| 00:00.32 | ManxPower | How would you fit all of Orange County into Asterisk? |
| 00:00.48 | jameswf-home | ~oc |
| 00:00.49 | Nivex | hey, no compressing my county! |
| 00:01.32 | ManxPower | Nivex: You live behind the Orange Curtain? |
| 00:02.10 | ManxPower | My condolences. 8-) |
| 00:03.03 | Nivex | (NC not CA) |
| 00:03.11 | drmessano | OC? |
| 00:03.14 | ManxPower | Ah. |
| 00:03.18 | ManxPower | Nevermind then. |
| 00:03.19 | jameswf-home | jbot: oc is an Optical Carrier, An OC1 has 672 channels or 44.736 Mbit/s |
| 00:03.20 | jbot | okay, jameswf-home |
| 00:03.34 | drmessano | oh |
| 00:04.01 | ManxPower | jameswf-home: You want Asterisk to support OCx connections? It's good to know you have a rich fantasy life. |
| 00:04.02 | drmessano | I have an OC-30, I just use 3 external chassis full of Rhino T-1 cards, and a bitch switch |
| 00:05.05 | Nivex | you'd need an awful lot of CPU power to process that |
| 00:05.17 | jameswf-home | I would use an 8088 |
| 00:05.40 | drmessano | Rhino cards have low CPU usage |
| 00:05.44 | drmessano | I run a few on 286s |
| 00:06.25 | ManxPower | drmessano: Rhino does not use the GPL Zapata design? |
| 00:06.26 | drmessano | If the call volume becomes an issue, I can hit the 6MHZ/12MHZ turbo button |
| 00:06.39 | ManxPower | Oh, nevermind. You never say anything true. |
| 00:06.52 | drmessano | What tipped you off, the 286 part? |
| 00:06.54 | jameswf-home | ctrl+alt+insert I think that was turbo been a while |
| 00:07.18 | drmessano | Something like that |
| 00:07.24 | jameswf-home | no Rhino isnt another clone |
| 00:08.22 | jameswf-home | i need to teach my dog some tricks |
| 00:08.45 | ManxPower | jameswf-home: you need to write up something technical as to why Rhino is a good card. |
| 00:09.31 | jameswf-home | 30 day gurantee feel free to try it if you dont like it return it.....been that way since day 1 |
| 00:09.31 | ManxPower | I'm just very skeptical about Zaptel compat cards. Most of them are just a copy of the GPLs Zapata design. In fact Rhino is the only one other than Sangoma that claim to not be based on that original board deisgn |
| 00:09.55 | ManxPower | jameswf-home: That is not technical, that is marketing. |
| 00:10.04 | hmmhesays | looks like xspf will be my player of choice |
| 00:10.42 | jameswf-home | marketing or not it lets you make your own judgements rather than a technical spec which is also marketing and would be seen as biased |
| 00:11.16 | drmessano | Technical specs are FUD anyway |
| 00:11.27 | ManxPower | jameswf-home: I see no reason to try a different brand of card when the Sangoma cards work just fine for us. |
| 00:11.28 | drmessano | Everything is as good as everything else |
| 00:11.38 | jameswf-home | If a technical compwrison is to be made it should be by a third party not by said companies if you want it to be credible |
| 00:12.05 | jameswf-home | besides james cant type on his korean keyboard |
| 00:12.16 | drmessano | If it was just about specs, the $50 TDM cards from that chinese place would kick the crap out of the Digium cards.. |
| 00:12.24 | drmessano | They CLAIM to be better heh |
| 00:13.01 | jameswf-home | maybe we will push some cards to the visidial guy let him do a write-up |
| 00:13.17 | ManxPower | And example would be "Digium and Sangoma cards use the Xlinx T-1/E-1 framing chip, which uses more CPU than SuperMegaChip T-1/E-1 framing chip that Rhino uses." |
| 00:14.03 | ManxPower | drmessano: those chineese cards are all the original Zapata design. |
| 00:14.09 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 00:14.16 | ManxPower | Which Digium moved AWAY from because of issues. |
| 00:14.18 | drmessano | I'm sure they are built JUST as well too ;) |
| 00:14.29 | drmessano | yeah |
| 00:15.29 | ManxPower | jameswf-home: What warranty do the Rhino cards have? |
| 00:15.35 | jameswf-home | I had a old digium T card that said Spartan on the Chip... I wanted todraw a stick figure of a guy kicking his pbx into a hols saying "This is sparta" |
| 00:15.48 | jameswf-home | 5 Years from day 1 |
| 00:15.56 | ManxPower | jameswf-home: the old Digium cards were not all that great. |
| 00:16.01 | ManxPower | jameswf-home: so, just like sangoma |
| 00:16.06 | ManxPower | and like Digium now. |
| 00:16.13 | outtolunc | .. |
| 00:16.29 | ManxPower | I still have not seen a reason to spend the work on the learning curve of using a new card. |
| 00:16.33 | jameswf-home | made in america (which probably isnt a selling point if your canadian) |
| 00:16.59 | jameswf-home | we arent using recomissioned window products |
| 00:17.02 | russellb | jameswf-home: how much does rhino contribute to asterisk? |
| 00:17.13 | russellb | (of course, i already know the answer) |
| 00:17.24 | ManxPower | jameswf-home: who uses "recomissioned window products"? |
| 00:17.30 | jameswf-home | as much as is allowed. this conversation has been had in private |
| 00:17.46 | russellb | you're allowed to contribute all you want |
| 00:18.18 | russellb | while we're trying to sell people on hardware brands, i figure it's all fair game |
| 00:18.34 | russellb | so while we're at it, i'll say again, there is one hardware company that actually contributes to asterisk, and the rest do not |
| 00:18.39 | jameswf-home | russellb: I dont discuss corprate politics in public.... |
| 00:18.51 | russellb | well you're making your sales pitch, so i'm making mine |
| 00:18.55 | jameswf-home | say as you will |
| 00:18.59 | russellb | i will |
| 00:19.09 | jameswf-home | I wasnt selling I was answering questions |
| 00:19.22 | russellb | and i'm offering counter-points :) |
| 00:19.30 | jameswf-home | no one asked :) |
| 00:19.50 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:19.52 | hmmhesays | anyone recommend me a better flash audio player than xspf? |
| 00:20.44 | hmmhesays | I'm building a voicemail web interface type thing |
| 00:21.00 | jameswf-home | hmmhesays: steal the google player |
| 00:21.09 | hmmhesays | I need a playlist |
| 00:21.32 | hmmhesays | http://musicplayer.sourceforge.net/ |
| 00:21.40 | hmmhesays | this is good looking, but it doesn't seem to play wav files |
| 00:22.05 | drmessano | Lets just all agree that the X100P is the best card ever, ok? |
| 00:22.13 | russellb | i disagree |
| 00:22.19 | drmessano | heh |
| 00:23.16 | hmmhesays | I guess i'm just going to have to lame my voicemail files |
| 00:23.16 | jameswf-home | well since you can only get x100p now as a clone I would say no as clones are evil.... ever see a positive clone role in movies no |
| 00:23.48 | jameswf-home | thats lame |
| 00:23.53 | drmessano | Fine.. the best card for Asterisk is the Intel 82550 10/100 NIC.. SIP FTFW4EVA |
| 00:24.01 | drmessano | I win |
| 00:24.40 | russellb | oic |
| 00:24.58 | drmessano | heh |
| 00:25.44 | jameswf-home | *CS |
| 00:26.09 | drmessano | I read that Win has something like 57 major bugs to be fixed before 1.0 |
| 00:26.11 | drmessano | Wine |
| 00:26.38 | russellb | that's pretty good |
| 00:26.39 | drmessano | hence the ever increasing number of .9 releases |
| 00:26.49 | jameswf-home | 57 total is better than windos 57 daily |
| 00:27.00 | drmessano | I think they squashed 2 this last update |
| 00:27.35 | drmessano | If anyone is interested, I noticed IPKALL has opened up IAX to everyone |
| 00:27.53 | hmmhesays | whoa lame really screwed with my file |
| 00:29.40 | egecko | MP3Play() function .. is there any way to grab the keypresses during play? |
| 00:37.31 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-11dca4c789f466b0) |
| 00:40.54 | ManxPower | egecko: does "show application mp3play" say you can grab keypresses during play? |
| 00:47.27 | jameswf-home | if you havent already you should get miro |
| 00:47.31 | egecko | ManXPower: The 'show application' command is deprecated and will be removed in a future release. Please use 'core show application' instead. and then using core I get: Your application(s) is (are) not registered |
| 00:47.58 | egecko | which is weird .. .since it does work |
| 00:49.56 | egecko | crap, typo it's mp3player |
| 00:50.21 | egecko | "User can exit by pressing any key on the dialpad, or by hanging up." |
| 00:52.06 | Nivex | "T1 over barbed wire" |
| 00:52.37 | Nivex | http://wiki.notacon.org/AnythingButEthernet/TimminsEntry |
| 00:52.52 | ManxPower | egecko: the answer was less than 5ft away from you the whole time. |
| 00:54.01 | egecko | ManxPower: problem is I can't detect what key was pressed |
| 00:56.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:04.40 | ManxPower | maybe you should just convert your MP3s to some format Playback, Background, etc support and avoid all these problems. |
| 01:06.13 | Nivex | 8K mono WAV. Disk space is cheap. |
| 01:06.40 | ManxPower | Nivex: or convert it to whatever codec is most used for calls on his system |
| 01:08.08 | *** join/#asterisk Uranellus (n=nnkvirc@unaffiliated/uranellus) |
| 01:10.32 | egecko | one of the features of mp3player though is it will play directly from a supplied URL: mp3player(htp://,,,/file.mp3) |
| 01:11.15 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 01:11.47 | frogonwheels | Anybody here know about MWI - message waiting indicator? |
| 01:12.17 | russellb | what about it? |
| 01:12.25 | frogonwheels | I've got connected to (iinet) my voip provider.. but just need to know more about how SIP MWI are read. |
| 01:12.41 | russellb | asterisk does not support receiving MWI over SIP |
| 01:12.43 | russellb | only sending it |
| 01:13.11 | frogonwheels | You sure? isn't there a mailbox= in the sip.conf? |
| 01:13.26 | russellb | yes, there is, but that's for sending MWI |
| 01:13.42 | russellb | i'm quite sure :) |
| 01:13.52 | frogonwheels | ah. |
| 01:13.56 | frogonwheels | bugger |
| 01:14.03 | Uranellus | hello, I'm trying to get an avm bri pci card to work with asterisk .. I used misdn-init scan / config to set everything up (also created simplest helloworld style misdn.conf dialplan).. but when I try to do misdn-init start .. I get http://pastebin.ca/916952 in dmesg, but no call is registered .. |
| 01:15.35 | frogonwheels | russellb: Do you reckon I could configure asterisk to crack open emails containing voice-mail? |
| 01:16.00 | frogonwheels | russellb: no actually - I'd get exim to do that.. put it in a directory somewhere.. |
| 01:16.12 | frogonwheels | russellb: then I should be able to get asterisk to pick them up possibly? |
| 01:16.36 | frogonwheels | (iinet has the option to email voice-mails). |
| 01:18.03 | frogonwheels | if anybody knows of this being done - could you send a url or a few google keywords my way please. |
| 01:18.59 | russellb | asterisk can't determine MWI by reading a mailbox unless you are using IMAP_STORAGE, which requires Asterisk itself storing the voicemails there, so it has special headers on the messages |
| 01:19.30 | russellb | but if you grab voicemail from email and drop them into the right directory structure, asterisk would pick it up |
| 01:20.25 | russellb | doesn't sound easy ... |
| 01:20.51 | *** join/#asterisk BeeBuu (n=beebuu@125.95.20.74) |
| 01:20.55 | frogonwheels | Couldn't I get exim to write the headers myself? |
| 01:22.29 | frogonwheels | - if you could point me at some doco/keywords I can always work out the answer to that myself. |
| 01:22.49 | russellb | maybe you could ... |
| 01:22.55 | russellb | i doubt the headers are documented |
| 01:23.03 | frogonwheels | .. of course.. |
| 01:23.05 | russellb | :) |
| 01:23.07 | frogonwheels | RTS then |
| 01:23.11 | russellb | basically |
| 01:23.16 | *** join/#asterisk iq (n=root@unaffiliated/iq) |
| 01:23.19 | russellb | normally nothing but asterisk internally cares ... |
| 01:23.24 | iq | Hi |
| 01:23.40 | *** join/#asterisk djs (n=djstillm@unaffiliated/djs26) |
| 01:23.51 | frogonwheels | russellb: thanks for your help .. you've probably just saved me 3 hours of googling.. |
| 01:23.52 | russellb | frogonwheels: if you're curious, search for "X-Asterisk" in apps/app_voicemail.c |
| 01:23.58 | russellb | frogonwheels: you're welcome |
| 01:24.00 | frogonwheels | thanks |
| 01:24.02 | russellb | np |
| 01:24.18 | iq | What is the right way to start asterisk so it does not creat so many processes? |
| 01:24.33 | russellb | iq: what you're seeing is threads, not processes |
| 01:24.33 | BeeBuu | anyone know what's the asteriskNOW's root password? |
| 01:24.35 | russellb | and that is expected |
| 01:24.41 | russellb | BeeBuu: no root password by default |
| 01:24.48 | russellb | BeeBuu: use "admin" |
| 01:24.58 | russellb | as the user ... |
| 01:25.03 | iq | russellb: so it is normal to see multiple asterisk when I do 'ps -ef' ? |
| 01:25.26 | russellb | iq: yes |
| 01:25.31 | iq | thanks russellb |
| 01:25.36 | tzanger | ugh mobile (cell) internet in canada sucks ass |
| 01:25.49 | BeeBuu | russellb: i mean the password at https://my-asterisknow:8003 |
| 01:26.24 | ManxPower | tzanger: I guess that would depend on the cell internet technology. Which one do you have? |
| 01:26.36 | tzanger | ManxPower: I'll go whichever way I can get a decent price |
| 01:26.46 | ManxPower | EVDO, 1xRTT, GMRS, or EDGE? |
| 01:27.00 | tzanger | Rogers (GSM) is $65/mo for 1GB |
| 01:27.09 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.102.30) |
| 01:27.15 | tzanger | Telus (CDMA) is $60/mo for the same (plus a $6.95 system access fee) |
| 01:27.19 | ManxPower | EVDO and EDGE are the fastest (EVDO is for CDMA networks, EDGE is for GSM networks) |
| 01:27.25 | tzanger | that's on top of whatever you do for basic cell service |
| 01:27.31 | russellb | BeeBuu: admin:password i think |
| 01:27.35 | tzanger | well HSDSPA is the new hotness for GSM |
| 01:27.40 | tzanger | er HSDPA |
| 01:27.48 | tzanger | but yeah |
| 01:28.03 | tzanger | I tether my Motorla L6 to my laptop, get GPRS and it's like $30/MB without a plan |
| 01:28.06 | ManxPower | I have EVDO Rev A on Verizon |
| 01:28.15 | BeeBuu | russellb: i got you! |
| 01:28.19 | tzanger | I grabbed about 10M this past Friday, that's not gonna be a nice bill |
| 01:28.27 | ManxPower | tzanger: GPRS is *slow* |
| 01:28.32 | tzanger | ManxPower: yes, it is |
| 01:28.36 | tzanger | the phone is EDGE capable |
| 01:28.56 | ManxPower | Go with the company with the best coverage in the places you will be. |
| 01:29.13 | tzanger | ManxPower: oh I'm with Rogers, wife with Telus, I don't much care at this point for data |
| 01:33.36 | bobnormal | gprs is ok here in china |
| 01:33.52 | bobnormal | im not sure but i think it competes with voice for bandwidth though |
| 01:33.59 | trnzmeta | I hear data is expensive in the n america |
| 01:34.10 | trnzmeta | it's getting way cheap in aus |
| 01:34.18 | bobnormal | how much in oz? |
| 01:34.40 | bobnormal | in china it's AU$35 a month unlimited transfer nation-wide |
| 01:34.54 | bobnormal | but you can get a 50 or 100MB plan for AU$3.50 |
| 01:34.55 | trnzmeta | $10/Gb |
| 01:35.03 | trnzmeta | mobile data |
| 01:35.10 | tzanger | fucking hell |
| 01:35.30 | trnzmeta | so yeah no excuses that the techies are uncontactable |
| 01:35.32 | trnzmeta | mwuhauhahua |
| 01:36.22 | bobnormal | hey can you use GSM modems as chan_mobile devices? |
| 01:36.56 | bobnormal | i've got mine rigged up to do incoming/outgoing SMS at the moment but all this talk of mobiles has me thinking about having a backup voice channel in case the E1 fails |
| 01:37.17 | tzanger | bobnormal: yes I've done that, both GSM and CDMA |
| 01:38.31 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 01:39.07 | bobnormal | tzanger: looking at http://www.voipinfo.org/wiki/index.php?page=chan_mobile it only mentions bluetooth |
| 01:39.20 | tzanger | bobnormal: yes, that's what I'm talking about |
| 01:39.25 | tzanger | I have a GSM phone with bluetooth |
| 01:39.29 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 01:39.51 | bobnormal | yeah this is a traditional GSM modem (box with external antenna) |
| 01:40.04 | bobnormal | guess its not covered |
| 01:43.16 | drmessano | chan_mobile is pretty cool |
| 01:43.36 | drmessano | I have some weird issue I am trying to work out, probably with my phone |
| 01:43.38 | tzanger | bobnormal: how do you use it normally? |
| 01:43.44 | drmessano | But other than that, it's cool |
| 01:51.42 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583194.dsl.bell.ca) |
| 01:52.01 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 01:52.01 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:00.27 | bobnormal | tzanger: the gsm modem? just for smstools sms daemon (incoming/outgoing sms with arbitrary charset) |
| 02:02.56 | tzanger | bobnormal: interesting |
| 02:04.31 | tzanger | bobnormal: do you have a model or anything I can look at? |
| 02:10.37 | *** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
| 02:12.07 | riddlebox | does anyone use astercrm? |
| 02:19.39 | *** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
| 02:20.37 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 02:22.58 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 02:33.05 | Uranellus | hey, is there a resource for some "playback" files in german? |
| 02:35.04 | bobnormal | tzanger: cant find the manufacturer name, i think its wavecom, ref no. is g900/1800, silver box with a blue strip. you can google for it. to be honest there's little point in buying one if you have a spare cellphone around though, the only benefit is the external antenna. |
| 02:38.26 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 02:44.37 | *** part/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
| 02:45.02 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 02:49.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:50.20 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 02:52.36 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 02:53.57 | tzanger | bobnormal: interesting, thanks |
| 02:56.39 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 02:58.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:03.25 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:06.47 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 03:07.13 | Robba | hi guys |
| 03:07.45 | Robba | has anyone setup asterisk to function over the internet publically? |
| 03:08.53 | WilliamK | what do you mean publically? |
| 03:08.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:09.00 | WilliamK | most * users use the internet |
| 03:13.30 | ManxPower | Don't you just love it when someone comes in, asks a question, then does not respond? |
| 03:15.07 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 03:16.19 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:20.42 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 03:21.08 | drmessano | What is the meaning of life? |
| 03:21.10 | drmessano | brb |
| 03:21.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:21.53 | drmessano | I get a small amount of amusement over some of the questions |
| 03:22.06 | drmessano | Everyone was a n00b once.. but... |
| 03:22.20 | drmessano | Anyone here using Asterisk with SIP or everyone just using IAX? |
| 03:23.40 | Wayhigh | no one here uses sip.. sip was found to not work well and there's no more devices that use sip at all.. it's now only iax |
| 03:23.50 | drmessano | oic |
| 03:24.08 | drmessano | I see a lot of stuff using SIP, I guess that's for Vonage and Cisco? |
| 03:28.40 | Robba | sorry i don't know how to explain our network topology, it kinda scattered |
| 03:29.18 | Robba | what we have done is put an asterisk box in a collocation building and are trying to connect to it remotely |
| 03:29.27 | Robba | this doesn't seem to be working |
| 03:29.27 | ManxPower | you asshole are going to make some newbie think wrong things because they don't know you are joking. |
| 03:29.45 | *** join/#asterisk Asterlinktechy (n=chito@124.104.101.90) |
| 03:29.46 | drmessano | Me? |
| 03:29.50 | ManxPower | yes you |
| 03:29.55 | ManxPower | and Wayhigh |
| 03:29.56 | drmessano | You need to chill out |
| 03:30.00 | ManxPower | but you are here more often. |
| 03:30.19 | ManxPower | you need to stop spouting out wrong information on the channel because you think it's "funny" |
| 03:30.47 | drmessano | lol |
| 03:30.55 | drmessano | You're a mess |
| 03:31.02 | Asterlinktechy | lmao |
| 03:31.55 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 03:32.46 | Robba | soo.... |
| 03:32.54 | Robba | what just happened here? |
| 03:33.21 | drmessano | Robba, do you have ports open and whatnot? |
| 03:33.24 | ManxPower | Robba: What you are trying to do is very common thing to do with Asterisk. It is complex and may take time to make work. |
| 03:33.58 | *** join/#asterisk JoeShmo (i=joeshmo@pool-129-44-76-164.bing.east.verizon.net) |
| 03:34.03 | JoeShmo | howdy |
| 03:34.25 | *** join/#asterisk Ridgeback (n=Ridgebac@65.127.64.55) |
| 03:35.28 | Robba | as stupid as it sounds its wide open |
| 03:35.53 | drmessano | Does it have a public IP address or is it NAT'ed? |
| 03:35.55 | Robba | i thought it maybe a problem with NAT |
| 03:36.09 | JoeShmo | has anyone ever played with a versa vx-168s phone w/asterisk? just wondering if it works... thinking about doing some playing.. ( http://www.versatek.com/products/vx168s.htm ) |
| 03:36.27 | Robba | the phones are on an internal network and are nat'd but the asterisk box is public |
| 03:37.02 | drmessano | JoeShmo: Is says it supports SIP, so it _sounds_ ok |
| 03:37.34 | drmessano | Robba, make sure nat=yes is set for each of the phones |
| 03:37.44 | Robba | ahhh |
| 03:37.52 | Robba | each phone you say? |
| 03:37.58 | drmessano | yeah |
| 03:38.06 | Robba | what if its set at the top of the sip.conf file? |
| 03:38.07 | *** part/#asterisk Ridgeback (n=Ridgebac@65.127.64.55) |
| 03:38.24 | Robba | does that work? |
| 03:38.38 | drmessano | I believe that's going to be counterproductive |
| 03:38.52 | JoeShmo | well, its the cheapest, most basic bottom of the barrel phone I could find... but it says nothing about full duplex, which makes me wonder about it.. |
| 03:39.18 | Robba | i see |
| 03:39.25 | Robba | i willl give it a try now |
| 03:39.26 | Robba | brb |
| 03:39.28 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 03:39.29 | mosty | JoeShmo, if they are so cheap, buy one and try it out |
| 03:39.56 | drmessano | They spec out ok, who knows how well they work.. but for $45.. its worth a shot |
| 03:40.03 | drmessano | Let me know if it does lol |
| 03:40.30 | JoeShmo | (nod)... then with a new install (not having touched asterisk at all), I wouldnt know if it was the phone, or asterisk if I have problems... |
| 03:40.34 | mosty | the audio quality has been very crappy on most cheap phones i have tried though |
| 03:40.35 | *** join/#asterisk Ridgeback (n=Ridgebac@65.127.64.55) |
| 03:40.38 | drmessano | "Major G.7XX audio codec" |
| 03:41.06 | ManxPower | JoeShmo: don't expect it to work well, but the cost of that problem is low. |
| 03:41.26 | ManxPower | Supaplex: I thought 1.6beta allowed listening on multiple ports. |
| 03:41.46 | Robba | will i have to restart asterisk in order for this to work? |
| 03:41.58 | *** part/#asterisk Ridgeback (n=Ridgebac@65.127.64.55) |
| 03:41.59 | drmessano | reload sip |
| 03:42.00 | Supaplex | the voip-info wiki is not always helpful... |
| 03:42.03 | ManxPower | Robba: no. reload chan_sip.so should work |
| 03:42.16 | JoeShmo | "Major G audio codec"... hmm.. kinda reminds me of that General Failure reading my hard drive.. |
| 03:42.27 | drmessano | lol |
| 03:42.35 | drmessano | Probably G711 only or something |
| 03:42.40 | ManxPower | Supaplex: voip-info is good for finding information that does not apply to the version of Asterisk you are running but doesn't bother to mention that. |
| 03:42.40 | drmessano | But who knows |
| 03:43.02 | Asterlinktechy | I want to connect LinkSysSPA on my public Asterisk... any good references? |
| 03:43.16 | drmessano | SPA what? |
| 03:43.17 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 03:43.34 | Robba | hmmm, i'm using xlite to try and connect to it but just keeps responding with 408 Timeout |
| 03:43.43 | Supaplex | well debian etch has 1.2.13 and I realiase it's ages old. anyways thanks :) |
| 03:43.56 | Asterlinktechy | SPA 3000 |
| 03:44.11 | drmessano | Robba when you said it's open.. is that UDP or just TCP? |
| 03:45.02 | ManxPower | Robba: make SURE there is no firewall on your asterisk server, nor a firewal at the hosting provider |
| 03:45.34 | MaliutaWrk | ManxPower: that's just stupid. How about "make sure your firewall is configured properly" |
| 03:45.51 | ManxPower | MaliutaWrk: That is for AFTER you get it working. |
| 03:46.33 | MaliutaWrk | I'd say before. considering the half life of an unpatched system on the 'net these days |
| 03:47.02 | Robba | yeah its open on both |
| 03:47.29 | Asterlinktechy | drmessano: I can successfully register to my public asterisk using softphones or ip phones(polycom)... |
| 03:47.32 | drmessano | Stupid question, sure they gave you the correct IP? |
| 03:47.41 | drmessano | Asterlinktechy: Sure you can |
| 03:48.05 | ManxPower | Robba: Next time get it all working on the local LAN with the phones and the server before moving the server to the COLO |
| 03:49.22 | Asterlinktechy | drmessano: any good reference? |
| 03:49.41 | Robba | brb i will load x-lite on a server in the collocation |
| 03:50.00 | drmessano | Shouldn't be much to it.. The extensions are NAT'ed, but the * box is on a public IP, correct? |
| 03:50.01 | Asterlinktechy | drmessano: Or I will just use one sip/iax account? |
| 03:50.03 | ManxPower | Robba: it is not a good idea to run any other SIP software on the same computer as Asterisk. |
| 03:51.02 | Robba | nah its a server on the same network as it. not the * server |
| 03:51.18 | *** join/#asterisk iPod-nano (n=david@c-68-43-60-193.hsd1.mi.comcast.net) |
| 03:51.24 | ManxPower | Robba: good. |
| 03:51.50 | iPod-nano | Can anyone explain why my connection cuts out when I use MP3Player() to connect to a SHOUTcast server? |
| 03:52.00 | ManxPower | Robba: this will allow you to eliminate any network, firewall, NAT issues. |
| 03:57.42 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 04:02.34 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 04:06.11 | drmessano | This VX-168S is the ONLY IP Phone this company makes |
| 04:07.03 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 04:12.39 | FireMac | anyone here has fxo fxs adapter for sale? |
| 04:12.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a06a9f5ea5c7d592) |
| 04:13.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-aeba231b14715c21) |
| 04:14.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:15.31 | ManxPower | FireMac: There's this guy called Sipura that has several for sale. |
| 04:16.02 | ManxPower | He has a web site, I think it's sipura.com |
| 04:19.48 | drmessano | FireMac: Your best bet is a new SPA-3102.. I know one place that has them for $70.. and considering how proud people are of their used VoIP gear, after they spent $80 for one, you're not likely to have them come off it for much less than $60.. |
| 04:19.54 | sudhir492 | ManxPower: Thanks for your hint about buddy watch. That worked well! |
| 04:24.16 | jameswf-home | wtf http://videos.fliqs.com/208/2006/04/uber-emo-queen.html |
| 04:25.22 | drmessano | lol |
| 04:25.22 | *** join/#asterisk Robba (n=rob@mail.ampwest.com.au) |
| 04:25.58 | Robba | ManxPower: sorry i forgot to disable IPTABLES |
| 04:26.08 | drmessano | Bingo |
| 04:26.21 | FireMac | thanks drmessano ManxPower |
| 04:26.31 | Robba | now i'm having another issue |
| 04:26.40 | drmessano | which is.. |
| 04:26.45 | Robba | the call can establish but nothing comes through |
| 04:27.28 | drmessano | You have nat=yes on the extension? |
| 04:27.36 | Robba | yep now... |
| 04:27.44 | drmessano | Box is on a public IP |
| 04:28.21 | Robba | i have just been informed now its on a 1-to-1 nat BARGH |
| 04:28.29 | drmessano | heh |
| 04:28.51 | drmessano | ~sipnat |
| 04:28.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:29.07 | Robba | thanks |
| 04:29.55 | FireMac | (drmessano): i have VoIP already but it doesnt support sip. im thinking to buy that 3102 then hook that VoIP adapter to my 3102 to get asterisk running in my house |
| 04:29.59 | FireMac | does that work? |
| 04:31.13 | drmessano | On paper it does |
| 04:31.42 | FireMac | why? |
| 04:32.08 | FireMac | the output of VOIP ATA is analog |
| 04:32.13 | drmessano | Your VoIP gateway is technically just another PSTN connection, and you're piggybacking the SPA-3102 > Asterisk.. So sure |
| 04:32.41 | FireMac | great |
| 04:33.19 | FireMac | which means, if asterisk is running, and configured to have extension numbers, then that's gonna be possible, right? |
| 04:33.30 | drmessano | Yeah |
| 04:34.15 | ManxPower | FireMac: Yes, you can link two systems together like that, but because it's all analog you will have issues. |
| 04:34.31 | FireMac | like what ManxPower? |
| 04:35.04 | ManxPower | Many voip boxes do not signal when the far end hangs up, so asterisk would not know when the call ended. |
| 04:35.37 | ManxPower | This is not usually an issue because YOUR side ends the call and all is good, but if the call goes into voicemail, the voicemail system will happily record hours of nothing. |
| 04:36.13 | ManxPower | you can sort of deal with this by using the callprogress and busydetect options in Asterisk, but they are buggy unreliable and often just randomly disconnect calls. |
| 04:36.15 | FireMac | that's if my voip doesnt signal hang up. right? |
| 04:36.56 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 04:37.58 | FireMac | does asterisk support freebsd? |
| 04:40.53 | bbryant | yes |
| 04:41.07 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 04:41.09 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 04:41.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:43.32 | FireMac | oki, coz im installing freebsd rightn now |
| 04:45.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:46.17 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 04:50.21 | drmessano | Must be a slow weekend.. VoIP-News has a story about how VoIP calls over the public internet aren't secure.. Yawn |
| 04:50.41 | drmessano | I guess they feel the need to reuse that story once every few months |
| 04:58.54 | WilliamK | I'd have to debate the security idea VS analog POTS |
| 04:58.56 | *** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 04:59.13 | WilliamK | alot less likely that someone knows howto capture packets than to use a butt set and jack in to your line |
| 05:00.15 | WilliamK | and if using an enecryption method with VoIP, it's probably even more secure than the telco handling the call |
| 05:00.41 | drmessano | Agreed |
| 05:01.14 | drmessano | It's also a little harder than plugging your laptop into comcast and turning on wireshark |
| 05:01.56 | WilliamK | alot harder, and with advances in technology even that could be circumvented |
| 05:02.00 | drmessano | Whatever happened to E-mail being insecure? |
| 05:02.01 | WilliamK | aka VLAN |
| 05:02.25 | WilliamK | depends on who's handling your emails :) |
| 05:02.30 | WilliamK | mine are encrypted |
| 05:02.30 | drmessano | People still use POP and SMTP without TLS |
| 05:02.41 | drmessano | But it's page 20 news |
| 05:03.16 | drmessano | I guess proclaiming VoIP as insecure makes for better reading that E-mail now |
| 05:03.20 | drmessano | than* |
| 05:03.58 | WilliamK | honestly media outlets think voip is this new kid on the block |
| 05:04.05 | WilliamK | most don't realize it's been around a while |
| 05:04.44 | JT | at least voip passwords aren't sent in the clear |
| 05:04.49 | WilliamK | kinda like why pick on the old kid on the block, always find the newer one that's more interesting |
| 05:04.53 | drmessano | Damn true, JT |
| 05:05.40 | drmessano | Did you hear that http isn't encrypted either? |
| 05:05.42 | drmessano | :) |
| 05:07.20 | drmessano | Sad thing is, once encryption is in place and becomes common, the media will draw a line in the sand between providers that care if youre secure and not |
| 05:07.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:09.11 | WilliamK | honestly I can't wait till the media picks up that some encryption methods can be unencrypted on the fly |
| 05:09.18 | WilliamK | that'll make for some news |
| 05:09.29 | WilliamK | maybe that is |
| 05:10.46 | WilliamK | anyway, bbl |
| 05:11.04 | drmessano | Later |
| 05:11.17 | *** join/#asterisk jeddev (n=jeds@74.85.65.58) |
| 05:13.48 | jeddev | This might be a dumb question, and happy to be pointed to any reference material that I have not found. I'm trying to compile one of our Asterisk 1.2 apps under 1.4, i've changed everything I found to match the framework. Got things to compile fine, but when loading the module I get ....asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_ourmodule.so: undefined symbol: symbol name , same message for each of our modules with the name and symbol changi |
| 05:14.50 | jameswf-home | ~dumb |
| 05:14.51 | jbot | ACTION thinks that sumone is drunk, idiotic or a robot abuser. |
| 05:17.41 | jeddev | ? |
| 05:21.21 | *** join/#asterisk ta^3 (n=tacvbo@189.136.30.223) |
| 05:22.01 | supa_disko | anyone with hands on experience converting cisco 79XX v7 to sip? |
| 05:22.12 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 05:22.51 | supa_disko | there's a plethora of info from cisco that doesn't work |
| 05:23.04 | supa_disko | it's like the're deliberatly making it hard :p |
| 05:32.52 | Corydon76-dig | supa_disko: Convert first to SIPv2... v3... v4... v5... v6... v7 |
| 05:33.11 | Corydon76-dig | No, I'm not joking |
| 05:41.12 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:41.12 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 05:41.45 | supa_disko | ok so I'm using image loader right, I have to find sipv2 image cisco |
| 05:42.31 | supa_disko | I read it somewhere but was hoping this isn't true |
| 05:42.45 | supa_disko | Corydon76-dig --> you wouldn't have these image files handy? |
| 05:44.06 | Corydon76-dig | No, I do not. I refuse to use Cisco phones |
| 05:44.26 | Corydon76-dig | Partially because of the crap they pull with licensing |
| 05:46.48 | supa_disko | cheers |
| 05:52.49 | *** join/#asterisk steliosk (n=Stelios@athedsl-279671.home.otenet.gr) |
| 05:56.18 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-73-120.tpgi.com.au) |
| 05:56.53 | Hyphenex | hi gang, I'm trying to follow the book on asterisk, but I'm stuck here |
| 05:56.54 | Hyphenex | *CLI> dialplan reload |
| 05:56.54 | Hyphenex | No such command 'dialplan' (type 'help' for help) |
| 06:04.26 | Quintin | So, I've had messing with asterisk and using it for my business on my todo list for awhile now. However at the moment my "phone" system is skype pro, for which I pay insanely low price of $3/ month for. What are some reasons I should switch to asterisk anyway? |
| 06:06.57 | Hyphenex | Quintin: because your not happy with Skype |
| 06:08.45 | lmadsen | Hyphenex: sounds like an version of 1.4, or more likely... you're running 1.2 |
| 06:08.55 | Quintin | Well, skype is fine.. I want all the cool hacks that can be done with asterisk though. I'd also like to keep my skypein number, which is used for business. don't know if that is doable or not though |
| 06:08.57 | lmadsen | s/an/an old |
| 06:09.17 | jameswf-home | ~skype |
| 06:09.18 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 06:09.48 | Quintin | all stuff I know... and you misspelled proprietary.. :D |
| 06:09.56 | Hyphenex | hyphenex:~# asterisk -V |
| 06:09.56 | Hyphenex | Asterisk 1.2.13 |
| 06:10.01 | Hyphenex | That'd do it I guess |
| 06:10.07 | Hyphenex | I just built with apt-get |
| 06:10.23 | lmadsen | Hyphenex: 2nd version of TFoT is based on 1.4. Use the first edtion of the book for 1.2 |
| 06:10.30 | jameswf-home | isnt the root word of skype a racial slur |
| 06:10.30 | lmadsen | 1.2.13 is VERY old |
| 06:10.52 | Hyphenex | ahh, I should be asking in #debian how to upgrade, right? |
| 06:11.02 | lmadsen | compile from source |
| 06:11.13 | jameswf-home | to upgrade ... make && make install |
| 06:11.28 | jameswf-home | doh ./configure |
| 06:12.03 | lmadsen | I think you can do make && make && make install to get around running ./configure (since basically the first make runs the ./configure for you :)) |
| 06:12.15 | lmadsen | which obviously makes no sense, but works |
| 06:12.54 | Hyphenex | so I can't use the super-easy apt-get? |
| 06:12.58 | jameswf-home | ~idnms |
| 06:12.59 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this channel? Nothing |
| 06:16.54 | jameswf-home | jameswf-home: has decided jameswf-home will talk about jameswf-home in the third person |
| 06:16.59 | MaliutaWrk | jbot: 2 words "midget fetish" |
| 06:23.05 | drmessano | lol |
| 06:24.19 | drmessano | So heres a question |
| 06:24.24 | Hyphenex | Garrgh, the new asterisk not not building from source, is there any easy way to install asterisk 1.4? |
| 06:24.46 | drmessano | If Skype opened up and allowed users to connect using something other than the Skype client, would then be less evil? |
| 06:26.50 | Hyphenex | configure: error: C compiler cannot create executables |
| 06:36.04 | *** join/#asterisk trnzmeta (n=bleh@people.guard.com.au) |
| 06:36.44 | Supaplex | Hyphenex: which distro are you on? |
| 06:37.06 | Hyphenex | Supaplex: Debian |
| 06:37.37 | Hyphenex | Supaplex: Dinner time, be back soonish |
| 06:37.38 | Supaplex | Hyphenex: apt-get build-dep asterisk will install builddeps of asterisk. |
| 06:37.56 | Supaplex | you're better off making a "simple sid backport". ask a bot in #debian how |
| 06:38.30 | Hyphenex | Thanks Supaplex :D |
| 06:41.58 | drmessano | http://www.destructoid.com/elephant/ul/13769-noscale-GH_SITAR_HERO.jpg |
| 06:42.42 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:45.52 | *** join/#asterisk happi (n=chatzill@unaffiliated/happi) |
| 06:55.10 | bobnormal | drmessano: nice poster :) |
| 07:01.49 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 07:02.24 | Hyphenex | Supaplex: Ever seen this one before? |
| 07:02.25 | Hyphenex | make: Warning: File `Makefile' has modification time 5.7e+08 s in the future |
| 07:02.51 | marc7 | chan_sip.c handle_response_invite: Received response: "Forbidden" ///// what am I missing in sip.conf? |
| 07:03.03 | Supaplex | Hyphenex: what's date -u show? is it close to mine? |
| 07:03.06 | Supaplex | Mon Feb 25 07:01:40 UTC 2008 |
| 07:03.31 | Hyphenex | Mon Jan 8 05:37:55 UTC 1990... Noooooo :P |
| 07:03.40 | Hyphenex | hyphenex:~/asterisk-src/asterisk-1.4.18# time |
| 07:03.40 | Hyphenex | real0m0.000s |
| 07:03.40 | Hyphenex | user0m0.004s |
| 07:03.40 | Hyphenex | sys0m0.000s |
| 07:04.23 | Supaplex | then FIX your date |
| 07:04.54 | Hyphenex | I might have to do that :P |
| 07:09.40 | marc7 | if I'm setting externip in sip.conf.... shouldn't all outbound SIP traffic be coming *from* that IP address? |
| 07:09.57 | marc7 | because i have that IP created as an ethernet alias |
| 07:10.12 | marc7 | so eth0 is 123.45.67.8, and eth0:1 is 123.45.67.9 |
| 07:10.44 | marc7 | regardless of externip, traffic seems to be leaving on 67.8 instead of 67.9 |
| 07:13.12 | *** part/#asterisk jst_home (n=jst@72.171.214.48) |
| 07:13.13 | Supaplex | outbound routing on linux is a odd creature |
| 07:14.21 | MaliutaWrk | how so? |
| 07:15.25 | MaliutaWrk | marc7: techincally you should nat to the address you set in "externip", but that will just tell asterisk what to put in the rtp portions of the packets. The other end should then use that IP to send stuff to |
| 07:15.52 | Supaplex | unless the app binds to that ip+port, the kernel just picks one (I forget the consice behaivor) |
| 07:16.09 | nixguy | marc7: your default gw will still just be assigned to one interface, even if they are all connected to the same network |
| 07:16.11 | marc7 | the problem is I need to bind to multiple addresses with the current configuration. |
| 07:16.15 | marc7 | ahhh |
| 07:16.21 | marc7 | see, that's my problem. |
| 07:17.11 | marc7 | anyway, it isn't the end of the world, I'll get around to reconfiguring my network one of these days |
| 07:20.28 | *** join/#asterisk J4zen (n=Jvan4zen@a82-95-153-17.adsl.xs4all.nl) |
| 07:21.32 | tzafrir | Supaplex, my personal backport: deb http://updates.xorcom.com/rapid etch main # i386 only. |
| 07:22.47 | Supaplex | tzafrir: which version is it? Hyphenex was the initial inquiror (yea, bad sp, too sleepy) |
| 07:22.58 | SwK | who's doing CNAM dips for asterisk boxes now? (ie for when you have a provider that doesnt send CNAM when you get DIDs from them) |
| 07:23.47 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.108.206) |
| 07:24.02 | tzafrir | I build it off of the version that goes to Unstable. With a few minor changes (no imap stuff) |
| 07:25.48 | Supaplex | ahh. I'm actually half way done backporting upstream |
| 07:26.19 | Hyphenex | Supaplex: that worked like a charm. She's building now :) |
| 07:28.52 | Supaplex | Hyphenex: a deb-src one (like simple sid backport) or just plain source? |
| 07:30.02 | Hyphenex | the deb-src, then built asterisk from source :) |
| 07:30.35 | nixguy | Hyphenex: i might be saying the opposite of some of the pro's here but i think using the debian builds has some pro's over source sometimes |
| 07:31.20 | Supaplex | by and large, ditto. |
| 07:31.22 | Supaplex | imo |
| 07:31.51 | nixguy | for example you dont need to spend the rest of your month trying to get h323 support working |
| 07:31.57 | Supaplex | but then i'm almost done getting asterisk-1.6.0-beta4 built as a backport for etch. :) |
| 07:35.47 | tzafrir | Supaplex, I'm interesting to see your work |
| 07:36.14 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 07:36.40 | Hyphenex | damn, I can't open it after a build |
| 07:36.42 | Hyphenex | ERROR: Could not open H.323 listener port on 1720 |
| 07:36.42 | Hyphenex | [Feb 25 18:37:40] ERROR[3433]: chan_h323.c:3162 load_module: Unable to create H323 listener. |
| 07:36.52 | Supaplex | tzafrir: http://riddle.daxal.com/~supaplex/etch-backports/builds/logs/?C=M;O=D |
| 07:37.03 | tzafrir | Hyphenex, another Asterisk is running? |
| 07:37.15 | Hyphenex | that's what I thought, hold on... I think I can check with ps |
| 07:37.32 | tzafrir | Supaplex, in fact, I suspect 1.6 is quite ready for an upload to experimental |
| 07:37.41 | Hyphenex | nothing shows up in ps -A | grep asterisk |
| 07:37.47 | Supaplex | quite possibly. |
| 07:37.58 | Supaplex | I'd love to collaborate with another $brain on the matter :) |
| 07:38.15 | Supaplex | maybe we can get a NMU into experimental |
| 07:40.56 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 07:40.59 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
| 07:41.02 | bobnormal | hows this for telecomms |
| 07:41.14 | bobnormal | i call my carrier to enable international calls on my e1 |
| 07:41.25 | bobnormal | and they go "sorry we cant do that, we have a problem with our system" |
| 07:41.35 | bobnormal | wtf? |
| 07:42.05 | SwK | bobnormal, sounds like most carriers heh |
| 07:42.17 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-110-161.bltmmd.east.verizon.net) |
| 07:44.11 | Hyphenex | ... help |
| 07:45.32 | Hyphenex | it doesn't look like asterisk is running |
| 07:45.41 | Hyphenex | maybe I should try and switch to Fedora |
| 07:47.27 | Hyphenex | asterisk is not running |
| 07:47.58 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:55.51 | tzafrir | Hyphenex, a more direct test: netstat -lntup | grep 1720 |
| 08:07.33 | *** join/#asterisk oej (n=olle@81.253.87.55) |
| 08:16.52 | WilliamK | bobnormal, how about being able to dialout but all your incoming numbers give an operator msg |
| 08:17.02 | WilliamK | :) |
| 08:18.51 | tzafrir | Hyphenex, if asterisk is not running, you'll probably find the direct cause for that in /var/log/asterisk/messages |
| 08:20.20 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:22.13 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 08:23.16 | *** join/#asterisk shtoom (n=godson@59.93.120.51) |
| 08:27.56 | bobnormal | williamk: that happened to me setting up the e1, the reason was that i hadn't configured an extension in the dialplan that matched exactly the incoming number |
| 08:28.16 | bobnormal | WilliamK: only found out when i ran 'asterisk -crvvvvvv' and tried to dial in. |
| 08:30.55 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:37.37 | Hyphenex | tzafrir: asterisk is not running... I don't know where to start. http://paste2.org/p/14281 |
| 08:38.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:41.41 | mvanbaak | I wonder, Digium T120p or sangoma A101 |
| 08:45.11 | Hyphenex | It's ok, you can all stop panicking now. All I had to do was re-load the samples |
| 08:45.56 | SwK | hah |
| 08:46.06 | SwK | who says we were panicing? |
| 08:46.50 | tzafrir | Hyphenex, do you really need h323? |
| 08:47.01 | bobnormal | mvanbaak: i've had all good experiences with sangoma |
| 08:47.23 | Hyphenex | tzafrir: Probably... not |
| 08:47.25 | SwK | must be a buncha aLawers in here |
| 08:47.42 | bobnormal | mvanbaak: their support people are really responsive |
| 08:47.56 | tzafrir | So just add another unload line in modules.conf |
| 08:48.03 | tzafrir | For a faster load time |
| 08:48.26 | Hyphenex | okies, I might do that later, for now, I can continue on that book |
| 08:48.27 | SwK | I know its a long shot... but anyone have GR1188 doc they wanna share a little info from? |
| 08:48.31 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 08:52.31 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:56.50 | g0mb0 | hi there |
| 08:57.11 | g0mb0 | I have symmetric NAT problem here |
| 08:57.25 | g0mb0 | does anybody know how to overcome symmetric NAT? |
| 08:59.02 | tzafrir | SIP? IAX? |
| 08:59.15 | g0mb0 | SIP |
| 08:59.25 | tzafrir | Can you control any of the NAT routers? |
| 09:00.00 | g0mb0 | I can control my NAT, I forwarded RTP port and it is working |
| 09:00.46 | g0mb0 | but is it possible to make it work without such config? |
| 09:01.58 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:02.05 | dandre | Hello, |
| 09:02.48 | dandre | I have a zaptel device, How can I know weather there is a hardware EC? |
| 09:03.18 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 09:10.28 | dandre | I have a zaptel device, How can I know weather there is a hardware Echo Canceler ? |
| 09:10.35 | Uatec | *whether |
| 09:11.50 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:14.36 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 09:19.52 | dandre | sorry ;-) |
| 09:20.13 | dandre | do you know the answer Uatec? |
| 09:20.26 | Uatec | no :( |
| 09:20.35 | dandre | ok |
| 09:20.49 | nixguy | u think you need it?! |
| 09:21.02 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:21.29 | dandre | I have echo during the first few seconds of a call |
| 09:21.48 | nixguy | you could add a delay before the call gets connected |
| 09:21.50 | nixguy | :) |
| 09:23.01 | dandre | humm will look at this but this doesn't really answer the question. Do I have EC on my tdm800 board? |
| 09:28.10 | Paladine | if you go look it up on digiums site it should tell you if it has echo cancellation |
| 09:28.36 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:29.07 | *** join/#asterisk RypPn (i=TuMbL@80.177.214.249) |
| 09:30.56 | *** join/#asterisk ArchSSM (n=tommy@host-81-191-139-130.bluecom.no) |
| 09:32.28 | BBHoss | anybody got any idea what this might be: http://flickr.com/photos/whitneygh/2275980572/sizes/l/ |
| 09:32.53 | obnauticus | Is it possible to get GSMFR for asterisk? |
| 09:33.11 | Paladine | whats gsmfr? |
| 09:33.16 | obnauticus | another codec. |
| 09:33.25 | Paladine | dunno |
| 09:35.25 | obnauticus | BBHoss that looks really weird. |
| 09:35.32 | BBHoss | tell me about it |
| 09:35.40 | obnauticus | Its the pulse of the future. |
| 09:35.54 | BBHoss | if you knew what time domain was youd be surprised |
| 09:36.02 | BBHoss | i think its this: http://www.timedomain.com/products/PLUSOverview.pdf |
| 09:36.11 | obnauticus | what does it do |
| 09:36.19 | obnauticus | WO |
| 09:36.24 | BBHoss | time domain makes UWB technology |
| 09:36.25 | tzafrir | obnauticus, patented? |
| 09:36.25 | obnauticus | That IS whta it is |
| 09:36.30 | obnauticus | gsmfr |
| 09:36.38 | obnauticus | i don't know i need to encode and decode from it thoug |
| 09:36.49 | tzafrir | wide-band gsm is patented, IIRC |
| 09:36.56 | obnauticus | Is that what it is? |
| 09:37.05 | BBHoss | i was just browsing through flickr and stumbled on to it |
| 09:37.07 | tzafrir | I have no idea |
| 09:37.09 | BBHoss | never seen one before |
| 09:38.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:39.18 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 09:40.14 | BBHoss | something else they make: http://www.youtube.com/watch?v=looiiLJLmJ0 |
| 09:40.19 | obnauticus | Also: In asterisk how do i tell what the value is of a certian device? |
| 09:41.05 | tzafrir | obnauticus, gsmfr is what we use |
| 09:41.09 | tzafrir | http://en.wikipedia.org/wiki/Full_Rate |
| 09:41.17 | obnauticus | how can i use it |
| 09:41.22 | obnauticus | like through asterisk. |
| 09:42.49 | dandre | How can I configure a zap channel to use alaw instead of ulaw? |
| 09:43.07 | obnauticus | tzafrir you mean on asterisk right? |
| 09:46.56 | nixguy | dandre: hmm maybe your sip.conf |
| 09:47.17 | nixguy | dissalow=ulaw |
| 09:47.26 | nixguy | allow=alaw |
| 09:47.38 | nixguy | disalow=ulaw |
| 09:47.41 | nixguy | one s |
| 09:48.03 | nixguy | and 2 ll |
| 09:48.06 | nixguy | i spell like crap |
| 09:48.07 | nixguy | :) |
| 09:48.49 | dandre | if I do this: |
| 09:48.49 | dandre | zap show channel 7 |
| 09:48.49 | dandre | I get this: |
| 09:48.49 | dandre | Default law: ulaw |
| 09:49.26 | dandre | I'd prefer to find a configure option for zap devices if available |
| 09:49.44 | nixguy | i understand, that knowledge is beyond me im afraid |
| 09:50.18 | obnauticus | tzafrir are you there? |
| 09:52.12 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 09:52.28 | tzafrir | obnauticus, yes, in Asterisk |
| 09:52.46 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 09:52.50 | tzafrir | dandre, in zaptel.conf |
| 09:53.06 | dandre | I haven't found |
| 09:53.11 | tzafrir | look at the sample zaptel.conf in the zaptel tarball |
| 09:53.21 | dandre | ok |
| 09:53.24 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:56.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:57.20 | kombi | http://pastebin.se/193454 <- could someone have a look? Simple problem but I am stalled... |
| 09:58.44 | kombi | the patterns in the beginning are of course skipped this way. How would it need to be? |
| 10:01.11 | *** join/#asterisk oej (n=olle@217.113.81.14) |
| 10:04.31 | kombi | found it, never mind! |
| 10:04.33 | sbingner | kombi, different contexts... |
| 10:04.42 | sbingner | and order matters |
| 10:05.12 | kombi | sbingner: actually, I just put the whole dispatch section on prio 5 now..;) |
| 10:05.50 | sbingner | I suspect you did something bad? can you show me? |
| 10:06.18 | kombi | sbingner: maybe I did, but it's working..;) |
| 10:06.20 | dandre | tzafrir I haven't seen in zaptel.conf any reference to a or mu law encoding |
| 10:06.42 | sbingner | kombi, you probably have something where it will just fall off the end of the dialplan if your called exten doesn't match |
| 10:07.53 | sbingner | well, this is how I would do what you're talking about anyway |
| 10:08.05 | sbingner | which doesn't exactly fix what I mentioned but heh |
| 10:08.25 | sbingner | http://pastebin.se/193456 |
| 10:08.25 | kombi | sbingner: go on then |
| 10:09.09 | kombi | sbingner: makes perfect sense though |
| 10:09.11 | sbingner | but you of course want an invalid-exten handler included in [dispatch] in case the called-number didn't exist ;) |
| 10:09.25 | kombi | sbingner: I was gonna say just that.. |
| 10:10.02 | kombi | sbingner: I believe there is some standard extension vor that? |
| 10:10.20 | sbingner | your way is probably not bad... but it just rubs me the wrong way to get to prio 5, THEN hit an invalid exten lol |
| 10:10.27 | sbingner | exten => i |
| 10:10.32 | sbingner | is invalid exten |
| 10:11.20 | kombi | sbingner: that's it! Gotta go now, thanks sbingner! |
| 10:11.46 | sbingner | np |
| 10:12.48 | *** join/#asterisk oej (n=olle@217.113.81.14) |
| 10:13.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:18.02 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:19.38 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:19.41 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:22.58 | *** join/#asterisk Asterlinktechy (n=chito@124.104.101.90) |
| 10:36.12 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 10:42.12 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:45.47 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 10:49.32 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:51.39 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-73-120.tpgi.com.au) |
| 10:51.51 | Hyphenex | hey, a little question about this section (in the book). what does this do? |
| 10:51.53 | Hyphenex | [internal] |
| 10:51.53 | Hyphenex | exten => 500,1,Verbose(1|Echo test application) |
| 10:51.54 | Hyphenex | exten => 500,n,Echo() |
| 10:51.54 | Hyphenex | exten => 500,n,Hangup() |
| 10:52.12 | mvanbaak | echo test application |
| 10:52.24 | mvanbaak | it will echo back to you whatever you yell into the handset |
| 10:52.25 | Hyphenex | what happens if it just keeps dialing and does not pick up? |
| 10:52.57 | mosty | there is no dial |
| 10:53.16 | mvanbaak | you have a sip entry with 'context=internal' ? |
| 10:53.29 | mvanbaak | are you using that sip account to dial 500 ? |
| 10:53.50 | Hyphenex | I don't think so... |
| 10:53.59 | Hyphenex | I see Executing [500@phones:2] Echo("SIP/1000-082108e8", "") in new stack in the log |
| 10:53.59 | mvanbaak | then it wont work |
| 10:54.02 | mvanbaak | ah |
| 10:54.06 | mvanbaak | ok, that works |
| 10:54.27 | Hyphenex | it never picks up though, it just keeps calling |
| 10:54.42 | Hyphenex | then exits on non-zero? |
| 10:54.56 | mvanbaak | try to add this after the Verbose and before the Echo: exten => 500,n,Answer() |
| 10:57.34 | Hyphenex | ahh cool |
| 10:57.36 | Hyphenex | that works :D |
| 10:57.44 | Hyphenex | ... I think the books needs updating then :P |
| 10:57.56 | Hyphenex | Thanks nvanbaak |
| 10:58.16 | Hyphenex | mvanbaak* |
| 11:00.06 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 11:02.23 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 11:04.43 | ruied | did any one tried the Siemens Optipoint VOIP phone with * and the siemens DECT/SIP phones ??? |
| 11:05.26 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 11:09.47 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 11:10.15 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 11:12.06 | mvanbaak | ruied: the dect phones were tested by a couple of our customers and they work great |
| 11:16.11 | ruied | mvanbaak, nice, the dect phones are stilysh ! I've seen yesterday that the optipoint also work with asterisk, but didn't find much info about those phones. They look really greate, if they work well with * they could end up to be my product of choice... |
| 11:26.55 | *** join/#asterisk guillote_GNU (n=guillote@host144.201-253-38.telecom.net.ar) |
| 11:27.26 | *** join/#asterisk chigital (n=chigital@91.90.144.8) |
| 11:27.52 | Hyphenex | I so love this PBX system |
| 11:31.24 | mvanbaak | ask it to marry you ;) |
| 11:32.16 | Hyphenex | very not funny mvanbaak |
| 11:32.40 | mvanbaak | gheh |
| 11:33.03 | Hyphenex | ahh well, I think that will do me for tonight |
| 11:33.09 | Hyphenex | I'm off to bed me thinks |
| 11:34.04 | mvanbaak | sweet dreams |
| 11:34.14 | Hyphenex | Thanks, you too |
| 11:34.30 | mvanbaak | Mon Feb 25 12:34:30 CET 2008 |
| 11:34.34 | mvanbaak | it's noon here |
| 11:34.37 | tzafrir | mvanbaak, back from fosdem? |
| 11:34.41 | mvanbaak | tzafrir: yup |
| 11:34.57 | mvanbaak | working from home today. slow day to recover ;) |
| 11:36.26 | Daviey | mvanbaak: Should have made yourself known, i would have said Hi |
| 11:37.04 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 11:38.03 | mvanbaak | Daviey: I did |
| 11:38.14 | mvanbaak | Daviey: I've been hacking on some asterisk code together with oej |
| 11:38.39 | mvanbaak | we had fun |
| 11:38.46 | oej | Me have been hacking with mvanbaak |
| 11:38.50 | oej | Still in Brussels |
| 11:39.13 | BBHoss | damn i wish i got to travel around |
| 11:39.42 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 11:39.46 | mvanbaak | oej: you visited thi qtopia talk ? |
| 11:40.06 | oej | Missed it |
| 11:40.11 | mvanbaak | ah, me too |
| 11:40.17 | mvanbaak | it was too early |
| 11:40.19 | mvanbaak | I arrived at 10 am |
| 11:40.33 | mvanbaak | but I did go to the xen and virtualbox talks |
| 11:40.37 | mvanbaak | very interesting |
| 11:41.38 | mvanbaak | expected ';' after SKREP |
| 11:41.39 | mvanbaak | HAHAHAHA |
| 11:41.45 | mvanbaak | nice |
| 11:43.31 | Daviey | xen was quite high level tho |
| 11:43.44 | mvanbaak | uhhuh. I liked that |
| 11:43.45 | *** part/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 11:43.48 | Daviey | i expected so new exciting stuff, and it was an intro |
| 11:43.54 | Daviey | so/some |
| 11:44.35 | mvanbaak | what was good was the promise virtualbox would stay virtualbox and not some dead project of sun |
| 11:44.40 | Daviey | But then, he was a good presenter for xen |
| 11:45.01 | mvanbaak | and the note that he is going to opensource the USB support in virtualbox |
| 11:45.03 | Daviey | were you fully convinced? :) |
| 11:45.32 | mvanbaak | pretty much |
| 11:50.03 | SteveTotaro | mornin |
| 11:50.22 | *** join/#asterisk quigon (n=matias@32.59.64.129) |
| 11:50.30 | *** join/#asterisk eharris_ (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 11:51.00 | mvanbaak | hey SteveTotaro |
| 11:51.42 | SteveTotaro | 0635 hours here, coffee is a brewing |
| 11:51.57 | SteveTotaro | too cold, i can't wait till spring |
| 11:56.05 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.242) |
| 11:57.55 | sudhir492 | where do you live Steve? |
| 11:58.09 | sudhir492 | it is pretty warm in Orlando |
| 11:58.52 | Uatec | hey, from the cli, is it possible to connect to ongoing calls together? |
| 12:04.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.53 | siya | ruied: which Siemens phones you referring to? |
| 12:06.02 | SteveTotaro | DC/Baltimore corridor |
| 12:06.20 | siya | I have a S450-IP but I wish they'd put a real WMI in instead of the pointless missed call thingy that there now |
| 12:07.01 | SteveTotaro | I used for work for a company based out of Oviedo, FL |
| 12:07.29 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:09.16 | sudhir492 | I just moved from Chantilly, VA to Lake Mary, FL in Oct. |
| 12:10.21 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 12:11.51 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:22.24 | Wayhigh | steve.. what the hell are doing doing awake? |
| 12:27.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:30.12 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 12:30.23 | nixguy | just curious |
| 12:30.25 | nixguy | with sip |
| 12:30.43 | nixguy | is all communication between endpoints routet through the asterisk servers? |
| 12:30.47 | nixguy | or is there a handover? |
| 12:30.54 | nixguy | so they talk peer to peer?! |
| 12:31.30 | Mw3 | it isnt called handover, but reinvite |
| 12:31.40 | nixguy | ok |
| 12:31.43 | nixguy | but it happens then? |
| 12:31.53 | Mw3 | two sip endpoints can send their rtp media directly to each other |
| 12:31.54 | nixguy | once you connect phones talk to each other directly? |
| 12:32.02 | nixguy | if they can that is |
| 12:32.19 | Mw3 | but it can happen the other way (throught the sip proxy) |
| 12:32.41 | Mw3 | you can control it in sip.conf. canreinvite=yes/no |
| 12:32.43 | mosty | nixguy, it's possible to do that, but NAT tends to break it easily, and so most people disable that feature |
| 12:32.55 | nixguy | im using layer3 without nat |
| 12:33.05 | nixguy | curious cause the siptrunk shanghai-sweden |
| 12:33.09 | Mw3 | some features of asterisk wont work with reinvite |
| 12:33.21 | nixguy | with one phone reigstered in shaghai calling through the trunk to a phone registered in sweden |
| 12:33.27 | nixguy | just sounds to good to be true |
| 12:33.31 | nixguy | (like no delay :) |
| 12:33.42 | *** join/#asterisk stoffell (n=stoffell@231.41-201-80.adsl-dyn.isp.belgacom.be) |
| 12:34.02 | nixguy | this probably just gives me a hardon cause im still sucha * noob |
| 12:49.31 | mvanbaak | if the phones are in the same subnet they can do RTP peer-to-peer |
| 12:53.36 | nixguy | mvanbaak: they are not |
| 12:56.45 | dandre | I have false incomming call detection on my zaptel fxo channel. When the call is hungup, I get rather imediately a false incimming detection. How should I investigated to find the reason and the correction for this?: |
| 12:56.45 | dandre | <PROTECTED> |
| 12:56.45 | dandre | <PROTECTED> |
| 12:57.51 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 13:00.35 | mvanbaak | nixguy: then the RTP will go through asterisk, and not peer-to-peer |
| 13:01.04 | nixguy | mvanbaak: ok but RTP is that like all communication? or just some controll-channel thingy?! |
| 13:01.25 | mvanbaak | nixguy: all |
| 13:01.28 | mvanbaak | actually |
| 13:01.34 | mvanbaak | RTP is the audio/video |
| 13:01.44 | nixguy | what does rtp stand for? |
| 13:01.50 | nixguy | real time protocoll? |
| 13:01.51 | JT | real time protocol |
| 13:01.52 | nixguy | :) |
| 13:01.53 | nixguy | k |
| 13:02.05 | nixguy | thnx |
| 13:02.07 | mvanbaak | the control channel thingie always goes through asterisk |
| 13:02.18 | mvanbaak | rtp == Realtime Transport Protocol |
| 13:02.40 | Uatec | SIP = Session Initiation Protocol |
| 13:02.51 | mvanbaak | yeah |
| 13:02.52 | Uatec | i.e. the one that sets up the RTP connections between the phones and stuffs |
| 13:05.48 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 13:07.18 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:08.45 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:18.08 | ruied | siya, siemens optipoing 500 |
| 13:18.22 | *** join/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 13:20.34 | ruied | optipoint |
| 13:25.54 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:26.01 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:26.26 | stansmith | hello |
| 13:26.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:27.17 | *** part/#asterisk vap0rtranz (n=jpittman@r75-110-17-157.gvllcmtc01.gnvlnc.ab.dh.suddenlink.net) |
| 13:28.13 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:28.58 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:30.55 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 13:32.07 | *** join/#asterisk eth01 (n=foobar@gentoo/user/eth01) |
| 13:43.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.119.187) |
| 13:43.14 | dandre | <PROTECTED> |
| 13:43.14 | dandre | -- Hungup 'Zap/7-1' |
| 13:43.14 | dandre | -- Starting simple switch on 'Zap/7-1' |
| 13:43.25 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:48.20 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:48.24 | stansmith | zaptek |
| 13:48.26 | stansmith | zaptel |
| 13:51.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:54.35 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.242) |
| 13:55.03 | tzafrir | dandre, what version of zaptel? |
| 13:56.47 | dandre | tzafrir 1.4.7 |
| 13:57.45 | tzafrir | What card? |
| 13:57.59 | dandre | tdm800p and tdm400p |
| 13:59.24 | tzafrir | both? |
| 14:00.40 | dandre | I had problems with tdm400 and asterisk 1.2 and now I have built another system with asterisk 1.4 and tdm800. |
| 14:00.40 | dandre | both show the issue |
| 14:01.44 | *** join/#asterisk ddunavant (n=David@pool-96-231-78-28.washdc.east.verizon.net) |
| 14:02.03 | *** part/#asterisk eth01 (n=foobar@gentoo/user/eth01) |
| 14:05.34 | stansmith | mm yummy |
| 14:10.40 | *** join/#asterisk ShaunWing (n=chatzill@196-209-54-238-rrba-esr-3.dynamic.isadsl.co.za) |
| 14:10.43 | *** join/#asterisk [gnubie] (n=[gnubie]@cm152.gamma183.maxonline.com.sg) |
| 14:11.08 | ShaunWing | Hi Guys, Anyone know what the exten => h |
| 14:11.09 | ShaunWing | does? |
| 14:11.21 | ShaunWing | I believe it can be used for debugging |
| 14:11.45 | shasta | ~book |
| 14:11.46 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:11.52 | shasta | ShaunWing, ^^^ |
| 14:11.54 | Katty | hai. |
| 14:12.08 | stansmith | exten => h? |
| 14:12.29 | ShaunWing | yes |
| 14:12.33 | oej | ~seen kpfleming |
| 14:12.36 | jbot | kpfleming is currently on #asterisk-dev, last said: 'please take a look through 'svn annotate' and figure out where that came from it possible, but as best i can tell the unlock/handle_init_event/lock stuff can just go away'. |
| 14:12.45 | ShaunWing | exten => h,1,NoOp(EXTEN : ${EXTEN}) |
| 14:13.31 | dandre | tzafrir I suspect that I must tune the timing configuration (debounce, wink, ...). Is there some doc on those parameters? |
| 14:13.47 | stansmith | ShaunWing: h is hang up |
| 14:13.55 | stansmith | ShaunWing: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 14:13.57 | ShaunWing | tx |
| 14:14.20 | stansmith | you could just replace "h" with "s" or some extension you created |
| 14:16.02 | Katty | jbot: hai? |
| 14:16.02 | jbot | Helicopter Association International. The Human Awareness Institute |
| 14:16.11 | Katty | jbot: :< |
| 14:16.12 | jbot | [<] redirection of stdin to a program |
| 14:16.17 | stansmith | LOL |
| 14:16.19 | Katty | jbot: /bonk |
| 14:18.07 | [gnubie] | anyone here familiar with the res_muted? muted.conf has a limited documentation |
| 14:21.58 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 14:23.35 | *** join/#asterisk Angel-D (n=angeldav@190.79.252.29) |
| 14:23.51 | stansmith | coffe hot |
| 14:25.20 | dandre | where can I find docs on timing parameters (wink, debounce, ...) for zap channels? |
| 14:25.53 | hi365 | <PROTECTED> |
| 14:25.53 | hi365 | where is that sert & how do i get rid of it? |
| 14:28.08 | [TK]D-Fender | hi365: unload the ztdummy module |
| 14:28.23 | hi365 | hmm, |
| 14:28.34 | hi365 | where is it set to run in the first place? |
| 14:28.44 | dandre | I have the following scenario: |
| 14:28.44 | dandre | - inbound call to my zap fxo channel |
| 14:28.44 | dandre | - routed to an extension within my dialplan |
| 14:28.44 | dandre | - if this extension goes on hook, the channel seems to be correctly hungup |
| 14:28.44 | dandre | - but is still active and it is routed another time though the dialplan |
| 14:29.40 | [TK]D-Fender | hi365: its a kernel module |
| 14:33.08 | hi365 | [TK]D-Fender: in /etc/sysconfig/zaptel i have this: #MODULES="$MODULES ztdummy" |
| 14:33.17 | hi365 | shouldnt this prevent it from loading? |
| 14:33.22 | [TK]D-Fender | hi365: "rmmod ztdummy" <- |
| 14:34.04 | hi365 | [TK]D-Fender: thanks. is that enough to prevent it from loading next time? |
| 14:34.13 | [TK]D-Fender | hi365: Should be. |
| 14:34.18 | hi365 | thanks m8! |
| 14:34.34 | Katty | mister fender. |
| 14:35.26 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:35.46 | [TK]D-Fender | Katty: Mew. |
| 14:35.59 | Katty | allo. |
| 14:36.08 | Katty | how're you? |
| 14:36.11 | stansmith | is it poor style to have your dialplan in production with every priority numbered, or should i be using |
| 14:36.14 | stansmith | ? |
| 14:36.16 | stansmith | "n" |
| 14:36.31 | stansmith | any advice? |
| 14:37.08 | [TK]D-Fender | stansmith: No issue with ahrd numbering and I always do it myself |
| 14:37.15 | [TK]D-Fender | hard* |
| 14:37.16 | stansmith | great thanks |
| 14:38.25 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:38.58 | Katty | [TK]D-Fender: how're you? |
| 14:39.14 | [TK]D-Fender | Katty: Doing pretty good, yourself? Still looking to move out? |
| 14:39.24 | Katty | [TK]D-Fender: move out? |
| 14:39.27 | Katty | [TK]D-Fender: oh, to canada? |
| 14:39.44 | [TK]D-Fender | Katty: Well you said you were think more like Europe. |
| 14:40.03 | Katty | yeah. uk or canada. |
| 14:40.12 | Katty | not sure, depends on foreign policy changes. |
| 14:40.21 | Katty | today i'm just trying to keep breakfast down. |
| 14:41.11 | stansmith | ive thought about moving to canada a few times |
| 14:41.13 | stansmith | seems safer up there |
| 14:41.58 | Katty | [TK]D-Fender: don't worry, it's not morning sickness. |
| 14:42.06 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 14:42.28 | Katty | unless morning sickness starts at 3am ^_- |
| 14:44.00 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 14:44.07 | ZaVoid | hey all |
| 14:44.20 | ZaVoid | whats the command similar to PlayBack that allows you to move on in your macro while playing? |
| 14:44.44 | nixguy | background |
| 14:44.47 | [TK]D-Fender | Katty: Whew! |
| 14:44.52 | nixguy | ZaVoid: |
| 14:45.13 | ZaVoid | hey nixguy |
| 14:45.16 | ZaVoid | background thanks |
| 14:45.21 | *** join/#asterisk awk (n=awk@security.web.za) |
| 14:45.23 | nixguy | np |
| 14:45.23 | ZaVoid | spaced there for a moment |
| 14:46.02 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
| 14:46.51 | stansmith | ah! |
| 14:49.07 | dandre | [TK]D-Fender: hello |
| 14:49.07 | dandre | do you have any tip for my false inbound detection on zap channels that I have decribed earlier? |
| 14:49.22 | [TK]D-Fender | dandre: If I did I would have answered you earlier |
| 14:50.28 | dandre | ok, is there any doc on the meaning of wink time, start time, ... parameters? |
| 14:50.37 | [TK]D-Fender | ~101 |
| 14:50.38 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 14:51.04 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:51.15 | dandre | ok thank |
| 14:52.45 | stansmith | asterisk is our friend |
| 15:01.36 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:02.27 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:02.43 | *** join/#asterisk Jocko (i=daemon@c-67-168-165-185.hsd1.wa.comcast.net) |
| 15:02.59 | dandre | I havenn't found relevant info for winktime or rxwink tme in this book |
| 15:03.57 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:05.28 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:06.21 | Jocko | Does anyone here know how to program an asterisk IVR so that users can key in responses before the playback has finished? |
| 15:06.43 | stansmith | background() or read() depending on what you want to do with that response |
| 15:06.48 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 15:08.09 | Jocko | Stansmith: I want to capture variables for the user, i.e., "enter you password" |
| 15:08.16 | stansmith | use read() |
| 15:08.37 | Jocko | How do I play them the prompt? |
| 15:08.52 | stansmith | s,n,Read(VAR_TO_SAVE,sound_to_play,,REPEAT,TIMEOUT) |
| 15:09.23 | Jocko | Thanks. I didn't know about he other options with read. |
| 15:09.29 | stansmith | np |
| 15:09.56 | stansmith | Jocko: i think you need to delimit the input with "#", i forget |
| 15:10.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:10.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:10.41 | Jocko | No. I can specify the number of characters I'm expecting. |
| 15:10.44 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:11.25 | Jocko | Example: Please enter a 6 digit date. I'll wait until they enter 6 digits. |
| 15:12.44 | stansmith | true |
| 15:13.14 | lmadsen | oej: I'm having a small problem across two separate boxes, where the one with the problem shows "SIP Options: replace replaces" in the sip show peers for the two asterisk boxes, but on the test servers (same dialplan code) it doesn't show that..... any idea why I might be seeing those replaces? file was thinking it had something to do with the polycom phones themselves, but I've been doing diffs between my configs and the p |
| 15:13.14 | lmadsen | roduction boxes configs, and can't seem to find a difference that would cause that. It seems the call-id's are not matching up or something, because when I do the REFER, to 3 way some phones together, then the phone that did the conference setup (conf button on the polycom), then hangs up, the other two partys are also dropped, but that does not happen in my test environment.... |
| 15:13.40 | jameswf | wow |
| 15:13.45 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:14.03 | lmadsen | oej: anything like that happen to you before? :) |
| 15:14.15 | oej | Where does it show "SIP Options: replace replaces"? I don't follow you |
| 15:14.26 | Jocko | Thanks again StanSmith |
| 15:14.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:15.21 | lmadsen | oej: sip show peer asterisk_primary |
| 15:15.32 | lmadsen | lmadsen: it would show it near the bottom of that peer |
| 15:15.48 | Qwell | stansmith: CIA using Asterisk now? |
| 15:16.06 | stansmith | i dunno? |
| 15:16.15 | Katty | hai Qwell |
| 15:16.21 | oej | lmadsen: I see. THat's a typical polycom. They mis-spelled the SIP option |
| 15:16.27 | jameswf | Asterisk is in alot of government fcilities |
| 15:16.32 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.110.133) |
| 15:16.36 | Katty | anonymouz666: :> |
| 15:16.45 | coppice | the US military in iraq uses asterisk |
| 15:16.48 | anonymouz666 | Katty! do you remember me? |
| 15:16.49 | lmadsen | file just showed me a bug in asterisk-bugs that might be relevant, but I can't seem to reproduce the issue on my test boxes... and I don't know why the production boxes show the 'replaces' in the SIP Options for the link between the two asteriskboxes |
| 15:16.51 | anonymouz666 | lol =) |
| 15:16.57 | Katty | anonymouz666: if course i remember you! |
| 15:17.02 | lmadsen | oej: ahhh I see... so you don't know then :) |
| 15:17.02 | Katty | anonymouz666: don't be redonkulous! |
| 15:17.08 | jameswf | I know we ship to the military all the time |
| 15:17.10 | *** join/#asterisk AndyGraybeal (n=andy@node180.39.251.72.1dial.com) |
| 15:17.22 | jameswf | we have a card on a nuclear sub :) |
| 15:17.46 | mvanbaak | dutch army uses asterisk in Afganistan |
| 15:17.59 | stansmith | i still love H.E.R |
| 15:18.37 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 15:18.53 | jameswf | Asterisk weather telemarketing for anti-war groups or guiding troops cause when it needs to get done you cant count on microsoft.... |
| 15:19.15 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 15:19.44 | stansmith | why microsoft? |
| 15:19.55 | stansmith | they prob run asterisk too |
| 15:20.25 | Katty | hmmhesays: ewwo? |
| 15:20.29 | jameswf | no M$ has their new responsepoint |
| 15:20.44 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:20.53 | stansmith | why would he divide by zero? |
| 15:21.23 | jameswf | some times chuck norris devides by zero |
| 15:21.38 | stansmith | haha |
| 15:22.12 | *** join/#asterisk ACiDV (n=joel@122-205-229.dr.cgocable.ca) |
| 15:22.43 | jameswf | speeking of chuck norris Ralph Nader is making his 4th run and the dems arepissed :)) |
| 15:24.23 | AndyGraybeal | is ralph nader the green party? |
| 15:24.31 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 15:24.44 | jameswf | he is but I think he is running as an indie |
| 15:25.18 | jameswf | Nader-Paul for 2008 woohoo |
| 15:25.27 | *** join/#asterisk rafiks (n=rafiks@c-68-56-23-83.hsd1.fl.comcast.net) |
| 15:25.30 | jameswf | that would be cool to see |
| 15:25.49 | jameswf | scary but cool |
| 15:27.02 | jameswf | if nader and paul join forces they would command like 2% of the vote and be like in your face |
| 15:27.09 | [TK]D-Fender | Ron Paul is nuking any chance for victory for his lack of advertising, presences, etc |
| 15:27.30 | stansmith | ive seen quite a few ron paul bumper stickers in columbus ohio |
| 15:27.43 | stansmith | but columbus is all indie rocker - types |
| 15:27.44 | jameswf | ah well I am voting for Mickey Mouse |
| 15:29.31 | jameswf | id vote for that :) |
| 15:29.43 | ZaVoid | i'd vote for barney |
| 15:30.13 | [TK]D-Fender | ZaVoid: Same thinking that got GWB a 2nd term. |
| 15:30.53 | jameswf | big even |
| 15:31.13 | stansmith | id rather vote for Rob n Big |
| 15:31.20 | ZaVoid | lol i know fender |
| 15:31.43 | jameswf | that would be cool amall horse in the whitehouse... |
| 15:31.48 | jameswf | damn |
| 15:31.50 | jameswf | small |
| 15:33.01 | ZaVoid | i voted for nader in 2000 and 2004 |
| 15:33.02 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 15:33.04 | ZaVoid | i won't this year |
| 15:33.24 | ZaVoid | in 96 i voted for Bob Dole.. because he thought it was his turn to be president :) |
| 15:33.59 | *** join/#asterisk Bambr (n=Bambr@81-208-51-12.ip.fastwebnet.it) |
| 15:34.37 | *** join/#asterisk momelod (n=smelo@bas2-toronto02-1279462672.dsl.bell.ca) |
| 15:34.43 | momelod | hello chanel. |
| 15:34.45 | jameswf | bob dole liked bob dole for president but bob dole couldnt get people to rally behind bob dole.. bob dole was sad and said wh dont people like bob dole, bobdole is good..... |
| 15:34.58 | Bambr | hi, is anybody able to give me link or another way to get the full and most up to date list of all operators with their network codes? preferably some way to update it in regular basis |
| 15:35.26 | jameswf | Bambr: Asterisk manager API |
| 15:35.31 | ZaVoid | jameswf: bobdole ended up making more money with viagra :) |
| 15:35.43 | momelod | can anyone tell me where i can download old firmware for my cisco 7960 ip phone? The cisco site only has the lastest firmware, but i need to do some incremental upgrades before i can install the latest version.. |
| 15:35.44 | jameswf | boing |
| 15:35.55 | Qwell | momelod: why incremental? |
| 15:36.19 | Bambr | jameswf, errm, it downloads some list of all the operators in the world? |
| 15:36.26 | momelod | well, i can't load the lastest firmware with out first upgrading to 5.x |
| 15:36.30 | jameswf | Bill Clinton should promote viagra... people would buy it fore sure if clinton used it |
| 15:36.47 | djs | lmao |
| 15:36.49 | momelod | i am at 3.x now, i need to go to 5.x so that i can update to 8.x lol |
| 15:36.52 | jameswf | viagra 32 interns in 4 hous....what... |
| 15:36.58 | jameswf | *hours |
| 15:36.58 | Qwell | momelod: why? |
| 15:37.09 | ZaVoid | i don't think clinton needs viagra |
| 15:37.10 | [TK]D-Fender | jameswf: That would be like promoting the abolishion of valuable opsitions like interns! |
| 15:37.17 | ZaVoid | he needs better taste though |
| 15:37.26 | stansmith | hey man, lil wayne sucks |
| 15:37.28 | momelod | when i boot up my phone, it doesn not request the correct files from my tftp server |
| 15:37.56 | momelod | the 5.x firmware updates the bootloader |
| 15:38.00 | jameswf | it doesnt request or you didnt create |
| 15:39.55 | Qwell | momelod: well, what is it requesting? |
| 15:40.19 | jameswf | holy crap XM is dedicating a whole month to Michael Jackson on one of their channels |
| 15:41.11 | *** join/#asterisk Woifi1988 (n=anon@M1340P027.adsl.highway.telekom.at) |
| 15:41.25 | stansmith | micheal jackson is the best dancer of all time |
| 15:42.00 | jameswf | does Michael Jackson deserve anything for Black history month? once apon a time he was black and semi normal.... then he turned 12 |
| 15:42.04 | *** join/#asterisk uwe (n=uwe@213.244.124.18) |
| 15:42.07 | momelod | Qwell: let me fetch the logs.. sec |
| 15:42.26 | jameswf | now he is a crazy white woman |
| 15:42.41 | Woifi1988 | what das the command alwaysauthreject=yes do? |
| 15:42.41 | mvanbaak | <PROTECTED> |
| 15:42.46 | mvanbaak | nice browser |
| 15:42.50 | Qwell | mvanbaak: only 761m? |
| 15:42.52 | *** join/#asterisk Jocko (i=daemon@c-67-168-165-185.hsd1.wa.comcast.net) |
| 15:43.00 | mvanbaak | god |
| 15:43.08 | mvanbaak | only 2 tabs open, no flash or anything |
| 15:43.10 | jameswf | yes son |
| 15:43.18 | jameswf | :) |
| 15:44.03 | ruied | can I see from asterisk if a phone is setted to DND ??? |
| 15:44.17 | jameswf | dwahahaha |
| 15:44.40 | jameswf | yesuh |
| 15:45.03 | momelod | Qwell: http://pastebin.ca/917643 |
| 15:45.04 | jameswf | im need caffiene holy crap brb |
| 15:45.13 | drmessano | caffeine is GOOD |
| 15:45.33 | Qwell | ^^ that's why Digium opened a coffee shop in the lobby :p |
| 15:45.37 | drmessano | heh |
| 15:46.01 | Qwell | momelod: so, I don't see the problem |
| 15:46.23 | momelod | Qwell: the files that the phone is requesting are not in my tftp root directory |
| 15:46.33 | momelod | its asking for files that dont exist |
| 15:46.41 | Qwell | P003-08-2-00.sbn |
| 15:46.43 | jameswf | coffee shop?? that spunds like it cost $$$ |
| 15:46.56 | Qwell | momelod: I see it requesting that, which you do have |
| 15:46.56 | *** part/#asterisk [gnubie] (n=[gnubie]@cm152.gamma183.maxonline.com.sg) |
| 15:46.59 | jameswf | our crack I mean caffien is free |
| 15:47.02 | momelod | lol! |
| 15:47.14 | jameswf | I need more bah |
| 15:47.17 | momelod | lemme check the phone, im soo stupid |
| 15:47.49 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 15:47.57 | ruied | I can see if it is busy from the command line if I dial the number, but is there any other way to see it from the command line, like 'sip....' |
| 15:48.16 | jameswf | the sign of a good admin is the ability to make their end users feel sooooo small |
| 15:48.23 | momelod | hrmm.. when i check the firmware on the phone, its still reporting the old firmware version |
| 15:48.40 | Qwell | momelod: so, you need to find out why it's failing, rather than guessing :) |
| 15:48.57 | JoseBravo | Can I connect my FAX to xp100 and use it like a FXS? |
| 15:49.27 | momelod | ill investigate further |
| 15:49.28 | momelod | thanx |
| 15:49.51 | ruied | JoseBravo, I think not... |
| 15:50.56 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:52.21 | ruied | JoseBravo, isn't xp100 an FXO one? |
| 15:52.46 | [TK]D-Fender | JoseBravo: No, the X100P is an FXO. |
| 15:52.49 | *** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 15:56.24 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:59.24 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 15:59.45 | mintee | I just got a polycom cx200 phone in, and other than the fact that the doc's say to install MS Office Communicator, is there some alternative software to configure the phone? |
| 15:59.54 | mintee | yeah, i realize that has nothing to do with asterisk |
| 15:59.56 | Woifi1988 | what das the command alwaysauthreject=yes do? |
| 16:00.58 | JoseBravo | I have my FAX connected to my asterisk by Linksys (Analog Phone Adapter) that connects by SIP using my LAN. Sometimes I can't receive errors. Its for a mistmake setup? |
| 16:01.16 | JoseBravo | * I receive errores |
| 16:01.51 | hmmhesays | faxing over ip sucks |
| 16:02.03 | *** join/#asterisk [gnubie] (n=[gnubie]@cm152.gamma183.maxonline.com.sg) |
| 16:02.06 | [gnubie] | hello all.. |
| 16:02.21 | [gnubie] | how do i show or list down the loaded modules on my asterisk 1.4.x? |
| 16:02.48 | Woifi1988 | try show modules? |
| 16:03.18 | hmmhesays | morning folks |
| 16:04.15 | Jocko | Does anyone know how to playback an MP3 and record a caller singing along with the playback? |
| 16:04.29 | [gnubie] | Woifi1988: thanks. i got 145 modules loaded |
| 16:04.46 | RoyK | Jocko: karaoke? :) |
| 16:04.53 | Jocko | Exactly. |
| 16:05.27 | Jocko | I want to record the two audio streams |
| 16:05.28 | RoyK | Jocko: perhaps with MoH with meetme... |
| 16:05.44 | RoyK | oh |
| 16:05.48 | RoyK | app_chanspy |
| 16:05.49 | *** join/#asterisk AndyGraybeal (n=andy@node180.39.251.72.1dial.com) |
| 16:06.02 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:06.20 | drmessano | Asterokie? |
| 16:06.29 | drmessano | Asteroke |
| 16:06.31 | [gnubie] | anyone tried the muted feature here? |
| 16:06.50 | stansmith | does anyone know of a good open source pbx software solution? |
| 16:07.07 | mvanbaak | stansmith: asterisk |
| 16:07.14 | stansmith | thx |
| 16:07.29 | anonymouz666 | haha |
| 16:07.45 | anonymouz666 | this was fun |
| 16:07.53 | hmmhesays | well there are other options also |
| 16:07.54 | mvanbaak | whehehehe |
| 16:07.59 | mvanbaak | yup |
| 16:08.11 | mvanbaak | but since this is #asterisk |
| 16:08.19 | RoyK | callweaver, freeswith, yate... |
| 16:08.25 | Jocko | RoyK: the MoH idea? Dump the caller into a conference with the music playing in the background and record the conference call? |
| 16:09.00 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 16:09.02 | RoyK | Jocko: yeah, but i think chanspy can do the same |
| 16:09.50 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 16:14.20 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:14.40 | *** part/#asterisk [gnubie] (n=[gnubie]@cm152.gamma183.maxonline.com.sg) |
| 16:18.15 | [TK]D-Fender | mintee: Does it come with any other software of its own? |
| 16:21.32 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:21.47 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:21.55 | *** part/#asterisk shtoom (n=godson@59.93.120.51) |
| 16:22.58 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:24.09 | JoseBravo | Is a bad idea connect my FAX to my asterisk using a Linksys PAP? Is better idea if I connect my fax directly to the asterisk using a FXS port? |
| 16:24.47 | [TK]D-Fender | JoseBravo: Fax over VoIP = trouble. You're best to use a completely separate line for this |
| 16:25.08 | AndyGraybeal | [TK]D-Fender: generally speaking, why is it trouble? |
| 16:25.25 | [TK]D-Fender | AndyGraybeal: Any jitter/delay, etc can kill a fax call |
| 16:25.33 | AndyGraybeal | ah okay good deal |
| 16:25.52 | *** join/#asterisk KillerQueen_86 (n=alan@201-212-119-77.prima.net.ar) |
| 16:26.04 | Katty | i'm tired of signing privacy contracts! |
| 16:26.07 | Katty | I WANT TO BURN THEM ALL |
| 16:26.30 | Katty | k, i'm over it. |
| 16:26.33 | stansmith | katty = a cat? |
| 16:26.45 | AndyGraybeal | kattykitty |
| 16:26.56 | Katty | as if that joke hasn't been madea billion times. |
| 16:27.16 | Katty | i'm sorry, i'm grumpy :< |
| 16:27.59 | *** join/#asterisk tychocaine (n=foo@86-41-203-93.b-ras1.lmk.limerick.eircom.net) |
| 16:28.07 | [TK]D-Fender | Katty: I managed to dodge signing my own companies "contract". I had "issues" with it, and the topic got "forgotten" in my delay :) |
| 16:28.36 | tychocaine | hi, has anyone here had experience setting up a b410p? |
| 16:29.38 | JoseBravo | [TK]D-Fender ok, but if I still want to have the FAX over my asterisk I can obtain better results if I get the FAX over FXS and not using a Analog Phone Adapter (SIP) |
| 16:31.10 | defswork | any know issues with BLFs going out of sync - showing busy when not etc.. (aastra 55i in particular) |
| 16:31.43 | Katty | [TK]D-Fender: nice. |
| 16:31.54 | Katty | [TK]D-Fender: this is a variation of the medical hippa agreement. |
| 16:31.59 | Katty | [TK]D-Fender: i'll be working with a lawfirm |
| 16:32.26 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-c260e6ac876d1862) |
| 16:32.42 | Katty | i wonder if i can claim that it would be a conflict of interest for me working with that lawfirm |
| 16:33.05 | XnOSX | hello friends, what its the correct sintax for a goto accion to a voicemail in other asterisk pbx¿ |
| 16:35.40 | XnOSX | exten => XXXXXXXXX,n,Goto(Voicemail/context:voicemailuser@context)? |
| 16:38.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:38.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:39.07 | *** join/#asterisk adker (n=chatzill@74-47-52-65.dr02.glvv.ny.frontiernet.net) |
| 16:40.59 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:41.42 | AndyGraybeal | if i wanted to record all calls; is there something that's already set up for that in asterisk? ... (say i want to record them in case there is a need to review the phone dialog within 6 weeks, and if not go ahead and delete the conversation after 6 weeks) |
| 16:41.45 | Katty | hi russellb (= |
| 16:41.59 | AndyGraybeal | i realize i can do that with russellb's asterisk-jack extension |
| 16:42.15 | AndyGraybeal | i'm wondering if asterisk already has it built in |
| 16:42.28 | russellb | Katty: greetings |
| 16:42.41 | russellb | AndyGraybeal: yeah, check out Monitor and MixMonitor |
| 16:43.00 | AndyGraybeal | <PROTECTED> |
| 16:43.04 | russellb | use MixMonitor, unless you want each leg of the call in it's own recording |
| 16:43.07 | russellb | yes |
| 16:43.12 | AndyGraybeal | rad thank you |
| 16:46.22 | JoseBravo | [TK]D-Fender ok, but if I still want to have the FAX over my asterisk I can obtain better results if I get the FAX over FXS and not using a Analog Phone Adapter (SIP) |
| 16:46.30 | AndyGraybeal | hah has someone done text to speech, then translated that and then a voice synth'd the translation? |
| 16:47.10 | AndyGraybeal | er.. i mean speech to text |
| 16:47.27 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 16:50.57 | *** join/#asterisk elguero (n=elguero@207.154.34.162) |
| 16:51.39 | stansmith | i HATE SUBVERSION |
| 16:51.52 | stansmith | im using asterisk to commit changes to subversion |
| 16:52.08 | Supaplex | you should use a svn client instead |
| 16:52.30 | *** join/#asterisk elguero (n=elguero@207.154.34.162) |
| 16:53.16 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:53.41 | deeperror | quick q....where does the "Unknown" come from? http://pastebin.ca/917714 |
| 16:55.39 | *** join/#asterisk miguel3239 (n=elguero@207.154.34.162) |
| 16:56.24 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 17:01.38 | scooby2 | Any way to use an IF state to exclude extensions from being recorded? |
| 17:02.59 | scooby2 | CEO wanted everyone recorded and is now freaking that his calls are being recorded |
| 17:05.24 | *** join/#asterisk miguel3239 (n=elguero@207.154.34.162) |
| 17:05.49 | plik | gotta love them "one rule for them, another for us" type execs |
| 17:06.04 | stansmith | lets get em! |
| 17:06.33 | *** join/#asterisk RoyK (n=roy@ip-32-4-149-91.dialup.ice.no) |
| 17:08.22 | scooby2 | plik: no kidding |
| 17:08.24 | deeperror | i would record his calls to another folder |
| 17:08.39 | scooby2 | help me do it:) |
| 17:08.41 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:09.15 | scooby2 | i was thinking set a variable if his extension and then if that variable, record to path X |
| 17:09.30 | scooby2 | no variable record to path Y |
| 17:09.33 | tzafrir | stansmith, you got it all wrong. The IF is intended for the destination of the recording. |
| 17:09.45 | stansmith | scooby2: ^^^^ |
| 17:10.50 | *** join/#asterisk joelsolanki (i=joelsola@220.224.52.214) |
| 17:11.19 | joelsolanki | any one steve thoma ? |
| 17:11.28 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 17:12.57 | RoyK | load '/Applications/X-Chat Aqua/Plugins/google.pl' |
| 17:13.53 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 17:13.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:14.29 | dandre | I have this situation: |
| 17:14.29 | dandre | - inbound call to my zap fxo channel |
| 17:14.30 | dandre | - routed to an extension within my dialplan |
| 17:14.30 | dandre | - if this extension goes on hook, the zap channel is hungup (console log) but doesn't seem to be destroyed (full log in /var/log/asterisk) |
| 17:14.30 | dandre | what could be the reason? |
| 17:14.42 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 17:14.43 | dandre | The consequence is that the line rings again |
| 17:15.40 | *** part/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 17:16.08 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 17:16.08 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:16.32 | robeph | ok silly question, as i'm not sure it has true real world application, but... I have two boxes, they're connected via iax trunks, I have a device on box 1, in the dialplan when I dial 2500, it dials out to box 2 ext.2500 which in turn calls back to box 1/ext.2501 which then dial the iax soft phone registered at box 1... I'm getting this error |
| 17:16.41 | robeph | Feb 25 10:35:28 WARNING[604]: chan_iax2.c:7366 socket_read: I don't know how to authenticate phone2 to 192.168.2.141 |
| 17:16.47 | robeph | .141 = box 2 |
| 17:16.56 | stansmith | ~pb robeph |
| 17:17.06 | robeph | nothing else to paste.. it's just that one line |
| 17:17.16 | stansmith | o, haha sorry |
| 17:17.37 | robeph | ;) I don't often come here, but I know the etiquette.. I'm active in a buncha other chans. |
| 17:18.07 | robeph | anyhow, basically I cannot figure what the cause of that erorr is I googled to no avail |
| 17:18.37 | robeph | ignore the fact that what I'm doing is completely pointless. |
| 17:21.47 | joelsolanki | Hi room |
| 17:21.52 | stansmith | ~hi joelsolanki |
| 17:21.52 | jbot | Many greetings, joelsolanki, most strange traveller, to this IRCdom of plenty. |
| 17:22.00 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 17:22.33 | joelsolanki | Is anybody using chan_ss7 + asterisk + T1 ? |
| 17:22.54 | Kobaz | in the lastest 1.4.x release, is the problem fixed where if you have no default route defined it can't get it's own ip |
| 17:23.12 | [TK]D-Fender | XnOSX: exten => XXXXXXXXX,n,Voicemail(voicemailboxnumberhere@context) |
| 17:23.22 | joelsolanki | I have sangoma A104D card and i want to use ss7 signalling with sangoma card using chan_ss7 |
| 17:23.35 | robeph | stansmith: http://rafb.net/p/K24Dxc61.html if you need more info from that, but then it hangs up |
| 17:23.35 | joelsolanki | but need to know wheather it is a stable solution or not / |
| 17:23.40 | joelsolanki | ? |
| 17:24.37 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:25.03 | stansmith | robeph: does phone2 have an entry in iax.conf? |
| 17:25.11 | robeph | in both boxes? |
| 17:25.22 | robeph | yeah |
| 17:25.23 | stansmith | is that link from box 1 or box 2? |
| 17:25.40 | robeph | this link is from .142 (box1) -> .141(box2) |
| 17:25.48 | *** join/#asterisk PepOSX (n=angeldav@201.243.76.220) |
| 17:26.20 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:26.51 | robeph | functions as such dial 2500 -> IAX2/iaxtrunk141/2500 -> 2500 Dial (iaxtrunk142/2501) -> 2501 Dial(IAX2/phone2) |
| 17:27.06 | robeph | well IAX2/iaxtrunk142 |
| 17:27.16 | jameswf | so I wrote a script that initiates and tears down like 30 calls a second... takes like 15 srconds to make asterisk lock up like a newb in a prison |
| 17:27.36 | *** part/#asterisk KillerQueen_86 (n=alan@201-212-119-77.prima.net.ar) |
| 17:28.06 | stansmith | robeph: post your iax.conf for box1 |
| 17:28.10 | robeph | stansmith: ok |
| 17:28.21 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:29.00 | jameswf | so I wonder if its the asterisk managers fault or asteisks fault.... |
| 17:29.06 | robeph | stansmith: http://rafb.net/p/VXahtK95.html |
| 17:29.33 | jm|laptop | what BT dongles are you guys using with chan_mobile ? |
| 17:29.49 | Qwell | jm|laptop: just about any should work |
| 17:29.52 | Qwell | "should".. |
| 17:30.02 | robeph | jameswf: prolly doesn't like that it's climbing linked lists everytime ya do something with a channel and doing that much that fast may not be sociable ;p |
| 17:30.05 | jm|laptop | I'm having 3-5s delay :/ |
| 17:30.10 | robeph | jameswf: I don't think it's the manager though |
| 17:30.23 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:30.38 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 17:31.06 | jm|laptop | I have one dongle that spews into syslog and another that gives this delay |
| 17:31.53 | jameswf | should I be nice and sleep 1 |
| 17:32.07 | robeph | k/iax.conf |
| 17:32.31 | robeph | http://rafb.net/p/VXahtK95.html stansmith |
| 17:32.35 | stansmith | robeph: im not sure, but you might want to try adding "register => iaxtrunk141:1234@192.168.1.141" |
| 17:32.45 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:33.06 | stansmith | how .142 is box 1, and .141 is box 2, is really throwin me off |
| 17:33.09 | robeph | stansmith: yeh, thing is, if I dial 2501 locally on box 2, it'll bounce across to box1 and dial phone2 |
| 17:33.11 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 17:33.22 | drmessano | Bluetooth can be a pain in the ass if you're a n00b to setting it up |
| 17:33.25 | drmessano | <-- n00b |
| 17:33.46 | jm|laptop | it installed okay from src |
| 17:33.49 | drmessano | I got it working by installing the RPMs on CentOS.. when I tried from Source, it was major fail |
| 17:34.08 | drmessano | See, I never got that far |
| 17:34.12 | drmessano | I got it "installed" |
| 17:34.19 | drmessano | But there was no bluetooth service |
| 17:34.25 | drmessano | and my dongles never showed up |
| 17:34.35 | drmessano | I removed it, reinstalled from RPM, working again |
| 17:34.45 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:34.57 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:39.19 | stansmith | wayhigh's name in real life is kevin lynn? |
| 17:39.36 | drmessano | I guess so lol |
| 17:41.38 | *** join/#asterisk radulovity (n=emil@77.46.189.24) |
| 17:41.42 | radulovity | Hi! |
| 17:42.05 | radulovity | i have one ISDN card |
| 17:42.11 | radulovity | lspci tell me Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 17:42.20 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 17:42.28 | stansmith | o yea? |
| 17:42.44 | radulovity | but, i phone.conf can i not put the device, because, i hav'n any /dev/isdn* |
| 17:42.52 | radulovity | how can i find the device? |
| 17:43.10 | stansmith | radulovity: dmesg | tail |
| 17:44.20 | *** join/#asterisk dj1066dj (n=chatzill@209.149.56.242) |
| 17:44.50 | x86 | is there an easy way to originate a call from an external Asterisk server? |
| 17:45.40 | x86 | for example, I've got a main Asterisk server at my headquarters that everyone gets their reports from (time spent on the phone, number of calls made, calls by account code, etc) |
| 17:46.08 | x86 | I'd like for someone to be able to click-to-dial from the reports, even though their extension is on a different Asterisk server |
| 17:46.31 | radulovity | stansmith: no info |
| 17:46.51 | stansmith | were you required to supply power to the card? |
| 17:46.54 | stansmith | wait nm |
| 17:47.05 | stansmith | radulovity: did you install drivers? |
| 17:47.21 | radulovity | stansmith: lsmod tell me isdn 121248 1 hisax |
| 17:47.43 | radulovity | uname -r 2.6.18-4 |
| 17:47.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:48.18 | stansmith | radulovity: ztcfg -vv ? |
| 17:48.29 | radulovity | stansmith: moment |
| 17:49.38 | radulovity | stansmith: 1 error(s) detected |
| 17:49.40 | radulovity | hmmm |
| 17:49.48 | radulovity | line 0: Unable to open master device '/dev/zap/ctl' |
| 17:50.12 | stansmith | is your zaptel service running? |
| 17:50.42 | radulovity | no, i hav't install it, i do it now |
| 17:50.42 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 17:50.54 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 17:50.55 | stansmith | if you havent installed it, i wonder how you had ztcfg... |
| 17:52.00 | radulovity | i have it install now, and i have now zaptel, but error bye starting |
| 17:52.15 | *** join/#asterisk SteveTotaro (n=root@static-71-251-146-11.hrbgpa.fios.verizon.net) |
| 17:52.22 | radulovity | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
| 17:52.31 | radulovity | need i to recompile the wole kernel? |
| 17:52.35 | SteveTotaro | no |
| 17:52.37 | stansmith | no just zaptel |
| 17:52.37 | radulovity | I have debian etch ... |
| 17:52.40 | SteveTotaro | joining late |
| 17:52.44 | radulovity | good :) |
| 17:52.59 | SteveTotaro | but open the makefile and uncomment # ztdummy |
| 17:53.56 | radulovity | SteveTotaro: ok, i download now the source, i don;t need the kernel headers? |
| 17:54.01 | radulovity | just the source from zaptel? |
| 17:54.05 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com) |
| 17:54.41 | stansmith | radulovity: if you have already compiled it before, just go back to the source directory and do "make clean" then "./configure" then "make menuselect" |
| 17:54.50 | stansmith | it = zaptel |
| 17:55.39 | radulovity | stansmith: no, i have it just install with apt-get, directly binary, now have i download the source |
| 17:55.47 | stansmith | o.... |
| 17:55.50 | *** join/#asterisk MicW (n=michael@dslb-088-074-180-216.pools.arcor-ip.net) |
| 17:56.00 | radulovity | problem? |
| 17:56.05 | MicW | hi |
| 17:56.25 | radulovity | debian Etch with xen image, but asterisk need to run on the host, not on virtual |
| 17:56.38 | stansmith | so.. |
| 17:57.03 | stansmith | as long as the sources are the exact same for the binary u got with apt-get, u should be ok |
| 17:57.12 | stansmith | feel free to correct me someone |
| 17:57.13 | MicW | are there any devices which can connect to 3 or 4 isdn ports and a network port and allows an asterisk server to access the isdn ports via ethernet? |
| 17:59.08 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 17:59.08 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:59.45 | x86 | MicW: you talking about PRI or BRI? |
| 18:00.02 | MicW | x86? |
| 18:00.08 | x86 | MicW? |
| 18:00.16 | stansmith | stansmith ? |
| 18:00.16 | MicW | what is PRI or BRI? |
| 18:00.34 | MicW | i thought bri are some 4-port-isdn cards |
| 18:00.36 | x86 | MicW: if you don't know that, you obviously do not need an ISDN bridge ;) |
| 18:01.36 | MicW | i have an asterisk server running in a virtual machine for ha-failover. the porblem is that the machine has no direct hardware access. so i cannot use a pci card or so |
| 18:01.47 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 18:01.48 | x86 | BRI == Basic Rate Interface, usually consisting of (1) 64kbps bearer (B) channel, and (1) 8kbps signalling (D) channel |
| 18:02.02 | jameswf | ~bri |
| 18:02.03 | jbot | i guess bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
| 18:02.05 | MicW | ah. isdn ;= |
| 18:03.00 | x86 | jameswf: jbot's BRI definition is wrong... standard BRI only uses (1) B channel... although in practice most times (2) B channels are used |
| 18:03.34 | JenniferAkemi | whenever we ordered BRI here (like 8 years ago) we always got 2 Bs |
| 18:03.44 | JenniferAkemi | here being canada |
| 18:03.54 | MicW | yes. that's hwat i'm talking about (an isdn s0 port has 2xB and 1xD) |
| 18:03.55 | JenniferAkemi | maybe that's where its' from |
| 18:04.15 | x86 | MicW: PRI == Primary Rate Interface, which can ride a T1, E1, or J1 circuit (or any variation thereof), which consists of (23) 64kbps bearer (B) channels and (1) 64kbps signalling (D) channel |
| 18:04.38 | *** join/#asterisk ffadaie (n=chatzill@CPE0013463f086f-CM0019475d7cb6.cpe.net.cable.rogers.com) |
| 18:04.40 | x86 | JenniferAkemi: most places worldwide will give you 2B |
| 18:05.01 | x86 | JenniferAkemi: but by standard definition, you only need 1B and 1D to have BRI :) |
| 18:05.15 | MicW | we'll have 4x BRI, all have the same phone numbers (MSNs), resulting in max. 8 parallel phone calls |
| 18:05.46 | MicW | i need some hardware that translates this to ethernet and some linux driver that translates it back to capi |
| 18:05.55 | x86 | so you want a quad-port BRI to SIP bridge |
| 18:06.14 | MicW | no, bri to ethernet to capi |
| 18:06.14 | x86 | why translate it back to CAPI? why not just use SIP? |
| 18:06.34 | x86 | not sure why you would do that |
| 18:06.48 | x86 | does not make much sense to convert it back |
| 18:06.49 | MicW | because 1. the sip translation will be done in the asterisk server and 2. there are some legacy capi applpications |
| 18:07.24 | x86 | good luck finding something like that |
| 18:07.33 | MicW | that's why i ask here |
| 18:07.36 | x86 | might have better luck upgrading your applications |
| 18:07.44 | ffadaie | guys, anybody has experience with jabber integration? |
| 18:07.49 | MicW | and are there quad-port BRI to SIP bridges? |
| 18:07.59 | x86 | MicW: perhaps... |
| 18:08.03 | radulovity | stansmith: zaptel running |
| 18:08.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:08.21 | radulovity | dmesg tell me Zapata Telephony Interface Registered on major 196 |
| 18:08.21 | radulovity | Zaptel Version: 1.2.11 Echo Canceller: MG2 |
| 18:08.22 | x86 | MicW: http://var.abptech.com/s.nl/it.A/id.1822/.f |
| 18:08.35 | radulovity | ztcfg -vv tell me 0 channels configured. |
| 18:08.46 | x86 | radulovity: wtf 1.2? |
| 18:08.52 | x86 | radulovity: 1.2 is no longer supported |
| 18:09.23 | radulovity | x86 hmmm, this was bye source able to download via dpkg |
| 18:09.27 | radulovity | debian etch |
| 18:09.39 | x86 | don't use dpkg |
| 18:09.42 | x86 | use wget ;) |
| 18:10.04 | radulovity | :) |
| 18:10.22 | radulovity | i look for someone unoffical mirro ... |
| 18:10.37 | MicW | x86: thx |
| 18:10.52 | radulovity | is runing somebody asterisknow under xen? |
| 18:11.15 | MicW | x86: one of this is a fax application (recieve only). will this also possible with such a gateway? |
| 18:11.23 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 18:11.23 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:11.24 | jm|laptop | [Feb 25 18:11:03] WARNING[25581]: pbx.c:2667 pbx_extension_helper: No application 'Dial' |
| 18:11.25 | jm|laptop | HA |
| 18:12.57 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 18:12.59 | file | Katty: play nice with the class |
| 18:13.13 | outtolunc | ooo baby |
| 18:13.47 | x86 | MicW: you can do fax with SIP, sure |
| 18:14.09 | x86 | MicW: just use G711u and be on the same LAN as the gateway, and you should be fine |
| 18:14.19 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:14.27 | MicW | nice :) |
| 18:15.07 | MicW | i'll try that first. the other application is a server which shows incoming calls to different applications. it's easy to change this to sip |
| 18:15.34 | MicW | (or to use ther asterisk's network interface) |
| 18:16.07 | radulovity | what is the lover hardware on that i am able to run asterisknow? |
| 18:19.43 | Katty | file: i have a hard time tolerating stupid :< |
| 18:23.05 | Katty | file: and there's a whole lot of that floating around my office today. |
| 18:24.20 | file | Katty: :( |
| 18:25.18 | momelod | hey, does anyone know where i can download older version s of cisco 7960 ip phone firmware? the cisco site only has the lastest release available |
| 18:27.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:27.58 | *** join/#asterisk ddunavant (n=David@pool-96-231-72-88.washdc.east.verizon.net) |
| 18:28.00 | stansmith | radulovity: sorry i stepped out to lunch for a second |
| 18:28.50 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:29.46 | file | eep! |
| 18:29.49 | radulovity | stansmith: no prob, i am patient :) |
| 18:30.18 | stansmith | zaptel running.... |
| 18:30.25 | stansmith | did u try ztcfg -vv again? |
| 18:30.29 | radulovity | yes |
| 18:30.47 | stansmith | and? |
| 18:30.58 | radulovity | Channel map: |
| 18:30.59 | radulovity | 0 channels configured. |
| 18:31.17 | stansmith | did u ever edit /etc/zaptel.conf? |
| 18:31.22 | radulovity | yes |
| 18:31.27 | radulovity | moment |
| 18:31.36 | radulovity | have't edit, i have edit one other file? |
| 18:31.46 | stansmith | /etc/asterisk/zapata.conf? |
| 18:32.28 | radulovity | Channel 30: Network HDLC (Default) (Slaves: 30 31 32 33) |
| 18:32.29 | radulovity | 4 channels configured. |
| 18:32.29 | radulovity | ZT_CHANCONFIG failed on channel 30: No such device or address (6) |
| 18:32.57 | radulovity | stansmith: can i speak with you in /query ? |
| 18:33.11 | stansmith | um |
| 18:33.23 | stansmith | we can try...im using irssi, and honestly, im not sure the command to switch between windows |
| 18:33.32 | stansmith | i thought it was ctrl+x |
| 18:33.48 | radulovity | you nee to use /window number_of_windo |
| 18:34.26 | stansmith | o ok sweet |
| 18:34.34 | stansmith | scooby2: sorry, i just now figured out how to change windows |
| 18:35.27 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:37.37 | J4k3 | stansmith: alt-number will change windows in irssi in most terminals |
| 18:38.52 | radulovity | bye me not :( |
| 18:41.53 | stansmith | µ |
| 18:44.47 | stansmith | that is cool² |
| 18:45.00 | stansmith | (cool to the second power) |
| 18:45.20 | radulovity | :) |
| 18:47.39 | *** part/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 18:52.43 | x86 | J4k3: there is another key combo where ALT-num doesn't work too... cant remember it though |
| 18:52.48 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 18:52.59 | *** join/#asterisk Q3Man (n=Steve@wsip-24-234-155-62.lv.lv.cox.net) |
| 18:57.34 | Q3Man | I'm having a problem with a fresh asterisk 1.4/freepbx install. I've purchased a digium TDM804E and genzaptelconf detects it fine and generated the proper configs. I've setup a 'g0' trunk and set an outbound route on that trunk. However, when I place a call asterisk informs me that 'all circuits are busy now'. The logs generated when I attempt to make a call are located at: http://pastebin.com/m36927ec6 |
| 18:59.25 | *** join/#asterisk elguero (n=elguero@207.154.34.162) |
| 18:59.35 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 18:59.48 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 18:59.48 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:00.44 | x86 | ~freepbx |
| 19:00.45 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:00.54 | x86 | Q3Man: read that |
| 19:07.17 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 19:07.28 | asteriskmonkey | i have an odd issue with call id on a pri |
| 19:07.41 | asteriskmonkey | all my callerids keep getting prepended with a dot ie .calleridname |
| 19:07.47 | asteriskmonkey | anyone seen this before? |
| 19:08.33 | outtolunc | sounds like a misguided 'cut' |
| 19:09.18 | asteriskmonkey | no cuts... just a straight dial(zap/g1/yadda) |
| 19:09.42 | outtolunc | watch the manager session, and your pri debug |
| 19:09.51 | outtolunc | see what changed it and where |
| 19:09.58 | asteriskmonkey | nothing is changing it |
| 19:10.11 | asteriskmonkey | havnt seen the pri debug, through in a ton of no ops though |
| 19:10.14 | outtolunc | it got 'set' somewhere |
| 19:10.15 | asteriskmonkey | nothing is changing calleridname |
| 19:10.19 | *** join/#asterisk oej (n=olle@212.35.97.249) |
| 19:10.25 | asteriskmonkey | ok |
| 19:10.27 | *** join/#asterisk ruied (n=ruied@bl9-255-200.dsl.telepac.pt) |
| 19:10.53 | outtolunc | the callerid just like the rest of the channel doesn't just APPEAR out of the ether with full info intact |
| 19:11.45 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 19:12.42 | mchou | anyone have a pap2 handy? I'd like to know what power supply is supposed to be used with it. (Voltage, current) |
| 19:14.26 | mchou | the specs say 5V 2.0A, but the unit I got says 12V 1A. Dont want to plug it in and fry the HW. |
| 19:15.05 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 19:15.13 | Yourname`` | Hi. How can I add SipAddHeader to a callfile? |
| 19:16.17 | [TK]D-Fender | Yourname``: You can't |
| 19:16.56 | Yourname`` | [TK]D-Fender: How about when I send a call throug manager APU? |
| 19:16.58 | Yourname`` | API* |
| 19:17.06 | jameswf | jbot: no bri is <reply> BRI == Basic Rate Interface, usually consisting of (1) 64kbps bearer (B) channel, and (1) 8kbps signalling (D) channel |
| 19:17.06 | jbot | okay, jameswf |
| 19:17.07 | [TK]D-Fender | Yourname``: Ditto |
| 19:18.05 | Yourname`` | [TK]D-Fender: When a call is placed through a callfile/Manager, it only goes through the extensions file when the caller picks up/voicemail, etc. If there is no answer, or if it's disconnected.. the call never comes in to the extensions file, which is why SipAddHeader addition is a little confusing. |
| 19:18.12 | Yourname`` | [TK]D-Fender: So what's a good way? |
| 19:18.41 | [TK]D-Fender | Yourname``: Think about the channel TYPE you are placing your call against. |
| 19:18.42 | x86 | jameswf: I was wrong about the D channel, it is 16kbps |
| 19:18.50 | jameswf | argh |
| 19:18.52 | jameswf | :) |
| 19:19.06 | jameswf | jbot: no bri is <reply> BRI == Basic Rate Interface, usually consisting of (1) 64kbps bearer (B) channel, and (1) 16kbps signalling (D) channel |
| 19:19.06 | jbot | okay, jameswf |
| 19:19.21 | x86 | :P |
| 19:19.26 | Yourname`` | [TK]D-Fender: Local? |
| 19:19.46 | [TK]D-Fender | Yourname``: Wow, 1st guess... I'm inpressed. |
| 19:19.52 | [TK]D-Fender | impressed even |
| 19:20.44 | Yourname`` | lol |
| 19:20.49 | Yourname`` | I'm getting better, eh? |
| 19:22.26 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 19:27.05 | stansmith | DaPrez: why were you asking? |
| 19:27.10 | J4k3 | jameswf: uh, james, could you be a little more incorrect |
| 19:27.32 | J4k3 | BRI is a 144k circuit, straight up. |
| 19:27.53 | J4k3 | you've got two B's and one D, how you activate it is up to you. |
| 19:28.15 | x86 | J4k3: by definition, BRI can have a single B channel |
| 19:28.22 | deeperror | anyone in here familiar with etelescript from digisoft? |
| 19:28.27 | J4k3 | x86: its still a fixed 144k circuit. |
| 19:28.34 | x86 | true |
| 19:28.36 | J4k3 | x86: theres still a timeslot, its up to *you* to activate it. |
| 19:28.42 | x86 | *nod* |
| 19:28.49 | J4k3 | thats like saying a frac-T1 is only running [frac] speed |
| 19:29.06 | J4k3 | when in reality its always running 1544k, you just turn up/down channels |
| 19:29.14 | x86 | right |
| 19:29.51 | J4k3 | (Which is fun if your telco is clueless... order 128k frac-t1, get 1408kbit free! |
| 19:30.12 | ManxPower | J4k3: of order a 128K frac T-1 and they hand you IDSL |
| 19:30.43 | x86 | yep |
| 19:30.53 | x86 | IDSL _sucks_ |
| 19:31.02 | x86 | HDSL2 is way better... |
| 19:31.02 | J4k3 | ManxPower: no, they don't. |
| 19:31.05 | stansmith | tommy tech |
| 19:31.08 | ManxPower | I would almost prefer dialup. |
| 19:31.20 | x86 | J4k3: i've been given IDSL on a frac T1 order before |
| 19:31.44 | x86 | J4k3: I made them rip it out and use HDSL2 :p |
| 19:31.44 | J4k3 | x86: when you order a *T1* you should get a *T1* |
| 19:31.49 | ManxPower | Or even better, you order 3Mbps of Bonded T-1s and they hand you an ethernet port. |
| 19:32.08 | x86 | J4k3: it's all DSL anyway these days... either IDSL or (way more common) HDSL2 |
| 19:32.20 | x86 | J4k3: and now they are starting to do HDSL4 also |
| 19:32.28 | ManxPower | and SDSL |
| 19:32.36 | J4k3 | when I order a T1 circuit, I get a T1 |
| 19:32.46 | J4k3 | I dunno what the fuck you're ordering, but a T1 is a T1 |
| 19:32.48 | x86 | J4k3: are you in .us? |
| 19:32.50 | ManxPower | J4k3: Yes, that is the way it is SUPPOSED to work. |
| 19:32.53 | J4k3 | yes, in the USA. |
| 19:32.58 | J4k3 | no, its not 'supposed' |
| 19:33.01 | x86 | J4k3: then they are running DSL to you |
| 19:33.07 | J4k3 | YOU ORDER WHAT YOU NEED, AND THEY HAVE TO DELIVER, AND YOU HAVE TO PAY |
| 19:33.15 | x86 | right |
| 19:33.20 | J4k3 | x86: yes, but the customer interface is a T1. |
| 19:33.26 | x86 | they deliver a T1 over DSL |
| 19:33.27 | x86 | right |
| 19:33.32 | x86 | RJ48C |
| 19:33.34 | J4k3 | x86: I could give a fuck what they're running on their wire. |
| 19:33.45 | J4k3 | I care what my gear gets |
| 19:34.06 | x86 | J4k3: err, my gear still got T1 signalling / framing when they brought it in over IDSL too ;) |
| 19:34.31 | J4k3 | and, quite simply, no telco hangs an IDSL-to-T1 on your wall, cuz that'd basically be a DCS, which costs a bit more than a your typical NIU board. |
| 19:34.37 | J4k3 | or its a really awful hack and should be avoided at all costs. |
| 19:34.50 | J4k3 | x86: wtf, I'da shot the telco man |
| 19:35.00 | mchou | haha |
| 19:35.10 | J4k3 | x86: and then shoved the spent shell in the supervisors ass |
| 19:35.15 | J4k3 | then shoved the gun up the engineers ass |
| 19:35.16 | mchou | lol |
| 19:35.33 | mchou | where's the can of whup-ass? |
| 19:35.48 | J4k3 | mchou: its inside this brass case that says "9MM PARA" |
| 19:36.02 | J4k3 | :D |
| 19:36.16 | x86 | EXIT ONLY |
| 19:36.17 | mchou | haha!! |
| 19:36.39 | J4k3 | x86: you're safe as long as you don't work for the local ILEC :) |
| 19:37.01 | mchou | J4k3: which ILEC is that? |
| 19:37.09 | x86 | J4k3: ILEC? where I live we've just got a shit ton of CLECs... there is no de-facto ILEC here |
| 19:37.22 | J4k3 | mchou: Windstream around here. |
| 19:37.25 | ManxPower | x86: where do you live? |
| 19:37.31 | x86 | J4k3: you can get POTS service from like 5 companies here :P |
| 19:37.35 | J4k3 | x86: there are some operating CLECs here, but they're all in the projects. |
| 19:37.35 | x86 | ManxPower: Peoria, IL |
| 19:37.37 | J4k3 | and I'm white. |
| 19:37.50 | x86 | ouch... profiling... |
| 19:37.56 | ManxPower | x86: So Verizon is your ILEC unless you are in an Independent area |
| 19:38.01 | stansmith | x86: LOL |
| 19:38.08 | J4k3 | Verizon, eww. |
| 19:38.15 | mchou | nas-t |
| 19:38.16 | x86 | ManxPower: actually verizon is like the only company you can't get service through here |
| 19:38.26 | J4k3 | hmm, I wonder if VErizon still has their entirely incompetent 'isp/clec' office |
| 19:38.33 | ManxPower | J4k3: Formerly Illinois Bell, formerly Ameritech |
| 19:38.38 | J4k3 | they had back when they called themselves GTE |
| 19:38.48 | x86 | ManxPower: Illinois Bell / Ameritech / SBC / AT&T |
| 19:39.02 | ManxPower | x86: Ah, yes, sorry. |
| 19:39.06 | x86 | ManxPower: that's the path of migration... Verizon was never in that mix |
| 19:39.10 | ManxPower | so SBC/AT&T is your ILEC |
| 19:39.24 | x86 | they are a CLEC here |
| 19:39.28 | J4k3 | ahh, AT&T is probably the least evil but most difficult to work with. |
| 19:39.38 | J4k3 | AT&T = no matter who you're talking to, whatever you need is not their job. |
| 19:39.40 | ManxPower | J4k3: That is very true. |
| 19:39.45 | jm|laptop | bah |
| 19:39.53 | jm|laptop | <PROTECTED> |
| 19:40.00 | x86 | J4k3: no kidding, i'm doing a huge managed MPLS implementation through the,... |
| 19:40.02 | ManxPower | AT&T always wants to hand us service as an ethernet service. |
| 19:40.06 | jm|laptop | I keep getting this when asterisk tries to connect to my mobile |
| 19:40.11 | J4k3 | buuut, once you find who you need they're like "oh, no problem!" and its done promptly and professionally |
| 19:40.50 | J4k3 | ManxPower: in some cases that could be cute, but only if they're willing to offer real redundancy behind it |
| 19:41.01 | J4k3 | ie - wiring you through two COs on different fiber |
| 19:41.02 | x86 | ManxPower: to me, an ILEC is a company that you're forced to go through for POTS service / the company that is always going to be the last-mile from CPE to CO |
| 19:41.17 | x86 | ManxPower: here, that could be 5 different companies, so to me they are all CLECs |
| 19:41.42 | J4k3 | well, I Doubt that |
| 19:41.55 | J4k3 | I seriously doubt there are 5 different LECs with wire in the ground in front of your house. |
| 19:42.07 | ManxPower | J4k3: We let them install the "VINA" box or the Adtran box, then we disconnect the box and put plug everything into our own CSU/DSU to split the voice and data. |
| 19:42.10 | x86 | they often share wiring here |
| 19:42.28 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 19:42.32 | x86 | J4k3: there are 5 different LECs with 5 different SONET rings around town :) |
| 19:42.49 | J4k3 | x86: yeah, but you live in Chicago. |
| 19:42.51 | J4k3 | 'live' |
| 19:43.01 | x86 | 3 hours from it, yeah |
| 19:43.02 | J4k3 | its the burbs, but damn, f that. |
| 19:43.15 | J4k3 | it takes 3 hours? f that traffic :) |
| 19:43.19 | x86 | I'm at least 2.5 hours from the closest chicago burb |
| 19:43.36 | x86 | i'm in the central part of the state, chicago is way the hell up north |
| 19:43.39 | *** join/#asterisk angryuser (n=angyuser@df01t2-212-194-87-141.d4.club-internet.fr) |
| 19:43.59 | angryuser | hi @all |
| 19:44.08 | J4k3 | http://en.wikipedia.org/wiki/Image:Network_Interface_Unit.jpg |
| 19:44.10 | stansmith | ~hi angryuser |
| 19:44.10 | jbot | Many greetings, angryuser, most strange traveller, to this IRCdom of plenty. |
| 19:44.19 | J4k3 | ghetto, their telco didn't give them the metal backplate for those :P |
| 19:45.23 | iCEBrkr | Holy crap. Ekiga works with Asterisk. |
| 19:45.53 | angryuser | i have reinstalled my asterisk server, well all is working fine, i have never used "make config" command in asterisk(not zaptel) install, what kind of script make config install ? |
| 19:45.55 | jameswf | ekiga sounds like that woman exersize |
| 19:45.58 | ManxPower | iCEBrkr: everyone except you knew that already |
| 19:46.03 | stansmith | sounds like a cheap automobile |
| 19:46.05 | iCEBrkr | ManxPower: haha |
| 19:46.20 | jameswf | oh like ekiga furnature? |
| 19:46.41 | angryuser | show me that script plzzzz |
| 19:46.42 | iCEBrkr | ha |
| 19:47.07 | iCEBrkr | It doesn't work all that well, but I was able to check voicemail. |
| 19:47.09 | iCEBrkr | whoopie |
| 19:48.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:48.15 | x86 | OMIGODHAHAHA XLITE WORKS WIF ASTERICKS LMAO ROFFLECOPTER |
| 19:48.24 | stansmith | LOL |
| 19:48.27 | iCEBrkr | ZOMGWTFBBQ?! |
| 19:48.37 | x86 | FOR GREAT JUSTICE! MOVE EVERY SIP! |
| 19:48.47 | x86 | SOMEONE SET US UP TEH IAX! |
| 19:48.51 | iCEBrkr | haha |
| 19:49.05 | ACiDV | you know what feeling I got when I discover that IAXy are compatible w/ * |
| 19:49.22 | ManxPower | I wish I could audomatically kick/ban people that do that |
| 19:49.41 | x86 | heh |
| 19:49.43 | iCEBrkr | ManxPower: Don't you have apwzor powerz? |
| 19:50.06 | ManxPower | iCEBrkr: not that I know of (assuming I even could understand what you just said) |
| 19:50.25 | iCEBrkr | ManxPower: you don't have ops? You've been here long enough, I guess I kind of figured you did. |
| 19:50.26 | x86 | ManxPower: aka opwhore |
| 19:50.35 | ManxPower | I guess I should pull up my "teen to adult translation service" web site. |
| 19:50.49 | JenniferAkemi | ouch |
| 19:51.09 | iCEBrkr | Hrrm, if I were only a teen again.. |
| 19:51.10 | ManxPower | iCEBrkr: I have never had ops. I don't know if I am authorized or not, as I have no idea how to even be opped |
| 19:51.17 | iCEBrkr | The damnage^H^H^H^H^Htrouble I'd be in |
| 19:52.33 | x86 | ManxPower: /m chanserv list #asterisk |
| 19:52.40 | J4k3 | ManxPower, your attitude is very efnet compatable :) |
| 19:52.40 | x86 | ManxPower: /m chanserv op #asterisk ManxPower |
| 19:52.46 | x86 | J4k3: no kidding |
| 19:52.53 | iCEBrkr | efnet compatible. haha |
| 19:53.02 | x86 | J4k3: aka "sad old man" compatible |
| 19:53.08 | ManxPower | I figure if I don't have ops then I cannot be drawn into the politics of being a op |
| 19:53.12 | iCEBrkr | <PROTECTED> |
| 19:53.24 | iCEBrkr | ManxPower: How true |
| 19:53.24 | J4k3 | ManxPower: yep, thats my feelings on opdom these days |
| 19:53.30 | ManxPower | ChanServ: (notice) An access level of [10] is required for [OP] on #asterisk |
| 19:53.37 | JenniferAkemi | do i have this right? in terms of sip, user is outbound, peer is inbound and friend is both outbound and inbound? |
| 19:53.42 | ManxPower | so I guess the answer is "no" |
| 19:53.51 | ManxPower | JenniferAkemi: you have it reversed. |
| 19:53.55 | JenniferAkemi | ok |
| 19:54.20 | ManxPower | always from the perspective of the server. |
| 19:54.28 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:54.34 | JenniferAkemi | so like a user would be jack at his desk, and a peer would be like At&T voip provider service thing |
| 19:54.36 | JenniferAkemi | ? |
| 19:54.48 | JenniferAkemi | and if At&T gave me DID's they'd be a friend? |
| 19:54.56 | x86 | JenniferAkemi: right |
| 19:54.59 | deeperror | at&t has voip service? |
| 19:55.04 | JenniferAkemi | (that was an example) |
| 19:55.05 | x86 | deeperror: yep |
| 19:55.11 | deeperror | what do they call it? |
| 19:55.12 | x86 | JenniferAkemi: they do, though ;) |
| 19:55.16 | JenniferAkemi | cool :D |
| 19:55.29 | x86 | deeperror: not sure... they bring in a dedicated data circuit and let you run VoIP across it |
| 19:55.30 | deeperror | i've contacted account reps and regional managers and they all tell me it's not possible for asterisk |
| 19:55.47 | x86 | deeperror: that's because they have no clue what "Asterisk" is |
| 19:56.03 | deeperror | i was at VON/digium asteirsk world i would assume they might have some clue |
| 19:56.12 | x86 | deeperror: tell them you've got an Intertel with an IPRC card in it... they'll give you SIP :) |
| 19:56.18 | ManxPower | deeperror: tell them you have Cisco Call Manager or Avaya VoIP Thingy |
| 19:56.25 | x86 | yep |
| 19:56.42 | deeperror | blah att sucks the life from me |
| 19:56.58 | x86 | they are highly reluctant to say anything good about Asterisk, assuming you can actually talk to someone at AT&T that has ever heard of it |
| 19:57.12 | iCEBrkr | x86: You get what you pay for! |
| 19:57.12 | x86 | deeperror: they are expensive for LD... |
| 19:57.22 | x86 | right... |
| 19:57.35 | deeperror | we use att for internet and route calls to lvl3 reseller it seems to work very well at the moment |
| 19:57.45 | iCEBrkr | ...at Least that's what Brighthouse/TimeWarner told me when I complained about my cable connecting sucking ass. |
| 19:57.59 | iCEBrkr | Of cousre they want you to use their 'digital phone service' |
| 19:57.59 | x86 | deeperror: probably because AT&T uses a lot of level(3)'s fiber ;) |
| 19:58.00 | deeperror | hope when our contract goes out with att we can get lvl3 to provide us with service but att had to install the fiber to the building |
| 19:58.13 | angryuser | do i need to start misdn (misdn-init start) each time before starting asterisk ? |
| 19:58.30 | x86 | deeperror: AT&T is your LEC then |
| 19:58.39 | angryuser | or * init script handla that ? |
| 19:58.45 | angryuser | *handle |
| 19:58.52 | deeperror | have opti-man 10m connection here |
| 19:59.02 | deeperror | maybe opt-e-man is what they call it ha |
| 19:59.11 | x86 | deeperror: opt-e-man yeah... |
| 19:59.38 | deeperror | x86: i then terminate to sotelips |
| 20:00.04 | x86 | I've got (14) T1's sitting downstairs... I think I'm going to put in a DS3 over fiber and mux them all over that |
| 20:00.07 | deeperror | who is on the level3 network i get 14ms pings to them all day |
| 20:00.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:00.35 | x86 | deeperror: AT&T almost always uses level3 for long haul |
| 20:00.49 | x86 | deeperror: level3 has more fiber in the ground than AT&T does ;) |
| 20:00.52 | deeperror | x86: yea that is what i was seeing |
| 20:01.20 | deeperror | everything was fine until it left their network then i would get 180ms pings and horrible echo after we got to 20 concurrent calls |
| 20:01.44 | x86 | welcome to net nuetrality! |
| 20:01.46 | deeperror | do you know anything about their platinum/silver access levels? |
| 20:01.52 | x86 | nope |
| 20:01.57 | x86 | level3 does VoIP too |
| 20:02.18 | x86 | something like a million minutes per month minimum |
| 20:02.19 | deeperror | nor I, but i heard them talking about some crazy stuff like qos...the thing that gets me is they take all the standards and change the names so no one knows what is going on |
| 20:02.39 | x86 | QoS is a very standard term heh |
| 20:02.47 | *** join/#asterisk AndyGraybeal (n=andy@node180.39.251.72.1dial.com) |
| 20:03.14 | deeperror | yea i couldn't sign up with lvl3 direct so had to go with one of their resellers and things are rocking now pushed 160K in january |
| 20:04.04 | deeperror | x86: i know qos is standard but their sales staff call it platinum/silver/gold access levels or something crazy like that |
| 20:04.48 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:05.44 | JenniferAkemi | i'm confused by this section in the book |
| 20:06.03 | JenniferAkemi | chapter 12, the database part. page 271 |
| 20:06.04 | x86 | deeperror: that relates to the level of service they give you an SLA on |
| 20:06.15 | JenniferAkemi | actually 272 |
| 20:06.38 | x86 | deeperror: generally different levels of guaranteed packet delivery percentage, and on-net latency |
| 20:06.44 | JenniferAkemi | it shows the fields that are defined for the sip user table, but there isn't a context, or a type field. is that supposed to be like that? |
| 20:07.22 | x86 | JenniferAkemi: all of mine have context and type :) |
| 20:07.43 | JenniferAkemi | x86: ok that's what i thought would be normal |
| 20:07.46 | JenniferAkemi | thanks |
| 20:08.13 | x86 | JenniferAkemi: next time you see lmadsen let him know |
| 20:08.25 | x86 | JenniferAkemi: he's the primary author of that book |
| 20:09.03 | JenniferAkemi | does he upate it though? i don't want to keep bringing stuff that he already knows about to his attention. that could be really annoying |
| 20:09.09 | JenniferAkemi | i'm sure many people before me have read this :) |
| 20:09.58 | x86 | not sure what the process is, but just let him know anyway :P |
| 20:10.14 | JenniferAkemi | yes boss |
| 20:10.47 | twisted | has anyone played with updog integration into asterisk? |
| 20:11.23 | JenniferAkemi | i can't believe i've managed to get this working |
| 20:12.28 | stansmith | when you put your mind to it, theres a lot you can achieve |
| 20:12.33 | stansmith | write that down |
| 20:17.29 | JenniferAkemi | is the asterisk realtime database stuff commonly used? |
| 20:18.00 | x86 | I use it |
| 20:19.38 | ZaVoid | i use it all the time |
| 20:20.00 | JenniferAkemi | what database do you guy suse |
| 20:20.05 | JenniferAkemi | "guys use" |
| 20:20.12 | stansmith | postgresql |
| 20:20.50 | *** join/#asterisk miguel3239 (n=elguero@207.154.34.162) |
| 20:21.06 | JenniferAkemi | hmm. with dynamic realtime if your db server goes down, so does your vip service? |
| 20:21.10 | JenniferAkemi | that's kind of bad |
| 20:22.16 | mchou | heh, high availability |
| 20:22.29 | mchou | minimize single points of failure |
| 20:22.47 | JenniferAkemi | are you talking to me? |
| 20:22.48 | mchou | of course with voip, that's all a crapshoot anyways :) |
| 20:22.55 | JenniferAkemi | can you make it have a backup db server? |
| 20:23.00 | Duplicator | Hey folks - is there a way to 'segment' 4 zap channels on the same card to have different dial rules? |
| 20:23.26 | mchou | JenniferAkemi: failover is a bit expensive, but you can do it |
| 20:23.34 | JenniferAkemi | what do you mean expensive |
| 20:23.50 | JenniferAkemi | in terms of money or cpu i mean |
| 20:24.07 | mchou | redunancy expensive. 2.5x everything (HW, SW, admin) |
| 20:24.13 | JenniferAkemi | i already have 2 database servers that are redundant |
| 20:24.24 | JenniferAkemi | and replicating |
| 20:24.36 | mchou | JenniferAkemi: ok then, go ahead and use them |
| 20:24.42 | JenniferAkemi | definately want redundant :D |
| 20:24.52 | mchou | JenniferAkemi: just make sure failover works reliably and you're good to go |
| 20:24.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:25.16 | JenniferAkemi | i'll have to do some googling about "failover and asterisk" ;) |
| 20:25.44 | mchou | many howtos exist |
| 20:25.52 | JenniferAkemi | i'm sure they do |
| 20:26.15 | JenniferAkemi | i have to say |
| 20:26.23 | deeperror | Duplicator: set them to different contexts? |
| 20:26.38 | JenniferAkemi | this is such a refreshing change from working with Harris Switches where there is absoultely NO information available online |
| 20:27.41 | Duplicator | deeperror: yes - basically I've got a Sangoma card, and I want to reference by channel, not by group |
| 20:28.10 | Duplicator | deeperror: But I don't know how to reference it by port? |
| 20:28.23 | [TK]D-Fender | Duplicator: What kind of channels are they? |
| 20:28.41 | Duplicator | [TK]D-Fender: FXO's |
| 20:29.07 | stansmith | they have a lot of phiber running under downtown columbus |
| 20:29.09 | [TK]D-Fender | Duplicator: Then in zapata just set each channel into its own context for inbound calls. For outbound calls just use the channel you like. |
| 20:29.12 | mchou | JenniferAkemi: I'd be much more concerned with VSP availability :) |
| 20:29.42 | Duplicator | [TK]D-Fender: But where do I know how to reference each channel? That's where I'm stuck. |
| 20:29.54 | JenniferAkemi | mchou: i don't eve know what that is |
| 20:29.55 | [TK]D-Fender | Duplicator: Dial(Zap/1/1234567) |
| 20:30.11 | mchou | JenniferAkemi: Voip service provider |
| 20:30.11 | JenniferAkemi | mchou: unless you mean virtual single processor, and i still don't know what that is :P |
| 20:30.13 | [TK]D-Fender | Duplicator: There. Dials out port 1. |
| 20:30.21 | JenniferAkemi | mchou: not going to be using those |
| 20:30.47 | Duplicator | [TK]D-Fender: Ahhh! Thank you! |
| 20:30.53 | JenniferAkemi | just going to go out on the PSTN and in on voip |
| 20:30.53 | mchou | JenniferAkemi: who provides call termination for you? |
| 20:31.11 | mchou | JenniferAkemi: ok. You're rolling your own |
| 20:31.15 | JenniferAkemi | mchou: we do heh. it will be connected to our "big" switch |
| 20:31.38 | JenniferAkemi | is finding vsp's hard? |
| 20:31.50 | mchou | JenniferAkemi: it's a PITA |
| 20:31.58 | JenniferAkemi | maybe we could become one |
| 20:32.12 | mchou | unless you are "enterprise" class |
| 20:32.41 | JenniferAkemi | I know MCI sells it |
| 20:32.41 | [TK]D-Fender | mchou: I don't roll my own PITAs, I buy the value-meal trio and enjoy my free time :) |
| 20:32.48 | mchou | JenniferAkemi: where do I sign up? :) |
| 20:33.04 | JenniferAkemi | what rates would you need |
| 20:33.14 | mchou | JenniferAkemi: minmal :) |
| 20:33.16 | JenniferAkemi | haha |
| 20:33.34 | mchou | JenniferAkemi: 2 simultaneous call apperances :) |
| 20:33.42 | JenniferAkemi | what does that mean |
| 20:34.09 | mchou | means you only need to handle 2 simultaneous calls for me |
| 20:34.09 | JenniferAkemi | do you think any vsp's use asterisk? |
| 20:34.16 | mchou | plenty |
| 20:34.20 | JenniferAkemi | that's not enough revenue :P |
| 20:34.31 | JenniferAkemi | you can fit hardly any minutes on 2 channels |
| 20:34.48 | JenniferAkemi | i guess it's not a wholesale thing, more a retail thing |
| 20:34.55 | mchou | bingo! |
| 20:35.09 | JenniferAkemi | actually what's the difference between a vsp and just signing up for voip service |
| 20:35.22 | JenniferAkemi | the plan is to sell voip service |
| 20:35.29 | JenniferAkemi | that's pretty much what you're talking about i think |
| 20:35.35 | mchou | same deal, pretty much |
| 20:35.45 | JenniferAkemi | i'll let you know when we're ready for customers ;) |
| 20:35.48 | JenniferAkemi | first i have to get it all working |
| 20:35.49 | JenniferAkemi | haha |
| 20:36.14 | *** join/#asterisk lateil (n=oz@88.239.73.53) |
| 20:37.55 | *** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net) |
| 20:39.06 | hmmhesays | ahh customers |
| 20:39.11 | hmmhesays | they're always fun to deal with |
| 20:39.36 | hmmhesays | holy carp I haven't heard this song in forever |
| 20:39.44 | stansmith | p. diddy? |
| 20:39.47 | mchou | everyone wants to shun the customer |
| 20:39.51 | hmmhesays | "Lit - Miserable" |
| 20:40.06 | J4k3 | hah first concert I ever went to was Lit+Garbage |
| 20:40.06 | mchou | but maybe that's Vonage's value-add |
| 20:40.22 | hmmhesays | Lit used to rock, garbage was so/so |
| 20:40.30 | hmmhesays | and what are you like 12? |
| 20:40.39 | mchou | if they go out of bizness I'm buying all their HW :) |
| 20:40.44 | stansmith | i was 12 when lit came out |
| 20:40.56 | J4k3 | hrm that was 97, I'm 29 now... |
| 20:41.02 | J4k3 | so 18 |
| 20:41.02 | stansmith | o wait |
| 20:41.05 | stansmith | no i was 10 |
| 20:41.15 | deeperror | my first show was pantera ha |
| 20:41.15 | hmmhesays | I suppose lit is older now |
| 20:41.19 | J4k3 | girlfriend got the tickets, 3rd row.. ended up against the stage. |
| 20:41.21 | hmmhesays | I just turned 26 a few days ago |
| 20:41.30 | stansmith | happy belated |
| 20:42.01 | hmmhesays | lit had to be pretty awesome live |
| 20:42.14 | J4k3 | suprisingly so |
| 20:42.33 | J4k3 | everyone was like "ho-hum we're waiting for garbage" at first but by the time they left the stage everybody was pumped |
| 20:42.35 | *** join/#asterisk ellanero (n=marc@183-230-221-213-pool.cable.fcom.ch) |
| 20:42.52 | hmmhesays | if you're a pop/punk band you gotta do that otherwise you suck |
| 20:42.55 | J4k3 | but, I also watched them at a school where I'd guess at least 50% of the population is lesbians. |
| 20:43.03 | J4k3 | its like 85% female. |
| 20:43.05 | hmmhesays | every time my band plays "myown worst enemy" everyone still goes nuts |
| 20:43.17 | mchou | J4k3: which school was this? |
| 20:43.18 | lateil | hello, anyone used spandsp? i am trying to write a tone detection application with it .. but no succes |
| 20:43.22 | J4k3 | mchou: sfasu.edu |
| 20:43.51 | x86 | hmmhesays: it's one of my favorites :) |
| 20:43.59 | x86 | hmmhesays: (my own worst enemy) |
| 20:44.00 | hmmhesays | x86: yeah that tune just rocks |
| 20:44.00 | [TK]D-Fender | hmmhesays: Easy songs to follow that would be "Movies" by AAF, "Basket Case" by Green Day |
| 20:44.05 | mchou | J4k3: lol, there are lesbians in TX??? |
| 20:44.10 | hmmhesays | lateil, try talk to coppice |
| 20:44.12 | [TK]D-Fender | hmmhesays: Both in the same key |
| 20:44.26 | x86 | [TK]D-Fender: AAF? |
| 20:44.30 | J4k3 | mchou: dude, put it this way... its damned hard to find a girl that *isn't* bi around here. |
| 20:44.34 | [TK]D-Fender | x86: Alient Ant Farm |
| 20:44.34 | stansmith | LOL@AAF |
| 20:44.40 | J4k3 | well, at least, isn't claiming bi |
| 20:44.44 | hmmhesays | [TK]D-Fender, actually I usually end up playing basket case in Eb, mowe is in E |
| 20:44.46 | J4k3 | a think a lot of them claim it for attention |
| 20:44.58 | mchou | J4k3: for real? |
| 20:45.01 | x86 | [TK]D-Fender: ah... was thinking you perhaps meant AHF (american hi-fi) |
| 20:45.02 | [TK]D-Fender | hmmhesays: Shouldn't detune... |
| 20:45.03 | J4k3 | they make out with chicks to get dudes to buy them drinks... I'm like fthat, get that bitch to buy you a drink. |
| 20:45.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:45.32 | hmmhesays | I can sing harmonies so much better in Eb |
| 20:45.46 | lateil | hmmhesays: thnx |
| 20:45.48 | mchou | J4k3: damn, I always thought TX was up tight with evangelicals (except Austin) |
| 20:46.07 | hmmhesays | I show chicks my massive ego to get them to buy me drinks |
| 20:46.18 | J4k3 | hmmhesays: pimp. |
| 20:46.18 | mchou | hmmhesays: lol |
| 20:46.32 | [TK]D-Fender | hmmhesays: Basket Case is in a comfortable octave. A lot of the stuff I like is one higher at which point if they scale up, my voice cracks on it. |
| 20:46.39 | J4k3 | I'm like... lets skip the drink, lets move on to sex and breakfast. |
| 20:46.43 | [TK]D-Fender | hmmhesays: so the natural tuning is fairly easy |
| 20:47.04 | hmmhesays | [TK]D-Fender, true it is. I alway tune to standard Eb though, |
| 20:47.21 | [TK]D-Fender | hmmhesays: I should keep a guitar in spare for that tuning myself..... |
| 20:47.31 | [TK]D-Fender | hmmhesays: now that I've got 3 I can certainly afford it. |
| 20:47.41 | hmmhesays | if we really need to play a song in the original key I usually just power chord it |
| 20:48.09 | *** join/#asterisk lvl- (n=lvl@145.52.248.101) |
| 20:48.19 | tzafrir_home | lateil, well, coppice is not here right now. He has been around lately |
| 20:48.36 | mchou | sigh. POS motherboard |
| 20:48.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:48.56 | hmmhesays | I'm considering a 7 string for B stuffs |
| 20:48.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:49.09 | lateil | tzafrir_home: thnx, i will wait him |
| 20:49.12 | hmmhesays | a shecter omen 7 is only like 300 bucks |
| 20:49.28 | *** join/#asterisk AndyGraybeal (n=andy@node180.39.251.72.1dial.com) |
| 20:50.05 | tzafrir_home | lateil, I note that you haven't actually asked a specific question |
| 20:50.14 | hmmhesays | speaking of songs to follow my own worst enemy. 99 red balloons is the exact same chord progression, in the same key |
| 20:50.56 | hmmhesays | (goldfinger remake) |
| 20:51.47 | Katty | hmmhesays: do you have asterisk starting in rc.local? |
| 20:52.05 | hmmhesays | rc3.d |
| 20:52.17 | Katty | what's your line look like? |
| 20:52.18 | hmmhesays | on my debian boxes |
| 20:52.30 | hmmhesays | which line? |
| 20:52.36 | Katty | that starts asterisk |
| 20:52.48 | hmmhesays | i use the script out of the source |
| 20:53.10 | hmmhesays | ~src/contrib/init.d/ if memory serves |
| 20:53.22 | *** part/#asterisk elllanero (n=marc@183-230-221-213-pool.cable.fcom.ch) |
| 20:53.24 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:54.43 | robeph | ok I don't get what the problem is here. http://rafb.net/p/Gzc7Ad24.html |
| 20:55.00 | robeph | says I need zaptel, yet its there and was inserted when I ran asterisk |
| 20:55.16 | ManxPower | robeph: What was inserted? |
| 20:55.21 | robeph | the module |
| 20:55.30 | robeph | ztdummy / zaptel etc |
| 20:55.47 | robeph | just meant before someone says "restart asterisk after you insert the modules" |
| 20:55.57 | ManxPower | robeph: If you want to use IAX2 trunking (reduces the bandwidth when 3 or more calls are going between the two endpoints) then you need to have Zaptel installed when you BUILD Asterissk |
| 20:56.11 | robeph | ah |
| 20:56.20 | ManxPower | If you don't need trunking then you should not have trunk=yes |
| 20:56.23 | robeph | I had it on the machine, jusyt the modules not in place |
| 20:56.26 | robeph | i do |
| 20:56.34 | robeph | lemme try that |
| 20:56.59 | ManxPower | robeph: is zaptel and ztdummy listed in the output in "lsmod" |
| 20:57.01 | waKKu | folks.. does some incompatability between IAX and pickupgroups ? I'm trying to use it with IAX, but doesnt work.. with sip works perfect.. |
| 20:57.10 | robeph | its all in the pasted link |
| 20:57.40 | ManxPower | robeph: looks to me that you installed zaptel after you installed Asterisk |
| 20:57.56 | ManxPower | waKKu: iax2 does not support pickup groups |
| 20:58.06 | robeph | yeah rebuilding asterisk now to check it out |
| 20:58.19 | robeph | i may just have forgotten to use flag |
| 20:59.02 | lateil | tzafrir_home: i can't figure out how to send audio samples to super_tone_rx(). |
| 21:00.09 | stansmith | ? |
| 21:00.17 | [TK]D-Fender | robeph: After compiling Zaptel, you have to recompile * again. |
| 21:00.41 | [TK]D-Fender | robeph: And before running * do "ztcfg -vvvv" |
| 21:00.43 | robeph | [TK]D-Fender: yeah, I'm using gentoo, it looks like it has a use flag to enable |
| 21:00.48 | ManxPower | [TK]D-Fender: what IS the command in 1.4 to make ./configure re-read everything and find zaptel |
| 21:01.00 | robeph | that is the problem, I have zaptel installed already, just not telling it to use it when I emerge it |
| 21:01.09 | [TK]D-Fender | ManxPower: Not sure actually, I jsut trasht he entire folder and re-extract :) |
| 21:01.11 | waKKu | ManxPower http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups even with 1.4 ? |
| 21:01.13 | ManxPower | robeph: We ONLY support Asterisk installs from SOURCE, not a package or a patched Asterisk |
| 21:01.14 | stansmith | make menuselect? |
| 21:01.48 | ManxPower | waKKu: The Wiki is a terrible source for info. |
| 21:02.19 | robeph | ManxPower: .... |
| 21:02.42 | ManxPower | waKKu: "group" is found nowhere in 1.4's iax.conf.sample |
| 21:03.14 | waKKu | hmm |
| 21:03.16 | waKKu | damn |
| 21:03.30 | [TK]D-Fender | waKKu: Pastebin what you've done |
| 21:04.05 | ManxPower | waKKu: Asterisk's config parser does NOT report invalid options |
| 21:04.09 | tzafrir_home | robeph, hmmm.... actualy the erquirement to rebuild asterisk after building zaptel is not that strict, if the same zaptel was already there |
| 21:04.17 | tzafrir_home | (the same zaptel.h) |
| 21:04.36 | robeph | tzafrir_home: I know the problem ;p i got it workin now |
| 21:04.45 | ManxPower | tzafrir_home: and if zaptel was not installed when asterisk was built? |
| 21:04.49 | robeph | tzafrir_home: it's the lack of zaptel support via gentoos pkg management |
| 21:05.06 | ManxPower | robeph: Now you can see why we don't support packages |
| 21:05.06 | robeph | tzafrir_home: since I have to be explicit in how to build it via that. |
| 21:05.25 | JenniferAkemi | what's the difference between user and username in sip.conf |
| 21:06.00 | robeph | ManxPower: no need to keep on that subject, I understand why, however, regardless of that factor you also must realize that even though you don't support said packages, that nonetheless the answer was the same and provided the same assistance as it would have for somone building from source, in this case. |
| 21:06.19 | tzafrir_home | ManxPower, packages are a different but generally moer reproducable way of building. |
| 21:06.29 | ManxPower | JenniferAkemi: one is a valid option, one is not a valid option and one is not. |
| 21:06.42 | tzafrir_home | But then again, with a Gentoo package there are many possible reproductions |
| 21:06.46 | ManxPower | tzafrir_home: that is all well and good,. but WE should not be asked to support them |
| 21:07.07 | ManxPower | JenniferAkemi: One is a valid sip.conf option and one is not. That is the difference. |
| 21:07.28 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:07.37 | JenniferAkemi | ManxPower: do i use user in sip.conf and username in the database? |
| 21:07.56 | ManxPower | robeph: you are one of the few that can diagnose their pwn package problem. |
| 21:08.04 | ManxPower | JenniferAkemi: USER= IS NOT A VALID SIP.CONF OPTION. |
| 21:08.08 | JenniferAkemi | i thought the field names in the database where supposed to be the same as the categories in the sip.conf |
| 21:08.13 | JenniferAkemi | ManxPower: oops. ok |
| 21:08.20 | JenniferAkemi | ManxPower: i think it works though |
| 21:08.39 | ManxPower | JenniferAkemi: I cannot help you further. |
| 21:09.02 | JenniferAkemi | ManxPower: you are cracking me up. thanks |
| 21:09.07 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 21:09.09 | *** part/#asterisk Porks (n=Porks@201.62.79.12) |
| 21:10.31 | waKKu | [TK]D-Fender http://www.pastebin.ca/918030 .. when I call and try to pickup (*5), CLI shows me a message like I was trying to call *5@context from exten ... |
| 21:10.49 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 21:11.30 | Katty | i'm trying to dial out through a sangoma pri card, and it's tell me unable to create channel of type "zap" (cause 34 - circuit/channel congestion) |
| 21:11.34 | dmz | howdy y'all, i upgraded from 1.3 -> 1.4 and now my queues are not connecting or queuing properly (dropping call instead of adding, or when briding connected call it just sits there). is there anything about this upgrade tha ti have to change in my queues? |
| 21:11.37 | Katty | is that an error with my syntax, or an error on the pri? |
| 21:11.43 | [TK]D-Fender | waKKu: Show me your CLI output for the complete call attempt |
| 21:11.53 | waKKu | ok, 1 minute |
| 21:11.57 | ManxPower | Katty: I assume that zttool shows the line as green? |
| 21:12.07 | robeph | ls |
| 21:12.09 | robeph | ... |
| 21:12.19 | ManxPower | [TK]D-Fender: he's trying to use pickup groups on iax2 channels, I don't think iax2 supports that. |
| 21:12.25 | Katty | ManxPower: i don't know how to use zttool |
| 21:12.36 | [TK]D-Fender | ManxPower: Sure it should. |
| 21:12.38 | ManxPower | dmz: nothing in upgrade.txt was helpful? |
| 21:12.48 | ManxPower | [TK]D-Fender: Of course it SHOULD, but I believe it doesn't. |
| 21:12.51 | dmz | not that i saw for queues |
| 21:12.53 | dmz | i'll look again |
| 21:13.06 | ManxPower | Katty: cat /proc/zaptel/1 Do you see an alarm (assuming your PRI is on span 1 |
| 21:13.10 | [TK]D-Fender | ManxPower: Well I don't currently have reason to believe otherwise. |
| 21:13.27 | *** join/#asterisk neoalex (n=chatzill@user-0ccens8.cable.mindspring.com) |
| 21:13.29 | Katty | ManxPower: everything is labeled as Clear (In Use) |
| 21:13.37 | Katty | ManxPower: i'm not sure what i'm looking at. would you like a pastebin? |
| 21:13.41 | ManxPower | [TK]D-Fender: I encountered the issue trying to help a user with an IAXy |
| 21:13.43 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:13.45 | waKKu | [TK]D-Fender http://www.pastebin.ca/918035 at end page |
| 21:13.54 | ManxPower | Katty: I'll pastebin a working one for you. |
| 21:13.59 | Katty | ok |
| 21:14.18 | neoalex | asterisk should work on xandros right? |
| 21:14.30 | stansmith | neoalex: ya why wouldnt it |
| 21:14.49 | ManxPower | Katty: pastebin /etc/asterisk/zapata.conf |
| 21:14.50 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:14.50 | neoalex | what would I need to install so I can compile asterisk? |
| 21:14.58 | Katty | ManxPower: k |
| 21:15.02 | stansmith | neoalex: gcc |
| 21:15.17 | jm|laptop | bah |
| 21:15.20 | neoalex | do I need the kernel source too? |
| 21:15.23 | ManxPower | Katty: see http://pastebin.ca/918037 |
| 21:15.31 | [TK]D-Fender | waKKu: PB your entire features.conf, and have you completely restarted * after your last change? |
| 21:15.35 | ManxPower | neoalex: try reading the README or INSTALL |
| 21:15.53 | tzafrir_home | neoalex, what Linux distribution do you use? |
| 21:15.53 | ManxPower | neoalex: asterisk does not require kernel source, zaptel does. |
| 21:16.08 | ManxPower | ManxPower: Does chan_iax2.c support pickup groups? |
| 21:16.08 | ManxPower | file: no. |
| 21:16.09 | [TK]D-Fender | ManxPower: Thought it only needed the headers, not full source. |
| 21:16.09 | Katty | ManxPower: http://pastebin.ca/918039 |
| 21:16.18 | neoalex | I am looking at asterisk.org/support/install but the list seems to be short |
| 21:16.25 | waKKu | [TK]D-Fender nops.. just reload respective modules.. |
| 21:16.41 | Katty | ManxPower: i am not 100% confident the telco is giving me proper info |
| 21:16.48 | [TK]D-Fender | waKKu: Restart, test, them PB as requested if unsuccessful |
| 21:17.05 | ManxPower | Katty: most PRIs use switchtype=national |
| 21:17.09 | neoalex | tzafrir_home: xandros |
| 21:17.12 | stansmith | i jump at least 6 inches above project fences |
| 21:17.18 | bkruse | ManxPower: I would say that is the safest guess |
| 21:17.35 | Katty | ManxPower: i'll give that ashot |
| 21:17.50 | ManxPower | [TK]D-Fender: it's official from both file and kpflemming. chan_iax2 does NOT support pickup groups. |
| 21:18.11 | tzafrir_home | neoalex, drop that. I'm serious |
| 21:18.30 | tzafrir_home | Or use apt sources and upgrade it to a decent Debian |
| 21:18.41 | stansmith | LOL |
| 21:19.04 | tzafrir_home | Xandros has been badly performing in providing decent software, security updates and such |
| 21:19.44 | Katty | ManxPower: http://pastebin.ca/918042 <- i've tried calling some of the lines to see if i can get an incoming call. there's a message from the telco saying "the number you are calling is temporarily unavailable" |
| 21:19.45 | waKKu | [TK]D-Fender again : ) http://www.pastebin.ca/918043 ... more things |
| 21:20.06 | JenniferAkemi | Katty: sounds like you need to call your telco and tell them you want to "bring up the PRI" |
| 21:20.16 | ManxPower | Katty: for one thing don't use "w" with PRIs |
| 21:20.23 | JenniferAkemi | Katty: it's common for them to have all the channels busied out |
| 21:20.33 | ManxPower | Katty: also change G1 to g1 |
| 21:20.35 | Katty | ManxPower: i'll change that real quick |
| 21:20.52 | ManxPower | if you don't have all 23 channels enabled on the PRI using G1 would screw things up if your config is wrong |
| 21:20.53 | waKKu | [TK]D-Fender ok.. ty :) |
| 21:21.13 | neoalex | not my choice guys |
| 21:21.26 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:22.30 | Katty | ManxPower: argh, same thing. |
| 21:22.49 | Katty | ManxPower: the equipment the phone company is using is a box with what appears to be an old centronix connector on it |
| 21:22.54 | Katty | ManxPower: it's labeled as 'voice' |
| 21:23.27 | Katty | ManxPower: centronic connector... this being my first PRI..... |
| 21:23.58 | Katty | ManxPower: do you think mayhaps the telco has equipment in there i can't connect to? |
| 21:24.03 | stansmith | at least 6 inches above project fences |
| 21:25.30 | Katty | ManxPower: here's the back of what the telco gave us http://blog.tmcnet.com/blog/tom-keating/images/adtran-total-access-908-bk.jpg |
| 21:26.55 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:29.16 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 21:29.16 | stansmith | what kind of logging do you guys use? |
| 21:29.23 | stansmith | (i.e - full, debug, etc) |
| 21:29.30 | stansmith | for production * |
| 21:31.02 | *** join/#asterisk Ramblurr (i=rmblr@amarok/developer/Ramblurr) |
| 21:31.29 | Ramblurr | Is it possible to bind a key to change the hold song to the next one in the queue? |
| 21:32.59 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 21:33.52 | jm|laptop | I have 3 - 5s latency on my chan_mobile; using Motorola V3i |
| 21:34.01 | ManxPower | Katty: Centronics (old printer style) is ANALOG |
| 21:34.01 | jm|laptop | it all "works"; just with jitter and latency |
| 21:34.13 | ManxPower | unless it's going into a punchdown block. |
| 21:34.22 | ManxPower | and you would have a red alarm if that was the case. |
| 21:34.38 | Supaplex | them old school things :P |
| 21:34.47 | ManxPower | Katty: plug into the DSX-1 port |
| 21:34.54 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:34.55 | dbtid | if anyone is interested, i have asterisk -18 and the current zaptel driver compiled and (at least asterisk) running on an MPC5200B |
| 21:35.06 | robeph | ManxPower: ok got it all sorted, any idea what would cause this error? Feb 25 15:33:55 WARNING[6545]: chan_iax2.c:7366 socket_read: I don't know how to authenticate phone2 to 192.168.2.141 |
| 21:35.18 | ManxPower | robeph: no. |
| 21:35.25 | robeph | I've looked and looked but don't know why it'd need to auth with the foreign pbx |
| 21:35.32 | robeph | since it's going out a trunk |
| 21:35.37 | Katty | ManxPower: that's the one it's in. |
| 21:36.03 | ManxPower | robeph: trunk just puts more than one call per packet. It does nothing else. |
| 21:36.03 | Katty | ManxPower: i'm going to take my concerns to telco and ask them why our numbers are Temporarily Unavailable. |
| 21:36.18 | ManxPower | Katty: in the asterisk console do a "pri debug span 1" |
| 21:36.22 | ManxPower | do you get lots of output? |
| 21:36.54 | Katty | lots of Sending Set Asynchronous Balanced Mode Extended |
| 21:37.07 | ManxPower | but no receiving of stuff? |
| 21:37.11 | Katty | nada |
| 21:37.23 | Katty | same line, over and over and over |
| 21:37.26 | Katty | for a bazillion lines |
| 21:37.45 | ManxPower | Call the telco and say "we have no traffic/data on the D-Channel". Telco will either say "oops!" and fix it or "everything seems fine on our end" and it will magically start to work. |
| 21:38.02 | Katty | mhmm |
| 21:38.05 | Katty | sounds like our telco allright |
| 21:38.11 | ManxPower | Katty: on the actual card, is there a red light or a green light on the port the line is plugged into? |
| 21:38.25 | Katty | lemme go look |
| 21:38.44 | Katty | red |
| 21:38.58 | ManxPower | then you have a PHYSICAL level issue. |
| 21:39.19 | ManxPower | Katty: You usually need a CROSSOVER T-1 cable for Asterisk<->Channel Bank. |
| 21:39.28 | ManxPower | that is NOT the same as an Ethernet Crossover |
| 21:39.58 | Katty | oh, i see. |
| 21:40.02 | Katty | hrmm. |
| 21:40.10 | Katty | where would you suggest getting one of these? |
| 21:40.16 | ManxPower | make one 8-) |
| 21:40.28 | ManxPower | Katty: is there anyone on site that can make ethernet cables? |
| 21:40.28 | mercutioviz | Katty: do you have a crimper and some RJ45 plugs? |
| 21:40.40 | Katty | not that i trust. |
| 21:40.46 | Katty | and i don't have the grip to crimp. |
| 21:41.03 | Katty | the only other guy here that does it has to make 3 to get 1 good one :/ |
| 21:41.07 | Katty | maybe newegg? |
| 21:41.12 | Qwell | likely |
| 21:41.21 | mercutioviz | Katty: do U need one right now, or can you order and wait? |
| 21:41.47 | ManxPower | Katty: expect them to send the wrong cable if you buy one. |
| 21:42.04 | Katty | well..hrmm |
| 21:42.06 | Katty | have a digram? |
| 21:42.11 | ManxPower | Katty: how long of a cable do you need? |
| 21:42.12 | *** part/#asterisk Ramblurr (i=rmblr@amarok/developer/Ramblurr) |
| 21:42.30 | Katty | not too long, 10ft |
| 21:42.54 | Katty | can't find one on newegg tho |
| 21:43.01 | mercutioviz | http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 21:43.14 | Katty | yeah i'm on that page now |
| 21:43.15 | mercutioviz | Katty: are you in a big city? |
| 21:43.23 | Katty | mercutioviz: hahahaha, no |
| 21:43.29 | Katty | mercutioviz: southeast missouri! ;) |
| 21:43.33 | mercutioviz | Do you have a home depot or lowe's or something? |
| 21:43.45 | mercutioviz | Oh no, misery! |
| 21:43.51 | denon | Katty: I'm sure I or someone else here could make and mail you one if you can't find one anywhere else |
| 21:44.14 | Katty | denon: that won't be required. our company is quite happy to pay for the things we need :P |
| 21:44.21 | Katty | just not my edjimication in voip stuff, apparently. hehe |
| 21:44.24 | denon | I didn't say for free :) |
| 21:44.27 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 21:44.33 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:44.35 | ManxPower | Katty: do you have a fedex or ups account number to bill to. I'll make one up for you for free if you pay shipping |
| 21:44.40 | Katty | i'm going to see if one of our copier techs can make us one |
| 21:44.45 | robeph | ok here's the issue, http://rafb.net/p/dkkazF74.html , when I call from 141 ext 2501, it'll ring the iax phone connected to 142, however, calling 2500 on 142 will not ring the phone, even though it's but one step, I just really want to understand what is going on here. |
| 21:45.01 | robeph | and please avoid flaming me for the obviously silly thing I'm doing here =) |
| 21:45.16 | ManxPower | Katty: I'll look for a better diagram than the one on voip-info |
| 21:45.22 | robeph | I'm just trying to understand what is occuring here. cos I really can't figure out why the call fails |
| 21:45.38 | JenniferAkemi | is realtime included in the current release of asterisk? |
| 21:45.46 | mercutioviz | Katty: how about this: http://www.connectworld.net/iss/t1-cables.html |
| 21:46.31 | ManxPower | Katty: I wire them up like this: http://www.voip-info.org/wiki/view/crossover+T1+cable but put the unused pairs in the same place they are on ethernet cables, they are not connected internally so it's not a problem |
| 21:46.48 | mercutioviz | Ditto |
| 21:48.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:50.50 | JenniferAkemi | grrr i don't know what is wrong with my database setup that makes my sip users not work. I can do the static database stuff fine, but for some reason not the sip susers part :| |
| 21:51.59 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 21:52.07 | *** join/#asterisk mvsn (n=four80-6@toroon63-1177846144.sdsl.bell.ca) |
| 21:52.15 | JenniferAkemi | "*CLI>realtime load sipusers name 1304" works fine, but all my users aren't loaded |
| 21:52.26 | mvsn | wondering if someone could help out... for some reason whether I call out, or have a call come in, I can talk for about 4 minutes and 30 seconds, then the phone just disconnects... (its usually between 4mins 20secs - 50secs).. any ideas? (trixbox ver) |
| 21:52.30 | JenniferAkemi | does anyone have any ideas? |
| 21:53.09 | robeph | mvsn: you watched the console to see why you get hungup? |
| 21:53.27 | *** part/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:53.42 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:54.21 | grandpapadot | What sucks is when your ITSP runs out of g729 licenses in the middle of the day when you're sending about 1000 calls an hour to them. |
| 21:54.23 | mvsn | robeph: unfortunately not, the machine doesnt have a screen attached |
| 21:54.51 | robeph | hrm |
| 21:55.02 | JenniferAkemi | mvsn: ssh to it and run asterisk -r |
| 21:55.21 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:55.27 | robeph | well ya could do that, but I'd have assumed you knew that also... |
| 21:55.28 | mvsn | ok sec, its been a while so have to remember pass :P |
| 21:55.32 | robeph | lol |
| 21:56.28 | mvsn | im in :P |
| 21:56.50 | mvsn | although asterisk -r switched into asterisk1*CLI> prompt |
| 21:57.00 | robeph | which it should |
| 21:57.03 | JenniferAkemi | yeah that's where you watch from :P |
| 21:57.08 | robeph | when ya make a call you'll get all kindsa pretty things. |
| 21:57.10 | JenniferAkemi | now make a 5 minute call |
| 21:57.14 | robeph | heh |
| 21:57.17 | ManxPower | mvsn: We can't help with Trixbox. What you are experiencing is a classic symptom of busydetect=yes or callprogress= |
| 21:58.05 | JenniferAkemi | ok my thing works now that i made the phones all be friends instead of users. what's up with that. |
| 21:58.06 | robeph | I love how he prepends everything with how no one can help, then provides an answer |
| 21:58.44 | clyrrad | Did anyone else have problems compliling asterisk-addons-1.4.6?? I am getting this error...... chan_ooh323.c:3343: error: expected ‘)’ before string constant |
| 21:59.14 | clyrrad | when I run 'make' |
| 21:59.33 | robeph | heh, someone leave out a closure? |
| 21:59.40 | clyrrad | looks like it |
| 21:59.48 | clyrrad | not sure where to look / how to fix this one |
| 21:59.50 | robeph | vi chan_ooh323.c, :3343 add the ) ;) |
| 22:00.20 | clyrrad | ok let me take a peak in there - I have never played with the asterisk code before :D |
| 22:00.39 | robeph | well I assume it's trivial in this case since it appears to be code independant and just a typographical error |
| 22:00.57 | clyrrad | line 3343 is actually the very last line... |
| 22:01.02 | clyrrad | the error is somewhere higher in the file |
| 22:01.15 | tzanger | there was mention of a utility that I could feed a wav file to and it would decode the dtmf... perhaps in spandsp? Does anyone recall anything more specific about what I am talking about |
| 22:02.07 | clyrrad | robeph: have you bulit this yet? |
| 22:03.01 | robeph | no |
| 22:03.06 | robeph | lemme grab it I'll build it |
| 22:03.26 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:03.52 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:03.53 | robeph | I had a script I wrote a long time ago somewhere lol, that would search for likely unclosed parans and bracks |
| 22:03.56 | grandpapadot | The ITU needs to let loose on g729 |
| 22:03.56 | clyrrad | robeph: thanks, im looking in the c file right now for the missing bracket as well |
| 22:04.14 | clyrrad | robeph: yea, this file is huge, that script would be quite handy right now :D |
| 22:04.37 | clyrrad | since it errors out on the last line its got to be a missing bracket from somewhere higher in the code |
| 22:04.43 | mvsn | ManxPower: curious, if its busydetect, or callprogress=, how does something like this happen? Been using this exact setup for over 6 months, it just started doing it.. nothing was ever changed. |
| 22:04.45 | ManxPower | robeph: That is not an answer. Trixbox does so many horrible things with the config files we have NO idea how/where to set the option. |
| 22:04.55 | ManxPower | I provided the answer for people running stock Asterisk |
| 22:05.07 | ManxPower | mvsn: I don't know. |
| 22:05.07 | robeph | it's just a perl script that would spit out the ()s that exceeded newlines / lengths , based on Cs functions that they would likely be part of heh |
| 22:05.23 | robeph | ManxPower: ah |
| 22:05.32 | ManxPower | ~trixbox |
| 22:05.32 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:05.43 | clyrrad | ManxPower: you refering to me? |
| 22:05.46 | robeph | no |
| 22:05.52 | robeph | talking bout that other cat. |
| 22:05.57 | clyrrad | ah |
| 22:06.09 | robeph | now which version was this? |
| 22:06.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:06.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:07.08 | grandpapadot | Do many of you use GSM for Internet based telephony? |
| 22:07.43 | robeph | ouch |
| 22:07.46 | clyrrad | robeph: 1.4.6 of Asterisk Addons |
| 22:07.48 | robeph | yeah , I can't build it either |
| 22:07.57 | clyrrad | yea there is a syntax error in the file here |
| 22:08.00 | clyrrad | im trying to locate it |
| 22:08.16 | robeph | well I think its another reason for me heh |
| 22:11.53 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.44) |
| 22:12.05 | robeph | clyrrad: is the ) thing your topmost build error? |
| 22:12.37 | Daejeo | hello anyone can help>????? http://pastebin.ca/918092 |
| 22:13.05 | Daejeo | i want to change the parameter " unknown" with some name |
| 22:13.12 | clyrrad | robeph: no the top most error is it cant find Asterisk.h |
| 22:13.22 | Daejeo | what should i change in SIP.conf |
| 22:13.29 | grandpapadot | Daejeo: I think just adding a user to your sip.conf |
| 22:13.30 | ManxPower | you need to upgrade your asterisk-addons |
| 22:14.00 | Robba | Hi guys |
| 22:14.15 | grandpapadot | Yes, add a username= |
| 22:14.29 | mvanbaak | meh |
| 22:14.31 | Robba | has anyone successfully made the beep sound work when doing an attended transfer? |
| 22:14.42 | clyrrad | robeph: channels/chan_ooh323.h:19:22: error: asterisk.h: No such file or directory |
| 22:14.45 | mvanbaak | and people are complaining about memory leaks in asterisk |
| 22:14.46 | mvanbaak | <PROTECTED> |
| 22:14.51 | mvanbaak | how about that ! |
| 22:15.08 | Daejeo | grandpapadot: let me try |
| 22:15.10 | mvanbaak | I only have bugs.digium.com open |
| 22:15.11 | ManxPower | clyrrad: and the updated asterisk-addons fixes the ooh323 problem |
| 22:15.12 | mvanbaak | nothing else |
| 22:15.21 | mvanbaak | no flash plugin installed or whatever |
| 22:15.29 | robeph | yeh |
| 22:15.32 | robeph | thats the one I'm getting |
| 22:15.37 | robeph | I dont' have the ( error |
| 22:15.39 | robeph | but same on there |
| 22:15.42 | clyrrad | ManxPower: Is 1.4.6 not the updated asterisk-addons ? |
| 22:15.46 | robeph | look also for missing asterisk.h |
| 22:15.56 | robeph | for somereason thought its in my includes, its not seein it |
| 22:16.31 | robeph | yeh |
| 22:16.37 | ManxPower | clyrrad: and what version of asterisk do you have? |
| 22:16.39 | robeph | you prolly having same issue I am |
| 22:16.49 | clyrrad | ManxPower: 1.4.18 |
| 22:17.01 | clyrrad | robeph: yea looks like it |
| 22:17.38 | robeph | although, I'm nto actually trying to build it, I just have all the 1.4 includes in path ;p |
| 22:17.46 | robeph | since I use 1.2 |
| 22:17.59 | clyrrad | ManxPower: I am in the process of upgrading from 1.4.17 to 1.4.18, also asterisk 1.4.18 built with out issue, its just the addons that I cant build.... |
| 22:18.38 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 22:18.40 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 22:19.54 | clyrrad | Its acutually the chan_ooh323.h file that has the issue |
| 22:20.13 | clyrrad | its included from chan_ooh323.c |
| 22:20.13 | robeph | is it direct linking to a file for include? |
| 22:20.29 | robeph | eg not canonical.... |
| 22:20.34 | clyrrad | chan_ooh323.c has this #include "chan_ooh323.h" |
| 22:20.50 | clyrrad | and on line 19 of "chan_ooh323.h" is where the error happens |
| 22:21.07 | clyrrad | line 19 of chan_ooh323 is #include <asterisk.h> |
| 22:21.14 | Qwell | clyrrad: what error? |
| 22:21.14 | clyrrad | which it can not find for some reason |
| 22:21.28 | clyrrad | Qwell: channels/chan_ooh323.h:19:22: error: asterisk.h: No such file or directory |
| 22:21.48 | clyrrad | Qwell: thats the top most error message on 'make' of the asterisk addons for 1.4.6 |
| 22:21.49 | Qwell | What OS? |
| 22:21.54 | clyrrad | CentOS 4.6 |
| 22:22.04 | Qwell | did you install asterisk? |
| 22:22.05 | clyrrad | Asterisk 1.4.18 |
| 22:22.14 | clyrrad | is installed and compiled with out issue |
| 22:22.24 | clyrrad | its just the adddons that wont build |
| 22:22.38 | Qwell | and did you run configure after you installed asterisk? |
| 22:22.48 | robeph | i did, same result |
| 22:22.53 | clyrrad | Qwell: yes |
| 22:23.02 | robeph | I just downloaded the src for addons, and I get same errors |
| 22:23.07 | clyrrad | Qwell: also 'make' dies with an error of channels/chan_ooh323.c:3343: error: expected ‘)’ before string constant |
| 22:23.12 | Daejeo | grandpapadot: |
| 22:23.20 | robeph | clyrrad: mine doesn't do that though |
| 22:23.22 | Daejeo | grandpapadot: |
| 22:23.26 | ManxPower | robeph: expect to get that error with 1.4.x addons compiling against 1.2.x Asterisk |
| 22:23.33 | grandpapadot | daejeo : grandpapadot {} |
| 22:23.35 | Daejeo | grandpapadot: still unknown |
| 22:23.44 | grandpapadot | no clue then, sorry |
| 22:23.50 | robeph | ManxPower: I'm not |
| 22:23.57 | clyrrad | But since I am running 1.4.18 these addons should work |
| 22:23.59 | robeph | ManxPower: it's compiling against 1.4 libs |
| 22:24.14 | robeph | ManxPower: and he has 1.4 asterisk, and has same errors I do |
| 22:24.38 | ManxPower | I guess it's time to report a bug |
| 22:24.38 | clyrrad | One of my associates has also confirmed they are getting this same error when they build the addons for 1.4.6 |
| 22:25.06 | Katty | ManxPower: heh, the telco didn't have the lines 'active' |
| 22:25.06 | clyrrad | ManxPower: Ok i'll do that now |
| 22:25.10 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 22:25.19 | Qwell | okay, it's just #include <asterisk.h> |
| 22:25.25 | dmz | yeah found my queue problem |
| 22:25.27 | Qwell | if it can't find that - something is broken on your system |
| 22:25.29 | DrkShadow | Got SIP response 415 "No Compatible Routes Found" back from <provider> <-- anyone have any info on that? google doesn't. |
| 22:25.34 | dmz | apparently Queue(queue,Tt) doesn't work anymore |
| 22:25.40 | dmz | got rid of Tt and it is happy again |
| 22:25.52 | Qwell | Did you do something silly when installing asterisk, like use --prefix or DESTDIR? |
| 22:26.01 | x86 | how do i do a NoOp from an AGI? |
| 22:26.09 | x86 | or would something like Verbose be better? |
| 22:26.10 | Qwell | x86: NoOp does nothing. |
| 22:26.13 | Qwell | Just put in a newline |
| 22:26.26 | clyrrad | Qwell: I did configure like this ./configure --prefix=$HOME/asterisk-bin --sysconfdir=$HOME/asterisk-bin --localstatedir=$HOME/asterisk-bin |
| 22:26.28 | x86 | Qwell: I always NoOp from my dialplan to give some info about the call in CLI |
| 22:26.34 | Qwell | of course you did |
| 22:26.35 | grandpapadot | x86: .. but will parse out your variables before it does nothing so you can see them in the cli |
| 22:26.47 | Qwell | so how, exactly, is addons going to find asterisk, if you don't tell it where it is? |
| 22:26.59 | clyrrad | Qwell: Asterisk addons 1.4.5 builds no problem |
| 22:27.06 | x86 | Qwell: i want to NoOp from an AGI so I can spew info to the CLI, that's all my AGI does |
| 22:27.13 | Qwell | x86: verbose |
| 22:27.36 | mvanbaak | Qwell: you abandoned chan_skinny.c ? |
| 22:27.42 | clyrrad | Qwell: I tried to pass the same configure options to the configure of the addons, perhaps I am doing something wrong? But I am doing the same thing I did to build 1.4.5 addons.... |
| 22:27.42 | russellb | for exec noop my info here |
| 22:27.45 | Qwell | mvanbaak: no, just low priority... |
| 22:27.46 | russellb | or whatever it is |
| 22:27.50 | russellb | s/for/or/ |
| 22:27.51 | Qwell | clyrrad: --with-asterisk |
| 22:28.01 | mvanbaak | Qwell :( |
| 22:28.01 | ManxPower | DrkShadow: that means the codecs you want and the codec they want have nothing in common |
| 22:28.12 | DrkShadow | manx: I suspected as much.. |
| 22:28.27 | mvanbaak | it has some serious issues |
| 22:28.29 | DrkShadow | why the hell doesn't asterisk give meaningful, informative errors? |
| 22:28.52 | grandpapadot | DrkShadow: ... as in beer |
| 22:29.02 | *** join/#asterisk miltux (n=miltux@62.1.19.195) |
| 22:29.07 | ManxPower | x86: noop is just an app, run it in the agi just like any other agi->exec'd app |
| 22:29.14 | clyrrad | Qwell: do I understand you correctly? I should configure the addons as ./configure --with-asterisk? |
| 22:29.19 | Qwell | yes |
| 22:29.24 | outtolunc | be thankful they didn't follow m$ and give you all that helpful info a BSOD gives you <G> |
| 22:29.32 | Qwell | and of course --prefix, et al |
| 22:29.46 | grandpapadot | It's open source, you can add whatever informative error messages you like ... |
| 22:29.48 | DrkShadow | BSOD is actually quite useful. |
| 22:29.57 | clyrrad | Qwell: ok trying that |
| 22:30.05 | x86 | ManxPower: yeah that's what I was doing -- not working correctly |
| 22:30.06 | DrkShadow | grand: and all it takes is a great amount of time that I don't have! |
| 22:30.20 | grandpapadot | DrkShadow: You can always pay someone to do it for you ... |
| 22:30.27 | DrkShadow | The typical argument.. "It's open source; do it yourself." |
| 22:30.33 | ManxPower | x86: Don't expect it to work as you expect, but it should work. |
| 22:30.46 | Qwell | DrkShadow: the other option is to hire a consultant |
| 22:30.54 | *** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 22:31.16 | x86 | ManxPower: tried Verbose as well as NoOp, niether seem to be showing me anything in CLI |
| 22:31.26 | grandpapadot | DrkShadow: Clearly. It's insulting to bitch about something that's for all intents and purposes free of charge, especially in a channel where the people that are doing the actual work hang out. |
| 22:31.31 | ManxPower | i.e. Since the app isn't executing in the dialplan don't expect all the dialplan variables to be availabe |
| 22:31.37 | x86 | ManxPower: I see where it execs the AGI, then seems to immediately return 0 |
| 22:32.06 | ManxPower | x86: looks like you have a problem with your AGI, not with Noop or Verbose. What version of Asterisk? |
| 22:32.10 | x86 | ManxPower: oh no I'm just using my AGI's variables, not trying to make it spit out channel variables |
| 22:32.19 | x86 | ManxPower: 1.4.12.1 |
| 22:32.32 | ManxPower | x86: "core agi debug" or something like that |
| 22:32.34 | clyrrad | Qwell: interesting, it sais "The Asterisk installation on this system appears to be broken", yet Asterisk 1.4.18 is up and running. Any ideas? |
| 22:32.38 | x86 | ManxPower: I'm certain it's a problem with my AGI... that's what I'm seeking help with ;) |
| 22:32.56 | Daejeo | grandpapadot: |
| 22:32.58 | x86 | ManxPower: agi debug, you mean :) |
| 22:32.58 | Daejeo | grandpapadot: |
| 22:33.03 | clyrrad | Qwell: thats what ./configure reports with the new flag added to it that you mentioned |
| 22:33.09 | Daejeo | grandpapadot: fromuser |
| 22:33.16 | grandpapadot | Cool. |
| 22:33.18 | ManxPower | x86: in 1.2 you had to run Asterisk in console mode to see the AGI error messages, that was supposed to be fixed in 1.4 |
| 22:33.30 | Daejeo | working now |
| 22:33.33 | Daejeo | :) |
| 22:33.40 | grandpapadot | I knew it was something like that, lol |
| 22:33.48 | Daejeo | right |
| 22:34.07 | Daejeo | wireshark helped me |
| 22:34.19 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:34.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:34.34 | robeph | <PROTECTED> |
| 22:35.06 | *** join/#asterisk infinity3 (i=brendon@saleen.netcal.com) |
| 22:35.27 | clyrrad | hrm |
| 22:35.56 | ManxPower | robeph: perhaps a pastebin'd iax.conf would be helpful |
| 22:36.19 | x86 | ManxPower: see a bunch of Tx's, but no Rx |
| 22:36.49 | robeph | http://rafb.net/p/hb4MG330.html |
| 22:36.50 | ManxPower | x86: is this a production serveR? |
| 22:37.11 | x86 | ManxPower: yep |
| 22:37.18 | infinity3 | <PROTECTED> |
| 22:37.21 | ManxPower | robeph: maybe if you removed all the auth= crap |
| 22:37.23 | x86 | want me to pastebin it? |
| 22:37.47 | ManxPower | x86: no. Can't help you further unless you can stop asterisk and start it as "asterisk -cvvv". |
| 22:38.09 | clyrrad | Qwell: you still here? |
| 22:38.10 | Qwell | clyrrad: *shrug* works on my box |
| 22:38.12 | ManxPower | There ARE other ways to find the info we need in 1.4, but I only know of the pre-1.4 way of running asterisk in console mode. |
| 22:38.17 | robeph | ManxPower: I added that just to see if it'd help actually, |
| 22:38.32 | clyrrad | Qwell: why would it say my asterisk is broken? How can I correct this? |
| 22:38.36 | ManxPower | robeph: there is 5 mins of my life I will never get back. |
| 22:38.39 | x86 | ManxPower: why would that help? |
| 22:38.45 | Qwell | rm -r /usr/include/asterisk* |
| 22:38.47 | robeph | ManxPower: ? |
| 22:38.48 | x86 | ManxPower: set verbose 9999999999 ftw ;) |
| 22:38.50 | ManxPower | clyrrad: I suspect Asterisk-Addons do not honour --prefix |
| 22:38.53 | Qwell | clyrrad: then make dist-clean in -addons |
| 22:39.01 | robeph | ManxPower: heh I removed it, still, same exact error, reloaded iax2 etc. |
| 22:39.06 | x86 | Qwell: you deal with AGI much? |
| 22:39.08 | ManxPower | x86: it will give you the stderr, which is where any language error messages would be. |
| 22:39.12 | clyrrad | Qwell: ok i'll try that |
| 22:39.23 | x86 | ManxPower: I ran it in a console just fine |
| 22:39.33 | Qwell | x86: no, never |
| 22:39.34 | x86 | ManxPower: the problem is how i'm trying to pass back the stuff |
| 22:39.41 | ManxPower | x86: as "asterisk -cvvv"? |
| 22:39.55 | x86 | ManxPower: nope, but I know it's not a language error |
| 22:40.07 | ManxPower | x86: I cannot help you further |
| 22:40.17 | x86 | http://pastebin.ca/918123 |
| 22:40.28 | clyrrad | ManxPower: yea, its interesting why 1.4.5 builds but 1.4.6 does not |
| 22:40.36 | infinity3 | ManxPower: can you give me a few tips with iax2 when using voipjet and ast 1.4.11 ? |
| 22:40.40 | x86 | can someone help me debug this AGI, about lines 43 and 44? |
| 22:40.59 | ManxPower | infinity3: no. |
| 22:41.05 | x86 | i'm printing it out twice, as I was trying one way on the first line (43) and another way on the second line (44) |
| 22:41.12 | x86 | both are fail, it seems |
| 22:41.13 | infinity3 | ManxPower: well, thanks for the response ... |
| 22:41.24 | ManxPower | infinity3: We use 1.2.x |
| 22:41.28 | clyrrad | Qwell: hrm, back at square one it still reports that asterisk is broken. Is it looking somehere for an old file left by the previous version? Perhaps I need to remove something? |
| 22:41.32 | ManxPower | we don't use voip service providers |
| 22:41.45 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 22:41.48 | infinity3 | ManxPower: you use a lot of polycom , right? |
| 22:41.59 | robeph | ok now thats odd.. Feb 25 16:41:39 WARNING[6545]: chan_iax2.c:7366 socket_read: I don't know how to authenticate iaxfwd to 192.168.2.141 getting this now |
| 22:42.07 | Qwell | clyrrad: did you do both things I said to do? |
| 22:42.50 | clyrrad | Qwell: yes I did the make dist-clean and also added the --with-asterisk and prefix jazz that you said |
| 22:43.03 | Qwell | did you remove /usr/include/asterisk* ? |
| 22:43.23 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 22:43.44 | clyrrad | Qwell: ah sorry I missed that, you didnt prefix my nick on that message, let me try that now |
| 22:44.04 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 22:44.15 | grandpapadot | Does the jitter buffer in 1.4 help with Internet latency using g729 at all? |
| 22:45.16 | *** join/#asterisk ludan (n=daniele@56-123-dyn.ipact.nl) |
| 22:45.47 | clyrrad | Qwell: ok thats done, do I need to rebuild asterisk now? Or just try the addons again? |
| 22:45.49 | ludan | hello everybody |
| 22:45.56 | Qwell | redo addons entirely |
| 22:46.04 | clyrrad | ok doing that now |
| 22:46.14 | ludan | i've a problem, difficult to bound |
| 22:46.32 | ludan | i had an asterisk configuration that was working pretty good |
| 22:46.36 | Daejeo | grandpapadot: did u ever try vtwhite.com? |
| 22:47.02 | ludan | then the pc crashed and my sysadmin managed to recover some configuration files for the PBX |
| 22:47.34 | clyrrad | Qwell: hrm no luck it still seems to think Asterisk is broken |
| 22:47.39 | ludan | now he gave me the root password and as far as I can see this machine is a debian with asterisk 1.2.13 |
| 22:47.46 | ludan | but nothing is working |
| 22:48.02 | ludan | and if I try to start with /etc/init.d/asterisk debug |
| 22:48.23 | ludan | it gives me some warnings but then with /etc/init.d/asterisk status => stopper :( |
| 22:48.31 | ludan | i checked the logs |
| 22:48.35 | ludan | nothing |
| 22:48.43 | ludan | what can I do? |
| 22:49.08 | Qwell | clyrrad: pastebin the entire output of make distclean, ./configure --whateverstuffhere, and then config.log |
| 22:49.40 | clyrrad | Qwell: ok no prob |
| 22:50.38 | ludan | Qwell: any clue? |
| 22:53.53 | clyrrad | Qwell: here it is: http://rafb.net/p/SDj1LT66.txt |
| 22:54.31 | *** part/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 22:54.53 | Qwell | clyrrad: you need to tell it where asterisk is |
| 22:55.02 | Qwell | --with-asterisk=/path/to/asterisk/ |
| 22:55.45 | clyrrad | Qwell: gotcha |
| 22:56.42 | clyrrad | Qwell: that should be --with-asterisk=$HOME/asterisk-bin correct? |
| 22:56.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:57.02 | clyrrad | or is it needing the path to the source files? |
| 22:57.45 | [TK]D-Fender | Someone got a minute in PM to help me with a (hopefully) minor Linux problem? |
| 23:00.49 | *** join/#asterisk psiforce (n=psiforce@ppp191-34.static.internode.on.net) |
| 23:01.09 | psiforce | hi all |
| 23:02.00 | *** part/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 23:10.21 | infinity3 | do people use the term "re" in chat anymore? |
| 23:10.32 | Supaplex | [TK]D-Fender: ##linux |
| 23:11.37 | sweeper | someone needs to make a dirt cheap cellphone that just does calls + really basic mp3 playing, with a microsd slot and a standard headphone jack |
| 23:13.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583194.dsl.bell.ca) |
| 23:14.23 | sweeper | oh, and make it charge from mini-usb |
| 23:14.45 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 23:17.48 | *** join/#asterisk andresmujica (n=andresmu@190.25.106.104) |
| 23:18.03 | andresmujica | hi! |
| 23:18.11 | andresmujica | anyone has ever used a linksys spa8000 ? |
| 23:18.21 | andresmujica | i've got some doubts about the electrical connection for the device... |
| 23:19.38 | jm|laptop | does anyone have ideas why I have this latency with chan_mobile ? |
| 23:20.06 | sweeper | andresmujica: what kind of doubts? |
| 23:20.24 | clyrrad | Qwell: you still with me? |
| 23:20.49 | Qwell | did it work? |
| 23:20.54 | clyrrad | nope :( |
| 23:20.54 | *** join/#asterisk ghenry (n=ghenry@80.187.215.183) |
| 23:20.57 | Daejeo | is there anyway to change useragent in pap2-na? |
| 23:21.06 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 23:21.06 | clyrrad | Qwell: it still sais Asterisk is broken, I cant figure out why |
| 23:21.11 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 23:21.20 | clyrrad | Qwell: Asterisk is running fine 1.4.18 |
| 23:21.28 | clyrrad | I just cant build the addons |
| 23:22.25 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:22.25 | clyrrad | Qwell: I did this ./configure --prefix=$HOME/asterisk-bin --sysconfdir=$HOME/asterisk-bin --localstatedir=$HOME/asterisk-bin --with-asterisk=$HOME/asterisk-bin |
| 23:22.56 | Qwell | pastebin config.log |
| 23:23.01 | clyrrad | Qwell: ok |
| 23:23.04 | andresmujica | sweeper: AS this unit provides ring tone to 8 phones, i wonder if it could be connected to an UPS, i mean the peak current with all the phones ringing could thake the UPS down, am i right? ... |
| 23:24.10 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 23:24.17 | husimon | um does anyone use 7935 or 7936 with asterisK? |
| 23:24.37 | husimon | cisco.. |
| 23:24.41 | clyrrad | Qwell: http://rafb.net/p/1tF9DZ52.txt |
| 23:24.47 | trnzmeta | hahaha husimon |
| 23:24.50 | [TK]D-Fender | andresmujica, its a glorified ATA, don't think too much on it |
| 23:24.56 | trnzmeta | I'm trying to convert them myself |
| 23:25.02 | trnzmeta | sccp to sip, pain in the ass |
| 23:25.17 | husimon | trnzmeta, i don't think they can run sip |
| 23:25.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:25.27 | husimon | do you have them successfully working on sccp? |
| 23:25.54 | andresmujica | hmm, what do you mean with glorified? |
| 23:25.54 | Qwell | clyrrad: ls -lR /home/asterisk/asterisk-bin/include/ |
| 23:26.03 | Qwell | pastebin the output |
| 23:26.04 | *** join/#asterisk whitehat (n=whitehat@unaffiliated/whitehat) |
| 23:26.15 | trnzmeta | sccp yes, convert to sip... no |
| 23:27.14 | ManxPower | I shall check to see if those phones support SIP |
| 23:27.19 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 23:27.22 | husimon | ManxPower, they don't |
| 23:27.28 | husimon | trnzmeta, can you show me your skinny.conf entry and how did you have to setup your tftp and files? |
| 23:27.33 | clyrrad | Qwell: http://rafb.net/p/Fe4bGT90.txt |
| 23:27.39 | husimon | trnzmeta, also what ver of asterisk are you running? |
| 23:27.51 | Qwell | so then your asterisk install is very clearly broken |
| 23:28.17 | clyrrad | Qwell: we deleted everything from there before.... |
| 23:28.23 | Qwell | ... |
| 23:28.35 | clyrrad | shall I recompile Asterisk? |
| 23:28.42 | Qwell | obviously |
| 23:28.43 | [TK]D-Fender | andresmujica, Go look up its power requirements, it isn't much |
| 23:28.57 | clyrrad | Qwell: ok i'll try it again..... |
| 23:29.19 | ManxPower | It does not appear to support SIP |
| 23:29.28 | ManxPower | husimon: you have a CCO account? |
| 23:29.32 | husimon | yes |
| 23:30.20 | ManxPower | Qwell: I suspect asterisk-addons is not honouring the --prefix options |
| 23:30.28 | Qwell | ManxPower: it works fine. see above |
| 23:31.38 | husimon | trnzmeta, any chance you can help me with the tftp part? |
| 23:31.44 | husimon | and skinny.conf |
| 23:32.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:32.11 | clyrrad | Qwell: its rebuilding now, i'll keep you posted |
| 23:32.29 | whitehat | please excuse my ignorance.will asterisk work without any extra hardware.can someone point me to docs on this issue. thank you. |
| 23:32.35 | Qwell | ~book |
| 23:32.36 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:32.40 | Qwell | whitehat: ^^^ |
| 23:32.56 | husimon | whitehat, it will work without extra hardware if you use a VOIP provider |
| 23:33.17 | husimon | if you are using a telephone line then you need a hardware card |
| 23:33.28 | ManxPower | whitehat: It all depends on what you want to do. Congratulations! You have won the "Most Vague Question" award for this hour. |
| 23:33.55 | whitehat | ManxPower: thank you. I've never won anything in my life! I feel good today! |
| 23:34.00 | husimon | trnzmeta, what sccp software do you have loaded? |
| 23:34.03 | husimon | the version |
| 23:34.17 | trnzmeta | v7 |
| 23:34.26 | trnzmeta | it has the image/app loader |
| 23:34.27 | husimon | weird, i don't see that on the cisco site |
| 23:34.32 | whitehat | ManxPower: I'm just testing this stuff on a very very very limited budget(i.e. $0) |
| 23:34.33 | husimon | i must be on the wrong page |
| 23:34.49 | Qwell | I need to get my hands on a 7936, so I can fix chan_skinny |
| 23:34.51 | trnzmeta | apparently I have to go sccp -> sip 3 --> sip 4 --> sip5 ... |
| 23:34.52 | friedrich| | whitehat: yes, use it together with sip and stuff |
| 23:35.03 | Qwell | trnzmeta: 793x does sip now? |
| 23:35.11 | husimon | trnzmeta, i've been successful at skipping all those steps with my 7940s |
| 23:35.20 | husimon | Qwell, apparently it does but i can'tfind the the software for it |
| 23:35.22 | ManxPower | trnzmeta: As the 793x does not support SIP, expect to FAIL. |
| 23:35.27 | Qwell | husimon: I doubt it does |
| 23:35.43 | Qwell | though, I've been tempted to try flashing the polycom firmware on it |
| 23:35.47 | trnzmeta | oh I have 7960 and 7940 :) |
| 23:35.50 | Qwell | probably brick it, heh |
| 23:35.56 | whitehat | friedrich|: as you may have guessed, my shoes and socks are still on and thus my feet are not wet yet. |
| 23:35.57 | husimon | Qwell, let me see where I found that information |
| 23:36.01 | ManxPower | trnzmeta: If you are lucky, the phone will just reject the firmware -- if you are unlucky it will just brick your phone. |
| 23:36.41 | whitehat | I'm still wading through references I was given |
| 23:36.41 | siya | aaargh... |
| 23:36.58 | siya | my * server keeps bouncing my sip peers |
| 23:36.58 | jm|laptop | ~seen proteus74 |
| 23:37.01 | jbot | i haven't seen 'proteus74', jm|laptop |
| 23:37.07 | siya | no idea what's going on |
| 23:37.19 | ManxPower | siya: Define "bounce" |
| 23:37.20 | husimon | Qwell, k nm i was thinking about a different phone |
| 23:37.35 | [TK]D-Fender | whitehat, All you need is a Linux server to install * on and you're ready to get learning. |
| 23:37.46 | siya | Anyone here who can tell me why if I lose all my 6 ITSP lines my extensions will suddenly get marked unreachable too? |
| 23:38.02 | dmz | trnzmeta ever play with a 7940? i've got my 7960 working great but the 7940 just couldn't upgrde firmware :( |
| 23:38.04 | jm|laptop | http://bugs.digium.com/view.php?id=11627 I have this bug which has been "closed" - but I have tried their solution without success :( |
| 23:38.05 | ManxPower | siya: remove qualify= option from the problem entries in sip.conf |
| 23:38.24 | siya | ManxPower: that would be all of them then |
| 23:38.33 | whitehat | [TK]D-Fender: k. have a few of those. F8 has the asterisk rpms. |
| 23:38.40 | husimon | dmz i have my 7940s working with the latest sip |
| 23:38.56 | ManxPower | siya: qualify= in SIP in Asterisk will frequently incorrectly tag peers as "offline" |
| 23:38.59 | [TK]D-Fender | whitehat, it is advised to compile from source tarball as downloaded from asterisk.org VS using packages |
| 23:39.20 | husimon | trnzmeta, where did you find v7 of the sccp software for 7935? |
| 23:39.21 | whitehat | [TK]D-Fender: any particular reason |
| 23:39.27 | husimon | the cisco website only lists 3.2(18) |
| 23:39.28 | dmz | husimon really? hmm you gonna be on for a bit? i gotta get my kids to bed in an hour and would like to pick your brain :) |
| 23:39.38 | husimon | dmz sure |
| 23:39.46 | clyrrad | Qwell: ok asterisk is rebuilt, going to try the addons again |
| 23:39.52 | [TK]D-Fender | whitehat, ensures that its build for your kernel (zzaptel reqs), arch, etc. |
| 23:40.04 | [TK]D-Fender | whitehat, just trust us and do it from scratch. |
| 23:40.19 | [TK]D-Fender | whitehat, Start by reading THE BOOK (as linked) |
| 23:40.23 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 23:41.02 | whitehat | [TK]D-Fender: I'm not distrusting anyone. it's an issue of time. |
| 23:41.18 | [TK]D-Fender | whitehat, its worth the extra 5 minutes or so :p |
| 23:41.46 | ManxPower | whitehat: Expect to spend several weeks before you understand Asterisk enough to put it into production. |
| 23:41.58 | whitehat | [TK]D-Fender: I have a slow machine so doing a compile on a large package tends to affect other things I do on this box |
| 23:42.06 | ManxPower | Expect to spend several days (at least) before you know enough to make a simple phone call. |
| 23:42.34 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 23:42.37 | whitehat | any quickstart refs available. |
| 23:42.39 | ManxPower | whitehat: you should NOT be running extra stuff on an Asterisk server. |
| 23:42.40 | [TK]D-Fender | whitehat, how small? |
| 23:42.49 | ManxPower | whitehat: Telecom is complicated. Get used to it. |
| 23:43.01 | twisted | heh |
| 23:43.17 | twisted | telcomplicated |
| 23:43.20 | whitehat | ManxPower: don't have much of a choice. limited resources on this box. I'm not worried about the complicated issues its just an issue of time |
| 23:43.29 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:43.31 | ManxPower | whitehat: Then you will fail. |
| 23:43.56 | whitehat | [TK]D-Fender: this is onlyu a 1.6ghz box with 1g mem. it's the biggest server I have |
| 23:44.06 | ManxPower | If you don't have the time to devote to learning Asterisk,. you have 3 choices 1) give up 2) hire someone 3) make the time |
| 23:44.14 | whitehat | ManxPower: don't like to fail. |
| 23:44.19 | [TK]D-Fender | whitehat, no big deal. I've installed on systems HALF that size |
| 23:44.20 | husimon | whitehat, asterisk doesn't need much |
| 23:44.23 | jameswf-home | pick 2 |
| 23:44.29 | clyrrad | Qwell: WHOHA! It worked!!!!! |
| 23:44.38 | ManxPower | whitehat: then stop doing stuff that ensures failure. |
| 23:44.45 | clyrrad | Qwell: but ONLY with the --with-asterisk flag |
| 23:44.45 | husimon | whitehat, unless you are doing lots of codec conversion |
| 23:45.05 | clyrrad | Qwell: Interesting why asterisk-addons 1.4.5 did not need that, but 1.4.6 does.... |
| 23:45.39 | husimon | whitehat, for something as integral as the phone system i might consider a small investment in a separate server. |
| 23:45.39 | clyrrad | ManxPower: it seems the addons will honor --prefix :D |
| 23:45.43 | jameswf-home | [Atention] all #asterisk support questions are now being outsourced to #india If you can not find a solution on google please ask in #india |
| 23:45.49 | husimon | jameswf-home, lol |
| 23:46.06 | clyrrad | LOL |
| 23:46.07 | andresmujica | join number india |
| 23:46.29 | jameswf-home | #india is empty...:( |
| 23:46.40 | clyrrad | Qwell: thanks for your help troubleshooting that, much apprecated |
| 23:46.45 | whitehat | husimon: I have 11 boxes here, but most are old with limited resources. I'd just like to get it up and running to test it for now |
| 23:47.06 | [TK]D-Fender | whitehat, Fine, download source, compile, enjoy. |
| 23:47.09 | husimon | whitehat, k well once you are done testing dedicate a box is all i'm saying |
| 23:47.15 | Qwell | ~paypal |
| 23:47.16 | jbot | well, paypal is at http://www.paypalsucks.com and http://www.paypal.com |
| 23:47.16 | jameswf-home | jbot: tell whitehat about buybook |
| 23:47.17 | husimon | test away |
| 23:47.21 | Qwell | jbot: nice |
| 23:47.22 | jbot | extra, extra, read all about it, nice is prime example of SuperJuan, or a good term for GNOME a derogatory term meaning bland, boring, feeble, or just crap. Example: That's a nice haircut. a city in france, or a program that will run a program with a modified scheduling priority (from -20 to 19, where 19 is the lowest). |
| 23:48.40 | jameswf-home | I wisk the standard ubuntu network manager would auto fial over to kppp |
| 23:48.58 | *** join/#asterisk AndyGraybeal (n=andy@node194.39.251.72.1dial.com) |
| 23:49.13 | jameswf-home | wish even |
| 23:49.19 | Qwell | jameswf-home: people still use ppp? |
| 23:49.21 | jameswf-home | and fail |
| 23:49.28 | jameswf-home | cell card |
| 23:50.04 | whitehat | you've given me some great advice. I thank you very much |
| 23:50.11 | jameswf-home | its silly At&t requires cham with blank uid and pw |
| 23:50.20 | jameswf-home | *chap |
| 23:51.56 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 23:52.48 | jm|laptop | Qwell: do you use chan_mobile ? |
| 23:52.55 | whitehat | what is this skinny protocol |
| 23:53.06 | JT | sccp |
| 23:53.16 | *** join/#asterisk thrills3k3r (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 23:53.29 | jameswf-home | it is like tha fat protocol |
| 23:53.36 | jameswf-home | ~skinney |
| 23:53.43 | jameswf-home | ~skinny |
| 23:53.44 | jbot | it has been said that skinny is a common name for SCCP, the VoIP protocol used by many Cisco phones, or what people look like when they put computing above eating |
| 23:53.44 | jm|laptop | chan_orexic |
| 23:54.07 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 23:54.11 | *** join/#asterisk ruied_ (n=ruied@bl9-255-200.dsl.telepac.pt) |
| 23:54.13 | jameswf-home | oooh we need a chan_orexic everytime you make a call asterisk pukes |
| 23:54.17 | whitehat | ah. well thank you bot. |
| 23:54.32 | whitehat | the day is really getting bad when one thanks a bot for help |
| 23:54.33 | Qwell | jm|laptop: I have. I need to again, now that I've got a new phone. |
| 23:54.35 | clyrrad | Any of you guys seen asterisk segfault when calling Voicemail in a nested Macro? |
| 23:55.00 | jm|laptop | Qwell: did you have any latency issues; when you did (you mentioned BT sticks earlier) |
| 23:57.17 | thrills3k3r | Hi, I have a problem with a Digium TE120P. When a DTMF tone is played from a phone that is registered with the server the entire server freezes solid. Anyone have an idea of what could be the culprit? This is Asterisk 1.4.18, zaptel 1.4.9, libpri 1.4.3, all compiled from svn |
| 23:57.53 | Qwell | jm|laptop: none that I recall, besides standard cell latency |
| 23:58.01 | jm|laptop | :/ |
| 23:58.04 | Qwell | maybe half a second or whatever it is |
| 23:58.07 | ManxPower | thrills3k3r: sounds like it's time to file a bug |
| 23:58.09 | jm|laptop | this is like 2 2+ seconds |
| 23:58.11 | jm|laptop | er |
| 23:58.14 | jm|laptop | "2+ seconds" |
| 23:58.40 | clyrrad | I should prolly file a bug too, cuz Asterisk really should never seg fault |
| 23:59.30 | thrills3k3r | ManxPower: I would but there is pretty much no information to go on since the server seizes up instantly. I'm a bit hesitant to file a bug with no useful information... |
