have some really weird issues with 1 polycom 501 to get it to register to asterisk |
| 16:05.34 | jameswf-home | *moxphone |
| 16:05.40 | jameswf-home | doh |
| 16:05.48 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 16:05.49 | Nuitari | in the sip debug, I see the first attempt that always fail, but the 2nd one with the username and digest never happen |
| 16:06.02 | johndbritton | and i just purchased a snom 190 off ebay, and configured it, but ive got one way audio... i can make outgoing calls from the phone and get incoming calls to the phone, but in both cases the other end cant hear me |
| 16:06.06 | Nuitari | and the phone does pick up the proper config in the ftp and the password and username are right in all cases |
| 16:06.16 | johndbritton | so i can hear the person on the other end from the hard sip phone, but they cannot hear me |
| 16:06.22 | johndbritton | and i think that is everything |
| 16:06.49 | JT | johndbritton: is qualify set to yes and do they regularly register? |
| 16:06.53 | Zeeek | johndbritton sounds like it's sno-specific if your softphones are working. Have you searched the mailing list and googled for snom+Asterisk, etc? |
| 16:07.19 | [TK]D-Fender | johndbritton: pastebin your sip.conf masking only passwords |
| 16:07.21 | [TK]D-Fender | ~pb |
| 16:07.22 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:07.23 | johndbritton | voip-info.org says this phone works very well with asterisk... ive updated the firmware |
| 16:07.23 | johndbritton | as it says |
| 16:07.23 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 16:07.31 | johndbritton | ok |
| 16:07.45 | JT | johndbritton: does it say it works very well with NAT? |
| 16:08.04 | johndbritton | it doesnt say "very well" but it says it does work |
| 16:08.14 | Zeeek | johndbritton we're talking SIP, right? |
| 16:08.26 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 16:08.26 | johndbritton | do you want the whole sip.conf, or just the relevant entry |
| 16:08.29 | johndbritton | yes |
| 16:08.37 | [TK]D-Fender | johndbritton: All |
| 16:09.22 | Zeeek | this *might* be of use (not sure) http://www.voip-info.org/wiki/view/One-way+Audio |
| 16:10.05 | Zeeek | remove pw! [TK]D-Fender is actually a chinese hacker who will try to compromise your system! |
| 16:10.29 | debiano778 | JT: i must use fsx signal in my card? |
| 16:10.30 | johndbritton | i will |
| 16:10.33 | [TK]D-Fender | Yes, remove the passwords, I love Chinese food.... |
| 16:10.34 | johndbritton | thats waht im doing now |
| 16:10.56 | Zeeek | put the passwords in place of the username and vice-verse |
| 16:11.03 | Zeeek | no one will be any the wiser |
| 16:11.32 | johndbritton | http://pastebin.com/d1790e99c |
| 16:11.36 | Darthclue | just rot13 encrypt them |
| 16:11.41 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:11.49 | Zeeek | so, after six weeks, ebay will be refunding my money from the compromised account. Ebay is STILL the DEVIL though |
| 16:11.58 | Nuitari | better use ROT52, much more secure |
| 16:12.00 | johndbritton | Zeeek: agreed |
| 16:12.34 | jameswf-home | I use craigslist, I usualy get 15 cashiers checks a week :) |
| 16:12.47 | Zeeek | I vote ebay for the absolute worst online company ummm online. Also, the CEO seriously needs a new hairstyle |
| 16:13.31 | debiano778 | JT: are u there? |
| 16:13.33 | Zeeek | hmmmm BV, add looking at BV forums for the answer as well |
| 16:13.45 | johndbritton | and the hardphone is extention 101 |
| 16:13.48 | jameswf-home | is donald trump the ceo of ebay? |
| 16:13.52 | Zeeek | there's always been a lot of chatter about BV and asterisk |
| 16:14.10 | johndbritton | it works |
| 16:14.12 | Zeeek | NAw, some lady with permanently bad hair day |
| 16:14.25 | johndbritton | just not this phone, everything else has worked |
| 16:14.44 | Zeeek | http://upload.wikimedia.org/wikipedia/commons/thumb/2/22/Meg_Whitman.jpg/180px-Meg_Whitman.jpg |
| 16:14.48 | Zeeek | here ya go |
| 16:14.51 | johndbritton | which leads me to believe that its a nat problem, but i don't know how to diagnose it |
| 16:15.37 | Zeeek | reference: http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP |
| 16:15.53 | johndbritton | im looking at the asterisk console at the |
| 16:15.54 | Zeeek | any other hardware phones working on the same setup? |
| 16:15.56 | johndbritton | "sip set debug" |
| 16:16.24 | johndbritton | its the first one i bought... been using asterisk for about 6 months on soft phones, and finally bought a hardphnoe |
| 16:16.34 | [TK]D-Fender | johndbritton: Looks fine |
| 16:16.46 | debiano778 | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? |
| 16:17.10 | debiano778 | anyone can help me with this error? |
| 16:17.30 | jameswf-home | only if you can read the error |
| 16:17.34 | johndbritton | the phone has stun support, but im not so sure im using it correctly... could that have something to do with the problem |
| 16:18.02 | JT | debiano778: i've already told you what, 3 times, and asterisk has told you heaps - CHANGE YOUR SIGNALLING |
| 16:18.12 | debiano778 | ok thanks but if i configure zaptes as fxs module is ok |
| 16:18.43 | JT | fxo modules use fxs signalling. |
| 16:18.46 | debiano778 | but can i make a call with fxs module? |
| 16:18.47 | JT | ~thebook |
| 16:18.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:18.50 | JT | please read that |
| 16:18.51 | JT | now |
| 16:19.10 | debiano778 | ok thanks |
| 16:19.24 | jameswf-home | fxs modules connect to phones fxo modules connect to telcos and cannot be swapped |
| 16:19.44 | debiano778 | fxo module are the red module? |
| 16:20.46 | [TK]D-Fender | debiano778: Correct |
| 16:20.51 | jameswf-home | where i work all modules are red lol |
| 16:20.56 | johndbritton | Zeeek: it doesn't make sense that I cannot be heard, I would think that nat problems would allow audio out but not in... does that make any sense? |
| 16:21.01 | debiano778 | ok in my car i have 2 red module |
| 16:21.15 | debiano778 | but i have this erroer when i configure as fxo module |
| 16:21.18 | jameswf-home | johndbritton: what type of router do you use |
| 16:21.26 | [TK]D-Fender | johndbritton: Does a softphone work in the same location you put the SNOM in? |
| 16:21.47 | johndbritton | in my house im using a linux PC with Smoothwall installed... a linux routing package |
| 16:21.55 | jameswf-home | debiano778: if its fxo it uses fxs signalling |
| 16:22.40 | JT | debiano778: how many god damn times do you need to be told? |
| 16:22.47 | JT | debiano778: read the damn book |
| 16:22.54 | debiano778 | can i get incaming call from pstn line with fxs signalling? |
| 16:23.07 | jameswf-home | yes |
| 16:23.29 | debiano778 | good |
| 16:23.30 | JT | debiano778: that's what an FXO port does, so YES |
| 16:23.41 | debiano778 | i procede to configure zaptel.conf |
| 16:24.25 | debiano778 | thanks for all guy |
| 16:25.03 | *** join/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 16:26.10 | shinao1 | sorry whats the default dialling code to find out your extension number in asterisk? im trying to configure asteriskNOW please |
| 16:26.17 | [TK]D-Fender | debiano778: You use an FXO Module (red) for plugging in a phone LINE. It uses FXS (Foreign Exchange Station) signalling because it is acting as a STATION (telephone endpoint). This port is expected to take ring voltage from your telco. |
| 16:26.30 | [TK]D-Fender | shinao1: You're in the wrong channel. Please read the topic. |
| 16:26.31 | johndbritton | [TK]D-Fender: it used to, but now im not able to get it to work at all... i was using SJPhone |
| 16:26.56 | [TK]D-Fender | johndbritton: So not even a soft phone works at that site now? Thats definately networking then. |
| 16:27.20 | jameswf-home | I am working from home with mozphone it seems to do good and no fluff |
| 16:27.38 | johndbritton | well im not sure i have the softphone configured correctly... it worked about a week ago... before i upgraded to leopard and didnt back up my softphon config file DOH! |
| 16:29.48 | Zeeek | so, in one short hour, we'll be starting the VOIP Users Conference: http://voipusersconference.org |
| 16:29.58 | Zeeek | maybe even a pre conference in 45 min |
| 16:30.37 | Zeeek | I never got SJ phone working. Zoiper is a great client IMO |
| 16:30.49 | Zeeek | both SIP & IAX2 |
| 16:30.52 | Zeeek | small footprint |
| 16:31.03 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 16:31.07 | jameswf-home | zoipew i a memory hog. in linux atleast |
| 16:31.21 | Zeeek | aw, who here uses linux? |
| 16:31.37 | Nugget | only when I have to. |
| 16:31.48 | Zeeek | and then only wirth X |
| 16:31.56 | Zeeek | gotta have a GUI! |
| 16:32.09 | jameswf-home | I only use linux |
| 16:32.09 | Nugget | yuck. my linux boxes don't even have monitors attached. :) |
| 16:32.11 | Zeeek | CPM rocks, though |
| 16:32.12 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 16:32.26 | Zeeek | I don't even use a keyboard to program |
| 16:32.31 | cpm | Zeek rocks! |
| 16:32.35 | Zeeek | switches, man |
| 16:32.36 | cpm | Zeeek, even! |
| 16:32.55 | Zeeek | that S100 bus is too cool |
| 16:33.32 | ChrisHardie | Hi, all. Anyone done any interfacing between Asterisk and an irc channel via a bot, for call notifications, etc? |
| 16:34.35 | Zeeek | ChrisHardie I hear Jabber is good for that |
| 16:34.51 | ChrisHardie | yeah, I've heard the same. Trying to stick with IRC if I can, though. |
| 16:35.10 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:35.12 | Zeeek | with a local server? |
| 16:35.17 | ChrisHardie | yes, on a private net. |
| 16:35.28 | Zeeek | interesting idea |
| 16:35.42 | ChrisHardie | thought it would be useful for providing some context on incoming tech support calls, sales calls, etc. |
| 16:35.46 | Zeeek | what about a web server |
| 16:35.49 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 16:36.02 | ChrisHardie | We already kinda do that via the FOP to some degree... |
| 16:36.15 | ChrisHardie | Custom lookup script based on caller id, etc. |
| 16:36.16 | Zeeek | ChrisHardie funny you should mention that. We're talking about an AGI that does exactly that today on http://voipusersconference.org |
| 16:36.22 | Zeeek | Live in about 1 hour |
| 16:36.26 | ChrisHardie | Interesting! |
| 16:36.37 | Zeeek | The example uses just a URL |
| 16:36.50 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 16:36.58 | Zeeek | the URL goes to a page with data looked up in a mysql db |
| 16:37.09 | Zeeek | take a look |
| 16:37.12 | ChrisHardie | nice. That's what we're doing (PostgreSQL, though) |
| 16:37.25 | Nugget | yay postgresql. |
| 16:37.43 | Zeeek | http://vgtc.ibsonecall.com/ |
| 16:37.58 | Zeeek | ^^^^ could eaqsily use wxhatever db you like |
| 16:38.05 | Zeeek | nice typing, too |
| 16:38.19 | ChrisHardie | Sure. I guess that could be used to trigger a write to an IRC channel, if the bot were already connected. |
| 16:38.29 | ai-a | Zeeek: you slap your keyboard there ? |
| 16:38.53 | ai-a | :( |
| 16:38.57 | Zeeek | this is a bargain kb. I didn't realize when I bought it, the rows are in different places! |
| 16:39.31 | ai-a | Zeeek: i had a lovely 'full' keyboard, and got coffee over it, now every keyboard i get is about 4mm shorter, really anoying with typing. |
| 16:39.39 | ai-a | doesn't help being dyslexic either :) |
| 16:40.10 | Zeeek | my fingers are dyslexic |
| 16:40.35 | jameswf-home | i dislexia love much verry |
| 16:40.43 | ai-a | Heh. |
| 16:41.13 | Zeeek | that's why short URL are good: x2z.eu/voip |
| 16:44.07 | ai-a | 2xw.ue.vop ? |
| 16:44.20 | ai-a | bloody brain. |
| 16:44.29 | ai-a | not enough bran. |
| 16:44.35 | [TK]D-Fender | Zeeek: AZERTY KB :p |
| 16:45.11 | Zeeek | http://x2z.eu/voip |
| 16:45.20 | Zeeek | now it's got a line under it :) |
| 16:45.37 | Zeeek | en été, nous avons reįu les pâtes |
| 16:45.49 | *** join/#asterisk asdx (n=diego@adsl-158-12.click.com.py) |
| 16:45.50 | asdx | hi |
| 16:45.51 | Zeeek | Ô galčre |
| 16:45.56 | Zeeek | HI!!!! |
| 16:46.24 | asdx | i'm trying to use voipjet with iax2 and when i connect i get this: [Nov 16 16:45:05] NOTICE[3256]: chan_iax2.c:5252 register_verify: No registration for peer '27404' (from 190.52.158.12) |
| 16:46.40 | [TK]D-Fender | Zeeek: Arret-donc! |
| 16:46.46 | Zeeek | ta gueuele |
| 16:50.21 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:50.56 | Zeeek | bon ben... voilā quoi |
| 16:51.13 | dandre | hello, |
| 16:51.24 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 16:51.29 | Zeeek | dans 20 minutes, la conférence des utilisateurs de voIP commence: http://x2z.eu/voip |
| 16:51.51 | dandre | may I put channel = 1 instead of channel => 1 in zapata.conf? |
| 16:51.55 | jameswf-home | srpms assemble automaticly with rpmbuild so why would you delay their delay |
| 16:51.58 | jameswf-home | idiots |
| 16:52.21 | jameswf-home | s/delay/release/ |
| 16:55.49 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 16:55.58 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 16:56.11 | dandre | may I put channel = 1 instead of channel => 1 in zapata.conf? |
| 16:56.31 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 16:56.58 | muiro | I'm having trouble getting a connector to MySQL to work with unixODBC/asterisk. Can anyone point me at a usefull resource? |
| 16:59.17 | jameswf-home | the more i read the more i think hybrid hosting is a stupid idea |
| 16:59.21 | tzafrir_home | dandre, yes |
| 16:59.36 | dandre | ok thanks |
| 16:59.54 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:00.11 | Zeeek | but it looks way less cool! |
| 17:02.39 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 17:03.45 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-205-101.cmdnnj.east.verizon.net) |
| 17:09.51 | Zeeek | ok, promise this is the last time: IRC #voip-users-conference phone in http://www.voipusersconference.org/access.php - see you there |
| 17:12.09 | Zeeek | in about 8 minutes |
| 17:12.47 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 17:19.42 | outtolunc | .. |
| 17:20.02 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 17:21.07 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 17:23.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:27.20 | horsesgofaster | here is the link http://vgtc.ibsonecall.com/ |
| 17:27.41 | horsesgofaster | for the AGI script example for Caller Id Lookup |
| 17:31.35 | *** join/#asterisk Rhinoo_ah (n=ahonea@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:33.24 | *** join/#asterisk unixdog (n=unixdog@adsl-69-234-184-228.dsl.irvnca.pacbell.net) |
| 17:35.41 | jameswf-home | offtopic: Free barry Bonds |
| 17:36.03 | Maliuta | wtf? |
| 17:36.12 | Maliuta | that is non-sensical |
| 17:36.40 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 17:38.05 | deegan | Why does reload chan_zap.so say "Reconfigured channel X, PRI Signalling signalling" when i have set fxoks in the config. Am i not getting this right or is that actually a strange message to get? |
| 17:38.28 | jameswf-home | deegan check includes |
| 17:39.35 | *** join/#asterisk Tourinho (n=tourinho@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 17:39.48 | Tourinho | hello, good morning |
| 17:41.11 | jameswf-home | well morning anyway |
| 17:44.10 | deegan | thanks jameswf-home, there's nothing in there though that would suggest PRI was chosen. it used to be, so i guess restarting asterisk could solve it. |
| 17:46.33 | Tourinho | What tool can I use to test Asterisk (transcoding)? Is there someone like sipp? |
| 17:49.23 | *** join/#asterisk Aparikh (n=Aparikh@ip-64-32-240-131.iad.megapath.net) |
| 17:50.55 | Aparikh | hi I need help with creating a file that my phones cna ftp to adn that will configure them |
| 17:50.58 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:50.59 | Aparikh | is that difficuly? |
| 17:52.09 | [TK]D-Fender | Aparikh: ...HUH!? |
| 17:53.26 | Darthclue | Aparikh, you want your phones to get the config from an ftp server? if so, what kind of phones do you have? |
| 17:53.59 | Aparikh | I have Polycom 330 |
| 17:54.02 | Aparikh | i mean 301 |
| 17:54.37 | Nuitari | Aparikh: there is a very well written guide on voip-info.org look for Polycom Provisioning guide |
| 17:54.51 | Nuitari | Also on Polycom's support site you will find the 285 pages Administrator Guide |
| 17:55.58 | Darthclue | Aparikh, take a look at http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#7224HowtouseProvisioningCentralBootServe and get the Admin guide from polycom as Nuitari suggested. |
| 17:56.32 | Nuitari | you should also get SIP 2.2 if you are going to be using NAT |
| 17:56.51 | Aparikh | how do I get sip 2.2 ? |
| 17:57.01 | Nuitari | you need to ask whoever sold you the phone to send it to you |
| 17:57.13 | Nuitari | or look on voip-info to see if someone posted a link to it |
| 17:57.27 | Nuitari | polycom are a-holes with their firmwares |
| 17:57.43 | jameswf-home | rtfm... noooooooo |
| 17:57.45 | Aparikh | yea i dont think my reseller will have it |
| 17:57.45 | jameswf-home | lol |
| 17:57.55 | Nuitari | I used to get mine from Atacomm, but they closed :( :( |
| 17:58.43 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.15 | *** join/#asterisk bantu (n=Miranda@p54A32B0D.dip0.t-ipconnect.de) |
| 18:01.51 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 18:02.24 | DarKnesS_WolF | i have a question in sip termination the sip provieder dosn't detect that hte other party hangup and keep the channel open any idea? |
| 18:02.43 | [TK]D-Fender | Aparikh: The should be able to get it for you. |
| 18:02.57 | DarKnesS_WolF | [TK]D-Fender: hello dude :-) long time didn't see u :-) |
| 18:03.21 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:03.24 | [TK]D-Fender | DarKnesS_WolF: Well, I'm here pretty much every day for about 1/2 day as it is... |
| 18:03.32 | [TK]D-Fender | DarKnesS_WolF: No its jsut you :) |
| 18:05.06 | DarKnesS_WolF | [TK]D-Fender: yep true i have been crazy busy |
| 18:05.11 | DarKnesS_WolF | dude this astribank is amazingg |
| 18:06.42 | DarKnesS_WolF | [TK]D-Fender: now i have a sip termination if i did cancel the call "reject it " the channel still open between asterisk the sip provider is that normal ? |
| 18:07.10 | [TK]D-Fender | DarKnesS_WolF: Never ask things like that without showing us the evidence... |
| 18:07.37 | DarKnesS_WolF | [TK]D-Fender: there is no evidence :P i didn't kill anyone :P |
| 18:07.38 | DarKnesS_WolF | ok wait |
| 18:09.15 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.com/mc44937c and i didn't answer the call i just rject it and it feels noting |
| 18:09.55 | [TK]D-Fender | DarKnesS_WolF: thats * vs your PHONE. |
| 18:10.11 | Nuitari | I somehow a polycom phone that doesn't send the username and password with the register |
| 18:10.15 | [TK]D-Fender | DarKnesS_WolF: If * saw your "rejection" it would have killed the channel |
| 18:10.34 | [TK]D-Fender | Nuitari: Sure it does... |
| 18:10.54 | DarKnesS_WolF | [TK]D-Fender: the whole idea i'm trying to make a failover for termination ...so i have one IAX and one sip .. it dials the number using the IAX trunk if the IAX trunk DIALSTATUS busy or wahtever strange it goto failover SIP and the phone rings i rejct the call but the * still open teh channel wth the sip termination . |
| 18:11.04 | Nuitari | no it doesn't, all of the other polycom do, but not that particular phone, for some odd reason |
| 18:11.08 | DarKnesS_WolF | [TK]D-Fender: but how the asterisk will see me rject |
| 18:11.09 | Nuitari | and I'm putting the debug in pastebin |
| 18:11.36 | [TK]D-Fender | DarKnesS_WolF: how are you "rejecting' this call? |
| 18:11.42 | DarKnesS_WolF | it should be the sip termination telling asterisk i rject |
| 18:11.45 | DarKnesS_WolF | i just press cancel |
| 18:11.47 | DarKnesS_WolF | on my cell phone |
| 18:12.29 | [TK]D-Fender | DarKnesS_WolF: Wait... you mean on the ITSP side... well if you hit cancel that might jsut push them to VM or an automated announcment... |
| 18:12.42 | [TK]D-Fender | DarKnesS_WolF: Or your provider isn't sending the right signalling back. |
| 18:13.18 | DarKnesS_WolF | [TK]D-Fender: no VM in my cell phone i.. if i did taht on the IAX termination asterisk getting back BUSY status and in the SIP gets nothing |
| 18:13.26 | DarKnesS_WolF | u mean my SIP provider right ? |
| 18:14.01 | [TK]D-Fender | DarKnesS_WolF: Either |
| 18:14.30 | DarKnesS_WolF | [TK]D-Fender: either what? |
| 18:14.46 | [TK]D-Fender | DarKnesS_WolF: either of the 2 things you just mentioned. |
| 18:15.27 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:15.33 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 18:15.38 | DarKnesS_WolF | u think it might be my cell phone provider ? |
| 18:15.49 | DarKnesS_WolF | i don't thikn so cuz it works with the IAX .. i think it just the SIP termination :-s |
| 18:16.41 | DarKnesS_WolF | [TK]D-Fender: --^ |
| 18:16.53 | Nuitari | http://pastebin.ca/777296 http://pastebin.ca/777301 |
| 18:17.19 | Nuitari | first one is the polycom phone that doesn't send the password/digest, second one is a normal polycom phone |
| 18:18.09 | [TK]D-Fender | Nuitari: pastbin your phone configs & sip.conf entries. |
| 18:18.38 | Nuitari | Ok |
| 18:19.19 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 18:19.58 | Nuitari | [TK]D-Fender: do you want me to trim the comments? |
| 18:20.03 | dijungal | hi.. i have a System command to rename a recording after the Dial command, but the recording does not get renamed because apparently it's not created as yet when the command is ran |
| 18:20.35 | dijungal | any ideas? |
| 18:22.13 | DarKnesS_WolF | [TK]D-Fender: any idea where i can find a nice doc. for termination failover ? i did it using DIALSTATUS but i don't like it :D |
| 18:24.06 | lirakis | dijungal: ha ha.. uhh.. soo .. your recording isnt set up right |
| 18:24.31 | dijungal | lirakis: ? |
| 18:25.03 | lirakis | dijungal: if this is recording on demand.. make sure you have the feature code uncommented in features.conf also automon set in extensions.conf, and the wW parameters passed in EVERY dial string |
| 18:25.27 | dijungal | hmmm.. |
| 18:25.40 | dijungal | Ww? |
| 18:25.47 | lirakis | dijungal: if this is not recording on demand, you will have to give significant detail.. because saying "the file isnt created" is ... not helpful at all |
| 18:25.59 | lirakis | dijungal: yes go read up on the Dial application |
| 18:26.18 | dijungal | i don't want recording on demand.. i want ALL the calls recorded |
| 18:26.26 | dijungal | i know the Ww |
| 18:27.15 | lirakis | dijungal: well you didnt specify that.. or really anything for that matter... so give the nice people here some information and they might be able to help you |
| 18:29.08 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 18:29.36 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 18:30.23 | [TK]D-Fender | DarKnesS_WolF: Doesn't matter if you don't like it, matters if it works. |
| 18:30.28 | [TK]D-Fender | dijungal: pastebin is your friend. |
| 18:30.32 | Nuitari | [TK]D-Fender: http://pastebin.ca/777313 is the config |
| 18:31.12 | dijungal | ok it goes like this... i call a Dial command with gh, then i use the h extensions to rename the wave file created, problem is... the wave file has not been mixed together as yet, so the system command as nothing to rename |
| 18:31.18 | Nuitari | I've removed any option in the sip.conf that is commented out |
| 18:31.38 | dijungal | at what point does the in and out gets mixed together and saved the the HD |
| 18:32.41 | [TK]D-Fender | dijungal: ......... PASTEBIN. |
| 18:33.07 | lirakis | dijungal: dude... you havent even told us HOW you initiate the recording .. pastebin like [TK] said |
| 18:33.41 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:34.27 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 18:34.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:34.50 | deeperror | anyone familiar with rhinosterisk? |
| 18:35.19 | dijungal | http://pastebin.com/d1ef5eecd |
| 18:35.22 | deeperror | just wondering about the recommended procedure to recompile asterisk on a system similar to this |
| 18:37.20 | [TK]D-Fender | dijungal: exten => _23[5678]x,n,Hangup <--- bad. this hangs up the call 100% after even a "g" |
| 18:37.41 | [TK]D-Fender | dijungal: Now show us the call that fails, and not just half of the picture. |
| 18:38.06 | lirakis | dijungal: why do you not just setrecordagentcalls=yes in agents.conf ? |
| 18:38.20 | dijungal | it's set |
| 18:38.20 | DarKnesS_WolF | [TK]D-Fender: true :-d |
| 18:39.01 | lirakis | dijungal: so, that records all calls your agents make... what is the MixMonitor for? |
| 18:39.07 | dijungal | i want the recordings to have a custom name number-date-agent.wav |
| 18:39.17 | dijungal | i can't get the 'agent' part working |
| 18:39.39 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 18:39.52 | dijungal | ok i have setrecordagentcalls=no, because i could not get the name customized |
| 18:41.07 | lirakis | dijungal: you can do that too.. use urlprefix, savecallsin to change the prefix of the file and location |
| 18:41.11 | dijungal | ok so tell me.. if i set the setrecordagentcalls=yes how do i get the custom name? |
| 18:41.24 | lirakis | lirakis: just did ;) |
| 18:41.34 | *** join/#asterisk _pepo_ (n=c9eea608@190.10.187.20) |
| 18:42.22 | _pepo_ | Hi friends |
| 18:42.25 | magic_hat | hey everyone. we have a door-buzzer downstairs in our building... when someone wants in the buzzer calls our * box. The system is supposed to unlock the door if we press '6', but it's not understanding the keypad entries for some reason and doesn't open the door. thoughts? |
| 18:42.42 | _pepo_ | Is there some way to use LDAP with Asterisk 1.4 ? |
| 18:43.18 | lirakis | magic_hat: make sure you are using the same dtmf type .. audio, rfc2833, etc. |
| 18:43.41 | lirakis | _pepo_: can you clarify? |
| 18:44.48 | dijungal | lirakis: urlprefix isn't that for the CDR? |
| 18:44.58 | _pepo_ | I want use use LDAP to register SIP extensions, the same password of the e-mail, (windows) login, etc |
| 18:44.59 | magic_hat | lirakis: any way to tell what dtmf type the buzzer is using? |
| 18:45.35 | lirakis | dijungal: http://www.voip-info.org/wiki-Asterisk+config+agents.conf |
| 18:45.45 | dijungal | that's where i am |
| 18:46.08 | lirakis | dijungal: "The text to be added to the name of the recording. Allows forming a url link. " |
| 18:46.09 | dijungal | and it says "The text to be added to the name of the recording" |
| 18:46.28 | lirakis | dijungal: okay.. soo.. the text.. will be added to the name of the recording... |
| 18:46.30 | dijungal | ok question.. how do i get the date and agent code in there??? |
| 18:46.48 | dijungal | do i use ${EXTEN}...??? |
| 18:46.59 | lirakis | dijungal: the default already contains the EPOCH |
| 18:47.09 | dijungal | hmmm |
| 18:47.59 | lirakis | dijungal: here is a DEFAULT file name from a recorded agent of mine agent-103-1191525122-1866.wav |
| 18:49.08 | dijungal | now have u noticed that to a regular office person who has to look for recordings "1191525122" makes NO SENSE to them |
| 18:49.12 | lirakis | magic_hat: i have no clue.. i mean is the door buzzer thing actually a phone? .. .. short answer no.. i dont know, but fortunately there arent that many choices to try |
| 18:49.32 | lirakis | dijungal: sooo in your rename script.. convert the unix time stame (AKA EPOCH ) to a date |
| 18:49.39 | lirakis | sigh... |
| 18:49.41 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:49.52 | dijungal | hence my mixmonitor |
| 18:49.58 | lirakis | dijungal: no |
| 18:50.15 | dijungal | k |
| 18:51.08 | Nuitari | [TK]D-Fender: any ideas? |
| 18:51.17 | lirakis | dijungal: if you have a rename script that you use already... use the default file name... and let it record... then .. run your little scripty.. that will simultaneously copy the files elsewehere and convert epoch to timestamp... or just scan the directory and put the files into a mysql table..which auto converts timestamps to human readable formats |
| 18:51.25 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:51.30 | dijungal | ok i've added "urlprefix=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})" |
| 18:51.34 | dijungal | lets see what happens :) |
| 18:51.44 | lirakis | oy |
| 18:51.59 | dijungal | hmmm... interesting idea |
| 18:52.14 | [TK]D-Fender | Nuitari: not offhand... |
| 18:52.22 | lirakis | _pepo_: sounds very secure.. ha ha ;P |
| 18:52.31 | dijungal | well the urlprefix is not working so i'll have to go with the script rename idea |
| 18:52.48 | Nuitari | sigh, ok |
| 18:53.53 | Nuitari | I'll hard code the ip and hope it doesn't change too often for now |
| 18:55.30 | DrAk0 | if a company invites you on a .eu country, excluding the hotel, how much per day they should give you? |
| 18:56.27 | Nuitari | DrAk0: depends what you do per day... |
| 18:56.36 | Nuitari | and on the .eu country |
| 18:56.40 | DrAk0 | Nuitari, spain |
| 18:56.48 | DrAk0 | is it âŽ110 per day too much? |
| 18:57.07 | Nuitari | for expenses or as pay? |
| 18:57.19 | DrAk0 | expenses |
| 18:57.47 | Nuitari | car rental? |
| 18:57.50 | DrAk0 | no |
| 18:58.38 | DrAk0 | Nuitari, im staying for meetings |
| 18:58.40 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:58.54 | Nuitari | would be more then enough for me |
| 18:59.18 | Nuitari | I usually get ~100$ per day in for expenses in the US, and I got to fit in a car rental in there |
| 18:59.33 | destructure | nuitari! awesome nick |
| 19:00.20 | _x86_ | anyone familiar with Polycom VSX 7000 series video conferencing units? I know this probably isn't the best place to ask, but figured it'd be worth a shot... I need to setup a centralized directory server for a bunch of these things, and have no idea where to start... do they use LDAP? TFTP? what do i need to do? |
| 19:00.33 | _x86_ | or is it some proprietary server i have to buy from polycom? |
| 19:00.35 | DrAk0 | 100⎠then |
| 19:00.39 | DrAk0 | ? |
| 19:00.57 | hmmhesays | how difficult is it to move from asterisk 1.4 to 1.2? |
| 19:01.16 | *** join/#asterisk CVirus (n=GoD@196.205.192.118) |
| 19:01.22 | Nuitari | DrAk0: of course if they are ok with 110 that's fine too |
| 19:01.39 | Nuitari | hmmhesays: very, all the new stuff in 1.4 won't be in 1.2 |
| 19:01.52 | _x86_ | hmmhesays: you dont want to do that |
| 19:02.26 | hmmhesays | 1.4 is displaying some strange behavior with my ip 501's |
| 19:02.42 | Nuitari | hmmhesays: what does it do? |
| 19:03.01 | hmmhesays | we get a lot of dead air when they hit the answer button, but if you put the call on hold and pick it up again then there is audio |
| 19:03.05 | Nuitari | i got very strange behaviour with 1 of them |
| 19:03.14 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 19:03.15 | teknoprep | hey all |
| 19:03.32 | teknoprep | trying to setup asterfax... i have everything working except outbound fax from email |
| 19:03.36 | _x86_ | hmmhesays: i'm running 1.4.12.1 just fine with a _bunch_ of Polycom IP501's |
| 19:03.37 | magic_hat | anyone tried offsite * hosting? Our linux box keeps crashing and/or taking the phones down when our power goes out for a second. |
| 19:03.48 | teknoprep | when a fax is outbound... it translates it properly from the attachment source |
| 19:04.06 | teknoprep | then i get an error about not being able to leave the channel i have specified |
| 19:04.08 | [TK]D-Fender | magic_hat: cONSIDER THIS MAGICAL THING THEY CALL A ups... |
| 19:04.13 | Nuitari | magic_hat: UPS... |
| 19:04.18 | hmmhesays | I don't know wtf could be causing this then |
| 19:04.24 | hmmhesays | it sure is irritating me |
| 19:04.27 | Nuitari | magic_hat: but I'll be happy to host your pbx on our servers for a fee |
| 19:04.57 | Nuitari | hmmhesays: have you looked for codec issues? |
| 19:05.03 | lirakis | <PROTECTED> |
| 19:05.14 | hmmhesays | all ulaw |
| 19:05.19 | Nuitari | mine doesn't actually send the password to register |
| 19:05.27 | lirakis | [TK]D-Fender: sorry .. didnt see it again you beat me ;P |
| 19:05.32 | Nuitari | hmmhesays: does it happen with other brand of phones? |
| 19:05.34 | hmmhesays | it is strange because only on the initial answer does this happen |
| 19:05.42 | hmmhesays | we only have ip501's there |
| 19:05.57 | Nuitari | hmmhesays: what's the firmware version? |
| 19:06.05 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 19:06.06 | magic_hat | okay, UPS would solve one problem. Our linux box is also just sketch. |
| 19:06.36 | Nuitari | sketch? |
| 19:06.47 | *** join/#asterisk dx9s_home (n=Doug@c-24-18-99-15.hsd1.wa.comcast.net) |
| 19:06.55 | Nuitari | I didn't know Linux could run on an etch-a-sketch |
| 19:07.08 | teknoprep | nvm i figured it out |
| 19:07.15 | magic_hat | lol |
| 19:07.28 | teknoprep | yeah i forgot to add ZAP/ |
| 19:07.33 | teknoprep | in front of my channel name |
| 19:07.45 | teknoprep | actually i am using IAX2 |
| 19:07.47 | teknoprep | instead of ZAP |
| 19:07.56 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 19:07.57 | teknoprep | and have had a 100$ success rate with faxing over IAX outbound |
| 19:08.03 | *** join/#asterisk macros73_ (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net) |
| 19:08.04 | teknoprep | i did a 200 page fax just to test it |
| 19:08.13 | teknoprep | i faxed out a printed pdf book and it didn't fail |
| 19:08.16 | teknoprep | i was very impressed |
| 19:08.17 | Nuitari | teknoprep: over the internet? |
| 19:08.20 | teknoprep | yeah |
| 19:08.23 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:08.25 | Nuitari | teknoprep: how? |
| 19:08.27 | teknoprep | lol |
| 19:08.34 | teknoprep | over comcast cable modem too |
| 19:08.59 | ussrback | hi all |
| 19:09.08 | teknoprep | i have a Pfsense router with QoS properly setup.. don't use the wizard |
| 19:09.12 | teknoprep | then my asterisk box |
| 19:09.21 | teknoprep | for faxing i use voicepulse provider |
| 19:09.25 | ussrback | what is a config file for chan_ooh323 ? |
| 19:09.37 | teknoprep | over IAX2 - which pings for me is less then 20ms on avg... and never more then 22ms |
| 19:09.52 | teknoprep | thats why i used them |
| 19:09.58 | *** join/#asterisk init6 (n=chad@d198-53-149-206.abhsia.telus.net) |
| 19:10.23 | Nuitari | ah |
| 19:10.24 | Nuitari | lucky |
| 19:10.47 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:11.02 | dx9s_home | teknoprep, sent a pdf as a fax .. some software that converts the pdf into some fax equiv call? |
| 19:11.10 | init6 | Is it possible to reserve a number of outgoing channels for a group of users? |
| 19:11.14 | nny | anyone here have some experience with Snom 320s and 200 phones? |
| 19:11.28 | Nuitari | init6: yes |
| 19:11.39 | *** join/#asterisk russellb_ (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:11.39 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 19:11.46 | nny | 300* |
| 19:11.57 | init6 | I mean I use thinktel and have 5 channels but we want to split that over a couple groups in the office. |
| 19:12.13 | init6 | Nuitari, can you point me to some docs or command names? Thanks. |
| 19:12.18 | *** part/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 19:12.26 | dx9s_home | teknoprep, nm see you printed the pdf out .... |
| 19:12.50 | Nuitari | init6: voip-info.org , don't use the channels for other users. You can use contexts in extensions.conf |
| 19:14.13 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 19:14.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:14.37 | ussrback | what is a config file for chan_ooh323 ? |
| 19:14.48 | teknoprep | dx9s_home what do you mean i printed the pdf out? |
| 19:14.55 | Nuitari | teknoprep: in what area are you (so that I get an idea of where voicepulse is good) |
| 19:15.04 | teknoprep | philly / nj area |
| 19:15.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:15.26 | dx9s_home | teknoprep, you said: <teknoprep> i faxed out a printed pdf book and it didn't failu |
| 19:15.34 | Nuitari | ussrback: /etc/asterisk/h323.conf |
| 19:15.44 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:15.59 | dx9s_home | teknoprep, assume that means you had the pdf already printed? |
| 19:16.47 | *** join/#asterisk joelsolanki (i=joelsola@220.224.79.78) |
| 19:16.51 | ussrback | is it necessary to have openh323 installed? |
| 19:17.01 | hmmhesays | Nuitar 2.0 |
| 19:17.05 | Nuitari | ussrback: not if you use the other 323 |
| 19:17.16 | Nuitari | hmmhesays: latest is 2.2 |
| 19:17.26 | ussrback | i have compiled addon |
| 19:17.33 | ussrback | is it enough? |
| 19:17.41 | init6 | Nuitari, I have 3 connections for my provider in sip.conf (one for each phone number). Do I just have to send the Dial() calls in extensions.conf to different SIP connections? |
| 19:17.44 | ussrback | chan_ooh323c |
| 19:18.09 | Nuitari | init6: yes |
| 19:18.16 | hmmhesays | yeah 2.0 should work though |
| 19:18.21 | nny | if anyone has provisioned a snome 300/320 or knows more about them hit me up |
| 19:18.23 | Nuitari | ussrback: I wouldn't know the details as I don't use it |
| 19:18.24 | init6 | Oh, that's perfect. Thank you! |
| 19:19.28 | Nuitari | hmmhesays: I actually had this problem on one phone with 2.0 in an intermitent way |
| 19:19.47 | hmmhesays | yeah it is intermittent here too |
| 19:19.51 | hmmhesays | where do I get the 2.2 firmware |
| 19:20.26 | Nuitari | hmmhesays: you have to ask your polycom reseller |
| 19:21.00 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 19:21.07 | Nuitari | teknoprep: thanks |
| 19:21.40 | CrazyTux | Would anyone happen to know why the uri "89999business2_100" when called as ${EXTEN:5} EQUALS "iness2_100" and NOT "business2_100" like it should? |
| 19:21.46 | CrazyTux | using asterisk 1.4 |
| 19:21.53 | wothinn | hmmhesays: Nuitari: http://www.polycomnz.co.nz/software.html |
| 19:22.07 | Qwell | CrazyTux: because you're doing something else to it afterwards |
| 19:22.26 | CrazyTux | Qwell, hows that possible? |
| 19:22.37 | CrazyTux | Qwell, I'm not doing anything else? |
| 19:22.49 | Qwell | So, if you do |
| 19:22.56 | Nuitari | wothinn: thanks, I needed a new one after mine went tits-up |
| 19:23.01 | Qwell | NoOp(${EXTEN} - ${EXTEN:5}), what does it show? |
| 19:23.34 | wothinn | ... just noticed there's a new bootrom for my phone. Time to read release notes. :-) |
| 19:23.50 | dx9s_home | (open question/survey -- if there is one already point me at it via URL?) What hardware (model is not AS important, vendor is more - IP phones, any gateways to analog realm for legacy phones?, etc.) for the end point device (digital to analog conversion) |
| 19:24.51 | CrazyTux | Qwell, one second, testing right now |
| 19:25.24 | teknoprep | The fax transmission failed due to the following cause: 61 Not receiving expected signals from the far end. this is the error i get from asterfax when trying to send out a fax |
| 19:25.39 | teknoprep | i looked on google and not too much info |
| 19:26.21 | *** part/#asterisk joelsolanki (i=joelsola@220.224.79.78) |
| 19:26.51 | CrazyTux | Qwell, so then something else is matching up with it correct? |
| 19:26.58 | CrazyTux | Qwell, looks like your right |
| 19:32.16 | hmmhesays | 2.5.0(11500-040) & 3.2.2.0019 I guess that is the firwmare and bootroom version on the ip501 |
| 19:32.49 | CrazyTux | Qwell, no something else has to be happening |
| 19:33.01 | CrazyTux | Qwell, I made sure it was the only rule that matches, and it still cuts the 'bus' |
| 19:34.11 | CrazyTux | Qwell, exten => _spa-9000-vm.,1,Set(phone_num=${EXTEN:11}) |
| 19:34.24 | CrazyTux | and exten => _spa-9000-vm.,2,Voicemail(${phone_num}) |
| 19:34.47 | hmmhesays | can you downgrade ip 501 firmware safely? |
| 19:35.25 | [TK]D-Fender | hmmhesays: SIP yes, bootrom, only to a certain point depending |
| 19:36.09 | hmmhesays | ok |
| 19:37.03 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 19:37.40 | hmmhesays | [TK]D-Fender: have you ever had an ip-501 play no audio when the call picks up, but if the user puts the call on hold and picks it back up they have audio? |
| 19:38.11 | [TK]D-Fender | hmmhesays: On some 600's yes, and with no explanation... |
| 19:38.18 | [TK]D-Fender | hmmhesays: a long while ago mind you. |
| 19:38.20 | hmmhesays | hmm |
| 19:38.25 | hmmhesays | i'm having that very problem |
| 19:38.27 | hmmhesays | with 501's |
| 19:38.34 | [TK]D-Fender | hmmhesays: Oh, and 2.5.00 is the boot block. You'll want the SIP version ... |
| 19:38.48 | hmmhesays | ok |
| 19:39.36 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 19:41.47 | |Rain| | is there any way to pull the list of active channels in the dialplan? I need to determine how many calls a SIP phone has up (e.g. count up SIP/123-*). I was using groups in asterisk 1.4.2, but somewhere between 1.4.2 and 1.4.8 groups stopped being local variables and started being combined on transfers |
| 19:41.54 | CrazyTux | Qwell, bus, it is taking out, I'm passing exactly "business2_100" to the Voicemail() app. |
| 19:42.05 | Qwell | CrazyTux: show me the CLI log |
| 19:42.45 | [TK]D-Fender | |Rain|: not in pure dialplan. What you'd do is in AGI parse out "show channels concise" |
| 19:43.00 | CrazyTux | Qwell, http://pastebin.ca/777387 |
| 19:43.26 | [TK]D-Fender | CrazyTux....shmuck :) |
| 19:43.34 | CrazyTux | [TK]D-Fender, is it a bug? |
| 19:43.43 | [TK]D-Fender | CrazyTux : You have forgotten that "b & u are PREFIXES! |
| 19:43.55 | Qwell | ahahaha |
| 19:43.56 | [TK]D-Fender | CrazyTux : for Busy & Unavailable messages! |
| 19:43.56 | Qwell | so is s |
| 19:44.20 | [TK]D-Fender | CrazyTux : 4-star failure for hitting EVERY prefix in a ROW! |
| 19:44.21 | |Rain| | [TK]D-Fender: I need to do this in a context that will be called from app_queue -- spawning a subprocess for every agent for every queue member seems much less than ideal |
| 19:44.27 | CrazyTux | [TK]D-Fender, lol |
| 19:44.32 | CrazyTux | Can I disable it for this, somehow? |
| 19:44.33 | Qwell | CrazyTux: put a | at the end |
| 19:44.39 | Qwell | that should do it |
| 19:44.40 | [TK]D-Fender | CrazyTux : ^^^^^^ |
| 19:44.45 | CrazyTux | at the end of bus| |
| 19:44.47 | CrazyTux | or end of it all |
| 19:44.53 | Qwell | the whole thing |
| 19:44.54 | [TK]D-Fender | CrazyTuxthat'll make it not parse the 1st parm for values. |
| 19:45.06 | Qwell | ...hopefully |
| 19:45.07 | Qwell | :p |
| 19:45.13 | CrazyTux | lmfao wow |
| 19:45.17 | CrazyTux | <--- long WEEK |
| 19:45.19 | CrazyTux | very long week |
| 19:45.24 | Qwell | [TK]D-Fender: good catch |
| 19:45.32 | [TK]D-Fender | |Rain|: no clean way since you want to check this in real-time. |
| 19:45.34 | CrazyTux | you know whats funny |
| 19:45.49 | |Rain| | [TK]D-Fender: boo. thanks. |
| 19:45.50 | CrazyTux | I was actually (honestly) wondering about that, but I was like nooo. I thought the prefix was Voicemail(foo, s) |
| 19:46.11 | [TK]D-Fender | CrazyTux : it is... unless you pass it only 1 parm |
| 19:46.18 | Qwell | besides, why are you using named voicemail extensions? |
| 19:46.21 | [TK]D-Fender | CrazyTux : at which point the old rule sticks |
| 19:46.24 | Qwell | Use contexts - that's the whole purpose of them |
| 19:46.24 | |Rain| | does anyone know why group behavior was changed in 1.4.x to copy over the groups on masqueraded channels? it'll end up setting the same group twice, which is sort of a bug, but fixing that bug doesn't help me here |
| 19:47.20 | [TK]D-Fender | Qwell: Yogi > I'm smarter than the average bear! |
| 19:49.41 | CrazyTux | [TK]D-Fender, didnt seem to do the trick |
| 19:49.49 | CrazyTux | exten => _spa-9000-vm.,3,Voicemail(${phone_num}|) |
| 19:49.58 | Qwell | use contexts |
| 19:50.03 | [TK]D-Fender | CrazyTux : pastebin it. |
| 19:50.20 | [TK]D-Fender | CrazyTux : and pick a message to play |
| 19:50.29 | CrazyTux | [TK]D-Fender, a message to play? |
| 19:50.42 | CrazyTux | [TK]D-Fender, |audio ? |
| 19:50.46 | [TK]D-Fender | CrazyTux : pick which recording to play when leaving your VM |
| 19:50.54 | CrazyTux | whats the default one |
| 19:50.57 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 19:51.03 | [TK]D-Fender | CrazyTux : Do you seriously have no idea about customize VM messages!? |
| 19:51.23 | [TK]D-Fender | CrazyTux : "core show application voicemail" |
| 19:52.25 | *** part/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 19:52.32 | CrazyTux | vm-intro ? |
| 19:52.46 | [TK]D-Fender | ...... |
| 19:52.50 | [TK]D-Fender | CrazyTux : "core show application voicemail" <----------------- |
| 19:53.04 | jameswf-home | asterisk -rx "!!" |
| 19:53.14 | CrazyTux | [TK]D-Fender, it only shows me syntax, not available audio, /var/lib/asterisk/sounds | grep vm |
| 19:54.09 | [TK]D-Fender | CrazyTux : Go into voicemailmain and realize you can record your busy, unavail, temp messages, your name for the directory, etc... when you ENTER voicemail you are supposed to tell it it WHICH message to play based on whatever decision you'd make for such a thing. |
| 19:54.30 | Qwell | OR, make the whole issue moot, and use contexts |
| 19:54.44 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:55.50 | [TK]D-Fender | Qwell: I'd sooner see him learn how to use the applications we've had for years now than to name things the way we think they should be. |
| 19:55.53 | nestAr | !! |
| 20:00.47 | asdx | when i configure asterisk with sip/iax2 and a voip carrier, is there a way i can test the connection from the server? |
| 20:01.08 | asdx | from the asterisk cli |
| 20:02.55 | *** part/#asterisk dx9s_home (n=Doug@c-24-18-99-15.hsd1.wa.comcast.net) |
| 20:03.35 | *** join/#asterisk bobkare (i=bob@cakebox.net) |
| 20:03.51 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 20:05.18 | [TK]D-Fender | asdx: "help dial |
| 20:06.36 | asdx | [TK]D-Fender: thanks |
| 20:07.28 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 20:10.12 | bobkare | Does anybody know of any voip client that runs as a daemon and can be controlled over a pipe or something? Or alternately a module or something that makes it reasonably quick to code a client in perl? I'm making a phone interface but need the control interface to be fairly insulated from the actual client software |
| 20:11.21 | ai-a[gone] | why not use an iax / sip library bobkare ? |
| 20:13.06 | bobkare | module, library, whatever... But I don't have too much time for this, so it needs to have a fairly easy API that's understandable by somebody who doesn't really know anything about voip |
| 20:18.09 | rfxr | help please. I am following the AsteriskTFOT.pdf with asterisk 1.4.13 and cannot get an incoming call to get picked up with the Echo() test |
| 20:18.36 | bobkare | I've been looking at Net::SIP::Simple, but the docs aren't exactly clear on the RTP handling, I would need it to talk to alsa or something instead of raw sound files which is what the samples seem to be assuming |
| 20:22.18 | rfxr | I have a TDM22B, zttool good and ztcfg report channels properly on hardware |
| 20:23.24 | rfxr | followed examples in docs for zapata.conf and extensions.conf for [incoming] context |
| 20:23.56 | rfxr | everything else is source install with samples but no pickup and I am stuck :( |
| 20:24.12 | *** join/#asterisk jpeeler (i=jbpeele@nat/digium/x-f1346119c67f0ae1) |
| 20:24.52 | rfxr | is there a more up to date doc with setup for a TDM22B somewhere? |
| 20:25.19 | [TK]D-Fender | rfxr: pastebin your zaptel.conf, zapata.conf, and extensions.conf. Also the CLI output of your failed call attempt at verbose 10, and the output of "zap show channels" at * CLI |
| 20:25.21 | [TK]D-Fender | ~pb |
| 20:25.21 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:25.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:26.02 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 20:26.33 | rfxr | [TK]D-Fender, ok, I am not sure how to get the verbose 10 output, but when I started asterisk -vvvv, I saw no sign of call activity on console |
| 20:26.49 | [TK]D-Fender | rfxr: "core set verbose 10" |
| 20:27.04 | [TK]D-Fender | rfxr: And pastebin everything you can before concentrating on what you can't |
| 20:27.21 | rfxr | asterisk -r, core set verbose 10...? |
| 20:27.26 | rfxr | ok |
| 20:32.12 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 20:32.24 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 20:35.34 | rfxr | http://pastebin.com/m10730083 |
| 20:35.40 | rfxr | my configs |
| 20:35.48 | *** join/#asterisk ant (n=ant@70.91.72.42) |
| 20:36.56 | [TK]D-Fender | rfxr: Awaiting the rest... |
| 20:36.57 | ant | What's going on with the bsd version of asterisk? |
| 20:37.06 | Qwell | ant: what bsd version? |
| 20:37.10 | Qwell | Asterisk is Asterisk is Asterisk |
| 20:37.52 | ant | well, there was a asterisk-bsd group, with assorted mailing list and what not |
| 20:38.02 | Qwell | yeah, and it was still asterisk |
| 20:38.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:38.22 | ant | long story short, it doesn't build on freebsd right now, and I'm just wondering why |
| 20:38.32 | Qwell | you tell us.. |
| 20:38.36 | Qwell | it should work just fine |
| 20:39.06 | rfxr | [TK]D-Fender, I need to get that in a bit... on phone now :P sry |
| 20:39.28 | ant | right, it *should*, but it doesn't |
| 20:39.35 | Qwell | ant: pastebin the errors |
| 20:40.37 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 20:40.39 | [TK]D-Fender | Qwell: load res_psychic.so! |
| 20:41.15 | ant | the build is looking for asterisk-1.2.12.1-codec-negotiation which is a path file that used to be on unofficial.portaone.com, and on digium's ftp servers, but doesn't appear to be anymore |
| 20:41.26 | Qwell | eh? |
| 20:41.39 | Qwell | ant: get a release from downloads.digium.com |
| 20:41.47 | Qwell | don't use ports - it sucks |
| 20:44.41 | [TK]D-Fender | ant : And... OMG ANCIENT |
| 20:44.55 | ant | yeah, that's pretty old isn't it |
| 20:45.06 | ant | lol |
| 20:45.31 | *** join/#asterisk sebele67 (n=sebastie@des67-1-82-231-81-128.fbx.proxad.net) |
| 20:46.00 | ant | there's no way 1.4.13 will build on freebsd tho |
| 20:46.02 | ant | farg |
| 20:46.10 | sebele67 | hello all |
| 20:46.52 | Qwell | ant: why not? |
| 20:46.58 | Qwell | people do it all the time |
| 20:47.26 | ant | it'll be awesome if it does |
| 20:47.54 | ant | 'linuxy' things often need to be patched for bsd |
| 20:48.20 | sebele67 | quit |
| 20:48.59 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 20:49.14 | Qwell | that's because only like 3 people use bsd |
| 20:49.22 | Qwell | </bsd trolling> |
| 20:49.44 | jameswf-home | I would rather debug windows than boot bsd |
| 20:49.50 | ant | only three users of asterisk, use bsd |
| 20:50.02 | Qwell | no, only three people use bsd |
| 20:50.13 | jameswf-home | <<<<< #asterisk-bsd |
| 20:50.16 | ant | three plus one |
| 20:50.31 | [TK]D-Fender | 33% increase!!!! |
| 20:50.43 | [TK]D-Fender | Fastest growing OS ever! |
| 20:51.26 | jameswf-home | bsd its like linux without the progress... |
| 20:51.59 | jameswf-home | freakin stoner hippy college kids |
| 20:52.08 | ant | it's looking more like windows every day, that's for sure |
| 20:52.09 | Qwell | college kids? bsd? |
| 20:52.13 | Qwell | yeah, in 1970 |
| 20:52.24 | nestAr | vitelity is about to be banned in my world. |
| 20:52.32 | ant | oh yee of little knowledge |
| 20:52.48 | delmar | Digium downloads down? |
| 20:53.02 | Qwell | delmar: no |
| 20:53.10 | *** part/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 20:53.26 | jameswf-home | hey stupid linux is a kernel, the thingy that looks like windows is what we call an application.... |
| 20:53.31 | delmar | Qwell, tested from 2 locations. not serving http content. |
| 20:53.40 | nestAr | i tell them i'm having quality issues.. using IAX. they suggest SIP, I change to SIP it works for a day, then I can't register to their server.. So I switch back to IAX. Advise them of the problem, they reply "We do not suggest the use of IAX" and close the ticket. |
| 20:53.59 | nestAr | that's the way to serve customers, ignore the issue and point the blame. |
| 20:54.02 | delmar | Attempting to download zaptel-fw-oct6114-064-1.05.01.tar.gz, it sits there doing nothing |
| 20:54.06 | *** join/#asterisk DarkDlx (n=darkdll@252.pool85-53-222.dynamic.orange.es) |
| 20:54.14 | Qwell | blah |
| 20:54.20 | jameswf-home | i have to use iax due to the netgear router |
| 20:54.22 | jameswf-home | argh |
| 20:54.28 | Qwell | it's never saying it's done downloading content |
| 20:54.54 | jameswf-home | cheap netgears dont do sip |
| 20:55.04 | delmar | Qwell, if I try to browse to http://downloads.digium.com/pub/ I get nothing.. just 'waiting for downloads.digium.com...' how about you? |
| 20:55.15 | Qwell | I get a 200 OK, then it sits there |
| 20:55.44 | delmar | k. so it's not down but.. its buggered. |
| 20:55.58 | Qwell | something like that |
| 20:56.11 | Corydon76-dig | Comes right up for me |
| 20:56.15 | ant | I just downloaded from that URL 2 minutes ago. |
| 20:56.19 | Qwell | Corydon76-dig: vpn? |
| 20:56.21 | delmar | Qwell, how can I make my zaptel 'make install' work ? |
| 20:56.23 | ant | might be max users |
| 20:56.23 | Qwell | works fine for me too |
| 20:56.27 | Corydon76-dig | Qwell: oh, probably |
| 20:56.37 | Qwell | I think it's just being flakey |
| 20:56.56 | ant | it's everybody jumping on the asterisk bandwagon |
| 20:57.09 | Corydon76-dig | That must be it |
| 20:57.16 | ant | all at once, on a friday |
| 20:57.22 | ant | lol |
| 20:57.32 | jameswf-home | http://downloads.digium.com/pub/ works from here |
| 20:57.48 | *** join/#asterisk sebele67 (n=sebastie@des67-1-82-231-81-128.fbx.proxad.net) |
| 20:57.59 | ant | delmar: go to jameswf's house, it works from there |
| 20:58.10 | jameswf-home | bring beer |
| 20:58.20 | ant | jameswf-home++ |
| 20:58.21 | delmar | jameswf-home, brand? |
| 20:58.37 | sebele67 | hello all |
| 20:58.46 | delmar | ffs |
| 20:58.47 | jameswf-home | corona... mmmm monkey piss and lime |
| 21:00.36 | nestAr | anyone know what would cause choppy audio on a IP550/ |
| 21:00.43 | nestAr | works fine on my IP300 |
| 21:00.45 | delmar | lets assume the box has no internet connection.. what do people do to get the 'make install' part of zaptel to work and get past the issue of no connection to the digium downloads?/ |
| 21:01.31 | jameswf-home | comment the wgets? |
| 21:01.37 | delmar | im going to edit the Makefile. screw this. what a dumb ass idea downloading firmware and BS. |
| 21:01.42 | delmar | yep. |
| 21:03.35 | *** join/#asterisk russellb_ (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:03.35 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 21:03.37 | sebele67 | what type of ip phone did you prefer, what do you think about snom? |
| 21:04.37 | [TK]D-Fender | sebele67: Where are you located? |
| 21:04.51 | nestAr | anyone know what would cause a parked channel to call itself when it times out.. |
| 21:04.55 | nestAr | == Timeout for Zap/8-1 parked on 701. Returning to park-dial,Zap/8,1 |
| 21:04.56 | delmar | sebele67, Polycom |
| 21:05.06 | delmar | sebele67, snom are a bunch of fags. |
| 21:05.28 | sebele67 | [TK]D-Fender strasbourg (france) and you |
| 21:05.29 | jameswf-home | fags? like english smokes? |
| 21:05.41 | delmar | as in a bunch of utter fuckheads |
| 21:05.48 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 21:06.22 | BBHoss | sebele67: aastras are nice as well |
| 21:06.30 | [hC] | anyone have common iperf command lines for testing SIP/IAX from the client side? |
| 21:06.39 | [TK]D-Fender | sebele67: Linksys would probably be the best quality/cost option for you. Polycom is by most accounts the best choice, but I'm quite sure disproportionate where you are. |
| 21:06.58 | sebele67 | delmar sorry i did not understund what you say.. |
| 21:07.38 | delmar | sebele67, the price of Polycom tends to be a bit higher than most too, so I agree with [TK]D-Fender .. Linksys best bet. I use a linksys ATA for my cordless and I'm VERY happy with it. the old PAP2 |
| 21:07.57 | [TK]D-Fender | delmar: that was a region based comment. |
| 21:08.15 | [TK]D-Fender | delmar: Had he said "North America" I wouldn't have suggested Linksys first |
| 21:08.17 | hmmhesays | well i'm moving my ip 501's back to 2.1.2 |
| 21:08.31 | delmar | hmmhesays, really? why? |
| 21:08.56 | [TK]D-Fender | hmmhesays: Mine seems ok on 2.2.0 but I don't exactly have a high call volume going through it |
| 21:09.23 | hmmhesays | i'm trying to solve this freaking no audio on initial answer problem |
| 21:09.54 | delmar | ooo |
| 21:10.08 | delmar | my 501 is still running like.. 2.1.0.2708 |
| 21:10.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:11.19 | *** join/#asterisk sebele67 (n=sebastie@des67-1-82-231-81-128.fbx.proxad.net) |
| 21:11.24 | sebele67 | sorry |
| 21:11.25 | hmmhesays | I can't figure this out it seems so random |
| 21:11.28 | hmmhesays | asterisk 1.4.13 |
| 21:12.09 | sebele67 | delmar : why did you not like the snom? |
| 21:12.21 | sebele67 | i think it's good phones.. |
| 21:13.23 | rfxr | [TK]D-Fender, I think I have found the problem... no zap command via cli? |
| 21:13.40 | [TK]D-Fender | rfxr: sounds like you haven't installed zaptel first before compiling * like you are supposed to. |
| 21:14.10 | rfxr | I did it before, then after but didn't do make clean |
| 21:14.21 | delmar | sebele67, I have dealt with them personally and they are ass holes is why. |
| 21:14.26 | [TK]D-Fender | sebele67: Against snom : poor LCD usage, flakey firmware, second-rate audio quality. |
| 21:14.40 | rfxr | [TK]D-Fender, make clean should do the trick now? |
| 21:14.54 | delmar | They could have the best phones in the world but I wouldn't deal with them because of their attitude. |
| 21:15.03 | [TK]D-Fender | delmar: Please try to state something a bit more quantifiable rather that just saying "they are assholes" |
| 21:15.23 | delmar | [TK]D-Fender, best short version answer I have time for :) |
| 21:15.36 | [TK]D-Fender | delmar: And Polycom could be a total bunch of assholes, but I love their product. Try to seperate the people you speak with and the product they make. |
| 21:15.48 | [TK]D-Fender | delmar: Not worth giving when it says so little.... |
| 21:16.05 | [TK]D-Fender | delmar: If you want to rant on something do provide some specifics :) |
| 21:16.12 | sebele67 | for the sound quality i don't discuss, polycom are very good, but the rest is correct (for me) |
| 21:16.25 | hmmhesays | I wonder if iptables could be making this freak out |
| 21:16.54 | [TK]D-Fender | sebele67: If you already have them and like then, then congratulations. They are not what most would recommend. |
| 21:17.08 | sebele67 | maybe |
| 21:17.17 | [TK]D-Fender | hmmhesays: then your IP 301 would have issues too. Its the same stack... |
| 21:17.27 | hmmhesays | i don't have any 301's |
| 21:17.38 | hmmhesays | there are a couple 320's that displayed that behavior I just found out |
| 21:18.36 | jozu | Somebody knows a command type.. "s,1,save_cdr_here_to_call_later" |
| 21:18.36 | ZenBSDi | [TK]D-Fender, .. I did it!!! |
| 21:18.51 | ZenBSDi | I finally got a working java script that takes input from the keystrokes |
| 21:19.46 | [TK]D-Fender | hmmhesays: Sorry, mixup with nestAr there... |
| 21:19.52 | hmmhesays | oh |
| 21:21.01 | *** join/#asterisk Egonis (n=roman@207.245.216.9) |
| 21:21.32 | Egonis | When loading asterisk 1.2.21.1-r1 I get the message: loader.c:326 __load_resource: /usr/lib/asterisk/modules/res_clioriginate.so: undefined symbol: ast_module_unregister |
| 21:21.32 | Egonis | Nov 16 16:20:03 WARNING[16124]: loader.c:555 load_modules: Loading module res_clioriginate.so failed! |
| 21:21.35 | Egonis | What could cause this? |
| 21:22.50 | ZenBSDi | did you ask google? |
| 21:23.25 | ZenBSDi | google helped me put my asterisk system together and write a java script against the fastagi interface.. |
| 21:23.33 | Corydon76-dig | Egonis: res_clioriginate does not exist in 1.2 |
| 21:23.34 | ZenBSDi | you'd be surprised how smart google is =p |
| 21:24.34 | Corydon76-dig | You'd also be surprised how smart it would be to pay attention to the messages when you type 'make install', especially the lines that start with WARNING WARNING WARNING |
| 21:24.45 | De_Mon | jozu try again and use a real sentance |
| 21:25.41 | jozu | I need it before starting the ivr save the number you have dialed |
| 21:26.07 | De_Mon | jozu no seriously, start over. speak in whole sentances |
| 21:26.15 | Corydon76-dig | De_Mon: specifically, it's because he installed a later version of Asterisk, then installed 1.2 on top of it |
| 21:26.36 | _x86_ | ugh, i dont understand why people still use 1.2 |
| 21:26.38 | De_Mon | Corydon76-dig I mean the WARNING WARNING WARNING part, I don't remember seeing that |
| 21:26.41 | _x86_ | 1.6 is about to come out, no? |
| 21:27.01 | Corydon76-dig | De_Mon: you won't see it if you've cleaned out the modules before you install a different version |
| 21:27.04 | [TK]D-Fender | _x86_: this decade for sure! |
| 21:27.34 | Corydon76-dig | _x86_: more than a month, less than 2 years |
| 21:27.46 | De_Mon | Ohhh |
| 21:28.05 | [TK]D-Fender | heading home, BBIAB |
| 21:31.15 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:31.56 | jozu | someone can helpme, please?, i have a gsm-gateway, but when call to this, only save the CDR of the gsm-gateway, i want to save the CDR mobile made call. |
| 21:32.10 | De_Mon | [Nov 16 16:34:26] DEBUG[6612]: pbx_spool.c:412 scan_service: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/advantage-500.call' |
| 21:32.31 | De_Mon | Is this normal when using a .call file? I usually dont have debuggin going to console |
| 21:33.58 | hmmhesays | I wonder if changing my rtpend to 11000 would cause any issues |
| 21:34.20 | De_Mon | jozu try this exten => s,n,Set(CDR(userfield)=${CALLERID(all)}) I don't know where the phone number is comming from |
| 21:36.19 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:36.40 | iratik | How do you show current sip peers? (like the sip trunks i'm currently connected to? |
| 21:36.41 | iratik | )? |
| 21:37.36 | BBHoss | sip show peers |
| 21:37.45 | BBHoss | ~book |
| 21:37.45 | jbot | 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 |
| 21:38.16 | jameswf-home | ~rtfm |
| 21:38.17 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 21:38.48 | iratik | so this channel is for anything not in the manual? |
| 21:39.33 | jameswf-home | this channel is for asterisk ogers... |
| 21:40.55 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 21:41.16 | jozu | De_Mon dont work, i only see in "userfield" the extension number |
| 21:42.22 | _x86_ | Corydon76-dig: hah |
| 21:43.29 | *** part/#asterisk Egonis (n=roman@207.245.216.9) |
| 21:45.54 | BBHoss | iratik: i gave you the command, the book has all of the commands you need in it, any reasonable person would at least reference the PDF of the book, before asking such a simple question here |
| 21:47.34 | iratik | sorry... i didn't know there was a book |
| 21:47.39 | iratik | until i asked the question |
| 21:49.15 | BBHoss | np |
| 21:49.21 | BBHoss | great thing is its free |
| 21:50.31 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:50.53 | *** join/#asterisk [zender] (n=ender@72.52.23.3) |
| 21:51.15 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 21:51.44 | ant | well, asterisk ogers, and winders whores |
| 21:52.34 | ant | ~ogers |
| 21:52.44 | ant | heh |
| 21:54.25 | nestAr | if i wanted to connect two asterisk servers, but one of them is behind nat, do i use sip or iax? |
| 21:55.15 | nestAr | i'm thinking iax |
| 21:55.25 | BBHoss | nestAr: either protocol will work, but i like iax because it works with NAT better |
| 21:56.01 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-f1a0855144cc3cb0) |
| 21:56.56 | jameswf-home | jbot botsnack |
| 21:56.56 | jbot | jameswf-home: aw, gee |
| 21:57.46 | jameswf-home | jbot help |
| 21:58.31 | _x86_ | nestAr: also, IAX is better from a bandwidth point of view as well... |
| 21:58.53 | _x86_ | nestAr: if your're going to have a good deal of call volume between two machines, IAX trunking will save you a lot of overhead |
| 21:59.46 | muiro | alright, I have a odbc function that returns a date in the format YYYY-MM-DD, I want to use this with SayUnixTime. I'm trying to run it through STRPTIME to get the unix time code with as such ${STRPTIME(${TIME},America/Chicago,%Y-%m-%d)} but I get the error "ouput nothing?!!" |
| 22:00.31 | muiro | I've read as much documentation on STRPTIME as I can find, can anyone help? |
| 22:02.13 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 22:03.21 | jameswf-home | ~hmm... |
| 22:03.46 | jameswf-home | jbot:hmm... |
| 22:07.04 | muiro | no info on STRPTIME? |
| 22:07.11 | Corydon76-dig | muiro: on your console, 'core set verbose 3' and report back what executes when you run that |
| 22:07.28 | Corydon76-dig | muiro: I suspect TIME is not set |
| 22:08.22 | De_Mon | jozu you will need to replace ${CALLERID(all)} with something that has the number you want. maybe ${EXTEN} ? |
| 22:08.23 | muiro | should I pastebin? |
| 22:08.32 | Corydon76-dig | muiro: please |
| 22:08.48 | jameswf-home | jbot: dropdatabase; |
| 22:08.49 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 22:08.53 | jameswf-home | lmao |
| 22:09.29 | De_Mon | What is the recomended way of calling applications that want data, Background(Somefile) or background,Somefile? |
| 22:09.51 | Corydon76-dig | De_Mon: first one. Second one is deprecated. |
| 22:10.02 | De_Mon | Ooohreally |
| 22:10.16 | Corydon76-dig | De_Mon: second one, in fact, will stop working in the next major release |
| 22:10.16 | De_Mon | I just saw it in 1.4's channelvariables.txt |
| 22:10.23 | De_Mon | Good, I hate it. |
| 22:10.31 | muiro | Corydon76-dig: ah, yeah, you're right. The set doesn't have a value, only a key |
| 22:11.00 | muiro | Corydon76-dig: wait, no, that's after |
| 22:11.02 | muiro | pastebin still coming |
| 22:11.15 | [TK]D-Fender | muiro, pastebin your dialpla, the CLI output of your failed attempt, and NoOp all variables before calling your functions so we can see them |
| 22:11.38 | muiro | yeah, I am. The longest part is stepping through the ivr... please hold |
| 22:12.35 | *** join/#asterisk zavoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 22:13.12 | De_Mon | Corydon76-dig so what about option separators Dial(SIP/me|options) or Dial(SIP/me,options) |
| 22:13.20 | jozu | but, de_mon i dont know the mobiles numbers has making call's to the gsm-gateay |
| 22:14.04 | De_Mon | jozu if the caller is not dialing the mobile number into asterisk, then ya, it'd be pretty hard for asterisk to know what it is |
| 22:14.53 | De_Mon | jozu the Dial() Command has an option to send DTMF after connecting a call, maybe you can use that to enter number into * and let * dial number to gsm-gateway |
| 22:15.02 | Corydon76-dig | De_Mon: comma only in the next major release |
| 22:15.35 | De_Mon | hmm, ok. |
| 22:15.55 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 22:16.03 | De_Mon | so has someone gone through all the apps and replaced | with , throughout? |
| 22:16.09 | Qwell | in theory |
| 22:16.14 | Qwell | ... |
| 22:16.32 | nestAr | anyone know what causes asterisk to try and send a timed out parked call to the same channel the call is on? |
| 22:16.33 | jozu | i understand, but all mobiles making call to an asterisk extension |
| 22:18.13 | [TK]D-Fender | De_Mon, talks does funny yoda, hmmmMMMM?!?! |
| 22:18.34 | De_Mon | jozu the call is *phone > * (dial gsm-gateway?) > gsm-gateway (dial mobile) > mobile. |
| 22:18.51 | De_Mon | what am I missing? |
| 22:19.17 | jameswf-home | jbot: trixbox |
| 22:19.18 | jbot | well, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support, and thus you will find little help here for it. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:19.26 | jozu | the call is.. mobile -> gsmgateway (exten 300) -> DIsa trunk sip -> trunk |
| 22:19.40 | jameswf-home | i expected that to be worse |
| 22:19.45 | De_Mon | Ooooh |
| 22:20.27 | De_Mon | heh, get a different gsmgateway? |
| 22:20.57 | De_Mon | this isn't a * question, its a gsmgateway question, "how do I get mobile number from this stupid thing" |
| 22:21.15 | [TK]D-Fender | jameswf-home, Welcome to the kinder / gentler #asterisk. |
| 22:21.22 | [TK]D-Fender | jameswf-home, now GTFO :p |
| 22:21.50 | BBHoss | heh, that bot response sounds very tame |
| 22:21.52 | jameswf-home | jbot: kerry |
| 22:21.52 | jbot | from memory, kerry is something you eat |
| 22:22.16 | _x86_ | wtf... why did they change allison's voice from 1.2 to 1.4 in the voicemail |
| 22:22.23 | muiro | Corydon76-dig: nevermind, it was a stupid mistake. Screen split in the middle of a variable refence and spelled it wrong. Amateur. Thanks for trying to help though. |
| 22:22.24 | _x86_ | 1.2 allison sounded much better, imho |
| 22:22.55 | Qwell | _x86_: are you kidding? Those prompts sounded terrible |
| 22:23.11 | _x86_ | Qwell: i think they sounded better *shrug* |
| 22:23.22 | jozu | i think the question is "Who is calling the extension 300?" |
| 22:23.34 | _x86_ | now. she. sounds. like. she's. talking. with. a. period. between. each. word. |
| 22:23.42 | _x86_ | before it sounded a little more fluid |
| 22:23.50 | jameswf-home | allison has been drinking |
| 22:24.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:24.10 | jameswf-home | I like angry allison :) |
| 22:24.10 | _x86_ | i've always wondered if allison is hot |
| 22:24.10 | _x86_ | she sounds doable |
| 22:24.13 | jameswf-home | jbot allison |
| 22:24.14 | jbot | i heard allison is The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/ |
| 22:25.04 | jameswf-home | she has a honkin nose lol |
| 22:25.12 | jameswf-home | you can see her @ http://theivrvoice.com/ |
| 22:25.51 | De_Mon | jozu if the gsmgateway isn't telling you the mobiles ${CALLERID(all)} you have to fix the gsm gateway afaik |
| 22:25.57 | De_Mon | as far as I know... |
| 22:26.19 | nestAr | i'd hit it. |
| 22:26.21 | nestAr | :) |
| 22:26.25 | De_Mon | I want GladOS voice prompts |
| 22:26.29 | _x86_ | yeah she's doable |
| 22:26.45 | De_Mon | GLaDOS rather |
| 22:27.14 | muiro | actually, de_mon, I was about to mention something like that |
| 22:27.51 | muiro | as a matter of fact I thought of turning on festival and having a bad extension prompt say "you will be baked and then there will be cake" |
| 22:28.04 | De_Mon | hehe |
| 22:28.26 | muiro | also I wanted to see how well festival could do the ending theme |
| 22:29.12 | muiro | this also because I've been coding dialplan just shy of 8 hours today and my brain's about to give |
| 22:29.22 | *** join/#asterisk rjack (n=jack@adsl-ull-107-35.51-151.net24.it) |
| 22:31.52 | *** join/#asterisk MatBoy (n=Matt@wiljewelwetenhe.xs4all.nl) |
| 22:32.15 | *** part/#asterisk ant (n=ant@70.91.72.42) |
| 22:32.16 | MatBoy | hi guys ! Is there a way to call people on email name ? |
| 22:32.23 | MatBoy | so name@domain.com ? |
| 22:32.43 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 22:33.01 | MatBoy | where these email addresses will only be used for calling and not for email :) |
| 22:33.13 | [TK]D-Fender | MatBoy, No, not an e-mail name, but you can dial that as a SIP URI. the call will go to the host, dialing the "name" as the exten. |
| 22:33.38 | MatBoy | [TK]D-Fender, you mean... name.domain.com ? |
| 22:33.57 | MatBoy | or sip.name.domain.com ? |
| 22:33.58 | [TK]D-Fender | MatBoy, Dial(SIP/Joe@somehost.com) |
| 22:34.05 | rfxr | [TK]D-Fender, thanks, got an answer now... had syntax error in config :P |
| 22:34.13 | MatBoy | [TK]D-Fender, yep, understand |
| 22:34.16 | [TK]D-Fender | rfxr, ok |
| 22:34.46 | MatBoy | [TK]D-Fender, but are there no clients where you only need to add the name ? |
| 22:34.54 | MatBoy | so, I want to type as less as possible |
| 22:35.01 | MatBoy | to keep it simple |
| 22:35.45 | [TK]D-Fender | MatBoy, who are "they"? |
| 22:36.06 | De_Mon | as less as possible, how cute |
| 22:36.44 | MatBoy | [TK]D-Fender, where do I typed "they" ? |
| 22:36.55 | MatBoy | -d |
| 22:37.23 | [TK]D-Fender | MatBoy, Opps & whatever. who exactly is using this client, and is it for exclusive use with your server? |
| 22:38.21 | MatBoy | [TK]D-Fender, yep for exclusive use, so let's say I want to call you, I only want to type your name and it will connect to the server and calls you at name@domain.com |
| 22:38.52 | [TK]D-Fender | MatBoy, thats exactly what you'd have to enter into a soft-phone. |
| 22:39.13 | [TK]D-Fender | MatBoy, and it wouldn't be going through your server, it'd dial it direct |
| 22:39.28 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 22:39.35 | MatBoy | [TK]D-Fender, ok, now I have to find a good open source SIP phone :) |
| 22:40.17 | [TK]D-Fender | MatBoy, And who do you think you're going to be calling like that? |
| 22:40.19 | coolfreecode | hey people |
| 22:41.29 | MatBoy | [TK]D-Fender, people that are with the same company or somehting like that and use the same pre-configured client |
| 22:42.26 | [TK]D-Fender | MatBoy, If they're in the same company it doesn't sound like you are trying to call a remote server, but rather jsut another soft-phone associated with yours |
| 22:43.24 | MatBoy | [TK]D-Fender, you need "something" to recognize who you want to call |
| 22:43.34 | MatBoy | on the same network btw |
| 22:44.22 | De_Mon | I just did an 'svn up' on asterisk trunk (who know how old it was!) and it has some files that were somehow not under version control and had to be removed manually? channels/busy.h and channels/ringtones.h |
| 22:44.22 | [TK]D-Fender | MatBoy, Ok, I think there is a big schizm in the description of exactly who these people are you are planning on calling, and what everyone is going to be connected to. |
| 22:45.10 | [TK]D-Fender | MatBoy, Sounds more like a standard * PBX build where you dial by name instead of number and all devices are registered to the same * server |
| 22:45.42 | MatBoy | [TK]D-Fender, ok, let's say... you have 10 persons and they want to call each other, they only need a client where they need too type the name of who they want to call, but it has to be on a server to route it to the name |
| 22:45.47 | MatBoy | do you get it ? |
| 22:47.00 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 22:47.00 | [TK]D-Fender | MatBoy, Your terminology and description are very poor. |
| 22:47.19 | De_Mon | MatBoy are you trying to host multiple domains? |
| 22:47.21 | [TK]D-Fender | MatBoy, answer this : 1 server with all "people connected to it? |
| 22:47.29 | [TK]D-Fender | De_Mon, spare yourself.... |
| 22:47.32 | De_Mon | ie your server is going to have joe@mydomain and joe@yourdomain? |
| 22:47.33 | MatBoy | [TK]D-Fender, yep |
| 22:47.56 | MatBoy | De_Mon, only one domain |
| 22:47.57 | [TK]D-Fender | MatBoy, then all you need to do is have extens for their names and have those call the sip devices you set up. |
| 22:48.05 | [TK]D-Fender | matend of story. no domains, no nothing. |
| 22:48.09 | De_Mon | [TK]D-Fender yeah... you can have him :) |
| 22:48.29 | *** join/#asterisk apardo (i=apardo@68.64.220.87.dynamic.jazztel.es) |
| 22:48.43 | [TK]D-Fender | MatBoy, Same as any other * PBX but instead of dialing "123" on your phone, you can just type in "joe" |
| 22:49.14 | MatBoy | [TK]D-Fender, ok, true, but what if people want to use SIP on their mobile phone ? I need a DNS record in that case, so that was why I asked is it possible to dail to name@domain.com |
| 22:49.22 | MatBoy | [TK]D-Fender, yep |
| 22:50.11 | [TK]D-Fender | MatBoy, and why do you think you need DNS? You don't seem to understand how soft-phones work. You have them register to a SERVER and it sends its calls THERE. you don't have to specify anything when you dial except the exten you are dialing |
| 22:51.08 | MatBoy | [TK]D-Fender, true, I know how they work, bu never mind, I just want to have people don't remember numbersd |
| 22:51.10 | MatBoy | -s |
| 22:51.11 | MatBoy | -d |
| 22:51.14 | MatBoy | +s :) |
| 22:51.40 | [TK]D-Fender | MatBoy, then add named extensions for them. |
| 22:52.13 | MatBoy | [TK]D-Fender, ok, will do that :) thanks ! |
| 22:56.52 | russellb_ | [TK]D-Fender: have you set up a bed for yourself in here yet? |
| 22:56.53 | russellb_ | :-p |
| 22:58.05 | [TK]D-Fender | russellb_, hourly rate baby! |
| 22:58.07 | [TK]D-Fender | ;) |
| 22:58.12 | russellb_ | w00t |
| 22:58.28 | russellb_ | not where i was going with that, but ok :) |
| 22:58.32 | MatBoy | [TK]D-Fender, are you in a DC than ? |
| 22:59.09 | [TK]D-Fender | MatBoy, No, AC only :) |
| 22:59.23 | MatBoy | [TK]D-Fender, DC tastes better ! |
| 23:00.08 | MatBoy | [TK]D-Fender, I hope you up the usage of it ;) |
| 23:04.27 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 23:06.09 | IOscanner | Has anyone had an issue of the notification lighton Cisco 7940 phones not getting enabled. I looked at the SIP traffic and I don't even see the notify message. I have other sites and they are working fine, but not sure why it is not working at this location. |
| 23:06.13 | IOscanner | Any ideas? |
| 23:06.46 | coolfreecode | [OT] Hello guys, who knows how to integrated asterisk with outlook =) |
| 23:07.42 | russellb_ | coolfreecode: where is the cool, free code? |
| 23:08.40 | russellb_ | :( |
| 23:10.45 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:18.04 | teknoprep | can i use a regular fax modem pci card for outgoing faxes on my pbx ? |
| 23:18.12 | teknoprep | does it get recognized as a ZAP card ? |
| 23:20.58 | [TK]D-Fender | teknoprep, No. |
| 23:22.39 | jameswf-home | jbot faxing |
| 23:22.40 | jbot | hmm... faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
| 23:24.20 | pat2man | nice |
| 23:25.33 | asdx | lol |
| 23:26.06 | IOscanner | lol |
| 23:27.17 | jozu | I am struggling |
| 23:27.18 | jozu | <PROTECTED> |
| 23:27.21 | jozu | gsmgateay |
| 23:36.14 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 23:37.05 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 23:40.28 | beek | Hello all. I just set my voice mail greetings in comedian mail. I also set the temporary voice mail message, which is now played by default. How do I get rid of it? The menu tree doesn't seem to give any option for it. (* 1.4.13) |
| 23:41.17 | [TK]D-Fender | beek, it doesn't tell you, but when you go to do the recording again, it will erase it. |
| 23:41.20 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 23:41.42 | [TK]D-Fender | beek, the menu description is simply poor, but the functionalilty is there. |
| 23:44.37 | obnauticus | is there an AGI script out there to send SMS text messages? |
| 23:45.00 | obnauticus | Like via sms to email (ie. <mobile_num>@messaging.sprintpcs.com <-- for Sprint) |
| 23:45.37 | *** join/#asterisk pacneil (n=pacneil@68.15.17.81) |
| 23:45.59 | pacneil | manager.c: Manager received command 'Login' |
| 23:45.59 | pacneil | Nov 16 15:45:38 NOTICE[6851] manager.c: 127.0.0.1 failed to authenticate as 'remote_mgr' |
| 23:46.11 | pacneil | anybody recognize that error? |
| 23:46.56 | Corydon76-dig | It's not an error, it's a notice |
| 23:47.09 | *** join/#asterisk rnovotny22 (n=rnovotny@70.13.150.181) |
| 23:47.33 | pacneil | Corydon76-dig: well it's not saying everything is working ok |
| 23:47.39 | pacneil | so it's an error |
| 23:47.52 | Corydon76-dig | No, it's not |
| 23:47.55 | pacneil | Corydon76-dig: do you know how to fix it? |
| 23:48.03 | [TK]D-Fender | pacneil, And its pretty blatant that you are running an AMI based service on your * server and it isn't set up to auth properly. |
| 23:48.06 | Corydon76-dig | There's nothing to fix |
| 23:48.29 | [TK]D-Fender | pacneil, car to tell us what that process is? |
| 23:48.32 | [TK]D-Fender | care* |
| 23:48.39 | obnauticus | [TK]D-Fender, do you know of any (refer 3:45:04 PST) ^^^ |
| 23:48.56 | [TK]D-Fender | obnauticus, No, and I'd have answered if I did. |
| 23:49.08 | obnauticus | just making sure, you are a bundle of * knowledge |
| 23:49.28 | pacneil | [TK]D-Fender: if I knew what process it was, i'd be able to fix it |
| 23:49.37 | pacneil | it's in /var/log/asterisk/full |
| 23:49.48 | pacneil | and it's filling my logs |
| 23:49.58 | Corydon76-dig | obnauticus: I have one for receiving SMS messages |
| 23:50.08 | Corydon76-dig | but it requires a GSM modem |
| 23:50.12 | [TK]D-Fender | pacneil, ..... and? |
| 23:50.30 | pacneil | and what? |
| 23:50.40 | pacneil | you know as much as I do |
| 23:50.50 | beek | [TK]D-Fender: Thanks! I'll try that to see what happens. |
| 23:50.52 | [TK]D-Fender | pacneil, I asked if you were going to tell us what it is that you installed which is failing to connect to AMI properly... |
| 23:51.02 | Corydon76-dig | pacneil: provide remote root access to your machine and I'll 'fix' it |
| 23:51.14 | BBHoss | probably a gui |
| 23:51.19 | pacneil | that's not likley to happen |
| 23:51.28 | [TK]D-Fender | Corydon76-dig, taking "Mercestes" lessons? ;) |
| 23:51.34 | Corydon76-dig | Then we can't help you |
| 23:51.46 | pacneil | so you're as clueless as I am? |
| 23:51.51 | Corydon76-dig | [TK]D-Fender: no, just feeling generous |
| 23:51.59 | [TK]D-Fender | pacneil, No, we have a clue, and you've given us nothing useful. |
| 23:52.07 | BBHoss | pacneil: have you installed any asterisk tools lately |
| 23:52.16 | pacneil | and neither are the logs |
| 23:52.27 | pacneil | yes, I've installed asterisk software lately |
| 23:52.34 | BBHoss | what exactly |
| 23:52.39 | Corydon76-dig | you give me remote root access and I will find and stop that process |
| 23:53.01 | beek | [TD]D-Fender: Thanks very much -- the option was hidden one level below. |
| 23:53.08 | pacneil | Corydon76-dig: why don't you just tell me what to look for and I'll stop it |
| 23:53.21 | [TK]D-Fender | beek, yeah, it is a definite thing to improve on.... |
| 23:53.41 | [TK]D-Fender | pacneil, We don't know what kind of add-on you've thrown in there! Do you think we're PSYCHIC? |
| 23:53.44 | Corydon76-dig | pacneil: first, obtain 5 years of system administration experience with Linux |
| 23:53.50 | pacneil | sorry but I don't give root access to anything to someone I just started talking to in IRC |
| 23:53.57 | pacneil | Corydon76-dig: I have 10+ |
| 23:54.01 | Corydon76-dig | Then learn C |
| 23:54.17 | Corydon76-dig | If you had 10 years, you wouldn't be asking |
| 23:54.25 | BBHoss | see if netstat can tell you |
| 23:54.40 | *** part/#asterisk pacneil (n=pacneil@68.15.17.81) |
| 23:55.03 | Corydon76-dig | Cute. He sent me 'kiss my ass' via pvt msg |
| 23:55.08 | Strom_M | Lies lies lies yeah! (they're gonna get you) |
| 23:55.19 | [TK]D-Fender | Corydon76-dig, "Why do we always come here? I guess we'll never know. It's like a kind of torture.... to have to watch this show!" |
| 23:55.24 | *** part/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 23:55.28 | jozu | bye, and thanks for all |
| 23:55.29 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 23:55.29 | *** mode/#asterisk [+o file] by ChanServ |
| 23:59.25 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
