| 00:00.03 | Dr-Linux | Qwell[]: but they have 512kbps and have only 2 systems |
| 00:00.15 | Qwell[] | Dr-Linux: yes, but packets could be getting lost |
| 00:01.00 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:01.00 | Qwell[] | run a ping between the servers. See if you notice any packet loss |
| 00:01.45 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.98) |
| 00:01.55 | Dr-Linux | hhm.. |
| 00:02.14 | Dr-Linux | Qwell[]: any other possiblity in your mind? |
| 00:03.03 | Qwell[] | Dr-Linux: nope |
| 00:03.22 | Dr-Linux | ok thanks |
| 00:03.58 | aphexddb | any ideas on this wierd dropping behavior? I can make and send receive nat'ed calls over a SIP trunk. however as soon as either party picks up the phone the connection drops. local extenstions on * work perfectly. |
| 00:04.07 | Druken | so that's what now? 3 servers Dr-Linux? |
| 00:06.10 | *** join/#asterisk eBook (n=none@213.13.226.224) |
| 00:07.09 | aphexddb | my asterisk console shows the phone as picking up, so wierd: http://pastebin.com/699162 |
| 00:07.13 | Dr-Linux | Druken: well, as i said before, we started a call center in Dominican Republic |
| 00:07.21 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 00:07.35 | Dr-Linux | Qwell[]: we opened a callcenter there, but voice is very very bad, caller/callee can't unerstand each other and big noises |
| 00:07.44 | Dr-Linux | @ Druken |
| 00:08.19 | Druken | yeah i read that |
| 00:09.17 | Dr-Linux | Druken: so what you say about this damn happen? |
| 00:09.53 | aphexddb | anyone bored enough to help? my girlfriend is going to freak if i dont buy a long distance card to call her since the 'computer thing' is broken |
| 00:10.10 | ManxPower | Dr-Linux, What technology is the call center using? |
| 00:10.11 | Druken | well, did you do what Qwell said ? |
| 00:10.31 | aphexddb | aye captain, no packet loss |
| 00:10.36 | ManxPower | eBook is a bot |
| 00:10.48 | Qwell[] | ManxPower: Thank you Captain Obvious |
| 00:10.53 | Dr-Linux | ManxPower: technology? we different softphones |
| 00:11.14 | ManxPower | Dr-Linux, How are you connecting to the PSTN? |
| 00:11.18 | Qwell[] | aphexddb: So go buy one? |
| 00:11.18 | Dr-Linux | ManxPower: that's what we are using in other countries, but everywhere is just fine but in DR |
| 00:11.36 | ManxPower | Dr-Linux, sounds like you have a problem in the DR then. |
| 00:11.46 | Dr-Linux | ManxPower: my server is in USA |
| 00:12.02 | Dr-Linux | ManxPower: yeah this problem is only with DR users |
| 00:12.02 | ManxPower | Dr-Linux, as you said you don't have problems in other locations. |
| 00:12.16 | Dr-Linux | ManxPower: yeah all locations are just fine |
| 00:12.26 | ManxPower | Dr-Linux, When you send calls over the internet there is not much you can do to fix problems like that. |
| 00:12.27 | Dr-Linux | but DR ... |
| 00:12.58 | Dr-Linux | ManxPower: but we hired some guys and stuff .. we need to fix this |
| 00:13.09 | Dr-Linux | this happening since FEB |
| 00:13.14 | ManxPower | Dr-Linux, switch to a different ISP. |
| 00:13.35 | Dr-Linux | hhm... |
| 00:13.48 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 00:13.49 | Druken | what kinda fucken retard would have a kid over the age of 60 ? |
| 00:13.56 | Druken | that is just wrong |
| 00:14.03 | Qwell[] | Druken: The kid was born 60 years old? That's crazy |
| 00:14.09 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:14.10 | Dr-Linux | ManxPower: i asked here, i thought DR is near US, maybe someone knows their fuckin badnwidth issues |
| 00:14.17 | Druken | uhmm.. no.. the mom is 60+ |
| 00:14.32 | Qwell[] | ahh...well then |
| 00:14.36 | Qwell[] | the kind with a deathwish? |
| 00:15.08 | Druken | well, thats just having sex when your above 60 :) |
| 00:16.49 | *** part/#asterisk terrapen (n=cjs@166.70.183.109) |
| 00:17.33 | Druken | well, there's a law i know i'll be charged with a few times if passed.... |
| 00:17.39 | Druken | banning cellphone use while driving |
| 00:18.13 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 00:19.12 | aphexddb | Qwell[]: i loathe the idea of buying a phone card, specially when * is close to working |
| 00:19.58 | Druken | what's your problem? |
| 00:20.03 | key2 | is it possible to use STM optic fiber from a telco provider to Asterisk ? |
| 00:20.27 | Qwell[] | key2: Sure, just figure out a way to plug it in to *, and write whatever drivers are needed |
| 00:20.55 | Druken | or find an adaptor of some sort |
| 00:20.55 | key2 | Qwell: but can I use any fiber optic PCI card ? |
| 00:20.56 | Druken | hehe |
| 00:21.04 | Qwell[] | key2: If you write drivers, sure |
| 00:21.05 | orlok | key2: does it have linux drivers/ |
| 00:21.18 | Qwell[] | asterisk and kernel drivers |
| 00:21.20 | key2 | orlok: not sure it's even the same transceiver they use |
| 00:21.56 | aphexddb | Druken: pickup of a phone from a call over main sip trunk drops the call. whats wierd is if i call in from my cell, the cell stays connected, but the Polycom drops it |
| 00:21.57 | key2 | so I might be able to plugg on the OF but not sure that the transceiver is the same... |
| 00:22.06 | *** mode/#asterisk [+b *!*n=none@213.13.226.*] by russellb |
| 00:22.06 | *** kick/#asterisk [eBook!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 00:22.09 | key2 | plus it's not said that it uses MAC for sending packets |
| 00:22.34 | Druken | aphexddb: is there a nat in the middle? or a firewall ? |
| 00:23.17 | key2 | Qwell: do u think that it would use SS7 on an opticfiber ? |
| 00:23.43 | aphexddb | Druken: yes- im natted behind a netgear, had to setup the externip in sip.conf |
| 00:24.24 | Druken | aphexddb: local network between the polycom and asterisk ? |
| 00:24.57 | aphexddb | Druken: correct, they are on 192.168. |
| 00:25.10 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 00:25.18 | aphexddb | i can make softfone <-> polycom calls perfectly on the local |
| 00:26.33 | *** join/#asterisk terrapen (n=cjs@166.70.183.109) |
| 00:27.06 | Druken | is asterisk trying to reinvite the polycom ? |
| 00:27.12 | Druken | or is it staying in the path ? |
| 00:29.23 | aphexddb | im staring at the sip debug log, not sure what that means. |
| 00:30.24 | Druken | is the original call comming in sip ? |
| 00:31.24 | *** join/#asterisk mutilator (n=WebChat@65.111.201.81) |
| 00:31.33 | mutilator | OMFG |
| 00:31.37 | Druken | ? |
| 00:31.40 | mutilator | anyone know where to buy Tsunami Mp11's? |
| 00:31.47 | aphexddb | comes in from a Telasip trunk, -- Executing Dial("SIP/gallen-a1cf", "SIP/200|15|tr") in new stack |
| 00:31.56 | mutilator | proxim stoppe selling them |
| 00:31.57 | Druken | wtf is an mp11? |
| 00:32.04 | mutilator | wireless router |
| 00:32.47 | Druken | aphexddb: telasip doesn't tell me anything, is that sip or iax? |
| 00:33.10 | aphexddb | Druken: sorry, its sip |
| 00:33.28 | Druken | do you have canreinvite=no in your sip.conf? |
| 00:34.16 | aphexddb | thats a negative |
| 00:34.40 | Druken | put it in your general |
| 00:34.48 | Druken | and reload your sip.conf |
| 00:35.23 | *** join/#asterisk Crontibs (n=frankie@ool-43525f0d.dyn.optonline.net) |
| 00:36.24 | Dr-Linux | Druken: my * box uptime is 124 days and till yet i didn't try this command "reload" |
| 00:37.05 | Druken | so ? |
| 00:37.56 | aphexddb | Druken: just did a reload, same issue but getting a much longer sip log now in the console |
| 00:38.49 | Druken | want to pastebin it? |
| 00:39.27 | litage | hey guys, i'm experiencing some strange behaviour when using a grandstreamg gxp-2000 ip phone with asterisk |
| 00:39.33 | litage | when i make a call to the pstn from my linksys spa-941, through * and then through my VSP, everything works. but when i use my grandstream ip phone (gxp-2000) or ata (ht-286), it doesn't even ring |
| 00:39.39 | aphexddb | doing so now :) |
| 00:39.44 | mutilator | http://www.smh.com.au/news/unusual-tales/man-33-marries-woman-104/2006/05/03/1146335794619.html |
| 00:40.51 | aphexddb | Druken: this is calling in remotely, then picking up the Polycom: http://pastebin.com/699197 |
| 00:41.04 | Druken | why doesn't it supprise me the guys name is muhammed.... |
| 00:43.35 | Druken | aphexddb: well, that was my idea... join #freepbx and ask in there |
| 00:44.03 | Druken | i see it's amp, and i have no clue about what amp does.... except a bunch of extra shit that i'd never use |
| 00:44.29 | mutilator | :P |
| 00:44.30 | aphexddb | thanks for the advice, much appreciated |
| 00:44.34 | mutilator | man |
| 00:44.40 | mutilator | i am so bummed right now |
| 00:44.46 | Crontibs | NuFone: OOPS! Our Apologies |
| 00:44.47 | mutilator | like, no idea how bad |
| 00:45.04 | Crontibs | hrmm wonder when we will see the apologies about the tollfree's beeing down |
| 00:45.05 | Crontibs | hrmm |
| 00:45.07 | Druken | mutilator: guarenteed i have you beet |
| 00:45.12 | mutilator | um |
| 00:45.14 | mutilator | i doubt it |
| 00:45.20 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:45.20 | Druken | i don't |
| 00:45.28 | mutilator | we have a grid with 60 towers covering nortern michigan |
| 00:45.39 | mutilator | we can't buy subscriber units for ANY of those towers anymore |
| 00:45.56 | Druken | oh well, that's just business stuff |
| 00:46.03 | mutilator | neither the terabeam/hyperlink routers |
| 00:46.07 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 00:46.08 | Druken | i'm bummed on a personal level :) |
| 00:46.08 | mutilator | or the proxim mp11s |
| 00:46.20 | mutilator | this is a personal level aswell |
| 00:47.12 | mutilator | this is like a 50cal cannon to the chest |
| 00:47.19 | litage | any idea what could be happening that would enable a Grandstream GXP-2000 and HT-286 to make local PSTN calls, but not even ring when making international PSTN calls? |
| 00:48.11 | Crontibs | litage what comes up |
| 00:48.15 | Crontibs | on the console |
| 00:48.18 | Crontibs | when you try to call |
| 00:48.35 | Crontibs | also do you have dialplan setup on those devices to allow International Calls |
| 00:48.48 | Druken | :) |
| 00:49.16 | litage | Crontibs: yes, i have a dialplan that allows international calls. my Linksys SPA-941 can make international calls |
| 00:49.30 | ceeto | Hi all. Where do I specify where the asterisk sounds dir is in 1.2x? |
| 00:50.04 | litage | Crontibs: i've compared the *'s console output between making an international call with my spa-941 and with my gxp-2000, and they're virtually identical |
| 00:50.30 | litage | Crontibs: i've also inspected the sip packets that * sends to and receives from the ip phones, and they seem to be almost identical too |
| 00:51.24 | Crontibs | hrmm i'm guessing you turned up the logging in asterisk as well |
| 00:51.42 | Crontibs | also do you have the sip-context for each phone |
| 00:51.51 | Crontibs | going to the same one on the spa-941 |
| 00:51.58 | litage | Crontibs: yeah i turned up logging. what logging level would you recommend though? |
| 00:52.18 | litage | Crontibs: yeah both the spa-941 and gxp-2000 are using the same context |
| 00:53.57 | Crontibs | .. | i suggest packet snifffer |
| 00:54.01 | Crontibs | see what comes up |
| 00:54.11 | ckwall | ok... how do you make an extension forward to voicemail on busy/no answer? |
| 00:54.25 | ckwall | i have the box and the extesion set up, but i am failing to make the connection between the two |
| 00:58.25 | *** join/#asterisk MageOfChrisz (n=Mage@unaffiliated/mageofchrisz) |
| 00:59.15 | *** part/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 01:01.52 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-71-250.dsl.irvnca.pacbell.net) |
| 01:04.03 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 01:04.21 | asterboy | Anyone suggest a good outlet for Sangoma cards? |
| 01:07.01 | ckwall | i am also having a problem with my dial plan... I am unable to dial outside numbers. could any one please double check my files and help me to identify what the problem is? |
| 01:07.12 | ckwall | http://pastebin.com/699226 |
| 01:07.32 | *** part/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 01:08.32 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 01:08.42 | ckwall | anyone willing to look and see if I have any mistakes? |
| 01:13.09 | ceeto | If I have a TDM400P installed but no phone lines, should asterisk still run without errors? I'm getting a zapata.conf error about my channels, 1-4. |
| 01:13.39 | generalhan | ceeto: asterisk should still run with out lines connected to those channels |
| 01:13.46 | ceeto | hrm... |
| 01:13.56 | generalhan | you using a TDM40B ? |
| 01:14.02 | ceeto | Yep. |
| 01:14.08 | generalhan | any other cards ? |
| 01:14.13 | ceeto | Nope.. |
| 01:14.18 | generalhan | it took me a while to get my TDM up with my TE210 |
| 01:14.28 | generalhan | pb your zaptel.conf and zapata.conf |
| 01:15.13 | ceeto | http://pastebin.com/699241 |
| 01:15.21 | ceeto | Thanks for any help! |
| 01:15.41 | generalhan | ceeto: remember you need FX |
| 01:15.49 | generalhan | FXO signalling for your FXS lines |
| 01:16.09 | generalhan | fxoks=1-4 in stead of fxsks=1-4 |
| 01:16.38 | ceeto | They are 1fb lines from the telco... |
| 01:16.45 | generalhan | ??? |
| 01:16.46 | ceeto | I think it's permissions related, works when I load as root... |
| 01:16.52 | generalhan | ohh |
| 01:16.53 | generalhan | ok |
| 01:17.06 | ceeto | Any clue where to set them? |
| 01:17.10 | generalhan | thats still odd though ... if you are using FXS modules it shouldnt work with FXS signalling |
| 01:17.38 | ceeto | So if I have the little red daughter boards, I should be using FXO signalling in zaptel.conf? |
| 01:18.05 | generalhan | shoot ... im not really sure what color they are .... lol |
| 01:18.17 | ceeto | The ones that you plug into the phone company, not the phones. |
| 01:18.19 | asterboy | ceeto, no you will use FXS signalling |
| 01:18.41 | ceeto | Ok. Thanks. I'm stuck on trying to figure out where the asterisk user needs permission to access the hardware... |
| 01:18.45 | generalhan | ceeto: i appologize for misleading you asterboy would be better suited to help you ! |
| 01:18.54 | asterboy | try running 'ztcfg -vv' at a linux prompt |
| 01:19.08 | asterboy | sometimes there is a timing issue |
| 01:19.09 | ceeto | Shows 4 channels. |
| 01:19.13 | ceeto | FXS |
| 01:19.16 | asterboy | then there you go |
| 01:19.19 | ceeto | Thanks. |
| 01:19.27 | ceeto | You the man/woman/gender challenged. |
| 01:19.41 | asterboy | lol, depends on what drugs I'm on |
| 01:19.45 | generalhan | asterboy: how come i have to use kewlstart on FXO signalling on my TDM40B but he doesnt have to ? |
| 01:19.54 | ceeto | heh |
| 01:20.05 | asterboy | I'm sure he is |
| 01:20.27 | asterboy | if not, you don't necesarily HAVE to have kewlstart |
| 01:20.35 | asterboy | the Bible says so |
| 01:20.42 | ceeto | Ok, when running as root all is well. I want to run as user/group asterisk. I get errors regarding loading chan_zap.so and then an error about loading audio data. Any clue where I have permissions messed up? |
| 01:20.45 | generalhan | well no ... but i definately had to have it set to FXO not FXS |
| 01:21.18 | ceeto | Got it, nevermind. /dev/zap |
| 01:21.42 | litage | Crontibs: yeah i examined the packes in ethereal but i can't see any difference between what the spa-941 and gxp-2000 send to/receive from asterisk |
| 01:21.51 | asterboy | FXO for FXO hardware? |
| 01:21.55 | generalhan | no |
| 01:21.58 | generalhan | FXO for FXS hardware |
| 01:22.00 | generalhan | TDM40B |
| 01:22.05 | generalhan | 4 FXS modules |
| 01:22.22 | asterboy | ah, yes that is correct |
| 01:22.28 | generalhan | thats the same card he is using |
| 01:22.58 | asterboy | he said RED, thats FXO |
| 01:23.06 | generalhan | thats why i was telling him to chage his fxsks=1-4 to fxoks=1-4 |
| 01:23.18 | asterboy | nope other way around |
| 01:23.35 | asterboy | remember signalling is the opposite |
| 01:23.44 | generalhan | see i thought that the TDM40B meant that it had 4 FXS modules i didnt know you could get it with FXS modules |
| 01:23.48 | generalhan | yea i know that |
| 01:23.52 | ceeto | Red plugs into telco... |
| 01:24.02 | ceeto | Green plugs into phones... |
| 01:24.05 | asterboy | yep and you can have TDM40B or TDM04B |
| 01:24.12 | asterboy | RED or GREEN |
| 01:24.18 | asterboy | FXO or FXS |
| 01:24.23 | asterboy | Telco or Phone |
| 01:24.30 | generalhan | right ... so when he tells me he has a TDM40B then i was right to tell him fxoks right ? |
| 01:24.33 | ceeto | I have telco plugged into 4 lines, my zaptel.conf is set to fxsks=1-4... is that right or wrong? |
| 01:24.36 | ceeto | It's working... |
| 01:24.39 | asterboy | nop |
| 01:24.46 | asterboy | fxsks |
| 01:24.50 | asterboy | OPPOSITE! |
| 01:25.05 | generalhan | the TDM04B would be fxsks cuase the 04B is FXO and the 40B is FXS |
| 01:25.08 | asterboy | that nop was for gen |
| 01:25.17 | asterboy | OPPOSITE |
| 01:25.20 | generalhan | DAMNIT |
| 01:25.21 | generalhan | lol |
| 01:25.23 | ceeto | Ahh.. So I'm right? heh... |
| 01:25.25 | asterboy | ceeto has it right |
| 01:25.38 | ceeto | >sigh< thanks guys.. I have a headache now.. night. |
| 01:25.39 | asterboy | yes correct affirmative |
| 01:25.42 | *** join/#asterisk jaguiar (n=jaguiar@201.152.42.105) |
| 01:25.51 | ceeto | (thanks for the help) |
| 01:25.56 | asterboy | It's the only place you can get turned around in * |
| 01:26.09 | asterboy | well, as far as FXO/FXS go |
| 01:26.23 | generalhan | asterboy: im REALLY REALLY confused ... i have a TDM40B ... NOT a TDM04B, and my configs are set up with fxoks cause the 40B is a FXO card |
| 01:26.38 | asterboy | nope |
| 01:26.41 | asterboy | wrong |
| 01:26.42 | Qwell | fxsks for fxo |
| 01:26.43 | asterboy | incorrect |
| 01:26.51 | asterboy | yep what Qwell said |
| 01:27.33 | generalhan | i mistyped before ... i am using fxoks on my FXS card which is a TDM40B ... the TDM04B would use fxsks signalling |
| 01:28.04 | generalhan | right ? |
| 01:29.02 | ckwall | is no one willing to look at my config files? :( |
| 01:29.10 | ckwall | so sad |
| 01:29.27 | generalhan | cuase i had that problem when i got this card ... i was using fxsks and it wasnt working ... then i switched them to fxoks and they work. so i asked him if he had a TDM40B so i would know his configs would be the same as mine |
| 01:29.52 | generalhan | but now you guys told him to do it the opposite of mine and thats why im all kinds of confused |
| 01:30.31 | asterboy | yes TDM40B is the FXS and should have FXO signalling |
| 01:31.15 | asterboy | fxoks is correct |
| 01:31.47 | asterboy | ceeto has the TDM04B with RED FXO modules...so he will have the opposite of yours. |
| 01:31.49 | generalhan | thats what i have been telling him since the begining |
| 01:32.07 | generalhan | ok im not confused at all .. he just told me the wrong model number for his board is all |
| 01:32.09 | asterboy | for him fxsks is the correct designation |
| 01:32.24 | asterboy | yep, I think that is all |
| 01:32.32 | generalhan | damnit im going outta my mind ... |
| 01:32.33 | asterboy | now look at the sea of text he created! |
| 01:32.36 | generalhan | i gotta get outta work already |
| 01:32.43 | generalhan | better than no talking ! lol |
| 01:32.57 | asterboy | bad ceeto |
| 01:33.22 | generalhan | i wish [TK]D-Fender or Hmmhesays was here ... i want to tell them that i finished finally |
| 01:33.34 | asterboy | ya TK is a great resourse here |
| 01:33.55 | generalhan | they have been trying to help me with a way to turn off callwaiting on phones that you cant. and how to use the ASTDB to pass variables into queues and back |
| 01:34.09 | ckwall | I have TE410P... I have a pri T1 plugged into it. On the T1, I have an 800 number associated with it. Is there anyway to see if calls are getting in on that T1? when I call, all i hear is dead air, then after about 5 second it hangs up. I am trying to see if my t1 is recieving calls, because I cannot figure out how to place calls yet. |
| 01:34.28 | generalhan | it sux cause i used to use incominglimit=1 and now thats depriciated so i had to replace it with 10 lines of code in extensions.conf ... boo |
| 01:34.52 | generalhan | ckwall: you have the CLI running ? |
| 01:34.56 | ckwall | yes |
| 01:35.07 | generalhan | ckwall: cuase you should see an attempt at that number weather you get to ring a phone or not |
| 01:35.15 | ckwall | hmmm |
| 01:35.16 | ckwall | ok |
| 01:35.27 | ckwall | so if i see no attempt, it is probably not working? |
| 01:35.28 | generalhan | i see people hitting numbers that i dont have defined on my T1 ALL THE TIME |
| 01:35.40 | generalhan | ckwall: you only have one number to test ? |
| 01:35.46 | ckwall | two |
| 01:35.53 | generalhan | a local and a toll-free ? |
| 01:36.17 | ckwall | two toll free, they have not told me what the local numbers are yet |
| 01:36.26 | generalhan | ckwall: thats the issue i think |
| 01:36.33 | ckwall | ah, ok |
| 01:36.42 | generalhan | ckwall: i dont know FOR SURE ... but typically a toll-free is pointed to a DID on your switch |
| 01:36.50 | generalhan | and THAT DID is what you program in your extensions.confg |
| 01:36.53 | *** join/#asterisk MacDome (n=eseidel@64-121-195-96.c3-0.sfpo-ubr4.sfrn-sfpo.ca.cable.rcn.com) |
| 01:37.14 | ckwall | but wouldnt the attempt be made before hitting the extensions.conf? |
| 01:37.26 | ckwall | on another note... I have not been able to place a call either. |
| 01:37.35 | generalhan | ckwall: so i have a toll-free 888-888-8888 and it points to my local DID of 555-555-5555, so the 555-555-5555 would be in my extensions.conf to direct the traffic on that toll-free number |
| 01:37.52 | ckwall | correct |
| 01:38.02 | ckwall | I will hold off on that till i get the dids. |
| 01:38.10 | ckwall | are you willing to help me troubleshoot my outbound calling? |
| 01:38.13 | generalhan | ckwall: but if you dont have the DID t hat the toll-free is pointing to you cant get it to do anything |
| 01:38.40 | generalhan | ckwall: im only an novice ... i can try but i will promise nothing |
| 01:38.46 | ckwall | thank you. |
| 01:38.47 | generalhan | will = will not |
| 01:39.03 | ioerror | Is asterisk pretty well supported on AMD64? |
| 01:39.25 | ckwall | ok, i have posted a few things on pb... let me find the link really quick |
| 01:40.07 | ckwall | http://pastebin.com/699226 |
| 01:40.20 | ckwall | I can dial other extensions, but not outside numbers. |
| 01:40.36 | ckwall | this is what led me to wonder if my t1 was even working. |
| 01:41.04 | generalhan | ckwall: what happens if you pick your phone up and dial 3000 right now ? |
| 01:41.22 | ckwall | well, my extension is 3000.. .so it rings its self. |
| 01:41.32 | ckwall | ...ringing, and blinking light |
| 01:41.54 | generalhan | ckwall: you have this --- exten => 3000,1,Dial(Zap/g1/18015582352) so that means that if that is your personal extension then your T1 IS working |
| 01:42.07 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:42.13 | generalhan | cause that is a dial command that is telling it to dial out on your Group 1 ZAP channels |
| 01:42.32 | [TK]D-Fender | generalhan : Had fun with what I gave you? |
| 01:42.41 | generalhan | [TK]D-Fender: WHATS UP !!!! lol .... i got it all working now ... i had to do it for 3 more queues and in-office extensions |
| 01:42.53 | [TK]D-Fender | generalhan : And is it starting to look/feel a bit more natural? |
| 01:42.58 | ckwall | well...even without that line it dials other extensions |
| 01:43.08 | ckwall | i added that line to try to dial out. |
| 01:43.32 | hads | ckwall: All the priorities on your exten 3000 are 1 |
| 01:43.40 | generalhan | ckwall: well first of all you have 2 exten => 3000,1,Dial CMDs |
| 01:43.43 | hads | and you have two exten 3000 |
| 01:44.23 | *** join/#asterisk Olobola (i=Olobola@216.218.221.165) |
| 01:44.40 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
| 01:44.46 | MrDigital | anyone familiar with SATA Drives? |
| 01:45.17 | timscott | a little. |
| 01:45.18 | timscott | What's up? |
| 01:45.22 | generalhan | ckwall: put this line in there exten => 5000,1,Dial(ZAP/g1/180080080000) and then do an extensions reload in the CLI and hit 5000 from your phone and see what happens |
| 01:45.23 | MrDigital | i got a PC |
| 01:45.28 | MrDigital | has 2 hdds |
| 01:45.37 | MrDigital | 1 Sata drive 200gig and a PATA(IDE) drive 120gig |
| 01:45.41 | asterboy | So far it looks like iFax has the best prices on Sangoma cards...anyone else have a better source? |
| 01:45.43 | ckwall | ok |
| 01:45.46 | MrDigital | for some reason it wants to boot off IDE0 |
| 01:45.51 | MrDigital | instead of IDE2(Sata) |
| 01:45.57 | MrDigital | before i put in the ide hdd |
| 01:45.58 | Qwell | MrDigital: change it in the bios |
| 01:45.59 | MrDigital | it would boot fine |
| 01:46.03 | MrDigital | cant |
| 01:46.04 | *** join/#asterisk annonimous (n=annonimo@dsl-201-133-94-178.prod-infinitum.com.mx) |
| 01:46.07 | timscott | Are you sure? |
| 01:46.07 | annonimous | hello ther |
| 01:46.07 | Qwell | Then you're screwed. :p |
| 01:46.13 | Qwell | Put your bootloader on ide0 |
| 01:46.14 | timscott | I'm sure you can...there's gotta be a way. |
| 01:46.17 | [TK]D-Fender | asterboy : lemme look |
| 01:46.21 | MrDigital | how its a XP Bootloader |
| 01:46.23 | generalhan | [TK]D-Fender: sorry didnt see your message ... i dont know if its feeling NATURAL .. but i DEFINATELY understand what its doing now ! |
| 01:46.25 | [TK]D-Fender | asterboy : can you link them for me? |
| 01:46.30 | Qwell | MrDigital: fixmbr |
| 01:46.30 | asterboy | sure |
| 01:46.43 | MrDigital | it writes the MBR to D:\ |
| 01:46.44 | asterboy | http://shop.ifax.com/-c-32_33_47_48.html |
| 01:46.44 | Qwell | or install grub or something :p |
| 01:46.49 | MrDigital | instead of C:\ |
| 01:46.50 | MrDigital | not linux |
| 01:46.52 | MrDigital | Windows |
| 01:46.52 | [TK]D-Fender | generalhan : Yeah you just need to know that the key-pair is UNIQUE, and you use something variable to ensure that. |
| 01:46.54 | Qwell | so? |
| 01:46.56 | Qwell | grub is grub |
| 01:46.58 | ckwall | I just get a fast busy signal |
| 01:47.02 | annonimous | in a TDM2400P i can input a 24 fxo (trunk) modules?? |
| 01:47.06 | Qwell | annonimous: yes |
| 01:47.21 | generalhan | [TK]D-Fender: this is what i said to you RIGHT after you left "OMFG i understand now ... to replicate the UNIQUEID you save it as ANOTHER variable to call on that exact stop later .... OMFG it all makes soooooo much freaking sense now ... thank you soo much !" |
| 01:47.32 | annonimous | Qwell, ok another tech question in a computer i can input 3 pci tdm04b without problems? |
| 01:47.39 | Qwell | annonimous: I wouldn't, no |
| 01:47.46 | Qwell | 2 tops |
| 01:47.52 | annonimous | Qwell, 2? ok |
| 01:47.55 | Qwell | interrupts are a bitch.. |
| 01:48.06 | annonimous | Qwell, ah i see |
| 01:48.12 | Qwell | even 2 have trouble sometimes |
| 01:48.24 | [TK]D-Fender | generalhan : Ok well put it all to good use. Forwarding is another good use of ASTDB |
| 01:48.24 | Qwell | If I were you, I'd just get a tdm2400p instead of 2x tdm400p |
| 01:48.27 | annonimous | oh ok |
| 01:48.34 | [TK]D-Fender | asterboy : Yup, pricing looks good |
| 01:48.36 | Qwell | the cost is about the same at that ratio |
| 01:48.38 | generalhan | [TK]D-Fender: forwarding how so |
| 01:48.41 | *** join/#asterisk cr_0 (n=y@toronto-HSE-ppp4334596.sympatico.ca) |
| 01:48.43 | Qwell | buy more modules, and it ends up being less |
| 01:48.55 | asterboy | I need 8 FXO, so iFax has Sangoma for $604, the rest are about $650 and VoipWare.ca wants an outrageous $800 |
| 01:49.14 | Qwell | asterboy: get a Digium :p |
| 01:49.14 | annonimous | Qwell, ah i see |
| 01:49.22 | ckwall | generalhan: no go |
| 01:49.22 | [TK]D-Fender | generalhan : If you want * to control forwarding on extensions instead of the phones themselves, setting ASTDB values based on the extension is a great way and checking before you dial. |
| 01:49.23 | Qwell | get 2 4xfxo modules... |
| 01:49.32 | asterboy | lol, ya after dealing with Digium's IRQ issues, no way. |
| 01:49.39 | cr_0 | i just enabled voicmail via mysql. everything _looks_ good. is there a way from the console to verify that asterisk is reading in the data correctly? |
| 01:49.41 | Qwell | asterboy: all cards have irq issues :P |
| 01:49.41 | asterboy | lost a lot of money from that already |
| 01:49.42 | generalhan | [TK]D-Fender: ohh isee ... that makes sense |
| 01:49.47 | [TK]D-Fender | asterboy : Yeah, Voipware are BAD.... |
| 01:49.48 | asterboy | Qwell, not the Sangoma |
| 01:49.49 | annonimous | Qwell, thank you! =)! |
| 01:49.57 | Qwell | sangoma uses irqs too |
| 01:50.04 | Qwell | all pci cards use an irq |
| 01:50.09 | Qwell | afaik, anyhow |
| 01:50.11 | generalhan | I have to get outta here guys |
| 01:50.13 | generalhan | lol |
| 01:50.13 | asterboy | ya but does not have the irq share issuees |
| 01:50.20 | asterboy | cya |
| 01:50.24 | Qwell | asterboy: irq share issues are part of the x86 spec :P |
| 01:50.26 | [TK]D-Fender | Qwell : referring to interrupts/sec load |
| 01:50.37 | Qwell | [TK]D-Fender: But that's different from sharing IRQs |
| 01:50.41 | asterboy | plus the card uses 1 pci slot |
| 01:50.48 | Qwell | yeah, so does a tdm2400p |
| 01:50.56 | asterboy | cost a lot less comparatively |
| 01:50.57 | [TK]D-Fender | Qwell : indeed, but the statement was grey and both fit in. easy to think A instead of B. |
| 01:51.02 | cr_0 | is there a way to show the voicemail boxes configured via the console? |
| 01:51.03 | Qwell | asterboy: not so |
| 01:51.04 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 01:51.18 | Qwell | cr_0: show voicemail users |
| 01:51.48 | ckwall | what is the syntax to be able to forward no answer to voicemail? |
| 01:51.53 | cr_0 | Qwell: i don't have show voicemail. show ? doesn not give a voicemail option. |
| 01:51.56 | asterboy | Qwell, I'd have to go with 2 TDM04B cards or 1 TDM2400P |
| 01:51.57 | Qwell | ckwall: It's all dialplan logic |
| 01:52.03 | Qwell | cr_0: dunno |
| 01:52.08 | cr_0 | Qwell: i have version 1.0.x , does that matter? |
| 01:52.09 | Qwell | asterboy: so get a tdm2400p |
| 01:52.12 | Qwell | cr_0: it might |
| 01:52.47 | generalhan | ckwall: you kinda had it on you 3000,1 lines ... just add the next line as 3000,2,Voicemail(u$EXTEN@whatever) and if you dont answer that line it will just reoll over to the VM line |
| 01:53.16 | asterboy | ya the tdm2401b costs $845 compared to A20004's $604 |
| 01:53.30 | cr_0 | app_voicemail.so is loaded, i assume that is the right one... |
| 01:53.45 | [TK]D-Fender | asterboy :actually, you'd need 2 modules for 6 lines... thats TDM2402b <- |
| 01:53.47 | generalhan | im out see ya all tomorrow ! |
| 01:53.54 | [TK]D-Fender | asterboy : and 2 ports wasted |
| 01:53.54 | ckwall | ok, cool |
| 01:53.56 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 01:53.57 | ckwall | thanks |
| 01:54.08 | Qwell | he said 8 lines |
| 01:54.12 | [TK]D-Fender | Qwell : oops |
| 01:54.13 | asterboy | oh ya! |
| 01:54.14 | [TK]D-Fender | :) |
| 01:54.15 | asterboy | your right! |
| 01:54.19 | [TK]D-Fender | that too! |
| 01:54.25 | asterboy | DOUBLE the cost |
| 01:54.27 | [TK]D-Fender | mixed him up with another guy earlier needing 6 |
| 01:54.37 | [TK]D-Fender | asterboy : No, I am mistaken, nix that |
| 01:54.39 | Qwell | $837 without echo can |
| 01:54.52 | asterboy | shit ya...back to 1 |
| 01:54.56 | asterboy | I had it the first time |
| 01:55.07 | cr_0 | this echo cancellation, does it make a big difference? |
| 01:55.12 | [TK]D-Fender | but yeah still 2 modules... eek |
| 01:55.14 | Qwell | cr_0: it can |
| 01:55.17 | [TK]D-Fender | <- math failing!! |
| 01:55.30 | asterboy | wait, isn't that a quad meaning 4 |
| 01:55.34 | cr_0 | Qwell: can? under what situation would you say you need it? |
| 01:55.34 | asterboy | I do need the two |
| 01:55.47 | Qwell | cr_0: it lowers processor usage |
| 01:55.50 | [TK]D-Fender | Sensei had us beating ourselves to death today and I've got all sorts of blisters and a little blood to back it too... |
| 01:56.12 | cr_0 | Qwell: ah, cool. so, echo can is done anyway, but on hardware is more efficient on the CPU. |
| 01:56.28 | cr_0 | Qwell: i was under the impression that there was a difference in the echo cancellation quality. |
| 01:56.39 | Qwell | nah, I don't think so |
| 01:57.00 | cr_0 | Qwell: thanks.... back to my voicemail woes... :-) |
| 01:57.12 | asterboy | So I do need a TDM2402B for 8 lines at $1200! |
| 01:57.19 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:57.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:57.25 | asterboy | 2 QUAD modules |
| 01:57.31 | asterboy | that is double the cost. |
| 01:57.36 | asterboy | compared to sangoma |
| 01:58.29 | Qwell | wtf |
| 01:58.46 | Qwell | asterboy: no, it's $837 for a tdm2402B |
| 01:59.04 | Qwell | and, wtf does the sangoma require so many slots on the back panel? 1 slot per 4 ports? |
| 01:59.28 | [TK]D-Fender | asterboy : No, on iFax (your link), the TDM2402B = $837 <- |
| 01:59.49 | asterboy | no, it show $1209 |
| 01:59.53 | asterboy | http://www.voipware.ca/index.php?main_page=product_info&cPath=9&products_id=68 |
| 02:00.04 | asterboy | I want the price you guys are getting. |
| 02:00.08 | [TK]D-Fender | Qwell : because the care has pot jacks on it and doesnt' require to spend MORE on an RJ21 breakout panel. |
| 02:00.19 | [TK]D-Fender | Qwell : And can fin in LP systems. |
| 02:00.24 | [TK]D-Fender | fit* |
| 02:00.32 | [TK]D-Fender | Qwell : Each has their strong suit |
| 02:00.43 | Qwell | show me a LP system that has 6 slots :P |
| 02:00.55 | annonimous | Qwell, thank you for your support i have2 go |
| 02:00.55 | [TK]D-Fender | Qwell : many supermicro actually... |
| 02:01.03 | Qwell | note to self: *NEVER* buy from voipware |
| 02:01.06 | [TK]D-Fender | Qwell : Take a tour, plenty of options... |
| 02:01.12 | Qwell | Those prices are fucking rediculous |
| 02:01.17 | [TK]D-Fender | Qwell : Gotta agree with you on them... |
| 02:01.22 | *** part/#asterisk annonimous (n=annonimo@dsl-201-133-94-178.prod-infinitum.com.mx) |
| 02:01.31 | asterboy | ya voipware is vampireware |
| 02:01.51 | [TK]D-Fender | :F |
| 02:01.56 | asterboy | lol |
| 02:01.56 | sevard | what about voipsupply.com they have alrigh tprices on somethings |
| 02:02.25 | asterboy | iFax has best prices on Sangoma |
| 02:02.32 | asterboy | so far as I have found |
| 02:02.48 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 02:02.59 | [TK]D-Fender | asterboy : VoipSupply's Canadian division is kinda cool,m but it does ship for the same place co check your exchange rate before choosing whi division, because duty will be the same % :) |
| 02:03.03 | marl | can anyone tell me, is it posible to initate a call from an agi script, and have the script comunicate with the end caller without the incoming call hearing? e.g. ask the called party if they wish to accept the call? |
| 02:03.25 | sevard | why the hell are there so many canadians using asterisk |
| 02:03.28 | [TK]D-Fender | asterboy : Actually GenTek has much better pricing, but thats what wholesaling is for :) |
| 02:03.32 | sevard | up in da nort wuds |
| 02:03.43 | asterboy | GenTek...I have an account with them |
| 02:03.49 | [TK]D-Fender | asterboy : well WTF! |
| 02:03.52 | asterboy | In the days when I bought modems |
| 02:03.54 | sevard | WELL WTF |
| 02:03.54 | [TK]D-Fender | asterboy : GO THERE! |
| 02:03.59 | sevard | GAWD DANG BOIE |
| 02:04.05 | sevard | LOUD NOISES |
| 02:04.06 | asterboy | sweet |
| 02:04.18 | sevard | ^^ one of my favorite quotes. |
| 02:05.08 | [TK]D-Fender | I live close enough to GenTek to WALK there :) |
| 02:05.09 | asterboy | ah forget it...I'm buying from VampireWare instead...money to burn |
| 02:05.24 | asterboy | that would be great. |
| 02:05.29 | [TK]D-Fender | asterboy : Oh you mean somebody elses? :) |
| 02:05.29 | asterboy | to be able to pickup |
| 02:05.52 | ckwall | how do i adjust how long a phone rings before going to voicemail |
| 02:06.34 | [TK]D-Fender | ckwall : that the timeout option in Dial. |
| 02:06.43 | ckwall | ok |
| 02:06.47 | ckwall | here is another question... |
| 02:07.00 | asterboy | I use to live in Ontario...beautiful province |
| 02:07.01 | sevard | pick a notha numba |
| 02:07.04 | [TK]D-Fender | ckwall : Go to the Wiki. look up the page called "Complete command reference" and then go though ALL OF THEM. In detail. |
| 02:07.37 | ckwall | I guess i am getting a little lazy, sorry |
| 02:07.44 | sevard | lazy bitch |
| 02:07.54 | [TK]D-Fender | Ontario = wishy-washy middle-of-the-road-yet-think-we're-better-than-everyone-else province :) |
| 02:08.16 | sevard | I like Ontario... the west side |
| 02:08.19 | sevard | screw the east side |
| 02:08.21 | sevard | all of it. |
| 02:08.43 | [TK]D-Fender | All of Eastern Canada = soon to be third world country. |
| 02:08.59 | [TK]D-Fender | (NB onwards) |
| 02:09.09 | sevard | I've never stepped foot in eastern canada and I'm not ready to any time soon |
| 02:09.29 | sevard | and if I'm going to western canada i'm just going to be on my way to alaska |
| 02:09.40 | [TK]D-Fender | Alaska? eek |
| 02:09.44 | file | eh? |
| 02:09.47 | file | I live in NB. |
| 02:09.53 | sevard | compared to alaska, everything else is a butthole. |
| 02:09.55 | sevard | fuck it. |
| 02:09.57 | sevard | :) |
| 02:10.31 | [TK]D-Fender | BC is very nice geographically. They've got it all. Appalachia, mountains, plains, valleys, deserts, everything.... and not to much snow. |
| 02:10.56 | [TK]D-Fender | And a mostly-working socialist provincial party. |
| 02:11.00 | sevard | hahaha |
| 02:12.01 | [TK]D-Fender | If anywhere, thats where I'd go. |
| 02:12.09 | sevard | the farthest east i'll go is red rock onterio |
| 02:12.14 | [TK]D-Fender | Better on imports too :) |
| 02:12.33 | sevard | but if i had the choice of where to be it'd be in sitka alaska |
| 02:13.06 | sevard | apparently I don't decide where to live these days this green paper shit is in charge |
| 02:20.25 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 02:20.30 | techman97_andy | hey all |
| 02:21.54 | techman97_andy | I have an odd question. I have multiple xLite softphones that are working just fine. My system is G729 end to end. I have one GrandStream GXP-2000 phone I'm using here, and when I try to initiate a conference call - I can have both parties on individual lines, but when I press "conference", the line I want to conference in drops, and the CLI says "no compatible codecs"....any ideas? |
| 02:22.09 | techman97_andy | I've set all codec preferences on the grandstream to G729 only |
| 02:22.32 | [TK]D-Fender | techman97_andy : X-Lite does NOT support G.729 |
| 02:22.38 | techman97_andy | yeah it does |
| 02:22.41 | techman97_andy | I'm using it right now? |
| 02:22.45 | techman97_andy | well, not RIGHT now... |
| 02:22.53 | techman97_andy | but xLite isn't my problem here |
| 02:22.55 | techman97_andy | it works fine. |
| 02:23.01 | jql | free stuff does not g.729 |
| 02:23.13 | [TK]D-Fender | exactly |
| 02:23.59 | jql | the patent nazi stormtroopers would squish X-Lite like a bug. no g.729 |
| 02:24.02 | [TK]D-Fender | techman97_andy : Most phones that support G.729 cannot CONFERENCE it, meaning only once channel can use that codec, and because you've restricted the phone to it it gets droeed. Thats why |
| 02:24.17 | [TK]D-Fender | dropped |
| 02:24.43 | [TK]D-Fender | This is a chronic thing with Sipura ATA's and so much else |
| 02:27.18 | techman97_andy | 64.24.35.196 1651492664 3d2966675bc 00102/00000 g729 No Tx: ACK |
| 02:27.18 | techman97_andy | 192.168.1.101 12 ad469c1e715 00101/00002 g729 No Rx: ACK |
| 02:27.18 | techman97_andy | 64.24.35.196 1612845213 0467cfb03d7 00102/00000 g729 No Tx: ACK |
| 02:27.18 | techman97_andy | 192.168.1.101 12 db36450cb61 00101/00004 g729 No Rx: ACK |
| 02:27.29 | techman97_andy | that's a 3-way call right now with xLite....conferenced. |
| 02:27.30 | techman97_andy | it works. |
| 02:27.46 | [TK]D-Fender | :/ |
| 02:27.56 | [TK]D-Fender | I'd love to know how thats possible.... |
| 02:28.07 | jql | sounds like X-Pro |
| 02:28.08 | techman97_andy | dunno man, maybe I'm just magic...:P |
| 02:28.26 | asterboy | techman97_andy, I had the same problem when I was trying to connect a VOIP service provider with G729 and my Digium FXS port. |
| 02:28.36 | asterboy | Had to purchase the license and it worked. |
| 02:28.46 | asterboy | not saying that you have same thing obviously |
| 02:28.53 | techman97_andy | but either way, the Grandstream isn't free....and it does g729 calls as well. I can't understand why it won't do a conference call |
| 02:29.04 | techman97_andy | I did purchase 10 licenses of the G729 licenses from Digium |
| 02:29.13 | asterboy | but it may help to understand that * can't bridge the call because it can't find a codec to convert to for some reason. |
| 02:29.34 | Olobola | so does providing iax origination/termination make for a bad business model? Why haven't any big money investors jumped in yet? Or have they? |
| 02:29.56 | techman97_andy | asterboy: I don't see how it can't though - it's all g729? Is there any way I can enable further logging on * to see anything? |
| 02:30.14 | asterboy | sip debug peer |
| 02:30.25 | jql | get the sdp, look at the codec list |
| 02:30.28 | jql | see what's fubar |
| 02:30.45 | websae | techman97_andy: not interested in our service? |
| 02:31.28 | asterboy | you have the right idea...you need to dig deeper to find out where the failure is. |
| 02:32.05 | asterboy | even though you think its all g729, there may be something otherwise somewhere. |
| 02:32.15 | file | debug debug debug |
| 02:32.22 | techman97_andy | k - I'm trying the sdp |
| 02:33.28 | techman97_andy | May 4 21:32:57 NOTICE[24263]: chan_sip.c:3588 process_sdp: No compatible codecs! |
| 02:33.31 | techman97_andy | grrrrr |
| 02:33.49 | *** join/#asterisk selsky (n=selsky@dookie.cc.columbia.edu) |
| 02:34.16 | *** join/#asterisk espn (n=DA-MAN@adsl-69-106-102-73.dsl.irvnca.sbcglobal.net) |
| 02:34.39 | selsky | how do i find info on what specific branches in the teams svn directory are being used for? |
| 02:36.20 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 02:38.47 | techman97_andy | hmmm - any other advice on other commands other than sdp I can use? It came up with a lot of text, but it was all identical to the softphone conference call with the one exception of that chan_sip.c:3588 error message |
| 02:39.13 | file | selsky: it's up to the person |
| 02:39.31 | jql | techman97_andy: personally, I use ethereal. Decodes the packet for me |
| 02:40.26 | techman97_andy | I did that once (forgive me, I'm not quite the experienced Linux admin yet...*smile*), but I haven't a clue on how to do it again - can you drop me a hint on how to get the packet dump? |
| 02:40.41 | jql | tethereal -R "udp.port == 5060" -V |
| 02:40.50 | jql | will dump everything in spammy fashion |
| 02:42.06 | asterboy | I use tcpdump |
| 02:42.34 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:44.22 | selsky | file: i'm looking at team/jrothenberger/asterisk-imap/ |
| 02:45.37 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 02:45.59 | file | ah yes... that |
| 02:48.01 | Olobola | have any of you experienced a T1 outage? My DSL has been down for hours, thinking about getting a T1. |
| 02:48.16 | orlok | Olobola: not that irecall |
| 02:48.17 | Qwell | Olobola: T1's have SLAs |
| 02:48.32 | Qwell | They can still go down, but it'll usually get fixed in fairly short order |
| 02:48.32 | orlok | there were probably some outages of the provider of the line |
| 02:48.50 | orlok | Qwell: so does DSL, its just that the telco's are more careful about bri's |
| 02:48.54 | orlok | cos they are a bigger cash cow |
| 02:49.00 | orlok | well, our DSL does :) |
| 02:49.11 | orlok | anything gets backhoe fade |
| 02:50.54 | Olobola | ok, thanks |
| 02:51.31 | Olobola | We have a wimax connection here too, but ping rates seem a little funky. I guess wimax isn't that stable |
| 02:51.49 | Qwell | wimax isn't even a standard yet, is it? |
| 02:52.13 | Olobola | I just learned about when we moved to our new location |
| 02:52.37 | asterboy | it is a standard, just that adoption is slow |
| 02:52.43 | Qwell | You sure? |
| 02:52.53 | asterboy | wimax forums has a lot of info on it. |
| 02:53.02 | Olobola | 5 megabit symmetrical for the price of a t1 |
| 02:53.19 | jql | yeah, had t1s go down, but not for long |
| 02:53.54 | jql | they went down for suspiciously about the time it takes to reboot an embedded OS... |
| 02:54.16 | orlok | i would not use wireless |
| 02:54.17 | asterboy | http://www.wimaxforum.org/home |
| 02:54.33 | orlok | at least when fibre/copper breaks theres a connection/break it can be traced to |
| 02:54.42 | orlok | wireless can just stop working, no explination, no recourse |
| 02:55.20 | Olobola | flock of birds? |
| 02:55.22 | jql | yep. phase-of-the-moon outages |
| 02:55.39 | orlok | nah |
| 02:55.45 | orlok | its called "shit wireess provider" |
| 02:55.52 | orlok | they are so dodgy |
| 02:55.53 | Olobola | oh, I see |
| 02:56.05 | orlok | earlier this year they were getting mass packet loss for several days |
| 02:56.10 | orlok | due to radar being tested |
| 02:56.24 | orlok | they are not using their own spectrum, etc so theres nothing they can do |
| 02:56.37 | orlok | they cant set up vlans either |
| 02:56.37 | xachen | heh |
| 02:56.45 | xachen | Wireless can work out good |
| 02:56.52 | orlok | theres posts to netflow support lists from their techs trying to set up flow accounting |
| 02:57.04 | orlok | "uh we cant seperate internal and external traffic" |
| 02:57.08 | *** join/#asterisk myconid (n=myconid@69-164-122-221.sbtnvt.adelphia.net) |
| 02:57.14 | myconid | Does asterisk implement hotdesking somehow? |
| 02:57.27 | xachen | provided you do the job right the first time, use good equip, and waso people aren't screaching out your signals |
| 02:57.29 | justinu|laptop | agents |
| 02:57.34 | xachen | s/waso/watch/ |
| 02:57.46 | justinu|laptop | myconid: agent logins |
| 02:57.58 | myconid | justinu|laptop: google asterisk agent logins ? |
| 02:58.03 | justinu|laptop | try voip-info |
| 02:58.06 | orlok | heh, this company is also now doing * stuff it seems |
| 02:58.38 | justinu|laptop | ~docs |
| 02:58.40 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:58.40 | myconid | justinu|laptop: so i would give all users 'virtual' extensions? |
| 02:58.50 | justinu|laptop | myconid: sorry, i honestly don't know all the details |
| 02:58.55 | myconid | kk |
| 02:58.58 | justinu|laptop | not my line of work :) |
| 02:59.08 | myconid | Im looking for some way my users can 'login' to a phone |
| 02:59.14 | myconid | similar to hotdesking nortel and inter-tel provide |
| 02:59.17 | justinu|laptop | yeah - that's what you want to reserach |
| 02:59.30 | techman97_andy | ugh. I'm back. The tcpdump didn't warrant any results apart from normal packet traffic (I ran it softphone conf and then GrandStream conf). |
| 02:59.40 | techman97_andy | man this is pissing me off! =P |
| 02:59.58 | techman97_andy | it has to be within the Grandstream |
| 03:00.02 | jql | if you have the full sip INVITE, you can see the codec list |
| 03:00.06 | justinu|laptop | myconid: http://www.voip-info.org/wiki-Asterisk+agents |
| 03:00.24 | justinu|laptop | techman97_andy: what is your issue? |
| 03:01.53 | techman97_andy | justinu: softphone - can do a g729 conference call from the phone OK. GrandStream cannot - I call 1st party, place on hold. call 2nd party, press CONF...press 1st line, the 1st line drops off the SIP appearance but the call does not end - it's still on hold and the caller hears the moh. The call is completely unretrievable from the Grandstream. |
| 03:02.06 | techman97_andy | I get a message on the CLI about "No compatible codecs" |
| 03:02.35 | myconid | justinu|laptop: it looks as though there is no way for a phone to initiate the hot desk, but it is done via a website, or a script..? |
| 03:03.04 | techman97_andy | jql: I saw the invites in the sdp - I'm only receiving g729 from my SIP PSTN gateway, so it has that in the invite list (with a crapload others, but they don't work...) |
| 03:03.17 | justinu|laptop | myconid: as I understand, the agent can dial a certain extension on a phone, and enter a userid/password. that associates the physical phone device with that "agent" |
| 03:03.27 | [TK]D-Fender | myconid : A million ways to skin this cat. Are you talking about general hot-desking or in specific relation to call queues? |
| 03:03.40 | justinu|laptop | dude, dejavu |
| 03:03.55 | [TK]D-Fender | justinu|laptop : All over again, for the first time! |
| 03:04.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-33-163.red.bezeqint.net) |
| 03:04.12 | justinu|laptop | myconid: you're in good hands ,d00d |
| 03:04.14 | myconid | [TK]D-Fender: I have 'clinicians' who come in and all they do is sit at a phone and talk with suicidal clients.. these clinicians are employees, so they need their own extension and voicemail.. I want them to be able to walk into the generic 'clinician' office and grab any phone that is free. |
| 03:04.20 | [TK]D-Fender | justinu|laptop : :D |
| 03:04.27 | myconid | and convert an offline phone into extension xxx... |
| 03:04.33 | orlok | myconid: ooh, whats the number? |
| 03:04.35 | myconid | and when they are done their shift.. 'logout' |
| 03:04.40 | justinu|laptop | techman97_andy: you have a capture of the "no compatible codecs"? |
| 03:04.53 | techman97_andy | depends on what kind of capture you want? |
| 03:04.54 | orlok | i reckon i'd be great in that job |
| 03:05.01 | asterboy | techman97_andy, have you upgraded that doorstop to the latest firmware? |
| 03:05.01 | orlok | cut down on call costs so much |
| 03:05.05 | myconid | orlok: got a masters in psychology? |
| 03:05.07 | orlok | try get them down to one call each |
| 03:05.07 | [TK]D-Fender | myconid : ok, so far so good. How do they receive calls. Direct by their extension #? |
| 03:05.29 | *** join/#asterisk MacDome (n=eseidel@64-121-195-96.c3-0.sfpo-ubr4.sfrn-sfpo.ca.cable.rcn.com) |
| 03:05.35 | myconid | [TK]D-Fender: all calls hit a switchboard.. they can press 1 for an extension, or the operator might transfer them. |
| 03:05.36 | justinu|laptop | techman97_andy: you talked about tcpdump |
| 03:05.38 | techman97_andy | asterboy: rofl, yea - talked w/ support and they even gave me a beta release of FW |
| 03:05.46 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:05.56 | asterboy | k |
| 03:06.05 | myconid | [TK]D-Fender: but from my desk, I wantto be able to try and call someone who doesnt have a 'home' phone.. but might be 'hot desked' in anywhere at one of 6 sites |
| 03:06.19 | myconid | if they arent hotdesked (logged in) anywhere.. i should get their voicemail |
| 03:06.33 | techman97_andy | correct. I have the tcpdump and the sdp, but the only difference between the dumps (other than the obvious) is the CLI's message about incompatible codecs |
| 03:06.38 | asterboy | just curious what your 'show codecs' has |
| 03:06.42 | techman97_andy | and it only happens on the doorstop..er...Grandstream |
| 03:06.45 | asterboy | most likely the defaults |
| 03:07.02 | myconid | [TK]D-Fender: you do contract work? <g> |
| 03:07.03 | techman97_andy | show codecs is defaults except for the 10 channels of G729 I purchased from Digium |
| 03:07.19 | asterboy | I know the GXP has G729 |
| 03:07.29 | [TK]D-Fender | myconid : Ok, then yeah you can do what you're looking for. |
| 03:07.32 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 03:07.37 | [TK]D-Fender | myconid : And yes I do actually |
| 03:07.38 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:07.53 | asterboy | you don't even need the G729 from digium unless your using FXS modules |
| 03:08.03 | techman97_andy | I tried a test: Put the phone completely in GSM (all preferences) and tried the conference call. It did the same thing, but popped up two new errors on the CLI - Invalid GSM Data |
| 03:08.07 | [TK]D-Fender | myconid : Only challenging part is notifying the phone of voicemail for the logged in user. |
| 03:08.21 | techman97_andy | I have FXS modules - fax machines and two analog channels for another business |
| 03:08.34 | myconid | [TK]D-Fender: another thing to throw at you.. ideally I would have 5 seperate asterisk servers.. one for each site.. connected over IP |
| 03:08.34 | asterboy | I should try that as a pickup line at a bar..."Hey baby...does your GXP do G729?" |
| 03:08.37 | [TK]D-Fender | myconid : Being able to pick it up, sure. Getting an e-mail SAYING there is a VM, sure..... |
| 03:08.40 | techman97_andy | rofl |
| 03:08.48 | myconid | and i want a user from site a to be able to hotdesk to site b |
| 03:08.53 | myconid | w/o using bandwidth over the wan |
| 03:08.55 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 03:09.00 | myconid | unless its from site a to site b |
| 03:09.02 | [TK]D-Fender | myconid : could work, challenging, but I think doable |
| 03:09.18 | myconid | ie, a call from site b to caller from site a, but hotdesked to site b would stay internal |
| 03:09.28 | myconid | caller = callee |
| 03:10.18 | asterboy | techman97_andy, do you have another brand of phone to try, so you can verify where the problem lies? |
| 03:10.41 | [TK]D-Fender | myconid : Yeah actually Its doable and I can see how I'd do it. Thing is all users would have to be synchronized on all servers |
| 03:10.49 | myconid | [TK]D-Fender: i really want to go with a major vendor (ie, Nortel, Inter-tel) but their products dont seem to really give us much more than what we have now. |
| 03:10.50 | techman97_andy | unfortunately, I only have softphones to play with, and they work with the conferencing. |
| 03:11.01 | myconid | [TK]D-Fender: im looking at $100k upgrade costs for all my sites.. with no *real* benefits |
| 03:11.09 | [TK]D-Fender | myconid : Yeah, they paint you into a cookie-cutter corner.. |
| 03:11.18 | myconid | I cant do hotdesking w/ nortel unless there is a home phone. |
| 03:11.29 | myconid | as it seems as though the hotdesking is a phone to phone feature |
| 03:11.36 | [TK]D-Fender | myconid : how many lines & what kind? (per site), how many users per site? |
| 03:11.45 | myconid | mind if I pm you? |
| 03:11.51 | [TK]D-Fender | myconid : np |
| 03:11.55 | file | [TK]D-Fender: what's this hot desking he keeps talking of... translated into my terms |
| 03:12.16 | Qwell | file: dynamic desk allocations? |
| 03:12.28 | Qwell | ie; it don't matter where you are |
| 03:12.29 | [TK]D-Fender | file : phones != extensions. person at the desk tells * what ext they are at and thats what the phone becomes. |
| 03:12.33 | file | oh |
| 03:12.36 | [TK]D-Fender | Qwell : Thats pretty much it |
| 03:12.41 | file | that's actually not hard |
| 03:12.53 | [TK]D-Fender | file : more challenging when its multi-site roaming <- |
| 03:13.05 | Qwell | dundi! :D |
| 03:13.05 | [TK]D-Fender | file : which is a point on his list |
| 03:13.06 | znoG | myconid: i was following the needs of your setup :( |
| 03:13.11 | znoG | myconid: now you've gone private... ;) |
| 03:13.32 | myconid | i dont want to share my line data :) |
| 03:13.33 | myconid | with the world |
| 03:13.56 | znoG | bah! :) |
| 03:14.14 | file | actually I have an idea for an app that would make it really easy if used with IAX2 switching or DUNDi without caching enabled |
| 03:14.36 | znoG | the idea of hotdesking is pretty cool, too bad the Asterisk setup i manage at the university is not entirely Asterisk based.. there is a Lucent in the middle |
| 03:14.58 | Strom_C | dumb question: what is hotdesking? |
| 03:15.10 | znoG | its going from site a to site b with the phone |
| 03:15.13 | myconid | hot desking is the idea that a user may not be at their assigned phone. |
| 03:15.32 | myconid | joe from aaccounting might be working up in the service desk for the day |
| 03:15.32 | znoG | and somebody calling the same extension reaches the person no matter where they are physically |
| 03:15.36 | file | actually we talked about a better way to do this at VON... but uh |
| 03:15.36 | myconid | and doesnt want to carry his phone |
| 03:15.39 | znoG | (as long as the phone is connected, of course) |
| 03:16.00 | file | but here it goes folks: |
| 03:16.09 | znoG | myconid: how do you hotdesk with SIP phones? unless you can enter your user ID/pass at the phone |
| 03:16.32 | myconid | thats my question indeed. |
| 03:16.33 | znoG | myconid: i thought you meant taking the SIP phone to site B |
| 03:16.45 | file | you can make variables be set in the dialplan in a sip.conf user or peer entry, that will tell you the technology and peer name to reach the phone |
| 03:17.08 | file | could also do extension of the phone |
| 03:17.29 | file | extension forwarding rather... thing |
| 03:17.40 | file | I'm going through this in my head and slightly rambling |
| 03:17.58 | Qwell | this is also easily done with queues, heh |
| 03:18.07 | file | that's overkill |
| 03:18.11 | Qwell | of course it is |
| 03:19.54 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 03:20.28 | znoG | guys, question unrelated to this... say Asterisk receives a call on one of its Zap channels (TDM400 card), and it INVITEs an ATA .. and the caller ID info has some bogus info in the "name" part (the caller ID number is ok, but the caller ID name is garbled characters).. who is at fault? the card? asterisk? the telco? |
| 03:20.58 | Strom_C | znoG: hook up an analog phone to the POTS line and find out |
| 03:21.27 | znoG | analog phone with caller ID... right? |
| 03:21.33 | Strom_C | obviously |
| 03:21.38 | Strom_C | unless you can decode FSK by ear |
| 03:21.39 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:21.42 | znoG | :) |
| 03:21.57 | file | basically you'd need some elaborate method of determing where an extension is... using local extension check, or doing a DUNDi lookup... and if those fail then you'd need to know where to send it to go to voicemail... |
| 03:22.17 | file | mmm I have cool ideas |
| 03:23.02 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:23.03 | znoG | Strom_C: actually I have a caller ID device thingy, might use that to find out |
| 03:23.17 | Strom_C | znoG: sure, whatever works |
| 03:24.38 | file | okay - imagine a context that used pattern matching and started at priority 2, when an extension is active then an extension with priority 1 and application noop is created in that context... it gets executed, moves to the pattern matching and grabs the phone's technology and information to call... |
| 03:24.49 | file | and that context is accessible over DUNDi |
| 03:25.27 | asterboy | I have the craziest callID when I forward my Rogers cell phone. The CallID is valid looking, but the data looks randomly picked from some database. |
| 03:26.41 | file | I seem to be rambling to myself |
| 03:26.45 | asterboy | I don't subscribe to call ID on the cell phone, so it has to be some strangeness coming from the VOIP provider |
| 03:26.49 | Strom_C | file: also dongs |
| 03:27.41 | file | so how about that stuff eh? |
| 03:27.45 | Strom_C | yeah |
| 03:27.46 | Strom_C | that stuff |
| 03:27.49 | Strom_C | it's stufftastic |
| 03:28.03 | file | I bet |
| 03:28.34 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 03:29.54 | *** join/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 03:30.35 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
| 03:33.34 | *** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
| 03:40.44 | Olobola | d |
| 03:42.54 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:44.31 | *** part/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
| 03:45.05 | *** join/#asterisk ManiacLappy (n=ManiacLa@unaffiliated/ManiacMac) |
| 03:46.14 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 03:51.33 | *** join/#asterisk bmg505 (n=leon@c1-82-12.rndf.isadsl.co.za) |
| 03:51.50 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 03:53.12 | *** part/#asterisk selsky (n=selsky@dookie.cc.columbia.edu) |
| 03:55.50 | catch23 | can i use the asterisk agi to operate any of the asterisk applications? |
| 03:56.52 | mog_home | exec? |
| 03:57.24 | catch23 | oh, duh |
| 03:57.37 | mog_home | some people driver their whole dialplan in agi |
| 03:57.55 | catch23 | i want to drive meetme, not sure how possible that is though |
| 03:58.23 | catch23 | you think it's possible to call 2 numbers, and have those 2 put into a conference? |
| 03:58.40 | catch23 | meetme works the other way around, where the user has to initiate the calling |
| 03:58.41 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-185-148.lsanca.fios.verizon.net) |
| 03:58.59 | Strom_C | catch23: call files |
| 04:00.01 | catch23 | well lets see... at the moment i'm using ragi to handle that stuff... i guess I could still redirect the call to a meetme conference then? |
| 04:02.17 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:02.39 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:05.26 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 04:12.16 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 04:12.40 | iceyp | Hey guys, how can I get asterisk to call 2 parts and bridge the call? |
| 04:12.56 | file | pick up phone, dial number, talk |
| 04:13.05 | denon | iceyp: google for spool files |
| 04:13.09 | file | I'm so evil |
| 04:13.14 | denon | or look at call.sample or whatever it's called |
| 04:13.17 | iceyp | I'd like to text my pabx, it would then generate a /var/spool/asterisk/outbound file call the first party and say, "Pleasae hold for barry" then it will call me |
| 04:13.40 | denon | iceyp: that's exactly how you'd do it .. have the file drop them into a context |
| 04:13.44 | denon | and have the context play a file, and dial |
| 04:13.49 | denon | assuming your name is barry |
| 04:13.57 | iceyp | it's the context im stuck with |
| 04:13.57 | denon | if your name isnt barry, it's all together different |
| 04:13.59 | iceyp | or |
| 04:14.06 | iceyp | actually i could drop the user in a meetme, then call myself |
| 04:14.13 | denon | exten => s,1,play(hold-for-barry) |
| 04:14.15 | xachen | sec |
| 04:14.18 | xachen | woops |
| 04:14.19 | denon | exten => s,2,meetme() |
| 04:14.35 | denon | or exten => s,2,dial(sip/barry) |
| 04:14.51 | iceyp | but i dont want it to call me if it didnt reach the first number |
| 04:15.05 | denon | it wont do the second leg until the first happens |
| 04:15.20 | denon | you have two sides of a call file, a context and a channel |
| 04:15.29 | denon | as mem serves, the channel happens first, then the context |
| 04:15.39 | file | yup |
| 04:15.57 | denon | file's idea would work too, though |
| 04:16.11 | iceyp | ok let me test this |
| 04:16.13 | iceyp | thnx |
| 04:16.16 | denon | sure |
| 04:16.21 | denon | just send your paypals to .... |
| 04:16.25 | denon | :) |
| 04:17.21 | catch23 | so i wonder, do you guys have a bunch of cell phones to test your apps with? i'm always finding it tough to test stuff with just one phone |
| 04:18.17 | file | I don't do much of stuff that requires multiple phones thankfully |
| 04:19.50 | denon | catch23: nah, 6-line cisco phone works fine |
| 04:20.02 | jql | I have an office full of phones... |
| 04:20.10 | denon | only use a cell to test trunks from an unassociated network |
| 04:20.50 | denon | his laptop and his pda |
| 04:20.55 | file | my PBX, and then development... works well :D |
| 04:21.03 | [TK]D-Fender | I have 2 phones registered to 1 server which is in place connected to EVERYWHERE :D |
| 04:22.38 | file | [TK]D-Fender: crazyness |
| 04:23.07 | catch23 | i mean of course you could use asterisk and make a bunch of softphones... but that would defeat testing... |
| 04:23.18 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 04:23.38 | jql | certainly works for testing plans... |
| 04:25.04 | MrDigital | who here knows how to program Asterisk and everything |
| 04:25.17 | file | erm? |
| 04:25.24 | file | everything is very... big |
| 04:25.24 | catch23 | i haven't touched their c code yet |
| 04:25.37 | MrDigital | who here wants a couple of bucks to help me? |
| 04:26.01 | catch23 | how much is a couple? |
| 04:26.16 | MrDigital | catch23: PM me |
| 04:26.17 | file | more then 1 I would assume |
| 04:26.35 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 04:26.37 | catch23 | hopefully more than 2 |
| 04:26.43 | file | that might be pushing it |
| 04:27.02 | MrDigital | file wanna help too? |
| 04:27.08 | file | no. |
| 04:27.26 | jql | a couple in bangalore? warsaw? perth? oslo? pittsburgh? seattle? san francisco? it matters. :)P |
| 04:27.55 | MrDigital | ???? |
| 04:29.09 | catch23 | I guess euro would be worth the most... |
| 04:29.09 | [TK]D-Fender | MrDigital : PM |
| 04:29.43 | file | give the starving people the bucks! |
| 04:30.23 | jql | rent-an-*-coder |
| 04:33.49 | dlynes_ | catch23: he was offering $150USD earlier...might be more now |
| 04:34.38 | jql | ahh, moscow |
| 04:35.48 | catch23 | dlynes, 150 huh? sheesh... |
| 04:36.38 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:36.40 | dlynes_ | catch23: well, it's more than 2, ain't it? |
| 04:38.40 | Manipura | So where to I get Bison-Devel 1.0.x |
| 04:39.14 | [TK]D-Fender | Ok, I'm fried... later all |
| 04:39.56 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 04:40.00 | kernel20 | hi |
| 04:40.16 | kernel20 | would it be possible to connect to skype using my asterisk box? |
| 04:42.11 | hypa7ia | kernel20: no |
| 04:42.30 | russellb | hypa7ia: i see you! |
| 04:42.35 | hypa7ia | russellb: :p |
| 04:42.45 | hypa7ia | russellb: school -11 days, yay! |
| 04:42.51 | russellb | nice!!!! |
| 04:42.58 | russellb | i have my list final of the semester on saturday |
| 04:43.05 | russellb | and then, insanity |
| 04:43.17 | hypa7ia | what kind of insanity? |
| 04:43.39 | russellb | the kind where you run in circles and scream |
| 04:43.50 | hypa7ia | eeks! |
| 04:46.46 | `Sauron | Oops. |
| 04:46.49 | file | eeeeeeeep |
| 04:46.57 | russellb | soooooooooooo hypa7ia . |
| 04:46.59 | russellb | WHAT IS UP |
| 04:47.39 | hypa7ia | namuch |
| 04:47.45 | hypa7ia | i'm going to ottawa this weekend |
| 04:47.50 | russellb | ooh |
| 04:47.52 | file | OLS? |
| 04:47.56 | russellb | to see the parents? |
| 04:47.59 | *** join/#asterisk KaBewM (n=DA-MAN@pulse.landshark.net) |
| 04:48.02 | hypa7ia | file: is that this weekend?!? |
| 04:48.02 | file | or am I wayyyyy off on dates |
| 04:48.19 | russellb | file knows nothing! |
| 04:48.21 | file | nah not till July |
| 04:48.24 | hypa7ia | file: just a couple of months :) |
| 04:48.30 | file | pfft, I was close |
| 04:48.48 | hypa7ia | my parents are in australia for 3 weeks so i have to go check on my little bro and sis |
| 04:48.59 | hypa7ia | and by little i mean 16 and 20 |
| 04:49.03 | file | ha |
| 04:49.19 | russellb | hypa7ia: you know they're causing trouble |
| 04:49.36 | dlynes_ | kernel20: using the skype libraries, yeah |
| 04:49.48 | dlynes_ | kernel20: but you need to have your skype client running on the same machine |
| 04:50.15 | dlynes_ | kernel20: well, and you'd have to write your own skype channel driver to handle it |
| 04:50.27 | file | yes... I know nothing |
| 04:50.29 | russellb | heh, like that's such a trivial thing to do |
| 04:50.49 | dlynes_ | russellb: yes, yes it is ;) |
| 04:51.14 | russellb | yeah, i've already written like ... 15 of tme |
| 04:51.18 | russellb | them. |
| 04:51.37 | Manipura | So where to I get Bison-Devel 1.0.x |
| 04:51.42 | file | russellb: don't admit it, then they'll go to you for help |
| 04:51.47 | dlynes_ | Manipura: ftp.gnu.org |
| 04:51.52 | russellb | file: ee[ |
| 04:52.00 | russellb | i can't type! |
| 04:52.06 | file | russellb: I HAVE 1 WAY AUDIOZ, PLS HELP!!! |
| 04:52.18 | russellb | file: you're such t3h n00b |
| 04:52.37 | Manipura | I can't find devel packages there dlynes_ |
| 04:52.42 | russellb | hypa7ia is l33t, tho! |
| 04:52.48 | russellb | talk to hypa7ia ! |
| 04:53.19 | kernel20 | is there any available libraries for skype? |
| 04:53.24 | hypa7ia | i'll be 1337 after this summer |
| 04:53.30 | hypa7ia | with my ZOMG JAVA courses |
| 04:53.33 | kernel20 | my only aim is to connect to skype user account and make calls |
| 04:53.36 | russellb | you already are! |
| 04:53.45 | tecnico | anyone running on gentoo ?? could I see your init.d/asterisk script if you have one ? |
| 04:53.54 | russellb | hypa7ia: you're on my super l33t list |
| 04:53.57 | file | hypa7ia: be careful... that stuff will rot your brains |
| 04:54.04 | kernel20 | u can install asterisk from source |
| 04:54.13 | kernel20 | no need to worry of the damn gentoo |
| 04:54.34 | kernel20 | anybody here have tried connecting to skype? |
| 04:54.39 | tecnico | kernel20: I did install it from source.. but I want the startup script so that asterisk starts when the machine is rebooted |
| 04:54.42 | kernel20 | asterisk->skype |
| 04:54.51 | *** join/#asterisk aasmith (i=andy@68-171-39-75.vnnyca.adelphia.net) |
| 04:54.55 | kernel20 | asterisk -vvvc |
| 04:54.57 | kernel20 | will do |
| 04:55.11 | tecnico | kernel20: and what if I'm not there when it reboots ?? |
| 04:55.13 | hypa7ia | file: i know, i will be very careful |
| 04:55.22 | hypa7ia | russellb: <3 |
| 04:55.30 | file | night night all! |
| 04:55.31 | aasmith | as there doesnt seem to be a sip channel here, i hope asking here is ok. Does anyone know of a good SIP client for win32? |
| 04:55.33 | kernel20 | place it in ur damn startup scrupt |
| 04:55.34 | russellb | hypa7ia: ;) :D |
| 04:55.46 | tecnico | kernel20: I have the asterisk -iT command in the boot script but it's not calling it... |
| 04:56.05 | tecnico | kernel20: I mean.. it's not starting asterisk.. the script runs.. |
| 04:56.28 | Manipura | dlynes_ I must be stupid because in the bison folder, there are no devel packages |
| 04:57.06 | kernel20 | tecnico: /usr/sbin/asterisk -vvvgc |
| 04:57.25 | *** part/#asterisk aasmith (i=andy@68-171-39-75.vnnyca.adelphia.net) |
| 04:57.34 | kernel20 | tecnico: http://www.voip-info.org/wiki-Asterisk+Starting+and+Stopping |
| 04:57.49 | kernel20 | anybody here have ideas on asterisk to skype? |
| 04:58.12 | dlynes_ | Manipura: It's source code; you need to compile and install it |
| 04:58.44 | dlynes_ | Manipura: if you want a package, you'll need to state which distribution you're using so people can point you to a server |
| 04:58.55 | tecnico | kernel20: tnx. for the info.. it seems I'm not getting my point through though. it's not the asterisk command that I'm having problems with. |
| 04:59.44 | dlynes_ | tecnico: you want to run 'safe_asterisk' |
| 04:59.59 | dlynes_ | tecnico: if you followed default installation methods, it shoudl be in your /usr/sbin directory |
| 05:00.22 | Manipura | Linux version 2.4.20-021stab028.3.777-enterprise (st@current.eng.sw.ru) (gcc version 2.96 20000731 (Red Hat Linux 7.3 2.96-110)) #1 SMP Wed Feb 2 21:08:59 MSK 2005 |
| 05:00.25 | dlynes_ | tecnico: just throw it into one of your startup scripts (I just throw it in /etc/rc.d/rc.local, personally) |
| 05:00.44 | Manipura | I have no idea what that is... I was told this server was supposed to be FC2 |
| 05:00.49 | dlynes_ | Manipura: So you want an rpm for RHEL |
| 05:01.22 | dlynes_ | Manipura: erm...actually that's probably Centos |
| 05:01.37 | tecnico | dlynes_: tnx. .. I just wished I could see a working gentoo init script .. 'cause my problem is not the executable that I'm calling.. but more of the start() stop() functions... tnx. anyway to all |
| 05:01.51 | Manipura | Hey, now that helps.... |
| 05:02.01 | dlynes_ | tecnico: Doesn't gentoo come with a gnu autoconfigure script? |
| 05:02.18 | dlynes_ | Manipura: Yeah...I don't use centos though....lots of people on here do |
| 05:02.30 | dlynes_ | Manipura: you can try asking them where you can get an rpm |
| 05:03.00 | Manipura | Okay, but you know if "storage size of `zt_fops' isn't known" is related? |
| 05:03.07 | dlynes_ | tecnico: to start, run safe_asterisk |
| 05:03.09 | Manipura | when compiling zaptel |
| 05:03.29 | dlynes_ | tecnico: to shutdown, asterisk -rx "stop when convenient" |
| 05:03.56 | dlynes_ | Manipura: Do you have your kernel-dev package installed? |
| 05:04.15 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 05:04.26 | dlynes_ | wb, cp |
| 05:04.49 | CunningPike | Thanks, dlynes :) |
| 05:05.07 | kernel20 | dlynes: how about asterisk to pbx |
| 05:05.13 | kernel20 | dlynes: how about asterisk to skype |
| 05:05.15 | kernel20 | i mean |
| 05:05.21 | Manipura | nope, thans dlynes |
| 05:05.22 | CunningPike | Did you get your zttest problem sorted out? |
| 05:05.43 | dlynes_ | kernel20: the only way you're going to get that working, is to write your own channel driver for skype, kernel20 |
| 05:05.59 | dlynes_ | CunningPike: Not goign to try for about another 2 or 3 hours |
| 05:06.06 | kernel20 | hmmm |
| 05:06.08 | CunningPike | dlynes: k |
| 05:06.09 | kernel20 | how? |
| 05:06.11 | dlynes_ | CunningPike: Waiting for the evening traffic to die off, first |
| 05:06.18 | CunningPike | dlynes: I hear ya |
| 05:06.34 | dlynes_ | CunningPike: and, besides...a potential investor will probably be calling into it shortly |
| 05:06.40 | dlynes_ | CunningPike: i don't wanna fubar that :) |
| 05:06.52 | CunningPike | dlynes: No kidding |
| 05:06.54 | *** join/#asterisk achandra (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
| 05:07.28 | dlynes_ | CunningPike: actually...just remembered the acpi |
| 05:07.35 | dlynes_ | CunningPike: that's something i can fix now, possibly |
| 05:07.48 | CunningPike | Requires a reboot, no? |
| 05:07.56 | asterboy | yep |
| 05:08.00 | dlynes_ | CunningPike: not for a daemon process, if i've got one running |
| 05:08.46 | dlynes_ | ah...nvm...don't have one running, anyways |
| 05:09.33 | dlynes_ | ah...bleh |
| 05:09.39 | dlynes_ | had acpi compiled in, like a dummy :((( |
| 05:09.47 | kernel20 | dlynes: how? |
| 05:09.57 | dlynes_ | had almost all acpi features enabled, too |
| 05:09.57 | asterboy | you can add a grub parameter |
| 05:10.07 | dlynes_ | asterboy: you mean lilo :) |
| 05:10.31 | dlynes_ | kernel20: with your eight fingers, a good old fashioned text editor, a compiler, and a linker, and a little bit of luck |
| 05:10.41 | dlynes_ | kernel20: plus about ten years or so of C programming experience |
| 05:10.42 | CunningPike | dlynes: Either way, Reboot City |
| 05:11.12 | asterboy | lilo ignored the settings |
| 05:11.23 | dlynes_ | I'm not going to use lilo settings, anyways |
| 05:11.23 | asterboy | in grub I put acpi=off or noacpi |
| 05:11.26 | dlynes_ | I'll just recompile the kernel |
| 05:11.42 | dlynes_ | Why use a kludge, if you don't have to? |
| 05:11.56 | asterboy | yep |
| 05:12.59 | dlynes_ | Manipura: i guess you dont' have kernel-dev installed, eh? |
| 05:14.28 | dlynes_ | CunningPike: Yeah, interrupts have to be the problem |
| 05:14.43 | dlynes_ | CunningPike: As soon as I started doing a kernel compile, I started getting a lot of those errors |
| 05:15.03 | CunningPike | dlynes: There you go - at least you know what it is |
| 05:17.07 | dlynes_ | CunningPike: I'm guessing this log would mean an interrupt problem, right? : http://pastebin.com/699466 |
| 05:19.36 | CunningPike | dlynes: That would be my first bet - especially with zttest numbers as low as 93% |
| 05:20.04 | dlynes_ | CunningPike: yeah, so i'm guessing getting rid of all that acpi crap will probably go a good way in solving that issue, then |
| 05:20.40 | dlynes_ | but it seems ridiculous that it would be that susceptible to interrupt loss |
| 05:20.40 | CunningPike | dlynes: I would think so - although acpi didn't make an enormous difference to us, I know others have had significant issues with it |
| 05:20.59 | dlynes_ | there's no way one single card should be generating that many interrupts |
| 05:21.01 | CunningPike | dlynes: The Digium cards are very fussy about getting interrupts at 1000/sec |
| 05:21.15 | dlynes_ | i'd hate to see what that card's going to be like when i'm using all four spans |
| 05:21.16 | CunningPike | Sangoma cards are much less so, apparently |
| 05:21.22 | CunningPike | :D |
| 05:21.26 | CunningPike | Like ours |
| 05:21.37 | dlynes_ | You're only using one span right now though, right? |
| 05:21.38 | *** join/#asterisk mgob (n=goldenol@c-24-17-242-105.hsd1.wa.comcast.net) |
| 05:21.43 | mgob | hi |
| 05:21.58 | CunningPike | Although, I think the number of active spans makes no difference - it needs 1000 interrupts/sec regardless |
| 05:22.03 | dlynes_ | ah |
| 05:22.06 | CunningPike | We have 3 active spans |
| 05:22.18 | mog_home | yes that is true |
| 05:22.18 | dlynes_ | so you still have one more span to add yet? |
| 05:22.19 | CunningPike | Two to the Nortel and one to Allstream |
| 05:22.25 | mgob | 911 question... if I have 10 lines all combined over 3 buildings which have different addresses, and I want the PSAP to know if the call comes from a particular phone it is one of those three addresses.... can I set this somehow? Can I pass any info to the PSAP? |
| 05:22.30 | CunningPike | No - they don't do a 3-span card ;) |
| 05:22.31 | dlynes_ | mog_home: ah...thanks for the confirmation |
| 05:22.41 | dlynes_ | CunningPike: heh |
| 05:22.45 | mgob | I was thinking when someone called 911 it could outpulse the local FAX line number to the PSAP (same for all three sites) and they could determine the caller based upon this |
| 05:23.00 | CunningPike | mgob: Where are you? |
| 05:23.16 | mgob | well, this is in Missouri |
| 05:23.19 | mgob | rural Missouri |
| 05:23.39 | mog_home | but if you have a zttest score as low as 93% you probably have other problems |
| 05:23.53 | dlynes_ | mog_home: only one of the checks was that low |
| 05:24.08 | mgob | I guess I am just trying to just figure out how to give the PSAP the right address |
| 05:24.09 | dlynes_ | mog_home: and one was at 98.something percent...the rest were all 99.something or 100 |
| 05:24.23 | mog_home | a good card / good machine is 99.something |
| 05:25.05 | dlynes_ | so even that low on only one check indicates bigger problems? even if it doesn't occur on more tests? |
| 05:25.10 | mog_home | digium support might be able to help get your scores up |
| 05:25.19 | mog_home | and audio quality up |
| 05:25.21 | dlynes_ | mog_home: ah...ok |
| 05:25.21 | CunningPike | mgob: All my experience is from Vancouver, BC, Canada - we have a 1B at each location with an ATA on it. We route the calls to the appropriate ATA based on the original caller's location |
| 05:25.22 | mog_home | which card is this? |
| 05:25.29 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 05:25.35 | dlynes_ | mog_home: I'm not sure if it's a te405p, or a te410p |
| 05:25.54 | mog_home | ahh you can do hardware hdlc too which helps on bad machines |
| 05:26.10 | mog_home | you running X , serial or dma on your drives? |
| 05:26.20 | mgob | CunningPike: great idea, thank you |
| 05:26.25 | CunningPike | dlynes: dmesg will tell you |
| 05:26.37 | dlynes_ | It's a xilinx 0314, on a bus master slot, medium devsel, latency 64, irq 16, 32-bit [size=128] on a pci-x bus, if that means anything |
| 05:27.01 | mog_home | thats all not important |
| 05:27.08 | dlynes_ | mog_home: I wouldn't dream of running X on that machine |
| 05:27.14 | mog_home | good good |
| 05:27.18 | CunningPike | mgob: You're welcome. We used SPA-3000s, and have a red POTS phone attached, so they get PSTN dialtone in the event of a power failure |
| 05:27.20 | mog_home | what machine is it? |
| 05:27.20 | dlynes_ | mog_home: the serial devices are enabled, but not being used |
| 05:27.30 | dlynes_ | mog_home: Celeron 2GHz |
| 05:27.37 | mog_home | home built? |
| 05:27.47 | dlynes_ | Nah...MGC I think? |
| 05:27.58 | mog_home | mgc? |
| 05:28.01 | dlynes_ | It's a rack mount |
| 05:28.07 | dlynes_ | 2U |
| 05:28.23 | dlynes_ | Basically no-name brand |
| 05:28.27 | mog_home | hmm okies |
| 05:28.48 | mog_home | nother one for ya can the card pass patlooptest? |
| 05:29.03 | dlynes_ | mog_home: can i run that while asterisk is running? |
| 05:29.09 | mog_home | no |
| 05:29.17 | mog_home | well you can |
| 05:29.21 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:29.21 | mog_home | on an empty span |
| 05:29.31 | dlynes_ | i c |
| 05:30.31 | mgob | Thanks again CunningPike, good night! |
| 05:31.29 | dlynes_ | mog_home: ./patlooptest /dev/zap/1? |
| 05:32.10 | mog_home | and set the channels to clear for the span |
| 05:32.22 | dlynes_ | mog_home: how would I do that? |
| 05:32.35 | mog_home | vi /etc/zaptel.conf |
| 05:32.39 | mog_home | clear=1-24 |
| 05:33.08 | dlynes_ | I get a crapload of unexpected results |
| 05:33.37 | mog_home | hmm if youd like i could log on take a look |
| 05:33.40 | dlynes_ | doesn't matter /dev/zap/1, /dev/zap/2, ... |
| 05:33.47 | mog_home | but im about to head to bed |
| 05:33.57 | dlynes_ | mog_home: Ok, I'd need to get an ssh public key from you |
| 05:34.13 | mog_home | eep /me doesnt have one |
| 05:34.38 | dlynes_ | mog_home: ok, lemme temporarily disable them, then |
| 05:34.40 | mog_home | have been meening to make one |
| 05:34.55 | mog_home | but im sooo lazy when it comes to doing certain things like that or cleaning my apt |
| 05:35.06 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 05:36.01 | coppice | "FCC orders VoIP to allow snooping" - time for some serious work on SRTP :-) |
| 05:36.49 | Qwell | T.38 first :p |
| 05:39.40 | faberk | hi |
| 05:40.00 | faberk | I see patlooptest into zaptel folder |
| 05:40.07 | faberk | but is not compiled |
| 05:40.13 | mog_home | or use iax2 encryption coppice |
| 05:40.21 | faberk | only patlooptest.c is present |
| 05:40.31 | coppice | iax2 doesn't really have encryption |
| 05:40.55 | faberk | how must be compiled? |
| 05:42.40 | mog_home | umm yeah it does... |
| 05:43.52 | coppice | OK then, how does it work? last time I looked there was just a rather broken framework for encryption |
| 05:44.05 | catch23 | anyone using ztdummy on a 2.6 kernel? |
| 05:48.20 | mog_home | its pretty straight forward coppice, but im heading to bed, not that i wouldnt turn down srtp |
| 05:48.48 | catch23 | Nugget, how did you avoid the Invalid module format error? |
| 05:49.35 | catch23 | Nugget, i'm assuming you're installing it from the bristuff? |
| 05:50.59 | Nugget | no, you don't need bristuff for ztdummy. |
| 05:51.40 | catch23 | so you installed directly from zaptel stuff? make linux26? |
| 05:51.44 | Nugget | right |
| 05:51.52 | Nugget | I'm still on 1.2.6, though, not 1.2.7.x |
| 05:51.57 | Nugget | dunno if anything's changed |
| 05:52.47 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 05:54.01 | catch23 | Nugget, hm, i'm still getting an invalid module format, are you using ehci? |
| 05:55.16 | Nugget | I don't know what ehci is. |
| 05:55.32 | catch23 | the usb 2.0 driver |
| 05:55.36 | Nugget | I doubt it |
| 05:55.39 | catch23 | not sure it it actually matters |
| 05:55.41 | Nugget | I have no idea, though. |
| 05:55.59 | catch23 | which 2.6 kernel you using? |
| 05:56.08 | Nugget | whatever shipped with centos 4 |
| 05:56.48 | Nugget | 2.6.9-34.ELsmp |
| 05:58.15 | catch23 | hm, well i'm using 15, so i guess something must've changed |
| 05:59.04 | Nugget | I'm not a fan of Linux, so I tend to be pretty conservative when I have to use it. |
| 05:59.18 | Nugget | you're helping to reinforce my position. :) |
| 05:59.38 | catch23 | hehe, yeah these things can be annoying... |
| 05:59.49 | catch23 | sucks to stay on the bleeding edge |
| 05:59.56 | Nugget | ever since they stopped doing the whole "even numbered kernel" "odd numbered kernel" it's been a real pain in the ass for me |
| 06:00.42 | dlynes_ | catch23: make sure the kernel and module versions match, and make sure that the compiler version that compiled the kernel is also the one compiling your zaptel drivers |
| 06:00.56 | catch23 | oh they match alright.... |
| 06:01.16 | dlynes_ | catch23: then you shouldn't get the invalid module format error |
| 06:01.19 | catch23 | i looked at the make log to make sure it was going into the right directory for the files |
| 06:02.54 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:07.08 | catch23 | i am using the archck patchset though, so i'll try a vanilla kernel i guess |
| 06:15.49 | kernel20 | i have use sox to convert wav to gsm but the problem is it is so destorted it can't even been understood |
| 06:15.52 | kernel20 | any ideas? |
| 06:16.03 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:16.16 | kernel20 | i use sox mywav.wav mywav.gsm |
| 06:16.16 | dlynes_ | kernel20: for music on hold? |
| 06:16.16 | kernel20 | any ideas? |
| 06:16.24 | littleball | hello, anybody from Germany? |
| 06:16.40 | kernel20 | Playback |
| 06:16.57 | dlynes_ | kernel20: did you convert it to 8K? |
| 06:17.07 | kernel20 | 8k how? |
| 06:17.07 | dlynes_ | kernel20: 8K mono, that is? |
| 06:17.15 | dlynes_ | kernel20: that's why it sounds like crap then |
| 06:17.16 | kernel20 | sox mywav.wav mywav.gsm |
| 06:17.20 | kernel20 | just that command |
| 06:17.33 | kernel20 | how to convert that in 8k? |
| 06:17.35 | dlynes_ | kernel20: downsample your wav file to 8K under windows or whatever you're using first |
| 06:17.41 | dlynes_ | kernel20: then convert it to gsm |
| 06:17.54 | kernel20 | ic ok wait |
| 06:17.55 | kernel20 | thanks |
| 06:18.07 | dlynes_ | kernel20: audacity.sourceforge.net has a program you can download for windows or linux to do that |
| 06:19.14 | kernel20 | i use coolspeech for my attendant |
| 06:19.23 | kernel20 | no 8k only 8x |
| 06:19.31 | dlynes_ | huh? |
| 06:19.36 | dlynes_ | no idea wtf 8x is |
| 06:19.49 | kernel20 | familiar with coolspeech? |
| 06:19.57 | dlynes_ | nope...never heard of it, or used it |
| 06:20.18 | catch23 | sox $1 -r 8000 -c 1 (basename $1 .wav).gsm resample -ql |
| 06:20.22 | catch23 | there's your sox script |
| 06:20.49 | dlynes_ | catch23: yeah, but if you downsample the wav file first, before conversion, it sounds a bit better |
| 06:21.05 | dlynes_ | catch23: sox isn't terribly good at downsampling |
| 06:21.29 | catch23 | bah, good enough... over a gsm cell phone, they all sound bad |
| 06:21.39 | dlynes_ | heh |
| 06:22.03 | catch23 | i used that for lots of my sound files, over a cell phone you can't tell the diff between a sound file and a real human. so i'd say it's the same |
| 06:23.03 | kernel20 | sox $1 -r 8000 -c 1 (basename $1 .wav).gsm resample -ql <- still the same |
| 06:23.18 | catch23 | same as what |
| 06:23.32 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:28.43 | kernel20 | crappled |
| 06:30.15 | catch23 | dunno, that's what i've been using for all my voice/music file conversions, sounds pretty good to me |
| 06:30.40 | CunningPike | kernel20: Me too..... |
| 06:31.00 | CunningPike | What's the format of the original file? |
| 06:32.04 | catch23 | maybe you're using a low outbound format? like if you're using lpc10 it will sound like crap no matter what |
| 06:32.12 | *** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com) |
| 06:32.42 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:32.43 | catch23 | I use GSM and it's fine... but most people probably use the g.711 |
| 06:33.16 | kernel20 | how would i make my wav file not be crap when it becomes gsm file? |
| 06:33.39 | catch23 | how do you know it's crap? |
| 06:33.45 | dlynes_ | kernel20: Try this instead: sox $1 -r 8000 -c 1 `basename $1 .wav`.gsm resample -ql |
| 06:34.07 | dlynes_ | kernel20: your shell might not support the (...) format |
| 06:34.14 | kernel20 | k wait |
| 06:34.34 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 06:35.16 | dlynes_ | kernel20: also, what files do you have in that directory with the same basename, but different extensions? |
| 06:35.19 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:35.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:35.35 | CunningPike | dlynes: Good point |
| 06:35.38 | kernel20 | sox: Can't open input file '.gsm': No such file or directory |
| 06:35.54 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-15.claranet.co.uk) |
| 06:35.54 | catch23 | oh hah... you can't just paste that line into your terminal window :P |
| 06:36.02 | kernel20 | no i dont |
| 06:36.04 | catch23 | no wonder nothing has been working |
| 06:36.05 | dlynes_ | kernel20: dood |
| 06:36.15 | kernel20 | sox -r 8000 -c 1 `miscncs $1 .wav`.gsm resample -ql |
| 06:37.05 | dlynes_ | kernel20: try sox miscncs.wav -r 8000 -c 1 miscncs.gsm resample -ql |
| 06:37.46 | CunningPike | kernel20: Is this music or voice? |
| 06:37.54 | CunningPike | Music always sounds crap over gsm |
| 06:38.06 | CunningPike | Half the frequency range is missing |
| 06:38.32 | catch23 | bah, most music is crap anyway, so can't tell |
| 06:38.59 | kernel20 | its works but a little distorted |
| 06:39.11 | kernel20 | any ideas on how to adjust it to more finer? |
| 06:39.11 | CunningPike | Anyway, I'm off |
| 06:39.21 | CunningPike | Night all |
| 06:39.42 | dlynes_ | g'night, cp |
| 06:40.10 | dlynes_ | kernel20: record the wav file at a higher quality, and use audacity to downsample it |
| 06:40.50 | kernel20 | how would i do downsample in audicity? |
| 06:41.00 | dlynes_ | Download it, read the documentation, and find out |
| 06:41.12 | dlynes_ | Or try asking in #audacity, if there is such a channel |
| 06:41.33 | dlynes_ | the program is pretty straightforward, though |
| 06:42.23 | dlynes_ | kernel20: yeah..there is a #audacity channel |
| 06:43.07 | dlynes_ | holy fleeping kernel takes forever to compile |
| 06:43.09 | catch23 | damn, i found my kernel module problem... had the kernel directory compiled for a different 2.6.15 build than my current running one :-/ |
| 06:43.31 | dlynes_ | catch23: i told ya, but ya wouldn't listen...no.... *smirk* |
| 06:43.50 | catch23 | dlynes, well i *was* running the same version..... |
| 06:44.06 | catch23 | but it was compiled with different options, for a different architecture even |
| 06:44.28 | dlynes_ | catch23: exactly, so it's a different version as far as linux is concerned |
| 06:45.11 | dlynes_ | catch23: one thing you can do, which is what i do, is give every kernel build a slightly different suffix to differentiate what platform it was built for |
| 06:45.40 | dlynes_ | catch23: i've got a general low latency kernel, a general non-low latency kernel, a low latency kernel optimized for via, and a non-low latency kernel optimized for via |
| 06:46.04 | catch23 | huh? all for 1 computer? |
| 06:46.05 | dlynes_ | catch23: so i call them 2.6.15.5LL, 2.6.15.5NLL, 2.6.15.5VIA-LL, and 2.6.15.5VIA-NLL |
| 06:46.15 | dlynes_ | catch23: nah...i deploy across multiple computers |
| 06:46.24 | dlynes_ | catch23: so i just compile them once, on one machine |
| 06:46.29 | catch23 | oh ok... yeah i should probably do the same |
| 06:46.31 | dlynes_ | catch23: package them up in slackware packages |
| 06:46.37 | dlynes_ | catch23: and then just deploy |
| 06:46.51 | catch23 | well yeah, i use debian, but same deal. has the make-kpkg mechanism |
| 06:47.12 | dlynes_ | catch23: anothing thing you might want to consider to, is putting hte .config file for each kernel in the appropriate boot directory |
| 06:47.25 | dlynes_ | catch23: that way you only have to distribute one copy of the kernel source |
| 06:47.32 | catch23 | oh make-kpkg does that for me automatically already :) |
| 06:47.35 | dlynes_ | catch23: and then just do make config on a clean source tree |
| 06:47.54 | catch23 | make-kpkg includes the whole deal, including initrd image if i specified one |
| 06:48.01 | dlynes_ | ah..ok |
| 06:48.07 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 06:48.38 | catch23 | make-kpkg created a .deb file for the kernel... so you can give it to someone else to dpkg -i |
| 06:50.25 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 06:52.47 | *** join/#asterisk BugKham (n=BugKham@125.24.14.10) |
| 06:57.01 | kernel20 | question |
| 06:57.10 | kernel20 | i am a newbie as u have observed |
| 06:57.25 | kernel20 | now i want to have a groups in my pbx |
| 06:58.26 | kernel20 | say when somebody dials 800 and attendant would say for park press 1, for tea press 2, for shoes press 3 to disconnect press 4 |
| 06:58.32 | kernel20 | what should be the dial plan? |
| 06:58.41 | catch23 | you should use the AGI interface |
| 06:58.45 | catch23 | super easy |
| 06:58.58 | kernel20 | i've been reading around but dont find any concrete example |
| 06:59.02 | kernel20 | catch23? |
| 06:59.10 | catch23 | just use some prepackaged stuff from some language |
| 06:59.16 | kernel20 | ? |
| 06:59.23 | catch23 | http://www.voip-info.org/wiki-Asterisk+AGI |
| 06:59.37 | kernel20 | what agi? |
| 06:59.39 | catch23 | scroll down and see all the language implementations for fastagi |
| 06:59.43 | kernel20 | all i need is a dial plan |
| 06:59.49 | catch23 | asterisk gateway interface |
| 06:59.53 | *** join/#asterisk qdk_ (n=qdk@213.237.44.34) |
| 06:59.58 | catch23 | yeah agi solves all your dialplan needs |
| 07:00.02 | kernel20 | ? |
| 07:00.41 | *** join/#asterisk apardo (n=apardo@87.217.147.105) |
| 07:00.42 | kernel20 | no not agi |
| 07:00.50 | kernel20 | only dial plan |
| 07:01.37 | catch23 | you must as well use agi for the dialplan, i hate manipulating the extensions file to do complex things |
| 07:01.53 | kernel20 | i dont have much time for agi |
| 07:01.55 | kernel20 | anybody>? |
| 07:02.31 | catch23 | well what do you have time for? |
| 07:03.29 | kernel20 | dont ask me to do ur way |
| 07:03.38 | kernel20 | i appreciate ur help |
| 07:03.44 | dlynes_ | kernel20: you want the background() dialplan command, and then branch based on what dtmf tones the user sent |
| 07:03.45 | kernel20 | but it doesnt really helps |
| 07:03.56 | kernel20 | yeah |
| 07:04.02 | kernel20 | something like that dlynes |
| 07:04.07 | kernel20 | ur always my hero |
| 07:04.08 | kernel20 | heeheh |
| 07:04.18 | dlynes_ | kernel20: yeah, but sometimes you're annoying :) |
| 07:04.23 | kernel20 | heeheh |
| 07:04.25 | kernel20 | sorry |
| 07:04.26 | dlynes_ | kernel20: Today you've actually been pretty good |
| 07:04.37 | kernel20 | i've learned my lessons |
| 07:04.57 | kernel20 | what should be the dial plan dlynes? |
| 07:05.14 | dlynes_ | kernel20: Well, you're annoying because you seem to want everyone to give you exact examples instead of reading the documentation and figuring it out yourself |
| 07:05.23 | dlynes_ | kernel20: kinda like what you did just now |
| 07:05.40 | kernel20 | i have made couple of configs |
| 07:05.45 | kernel20 | but dont really get it |
| 07:05.57 | kernel20 | i am just a newbie please do understand |
| 07:05.58 | catch23 | why not just use something like asterisk@home? i'm sure they got builtin dial plan config tools and stuff :P |
| 07:06.05 | dlynes_ | kernel20: read up on the documentation for the Background() dialplan command |
| 07:06.06 | kernel20 | mybe u will shed light |
| 07:06.14 | dlynes_ | kernel20: the documentation has plenty of examples |
| 07:06.26 | dlynes_ | catch23: please don't get people to do that |
| 07:06.35 | catch23 | dlynes, why not? |
| 07:06.39 | dlynes_ | catch23: then they come in here, wanting us to solve their problems iwth freepbx config files |
| 07:06.54 | catch23 | dlynes, ah heh |
| 07:07.10 | kernel20 | actually catch23 doesnt really helps |
| 07:07.13 | catch23 | I don't actually know anything in asterisk@home, i figured they'd have a GUI tool or something to do that |
| 07:07.31 | dlynes_ | catch23: i'd rather help the newbs with asterisk config files...then eventually they can figure out stuff themselves, once they get comfortable with it |
| 07:07.46 | catch23 | kernel20, sorry for not helping, i'll try to refrain from speaking to you in the future. |
| 07:09.46 | dlynes_ | kernel20: Is your native language tagalog? |
| 07:10.53 | dlynes_ | holy flipping cow |
| 07:11.02 | dlynes_ | i can't believe the kernel hasn't finished compiling yet |
| 07:12.29 | catch23 | time to buy a faster machine |
| 07:13.01 | catch23 | i compiled 6 versions of it since i joined this channel a few hours ago |
| 07:13.36 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:16.33 | dlynes_ | i think maybe it was compiling a bunch of shit i told it not to |
| 07:16.38 | dlynes_ | probably forgot to do make oldconfig |
| 07:16.47 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:20.12 | dlynes_ | Does anyone know if Enabling kernel IRQ balancing is a 'bad thing' when it comes to Asterisk? |
| 07:26.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.40.150) |
| 07:27.56 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 07:28.50 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:29.13 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:30.24 | Druken | uhg.... |
| 07:30.38 | Druken | 3am sucks ass |
| 07:31.35 | catch23 | only in gmt -5! |
| 07:31.41 | dlynes_ | 12:30am sucks pretty harsh ass, too |
| 07:32.08 | dlynes_ | Druken: you know if enabling irq balancing in the kernel will negatively impact a pri card or not? |
| 07:32.29 | Druken | i guess... but going to sleep at 11.... and back up at 2:30 really sucks |
| 07:33.08 | dlynes_ | ah...someone's phone line went down? |
| 07:33.18 | Druken | i can't see why it would... but digium cards do NOT like to share.... |
| 07:33.42 | Druken | dlynes, no... just my sleeping pattern lately |
| 07:33.43 | dlynes_ | Yeah, i've noticed that |
| 07:33.57 | dlynes_ | Got a 93% on one of the tests from zttest |
| 07:35.34 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:35.37 | kernel20 | no dlynes |
| 07:35.38 | *** join/#asterisk Juggie (i=user@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 07:35.40 | Druken | can't say i've ever used zttest |
| 07:36.54 | dlynes_ | kernel20: no, what? |
| 07:37.53 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:38.10 | dlynes_ | man, all those interrupts that the zaptel card is taking up is making my compile process take super ass long |
| 07:38.52 | snitt | only 1000 interrupts per sec |
| 07:38.52 | tainted- | kernel20 nugget@slacker.com |
| 07:38.58 | snitt | for each zaptel card |
| 07:39.21 | dlynes_ | only? |
| 07:39.28 | Druken | tainted-: your just tryin to piss him off aren't ya ? |
| 07:39.40 | tainted- | lol |
| 07:39.41 | snitt | i think it is not too much |
| 07:39.46 | dlynes_ | tainted-: it's not obvious? |
| 07:39.50 | dlynes_ | erm |
| 07:39.55 | dlynes_ | Druken: it's not obvious? |
| 07:39.58 | snitt | even with ztd_eth sending sync packets to other asterisk boxes over ethernet |
| 07:40.04 | Druken | :) |
| 07:40.32 | dlynes_ | snitt: why would you have two timing sources? |
| 07:40.44 | snitt | dlynes_: it is not two timing sources |
| 07:40.50 | Druken | multipul cards? |
| 07:40.53 | Druken | hehe |
| 07:40.57 | snitt | the one with the zaptel card provides timing source via ethernet to the other box |
| 07:41.02 | dlynes_ | snitt: oh...you're using ztd_eth instead of ztdummy or wct4xxp, ... |
| 07:41.21 | snitt | yes, sort of |
| 07:42.11 | dlynes_ | snitt: so how does the ztd_eth box get the timing from the zaptel box? does it send it over a certain tcp port? |
| 07:42.17 | Druken | i just toss in a noname x100p for timing.... |
| 07:42.28 | snitt | dlynes_: it is layer2 protocol |
| 07:42.31 | Druken | ztdummy is a useless pos in my opinion |
| 07:42.46 | dlynes_ | snitt: i can't remember what the different layers are |
| 07:42.56 | snitt | dlynes_: so no ip addresses and ports here |
| 07:42.56 | dlynes_ | Druken: works fine for me |
| 07:43.00 | snitt | just mac addresses |
| 07:43.17 | dlynes_ | snitt: doesn't there have to be something on the zaptel box to pair up with ztd_eth though? |
| 07:43.33 | dlynes_ | I've never used ztd_eth, so I don't know what's involved with it |
| 07:43.47 | snitt | well i have asterisk running on both boxes |
| 07:43.59 | *** join/#asterisk RalfX (n=ralf@cpe-66-27-196-236.socal.res.rr.com) |
| 07:44.00 | snitt | but i guess a running asterisk is not a must have in this situation |
| 07:44.51 | dlynes_ | and then? |
| 07:45.03 | snitt | and then what? |
| 07:45.14 | dlynes_ | how does the ztd_eth time with the other box? |
| 07:45.26 | snitt | i dont know really |
| 07:45.33 | snitt | it sends a small packet in each ms |
| 07:45.35 | dlynes_ | it doesnt' actually do anything with asterisk...i'm sure of that |
| 07:45.58 | dlynes_ | drivers can't rely on applications to be running |
| 07:46.04 | dlynes_ | that would just be ludicrous |
| 07:46.08 | snitt | sure |
| 07:46.44 | RalfX | Added Incoming Route in Asterisk at home 2.8, made the mistake of doing the DID: Zap/2 instead of Zap2. Now I can't delete the Zap/2 one. I guess the / character is the problem. How can I get rid of it? I haven't found the file it is in. I suspect it may be in MySQL? |
| 07:46.57 | dlynes_ | RalfX: type /topic |
| 07:47.26 | Druken | ~amp |
| 07:47.28 | jbot | amp is probably "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 07:47.28 | RalfX | thnks. |
| 07:47.49 | dlynes_ | the reason freepbx/@home/amp isn't supported here is because they have a funky config file setup that nobody here wants to try to debug |
| 07:47.57 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 07:47.58 | dlynes_ | it's hard enough with a normal setup |
| 07:48.02 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 07:48.26 | dlynes_ | ~aah |
| 07:48.28 | jbot | [aah] AMP and amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp) |
| 07:48.37 | *** part/#asterisk RalfX (n=ralf@cpe-66-27-196-236.socal.res.rr.com) |
| 07:48.40 | dlynes_ | lol |
| 07:48.46 | dlynes_ | ~@home |
| 07:48.47 | jbot | @home is probably the most pathetic Internet Service Provider on planet Earth. Downtime hourly ... can't play one OLG without getting disco'd, they're just horrible. If you're contemplating using it, don't bother., or see home-scanning, or get portsentry NOW and set it to stealth mode. also see home-dhcp |
| 07:48.57 | dlynes_ | cool beans |
| 07:48.57 | snitt | ~*@~ |
| 07:48.58 | jbot | methinks *@~ is "Asterisk at Home" as found at http://asteriskathome.sourceforge.net/ |
| 07:49.20 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:49.23 | dlynes_ | I like the @home one better :) |
| 07:49.31 | dlynes_ | My cable provider used to be part of @home |
| 07:49.35 | dlynes_ | They're absolutely horrible |
| 07:49.40 | snitt | lol |
| 07:49.51 | dlynes_ | Thank god my cable provider's not using them any longer |
| 07:53.09 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 07:53.23 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:53.33 | hwt | No such command 'dial'. |
| 07:53.42 | hwt | which module is dial in? |
| 07:53.51 | hwt | (manual dialing from the asterisk console. |
| 07:53.53 | hwt | ) |
| 07:54.07 | dlynes_ | hwt: app_dial.so? |
| 07:54.26 | dlynes_ | oh...nvm |
| 07:54.34 | dlynes_ | that's for the dialplan |
| 07:56.34 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 07:57.54 | dlynes_ | hwt: i don't think asterisk has such a cli command |
| 08:00.06 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 08:02.12 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 08:04.09 | hwt | dlynes_: uhm, it has, okay. i can use it on another *-box that has autoload=yes. |
| 08:04.13 | hwt | but i don't want that. |
| 08:04.41 | dlynes_ | really? |
| 08:04.51 | dlynes_ | oh |
| 08:04.55 | dlynes_ | nvm....just remembered |
| 08:05.04 | dlynes_ | you need to load chan_alsa, or chan_oss |
| 08:05.12 | dlynes_ | but not both |
| 08:07.38 | hwt | ah, okay. |
| 08:08.41 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 08:11.53 | dlynes_ | ah...kernel's finally compiled :) |
| 08:12.21 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 08:12.33 | dlynes_ | now gotta wait forever for the damned modules :(( |
| 08:12.43 | dlynes_ | thankfully i only asked for ethernet modules :) |
| 08:13.57 | hwt | hm, doesn't work now either. |
| 08:19.49 | *** join/#asterisk tzafrir_home (n=tzafrir@station-131.gadot.org.il) |
| 08:20.16 | tzafrir_home | anybody here tried connecting * to ISDN(BRI) in Israel? |
| 08:21.21 | dlynes_ | hwt: do you have your alsa or oss drivers loaded and working under linux properly? |
| 08:23.51 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:29.56 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 08:32.22 | *** join/#asterisk Delvar (n=irc@217.14.137.33) |
| 08:33.23 | *** join/#asterisk motu (n=motu@192.165.166.190) |
| 08:37.18 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:38.10 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:38.47 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:52.53 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 08:53.55 | *** join/#asterisk Delvar (n=irc@217.14.137.33) |
| 08:56.40 | qdk_ | how do you guys carry info about a call over an IAX trunk? I need go check the status of call presentaion on "the other side". |
| 08:59.51 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:59.56 | *** join/#asterisk HatemZ (n=hzaghlou@213.120.90.67) |
| 09:00.14 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 09:04.42 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 09:05.10 | *** join/#asterisk apardo (n=apardo@87.217.146.80) |
| 09:08.04 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:13.23 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 09:19.42 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 09:25.49 | *** join/#asterisk gmaruz1 (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 09:29.55 | nfi|ermes | hi all |
| 09:30.29 | nfi|ermes | can i use bristuff-0.2.0-RC8r with asterisk 1.2.0 ? |
| 09:34.15 | *** join/#asterisk pashah (n=pashah@85-114-4-82.obit.ru) |
| 09:34.24 | pashah | hello everybody |
| 09:34.59 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 09:36.34 | pashah | having problems with dundi, all is setup as per voip-info.org I can do a dundi lookup, etc. but get 'No such host' when trying to dial, any pointers? |
| 09:36.44 | *** join/#asterisk bigmac2004 (n=Hype@62.76.244.194) |
| 09:36.53 | bigmac2004 | hi |
| 09:36.55 | bigmac2004 | there |
| 09:37.06 | bigmac2004 | hello |
| 09:40.35 | snitt | pashah: voip-info like setup works for me |
| 09:41.21 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 09:47.10 | bigmac2004 | hi |
| 09:50.55 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 09:51.37 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-53-dsl.end.estpak.ee) |
| 09:52.01 | shiznatix | hello. with WaitExten how can I make it stop 'waiting' when the user presses the # key? |
| 09:52.47 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:53.05 | MrChimpy | oooh. i want a button on my phone that gets me tea. |
| 09:54.49 | *** join/#asterisk zwelch (n=kumquat@pdpc/supporter/sustaining/zwelch) |
| 09:56.26 | bigmac2004 | any active chatters |
| 09:57.14 | *** join/#asterisk fulgas (n=fulgas@209.8.233.175) |
| 10:01.14 | zwelch | so, if i have a couple of broken sip channels, it looks like 'soft hangup' should do what i want to clean them up; however, i can't figure out what to pass as the channel identifier. |
| 10:06.43 | bigmac2004 | anyone knows how to connect 2 asterisk over WAN |
| 10:06.46 | bigmac2004 | please help |
| 10:07.39 | hwt | does digium/sangoma produce cards that let me connect ISDN-phones directly to Asterisk? ISDN "fxs" ports. |
| 10:10.49 | MrChimpy | bigmac: connect is rather non specific. for chucking calls between them i'd use IAX |
| 10:15.34 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 10:16.03 | stoffell | bigmac2004, look on voip-info, connecting 2 asterisk boxes |
| 10:16.27 | bigmac2004 | thanks |
| 10:16.29 | bigmac2004 | a lot |
| 10:16.57 | bigmac2004 | i read about iax |
| 10:17.03 | hwt | isn't it as easy to just do it with SIP? |
| 10:17.21 | bigmac2004 | but my situation is i have 2 building having asterisk server on each |
| 10:17.28 | hwt | if both of the boxes are online without NAT, there is no reason to use IAX imho. |
| 10:17.47 | bigmac2004 | i would like to connect it together but i dont want to lease T1 lines as it is expensive |
| 10:17.51 | bigmac2004 | is there any alternative |
| 10:18.04 | bigmac2004 | could it be done by analog phone lines |
| 10:18.18 | bigmac2004 | since over here local phone call are free |
| 10:18.20 | pashah | cheers |
| 10:18.23 | bigmac2004 | please advise |
| 10:18.47 | hwt | bigmac2004: you can do SIP over the net. |
| 10:18.52 | hwt | bigmac2004: a SIP trunk. |
| 10:18.56 | hwt | bigmac2004: or IAX trunk |
| 10:19.26 | bigmac2004 | i dont want to use internet ...because i have to pay for the banthwith..and its expensive....is there any other alternative |
| 10:19.45 | hwt | bigmac2004: okay, you said WAN. |
| 10:19.55 | bigmac2004 | i said wan |
| 10:20.10 | bigmac2004 | sorry |
| 10:20.22 | bigmac2004 | said wan...but not sure how to set up wan |
| 10:20.45 | bigmac2004 | is it possibe to set up wan through dsl modem |
| 10:21.02 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 10:21.15 | bigmac2004 | it will be used opnly for asterisk server |
| 10:21.23 | bigmac2004 | no data transfer or internet is required |
| 10:21.39 | bigmac2004 | hwt: please suggest alternative |
| 10:21.44 | Altair256 | bigmac2004, everyone else would just use IAX (data) over the interenet |
| 10:21.56 | Altair256 | if you are convinced you have to use analog lines for this (since they are free) |
| 10:22.15 | HatemZ | I've a strange sort of question... Is it possible to find out what a digium card is actually dialing? |
| 10:22.17 | Altair256 | put some FXO cards, or FXO bank cards into your box and pay for multiple phone lines |
| 10:22.47 | Altair256 | HatemZ, watching the Asterisk CLI might show you what you need |
| 10:23.02 | HatemZ | I want to see what happens after asterisk tells the card "dial this number" |
| 10:23.17 | HatemZ | I want to know if the card is putting "number" or "prefix+number" on the line |
| 10:23.24 | Altair256 | is this a regular analong line, sip, channel bank, etc? |
| 10:23.30 | HatemZ | analog |
| 10:23.44 | Altair256 | put a splitter on the outside and attach a regular phone |
| 10:23.46 | bigmac2004 | the main problem over here is that the internet is expensive...routing the traffic over iax or VPN will cost alot |
| 10:23.53 | Altair256 | pick up the phone first and listen to what it dials |
| 10:23.53 | HatemZ | can't do that... am in a different country |
| 10:24.08 | HatemZ | and, the guys don't wake up for a few hours |
| 10:32.41 | Altair256 | HatemZ, you could try the log files at /var/log/asterisk/full |
| 10:34.48 | nfi|ermes | can i use atxfer and blindxfer in asterisk 1.2.7.1 ? |
| 10:42.12 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 10:42.40 | *** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
| 10:42.58 | pycsusz | Hi Everybody! |
| 10:43.06 | pycsusz | I need some help |
| 10:43.23 | pycsusz | How can I activate the features.conf??? |
| 10:43.54 | pycsusz | Somebody can help me? |
| 10:44.49 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:45.37 | *** join/#asterisk kay2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 10:45.42 | kay2 | hi |
| 10:46.12 | pycsusz | hi! |
| 10:46.48 | kay2 | Witch what can I convert a .wav to .gsm ? |
| 10:47.05 | pycsusz | ??? |
| 10:47.31 | pycsusz | do you want to convert wav to gsm? |
| 10:47.39 | kay2 | yeah |
| 10:47.43 | kay2 | i guess i have to use sox |
| 10:47.46 | pycsusz | with sox |
| 10:47.47 | pycsusz | ohh |
| 10:47.55 | pycsusz | 1 moment |
| 10:48.36 | pycsusz | I give you a bash script what convert all of wav file in the actual directory to gsm |
| 10:50.36 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 10:50.41 | pycsusz | #!/bin/bash |
| 10:50.41 | pycsusz | # Convert wav2gsm |
| 10:50.41 | pycsusz | for i in *.wav |
| 10:50.41 | pycsusz | do |
| 10:50.42 | pycsusz | <PROTECTED> |
| 10:50.42 | pycsusz | <PROTECTED> |
| 10:50.44 | pycsusz | done |
| 10:51.10 | pycsusz | you have to type it to a file and give permit to runable |
| 10:51.55 | *** part/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
| 10:52.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 10:54.09 | stoffell | kay2, there's a sample on http://astrecipes.net also (Script to convert music-on-hold to native formats) |
| 10:58.37 | Zeeek | poof |
| 11:02.59 | *** join/#asterisk Delvar (n=irc@217.14.137.33) |
| 11:03.29 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 11:04.04 | boddy | hi all I need one port digium E1 card which model I have buy |
| 11:04.13 | boddy | te110P |
| 11:04.14 | boddy | ? |
| 11:04.54 | andrebarbosa | if you only need one, i think it's the best option |
| 11:04.55 | andrebarbosa | :) |
| 11:05.17 | andrebarbosa | check pci specifications for that card, and see it your MB support then ;) |
| 11:06.31 | boddy | ? |
| 11:10.30 | *** part/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 11:14.09 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 11:16.35 | MRH2 | where is the best place to put minor non technical sugestion( re: meetme join announcement behaviour) |
| 11:18.08 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 11:18.24 | MRH2 | basically you should hear your own join announcement instead of dead air |
| 11:19.07 | *** join/#asterisk Formater (i=Formater@dhcp-87-116-138-55.cmtsns-ns.customer.sbb.co.yu) |
| 11:20.20 | Formater | hi. I have some problems, maybe someone can help. I start a call with X-lite, when I hangup, X-lite sends CANCEL, but Asterisk doesn't forwards it to termination provider. Any idea?\ |
| 11:22.44 | Formater | I monitor the traffic, x-lite send cancel, but asterisk doesn't send it further to termination providers. nor on sip channel, nor on h323... |
| 11:25.37 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 11:28.51 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 11:29.12 | shiznatix | how can I reset the priority number that I am on like if I do 999,1,... then 999,2,... but then I want the priority to go back to 1 and continue on in that context, how do I do that? |
| 11:29.43 | dlynes | shiznatix: Goto(1) |
| 11:30.16 | *** join/#asterisk snowchyld (i=snowchyl@koifish.org) |
| 11:31.01 | snowchyld | Has anyone had experience with timezones + GMT vs BST and voicemail.conf [zonemessages] ? |
| 11:31.06 | dlynes | MRH2: You can try suggesting it on the asterisk-development mailing list, or in #asterisk-dev |
| 11:31.14 | shiznatix | dlynes, thanks |
| 11:31.50 | dlynes | Formater: so the channel never gets hung up? |
| 11:32.09 | Formater | dlynes: it hangs after timeout |
| 11:32.30 | dlynes | Formater: but the channel doesn't hang up when the caller hangs up? |
| 11:32.44 | Formater | dlyynes: yep, it keeps ringing |
| 11:33.30 | dlynes | Formater: So you see x-lite sending the cancel when you do a sip debug? |
| 11:33.43 | Formater | so I execute Dial... the called party starts ringing, I (the caller) press hangup, but the called party keeps ringing, till timeout occures, then I get back control from Dial.. |
| 11:34.03 | Formater | yep, X-lite sends Cancel.. and keeps sending it many time |
| 11:34.14 | Formater | but I tried with Sipura too, and the same thing. |
| 11:34.42 | dlynes | Formater: I've heard there's a sensitivity option or something for sip and iax |
| 11:34.49 | dlynes | Formater: that might effect it |
| 11:35.15 | Formater | and what is that option?:) |
| 11:35.37 | Zeeek | sensitivity=strawberries |
| 11:35.56 | dlynes | I have no idea |
| 11:36.01 | MRH2 | ok thanx |
| 11:36.08 | dlynes | Somebody on here was mentioning it earlier this evening |
| 11:36.17 | dlynes | but they couldn't remember what the exact option was |
| 11:36.23 | dlynes | they just remembered coming across it before |
| 11:36.39 | dlynes | they seemed to indicate it was a channel-agnostic setting |
| 11:37.06 | dlynes | it was some kinda setting to prevent zombies from happening |
| 11:37.25 | Formater | tnx |
| 11:37.29 | dlynes | He mentioned he seen it on the asterisk-users mailing list recently |
| 11:37.49 | dlynes | try doing a search for the keyword 'zombie' on the asterisk-users mailing list |
| 11:37.50 | snowchyld | hmm |
| 11:37.56 | Formater | tnx |
| 11:37.56 | snowchyld | asterisk doesn't follow DST does it ? |
| 11:38.29 | Sonderblade | anyone know how to change the No key entry setting of eHOP8T phones? |
| 11:39.12 | dlynes | snowchyld: where would it care whether it's DST or not? |
| 11:39.37 | snowchyld | the voicemail |
| 11:39.46 | snowchyld | received at 11 39 am |
| 11:39.50 | snowchyld | (thats UTC) |
| 11:39.57 | dlynes | snowchyld: voicemail tells you the time zone? |
| 11:40.06 | snowchyld | time now BST (summer time uk) is 12 39 |
| 11:40.10 | snowchyld | UTC+1 |
| 11:40.19 | russellb | snowchyld: you set the timezone in voicemail.conf for each user |
| 11:40.26 | snowchyld | the timezone is set |
| 11:40.29 | russellb | well, it's a global option as well |
| 11:40.44 | snowchyld | [zonemessages] |
| 11:40.44 | snowchyld | gmt=Europe/London|'vm-received' q 'digits/at' HM |
| 11:41.13 | snowchyld | user@host:~# date |
| 11:41.13 | snowchyld | Fri May 5 12:41:03 BST 2006 |
| 11:41.19 | dlynes | thought you were going back to sleep, russ? :) |
| 11:41.28 | russellb | oh yeah :) |
| 11:41.43 | russellb | zzz ... |
| 11:41.55 | dlynes | you haven't been getting very much sleep lately :) |
| 11:54.01 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-220-183.telkomadsl.co.za) |
| 11:54.18 | X-Gen | hey freaks |
| 11:59.39 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-224-153.dsl.snantx.swbell.net) |
| 12:01.36 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 12:02.02 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 12:05.55 | Druken | morning peoples |
| 12:08.22 | DarthClue | Buenos días |
| 12:08.50 | Zeeek | bonjour |
| 12:09.30 | MrChimpy | bom dia |
| 12:09.33 | Druken | salut |
| 12:09.35 | Druken | :) |
| 12:09.38 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 12:09.45 | Zeeek | guten Tag |
| 12:09.56 | Zeeek | (and so's yer mom) |
| 12:10.32 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:10.32 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 12:11.10 | BugKham | hi there, anyone knows where to find documentations on setting up asterisk to recieve external sip calls |
| 12:11.29 | Druken | ~docs |
| 12:11.31 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:11.31 | SpaceBass | BugKham, www.archatechs.com www.nerdvittles.com |
| 12:11.43 | Zeeek | Starter tutorial: |
| 12:11.43 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:11.43 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 12:11.43 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:11.43 | Zeeek | THE reference of the moment: |
| 12:11.44 | Zeeek | http://www.asteriskdocs.org |
| 12:11.54 | SpaceBass | and of course the wiki ^^^^^^^^^^^^^^^^^ |
| 12:11.55 | BugKham | Zeeek: thanks |
| 12:12.07 | Zeeek | and the mailing lists |
| 12:12.16 | Zeeek | and the many local forums |
| 12:12.30 | SpaceBass | BugKham, and get freepbx |
| 12:12.32 | SpaceBass | :) |
| 12:12.36 | Zeeek | nah |
| 12:12.50 | BugKham | Zeeek: yeah, I could call my other boxes but needed fromuser=guest all the time |
| 12:13.27 | BugKham | SpaceBass: I've been running * for some time now |
| 12:13.41 | SpaceBass | gotcha |
| 12:13.56 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-60.cybersurf.com) |
| 12:14.43 | *** mode/#asterisk [-f] by russellb_ |
| 12:14.56 | BugKham | SpaceBass: do you need to put fromuser=guest when calling another * box? |
| 12:15.28 | SpaceBass | BugKham, no, you shouldn't need it |
| 12:15.41 | SpaceBass | but you may need a legit user on the other bo |
| 12:15.42 | SpaceBass | x |
| 12:15.43 | BugKham | SpaceBass: I noticed that when forwarding a call from providers like didx, ipkall, etc |
| 12:16.00 | SpaceBass | some providers use guest sip calls |
| 12:16.08 | SpaceBass | www.freedigits.com does that |
| 12:16.11 | BugKham | they could send callid in a "name" <number> format |
| 12:16.13 | SpaceBass | and its annoying |
| 12:16.35 | Zeeek | providers are like women. They all do annoying things sometime |
| 12:16.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:16.48 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:16.48 | BugKham | but I could not call my * box without fromuser=guest |
| 12:16.57 | Druken | sometimes? |
| 12:17.07 | Zeeek | yeah, only sometimes |
| 12:17.17 | puzzled | hi |
| 12:17.21 | Zeeek | BugKham so you have a user named guest |
| 12:17.25 | Druken | i'm gonna go for more often than not |
| 12:17.43 | Zeeek | Druken don't be bitter. You'll find the one that doesn't someday |
| 12:17.53 | SpaceBass | leme just say one thing.... sake + beer + wine + 2 martinis != good idea |
| 12:18.01 | Zeeek | anyonther case where they're like providers |
| 12:18.01 | SpaceBass | I'm paying for last night like you read about |
| 12:18.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:18.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:18.19 | SpaceBass | BugKham, you could setup a real user (exten) and pass calls that way |
| 12:18.29 | BugKham | Zeeek: so, the username can only be "guest"? |
| 12:18.45 | Zeeek | no I was asking if you had one? Are you talking iax or sip? |
| 12:18.57 | Druken | uhm, just allow guest |
| 12:19.13 | Druken | or autocreate peer |
| 12:19.15 | SpaceBass | guest is fine, but its hard to route more than one guest call |
| 12:19.26 | SpaceBass | at least I cannot figure it out |
| 12:19.28 | SpaceBass | :D |
| 12:19.31 | Zeeek | because many iax providers just have a user entry with a host address |
| 12:19.32 | BugKham | Zeeek: sip of course, I'm ok with iax |
| 12:19.42 | Zeeek | ok |
| 12:20.21 | BugKham | Zeeek: I have username=myname and fromuser=guest to have it working |
| 12:20.33 | Zeeek | type= ? |
| 12:20.59 | BugKham | and in sip.conf of another box I put allowguest=yes even it's a default |
| 12:21.15 | BugKham | Zeeek: type=peer |
| 12:21.25 | Zeeek | for incoming call? |
| 12:21.28 | BugKham | Zeeek: I use it to call out only |
| 12:21.41 | Zeeek | and on the other end? |
| 12:21.56 | BugKham | Zeeek: and insecure=very |
| 12:22.01 | [TK]D-Fender | BugKham : Every * PBX I'v managed I've set up so you can use the employees name + domain to call them |
| 12:22.26 | BugKham | [TK]D-Fender: yes, I could also |
| 12:22.44 | BugKham | [TK]D-Fender: but the caller requires fromuser=guest |
| 12:22.55 | [TK]D-Fender | BugKham : No need for "guest". You just need to set a context in [general] and set "allowguest=yes" and unauthed callers go there. So I have an exten => andrew,1,Goto(ringall,s,1) for instance |
| 12:23.10 | [TK]D-Fender | BugKham : Not at all considering the above |
| 12:23.19 | BugKham | [TK]D-Fender: no problem on the callee side, it goes to the right context |
| 12:23.50 | [TK]D-Fender | BugKham : All I can say as this method has never failed me.... or the numerous other systems I've worked on. |
| 12:24.38 | BugKham | Zeeek: on the other side I'm having allowguest=yes , context=from-sip-external |
| 12:24.57 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 12:25.07 | [TK]D-Fender | BugKham : And whats in that context? |
| 12:25.28 | BugKham | [TK]D-Fender: so, on the caller's setting you don't need fromuser=guest, do you? |
| 12:25.37 | [TK]D-Fender | BugKham : nope, not at all. |
| 12:25.42 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 12:26.02 | [TK]D-Fender | BugKham : ANY old call with do, including Dial(andrew@aocomputing.cjb.net) |
| 12:26.11 | [TK]D-Fender | From a softphone, for anywhere. |
| 12:26.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:26.32 | BugKham | [TK]D-Fender: on the context "from-sip-external", I have something like exten => _2XXXX,1,... |
| 12:26.37 | [TK]D-Fender | as long as their callerID does not match a user on my system its all good to go. |
| 12:27.03 | BugKham | [TK]D-Fender: and the call was passed to 23456@mydomail.com without a problem |
| 12:27.04 | [TK]D-Fender | Then someone should be able to call 2000@ipornameofyourserver |
| 12:27.28 | BugKham | [TK]D-Fender: yes |
| 12:27.34 | [TK]D-Fender | BugKham : 1 digit short there on mine. but you get the idea. |
| 12:27.40 | BugKham | [TK]D-Fender: anyone could call |
| 12:28.04 | [TK]D-Fender | that is the idea, no? |
| 12:28.13 | BugKham | [TK]D-Fender: but when I tried to call from another box I need to put fromuser=guest |
| 12:28.22 | BugKham | [TK]D-Fender: to have it working |
| 12:28.22 | *** join/#asterisk gezick (n=gezick@208.255.178.130) |
| 12:28.45 | BugKham | [TK]D-Fender: do you get what I'm trying to explain? |
| 12:29.26 | BugKham | [TK]D-Fender: the call was passed ok with sip |
| 12:29.31 | [TK]D-Fender | BugKham : From another box? Then what DOES work without "fromuser"? |
| 12:30.08 | Zeeek | BugKham if you want to get somewhere, pasetbin the two sip sections with Box A and box B ? |
| 12:30.18 | Zeeek | ~pb |
| 12:30.42 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 12:30.43 | Druken | i'm not quite getting what it matters what user it came FROM... asterisk shouldn't gve two shits.. no? |
| 12:30.45 | [TK]D-Fender | Druken : Nope, it shouldn't |
| 12:30.46 | Druken | it should only care about the user it's going TO ? |
| 12:31.04 | Zeeek | it's a pice of software. It doesn't have any cares at all |
| 12:31.11 | [TK]D-Fender | Druken : Actually it WILL fail if the callerid matchs a known user |
| 12:31.14 | *** part/#asterisk Cereal_Killer (n=none@gentoo/contributor/pdpc.active.cokehabit) |
| 12:31.34 | Druken | really.... |
| 12:31.43 | BugKham | [TK]D-Fender: hmm, is it possible for you to call me? |
| 12:32.35 | [TK]D-Fender | BugKham : @work, sorry |
| 12:33.19 | [TK]D-Fender | BugKham : In many hours when I'm home, sure |
| 12:34.25 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 12:34.55 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:36.29 | BugKham | [TK]D-Fender: that's fine |
| 12:36.33 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 12:36.42 | BugKham | [TK]D-Fender: once I removed fromuser=guest |
| 12:36.46 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 12:36.52 | BugKham | I'll have "Failed to authenticate user" error |
| 12:37.31 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 12:39.19 | BugKham | [TK]D-Fender: that is "chan_sip.c:10326 handle_request_invite: Failed to authenticate user" |
| 12:40.04 | Druken | are you reinviting the call from the middle asterisk to a phone? |
| 12:41.44 | coppice | Would Middle Asterisk be something to do with the Lord of the Rings? :-) |
| 12:41.51 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:42.08 | russellb | coppice: haha |
| 12:42.09 | BugKham | Zeeek: one box is using realtime |
| 12:42.31 | BugKham | Zeeek: the other one uses sip.conf |
| 12:43.08 | BugKham | Zeeek: it should not matter but I'm thinking how can I show my sip conf of the realtime box |
| 12:43.23 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 12:44.15 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 12:44.49 | tdonahue-laptop | morning all |
| 12:44.50 | MRH2 | can someone point me in the direction of the link with instructions on how to make a "full trace" |
| 12:45.50 | tdonahue-laptop | i have a quick question... is Digium's 729 implementation 729A or 729AB, our carrier is asking which 729 codec we want to test... |
| 12:46.34 | MrChimpy | http://youtube.com/watch?v=MKX0RN19zc0&search=yep%20yep%20muppets - my asterisk course |
| 12:47.47 | MRH2 | when someone wants a full trace is that just the output from debug being on? |
| 12:48.52 | tdonahue-laptop | MRH2, they may be looking for a Ethereal/tcpdump packet capture |
| 12:49.13 | BugKham | Zeeek: Ok, here it is. http://pastebin.com/699923 |
| 12:49.34 | tdonahue-laptop | tcpdump -s 1504 -w CallTrace.pcap -i <interface> host <ip address of remote host> |
| 12:49.49 | BugKham | I just selected var_name and var_val from the table |
| 12:49.57 | *** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net) |
| 12:50.32 | MRH2 | thanx |
| 12:50.43 | tdonahue-laptop | MRH2, if they are just looking for the SIP trace then sip debug output would probably be addequate. |
| 12:51.01 | brif8 | dlynes_: sorry I had to cut last night but I got called away and only got back in late |
| 12:51.28 | MRH2 | ok i'll do what i can |
| 12:51.33 | MRH2 | gtg |
| 12:52.03 | MRH2 | thanx |
| 12:52.14 | aphexddb | is there a cmd i can type in the console to call an extenetion, then forward it to another ext for testing? im googled out. |
| 12:52.42 | mut | good training video for n00bs mrbnet |
| 12:52.45 | mut | MrChimpy even |
| 12:54.41 | MrChimpy | aye. "Cat!" "No, TELEPHONE!" |
| 12:54.49 | MrChimpy | "brrrreeeeeeng!" |
| 12:54.54 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 12:54.55 | mut | for setting up call blocking |
| 12:55.13 | mut | thinking the easiest way would be a ignorance check against the astdb |
| 12:55.15 | mut | something like |
| 12:55.27 | mut | User/PhoneNumber/BlockNumber = 'yes' |
| 12:55.31 | mut | and check if it exists |
| 12:55.41 | mut | any other ideas? |
| 12:55.42 | *** join/#asterisk YVS (n=naoeda@copernico.inatel.br) |
| 12:55.58 | ManxPower | aphexddb, the CLI Dial command works if Asterisk saw OSS/ALSA headers and libs when you built Asterisk. |
| 12:56.44 | YVS | Does anybody know about Zaptel drivers (C code) ? |
| 12:57.46 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 12:57.53 | [TK]D-Fender | YVS : Ask the mailinglist for that like we suggested yesterday. With your luck the only people who could assist you are in a completely unfriendly time-zone and/or maybe hate IRC. |
| 12:58.03 | [TK]D-Fender | asterisk-dev maybe |
| 12:58.16 | YVS | kool thx |
| 12:58.43 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 12:59.01 | [TK]D-Fender | YVS : Sorry, not to rain on your parade, just those are the real odds here... |
| 12:59.24 | YVS | np |
| 13:00.18 | [TK]D-Fender | *sigh* |
| 13:00.41 | [TK]D-Fender | YVS : But like coppice commented it does sound more like you are making a switch than an interface card. |
| 13:01.20 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:02.19 | russellb | there are very few people that know the zaptel code ... |
| 13:02.43 | YVS | yes thats true |
| 13:02.46 | kay2 | If I want the user to press "*" to do something, how am I supposed to write that in the dialplan ??? |
| 13:03.09 | YVS | and all links on net leads to the same place/or disabled page. |
| 13:03.25 | YVS | I just joined the mailing list |
| 13:03.30 | YVS | hope that helps me |
| 13:04.36 | [TK]D-Fender | kay2 : exten => *,1,DoSomethingUsefulHere() |
| 13:05.24 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:05.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:05.25 | *** join/#asterisk Icemon (n=icechat5@dsl-165-152-08.telkomadsl.co.za) |
| 13:08.30 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 13:08.41 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 13:08.41 | kay2 | [t4]D-Fender: but let say that With my SIP phone I dial a number, then I get a background saying "to continue press *", then how can I know that the * has been presses or not? |
| 13:09.15 | kay2 | [TK]D-Fender |
| 13:09.35 | [TK]D-Fender | kay2 : just like I wrote. |
| 13:10.06 | [TK]D-Fender | kay : pastebin the non-function dialplan after giving it a few tries. |
| 13:10.07 | [TK]D-Fender | ~pb |
| 13:10.15 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:10.34 | kay2 | [TK]D-Fender]: Could it be a DTMF issue ? |
| 13:10.35 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 13:10.49 | kay2 | do I have to be in inband or RFC |
| 13:11.22 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:11.39 | qdk_ | kay2: does it take you many sec to test both? |
| 13:11.55 | [TK]D-Fender | kay2 : Do other extens work? |
| 13:12.15 | [TK]D-Fender | kay2 : depends on what the other side uses. |
| 13:12.44 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:12.44 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 13:13.01 | aphexddb | ManxPower: thx for the cli info |
| 13:13.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:14.34 | kay2 | [TK]D-Fender, yeah it was a DTMF issue |
| 13:15.02 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 13:18.59 | *** part/#asterisk HatemZ (n=hzaghlou@213.120.90.67) |
| 13:22.03 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:22.59 | Hmmhesays | so I got my external voicemail notify shiat working |
| 13:23.30 | Druken | outgoing voicemail notify? |
| 13:23.47 | Druken | geez... that would be annoying after a while..... |
| 13:24.49 | Hmmhesays | we'll i've got an asterisk/ser setup here |
| 13:24.52 | ManxPower | Druken, Much less annoying than your users complaining about not having it. |
| 13:25.23 | Druken | ManxPower: i've never had someone bitch... hehe |
| 13:26.09 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 13:26.33 | Druken | Hmmhesays: i wasen't thinking in that manner, i was thinking of asterisk putting a call out telling you about voicemail... |
| 13:29.01 | *** join/#asterisk ToyMan (n=stuq@74-32-62-42.dsl1.mdl.ny.frontiernet.net) |
| 13:30.14 | ManxPower | Druken, You must have users that are not morons. |
| 13:30.39 | nfi|ermes | i have asterisk 1.2.7.1 bristuffed (0.3.0-PRE-1o) with florz patch (zaphfc_0.3.0-PRE-1o_florz-12.diff.gz). I receive there messages: |
| 13:30.45 | ManxPower | One of my users actually said "I don't want to be notified of voicemail via Text Message, it's too complicated to read them." |
| 13:30.54 | nfi|ermes | zaphfc[0]: b channel buffer underrun: 0, 0 |
| 13:30.54 | nfi|ermes | zaphfc[0]: b channel buffer overflow: 55, 55 |
| 13:31.05 | *** join/#asterisk ghost99 (n=neville@222-153-96-3.jetstream.xtra.co.nz) |
| 13:31.13 | nfi|ermes | is it a problem ? |
| 13:31.47 | kay2 | If I do a Gosub(Authercontext|1|1) , in [Authercontext], can I put a Return to come back ? |
| 13:32.27 | kay2 | because right now, as soon as it sees the Return() it hangs up :( |
| 13:32.28 | Hmmhesays | Druken you could do that |
| 13:34.06 | Hmmhesays | too complicated to read? |
| 13:34.07 | Hmmhesays | WTF |
| 13:37.43 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 13:38.07 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:38.09 | qdk_ | kay2: looks to my like MACRO is whatyou wanna use. |
| 13:39.15 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 13:41.46 | brif8 | someone care to enlighten me http://pastebin.com/699990 what is this saying and what is causing this ? |
| 13:42.46 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.142.Dial1.SanJose1.Level3.net) |
| 13:44.21 | Hmmhesays | external notify doesn't pass the voicemail box as an arguement? argh |
| 13:45.38 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 13:47.24 | Blackthorn | Hi, when I place a call from spa-2000 to * to outgoing pri local, I get a double ring. You hear a neormal ring then about 3/4 the way thorugh the ring you here another one a bit louder. What do I need to adjust? |
| 13:48.21 | mut | change call progress indication in your sip.conf |
| 13:49.10 | brif8 | anyone know what causes 'destroying call '.....@ip.address' |
| 13:50.53 | brif8 | I'm getting it repeatedly with my * box IP address ? |
| 13:53.33 | MrChimpy | from an AGI app I need to pass the call on to another number - I do a dial. But I need to record that call... what would be the typical method for doing that? |
| 13:53.53 | Hmmhesays | send the call back into your dialplan |
| 13:53.58 | Hmmhesays | use monitor or mixmonitor |
| 13:54.53 | brif8 | If I have port 5060 forwarded on the firewall direct to the * box , do I still need to use NAT = yes and externip = in sip.conf ? |
| 13:55.14 | MrChimpy | hmm: thanks |
| 13:55.32 | iulius | Yes - the NAT support is to help with the RTP stream which takes place on a random UDP port |
| 13:56.02 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-45-251.cust-adsl.tiscali.it) |
| 13:57.00 | iulius | More to the point, if you don't the remote phone will probably start sending RTP to the local address of the asterisk--which of course isn't available on its network |
| 13:57.06 | [TK]D-Fender | brif8 : yes, and you also need to forward ports for RTP to work, typically 10000-20000 |
| 13:57.39 | [TK]D-Fender | brif8 : You also need to declare your local subnets with "localnet=" lines in sip.conf |
| 13:57.56 | Hmmhesays | I've run into a snag |
| 13:58.22 | brif8 | [TK]D-Fender: ok I don't have it set yet I'm receiving calls fine, but I do get the odd dropped call. When I set it the last time I could not connect to the VoIP provider at all, did I perhaps do something else wrong ? |
| 13:58.26 | [TK]D-Fender | Hmmhesays : get some pliers to compress the barb :) |
| 13:58.34 | Hmmhesays | I tried |
| 13:58.41 | Hmmhesays | actually is anyone using sipsak in here? |
| 13:58.49 | Blackthorn | Hi, when I place a call from spa-2000 to * to outgoing pri local, I get a double ring. You hear a neormal ring then about 3/4 the way thorugh the ring you here another one a bit louder. What do I need to adjust? |
| 13:58.55 | Hmmhesays | I have my sip message i'm sending in a text file, but I need to pass a couple arguments to it |
| 13:59.11 | [TK]D-Fender | Blackthorn : You;ve already been answered on that.... |
| 13:59.40 | shiznatix | when I am using GoToIfTime then I have for the active days |mon-fri| but what I want to do is do a |mon,wed,fri| but I don't know the correct syntax for that to work |
| 13:59.48 | brif8 | [TK]D-Fender: is it possible that it would fail if I set it to yes ? |
| 13:59.51 | ManxPower | Hmmhesays, too complicated to get to the place in the phone you can read them |
| 14:00.03 | [TK]D-Fender | brif8 : set what to yes? |
| 14:00.19 | Hmmhesays | ahh ManxPower lusers rock |
| 14:00.44 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 14:00.50 | ManxPower | Blackthorn, talked about all the tome on the mailing lists, I think it's priinband=never or something like that. search for double ring in the mailing lists archives |
| 14:00.59 | brif8 | [TK]D-Fender: nat = yes. I set nat = yes and then couldn't connect I assumed that it stopped working because I forwarded the port direct to the * box |
| 14:01.38 | [TK]D-Fender | brif8 : It won't work because you don't have a localnet... |
| 14:01.44 | techman97_andy | hey all - I'm trying to troubleshoot a grandstream phone conferencing issue, and I need to get an Ethereal trace sent to grandstream. I've installed the RPM, but can't figure out which directory it was installed to - any hints from anyone who's used this before? |
| 14:01.52 | [TK]D-Fender | brif8 : pastebin your [general] section of sip.conf |
| 14:02.50 | brif8 | [TK]D-Fender: nat = yes. I set nat = yes and then couldn't connect I assumed that it stopped working because I forwarded the port direct to the * box |
| 14:02.57 | brif8 | [TK]D-Fender: http://pastebin.com/700024 sorry |
| 14:03.51 | ManxPower | nat= needs to be renamed remotedevicenatted=yes|no |
| 14:04.00 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:04.46 | [TK]D-Fender | brif8 : Why is it all commented out? and change your bind to 0.0.0.0 |
| 14:04.58 | ManxPower | or don't use bindaddr at all |
| 14:05.05 | [TK]D-Fender | brif8 : And make sure you are forwarding the right ports |
| 14:05.44 | ManxPower | [TK]D-Fender, Isn't all this stuff in the Wiki? |
| 14:05.46 | brif8 | [TK]D-Fender: as I said when I put it in I couldn't connect to the VoIP provider so I took it out. Yes I have always been forwarding 5060 and I have just opened now 1000-2000 |
| 14:05.50 | MrChimpy | where does asterisk put monitor recordings by default? |
| 14:06.44 | MrChimpy | ah /var/spool/asterisk/monitor |
| 14:07.05 | ManxPower | brif8, don't you mean 10000-20000 |
| 14:07.10 | brif8 | ManxPower: yes that's why I added it but then when connection failed I took it out and it connected. thus my assumption was it caused the connection to fail since I have already forwarded port 5060 to the * box |
| 14:07.19 | brif8 | yes sorry 10000-20000 |
| 14:07.38 | ManxPower | brif8, the instructions for this on the Wiki didn't work? |
| 14:08.22 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 14:08.55 | brif8 | I added the nat = yes , externip and localnet. then restarted * and got unable regsiter. I stopped * commented them out and restarted * and registered fine with VoIp |
| 14:08.59 | brif8 | provider |
| 14:08.59 | [TK]D-Fender | ManxPower : Aren't the answers to 99% of questions asked here in the WIKI. *yes* |
| 14:09.17 | [TK]D-Fender | brif8 : Thats 10000-20000, not 1000-2000 |
| 14:09.24 | brif8 | yes sorry 10000-20000 |
| 14:09.35 | ManxPower | brif8, If your provider is not natted then you do not put nat=yes. nat=yes for to tell Asterisk if the REMOTE device is behind NAT |
| 14:09.57 | ManxPower | localnet, externip, tell Asterisk it's behind NAT |
| 14:10.15 | Sedorox | NAT+VOIP=evil |
| 14:10.16 | Sedorox | lol |
| 14:10.34 | Sedorox | actually.. I guess its more.. PAT+VOIP=Evil |
| 14:10.42 | Sedorox | nat shouldn't be a problem.. as pure nat is 1-1 |
| 14:10.47 | brif8 | my provider Teliax is not natted to my knowledge. I have a firewall (which does NAT) however on the firewall I forward all port 5060 to the internet * box address 192.168.25.22 |
| 14:11.13 | brif8 | at present this part is working ( do get dropped calls now and again) so perhaps not working perfectly correctly |
| 14:11.26 | Sedorox | brif8: by any chance.. is this a cheap like netgear/linksys router? |
| 14:11.28 | ManxPower | brif8, then nat=yes in the [teliax] may break it. |
| 14:11.39 | brif8 | no it's a sonicwall pro 3060 |
| 14:11.50 | Sedorox | ok... and you made sure its UDP, not TCP, right? |
| 14:12.04 | brif8 | ManxPower: I don't have nat=yes in teliax |
| 14:12.09 | brif8 | ManxPower: I don't have nat=yes in [teliax] |
| 14:12.24 | brif8 | I have insecure = very |
| 14:13.07 | brif8 | 5060 UDP and 10000-20000 UDP are now forwarded to the internal address |
| 14:15.06 | ManxPower | brif8, and you have localnet and externip specified in [general] |
| 14:15.49 | brif8 | ManxPower: I did when it didn't work, and it is commented out as posted to pastebin |
| 14:16.04 | ManxPower | brif8, You MUST have those set. |
| 14:16.26 | ManxPower | If you do not set them, then asterisk will not work behind NAT |
| 14:17.14 | ManxPower | It's not rocket science. It takes all of 10 mins to set up Asterisk behind NAT. |
| 14:17.17 | brif8 | how come it is working now without them? I don't mean to be dumb, (1) I understand you MUST have it (2) when I put it in * failed, (3) when I took it out * worked ? |
| 14:17.49 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:18.12 | ManxPower | brif8, I don't know, but without those options Teliax will send it's audio and responses to address of the Asterisk server, not the external IP of your NAT router. |
| 14:18.20 | C4T3l | hello world! |
| 14:18.21 | brif8 | I will put it back in and sip reload but it will proable fail again |
| 14:19.27 | brif8 | ok changing bindaddr to 0.0.0.0 adding nat = yes , externip and localnet anything else ? |
| 14:19.33 | ManxPower | Teliax uses Asterisk. Maybe THEY have nat=yes in your config on their side. In which case we cannot help you and you should consult the "support" link from your account on Teliax |
| 14:19.50 | brif8 | what about allowguest = yes ?? |
| 14:19.57 | ManxPower | brif8, what are localnet and externip set to? |
| 14:20.13 | ManxPower | brif8, allowguest is not used in this situation |
| 14:20.30 | brif8 | localnet = 192.168.25.0/255.255.255.0 externip = 111.41.5.12 |
| 14:21.16 | ManxPower | brif8, you do realize that Teliax provides confiuration information for Asterisk on their support page, right? |
| 14:21.18 | brif8 | comment allowguest out or leave it in ? |
| 14:21.37 | ManxPower | brif8, it will make no difference for teliax if you remove allowguest or remove guest |
| 14:21.46 | brif8 | yes I have read their page and the WIKI following all the advice it gives. |
| 14:22.44 | brif8 | ok edited sip.conf doing sip reload |
| 14:23.09 | ManxPower | brif8, have you read this page? https://www.teliax.com/clients/?a=su&doc=aah |
| 14:23.58 | ManxPower | brif8, Why are you using SIP with Teliax? They also support IAX2 |
| 14:24.41 | ManxPower | I give up. I cannot help you further. |
| 14:25.12 | brif8 | because all the Cisco 7960 use SIP so staying with a common protocol. Also I forward calls person x calls my number and based on logic I forward the call to another number. in talking with Teliax SIP was the cleaner and more quality way to proceed |
| 14:25.19 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 14:26.18 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 14:27.21 | brif8 | ManxPower: I don't mean to agrivate you or anyone else. Yes I'm trying hard to learn, I just know what happened. why that's what I'm trying to find out. |
| 14:28.00 | brif8 | it makes no sense whatsoever that the WIKI says use nat=yes, and then when I do it fails. My first assumption is I have a config problem, not that WIKI it wrong. |
| 14:31.35 | brif8 | ok netstat is show 0.0.0.0:5060 |
| 14:32.23 | brif8 | sip show peers has teliax/legendary 207.174.202.4 N 5060 OK (52 ms) |
| 14:32.25 | [TK]D-Fender | brif8 : What kind of router do you have? |
| 14:32.46 | brif8 | now I'm real confused because when I set this up two weeks ago nat = yes failed ????? |
| 14:32.59 | brif8 | sonicwall Pro 3060 |
| 14:36.11 | brif8 | ok with nat = yes etc.. I lost all connections to my IP phones ?? |
| 14:36.29 | brif8 | now-one could phone out at all and calls coming in routed to unavaioable |
| 14:36.53 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 14:37.36 | *** part/#asterisk lawnjam (n=john@cpc1-broo2-0-0-cust675.renf.cable.ntl.com) |
| 14:37.40 | AsteriskAlbania | What is the best solution for PrepaidCallingCard with asterisk? |
| 14:38.15 | pif | has anyone had success with /var/lib/asterisk/agi-bin/receive_fax ? |
| 14:38.36 | pif | in the asterisk-app-fax debian package? |
| 14:39.45 | [TK]D-Fender | brif8 : if your localnet is defined right you should be fine |
| 14:40.19 | brif8 | localnet is defined correctly and it dropped all IP phones |
| 14:40.20 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 14:40.29 | Blackthorn | brif8 please read your private convo up top. |
| 14:40.33 | kay2 | when a call is queued, how can I know in the CDR how long it stayed in queue and how long the user was with the operator ? |
| 14:46.40 | *** join/#asterisk stack_ (n=stack_@63.239.190.202) |
| 14:46.40 | brif8 | [TK]D-Fender localnet = 192.158.25.0/255.255.255.0 |
| 14:46.47 | brif8 | which is correct subnet |
| 14:47.15 | AsteriskAlbania | [TK]D-Fender what is the best and comfort solution for Calling Card Application with asterisk |
| 14:48.13 | [TK]D-Fender | AsteriskAlbania : No clue. No experience in billing |
| 14:48.44 | asterboy | kay2, for the entire call duration I subtract the start time from the end time, as far as queue length and operator time goes, unless there is data in the CDR to do the math on, not sure how you can get that unless you increase the verbosity. Maybe something you can add in the dial plan? |
| 14:49.21 | MrChimpy | hmm. i'm trying to monitor a call in an agi app which does a DIAL to a phone |
| 14:49.25 | Hmmhesays | ok i just got a crash course on text replacement using sed |
| 14:49.27 | Hmmhesays | fun |
| 14:49.33 | kay2 | asterboy, the thing is that when u Queue a user to a dialplan, u don't know when it answeres... |
| 14:49.45 | MrChimpy | i get nowt recorded, or it stops before/on answer |
| 14:49.54 | MrChimpy | i'm using agi's EXEC function |
| 14:50.01 | MrChimpy | to call monitor |
| 14:50.21 | MrChimpy | is it broken such that I can only use monitor within a real dialplan? |
| 14:51.40 | Hmmhesays | I already told you a good way to do that |
| 14:51.45 | asterboy | kay2, does it show in cli? |
| 14:51.58 | MrChimpy | yeah, but chucking back to the dialplan isn't good for me |
| 14:52.33 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:53.02 | ManxPower | MrChimpy, call Monitor before Dial, of course |
| 14:53.14 | MrChimpy | s'what I do |
| 14:53.31 | *** join/#asterisk becks (n=becks@stepbuild.com) |
| 14:53.57 | ManxPower | MrChimpy, start asterisk as "asterisk -cvvv" then watch that screen and run your AGI. STDERR from AGI only gets output on the specific console terminal |
| 14:54.08 | MrChimpy | <PROTECTED> |
| 14:54.08 | MrChimpy | <PROTECTED> |
| 14:54.08 | MrChimpy | <PROTECTED> |
| 14:54.49 | MrChimpy | Monitor never seems to attempt to mix the output either - /var/spool/asterisk/monitor has two files there |
| 14:55.01 | asterboy | kat2,AppendCDRUserField() may be something you can add when you queue the user |
| 14:55.07 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 14:55.40 | MrChimpy | <PROTECTED> |
| 14:55.40 | MrChimpy | PASS (0) |
| 14:55.40 | MrChimpy | ********** would dial 07841870816 |
| 14:55.40 | MrChimpy | <PROTECTED> |
| 14:55.40 | MrChimpy | <PROTECTED> |
| 14:55.41 | MrChimpy | <PROTECTED> |
| 14:55.43 | MrChimpy | <PROTECTED> |
| 14:55.45 | MrChimpy | <PROTECTED> |
| 14:55.47 | MrChimpy | <PROTECTED> |
| 14:55.51 | MrChimpy | <PROTECTED> |
| 14:55.53 | MrChimpy | <PROTECTED> |
| 14:56.08 | asterboy | ~pastebin |
| 14:56.11 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com |
| 14:56.23 | techman97_andy | ok - I think I have a clue about my GrandStream conferencing issue. When the conf call is initiated from the GS-GXP2000, I did an ethereal trace and buried in the mix was: SIP/2.0 488 Not Acceptable Here |
| 14:56.42 | techman97_andy | any ideas? |
| 14:56.44 | MrChimpy | not sure what the b option is for monitor... what does "bridged to another channel" mean? will a dial make a bridge? |
| 14:57.28 | asterboy | techman97_andy, that is exactly the message I received when the GXP firmware was broken |
| 14:57.51 | techman97_andy | I called GS support this morning, and they pushed me through 3 firmware updates...=S...all the same thing each time |
| 14:58.18 | techman97_andy | how did you *fix* the firmware? |
| 14:58.54 | asterboy | It just worked for me. |
| 14:59.04 | asterboy | You may have to retire that bookend |
| 14:59.13 | ManxPower | techman97_andy, GS has a long history if bad firmware |
| 14:59.23 | techman97_andy | what did you do though when you found out the firmware was broken? |
| 14:59.45 | asterboy | Called tech support, did the latest upgrade and it worked there after. |
| 15:00.01 | asterboy | The crazy thing was the GXP was also breaking my Polycoms! |
| 15:00.17 | asterboy | Lost a lot of money trying to save money with that garbage |
| 15:00.25 | brif8 | ManxPower: if I may what causes the 'Destroying call' message on the console also found in the debug logfile ? |
| 15:00.29 | ManxPower | techman97_andy, that's a message where the two sides can't agree on a codec. Maybe you are using G729 and the phone can only do 1 G729 call at a time (prett common) |
| 15:01.02 | coppice | pretty pathetic, too |
| 15:02.57 | asterboy | actually, I would not want to display that phone on my shelf...and bookend is to high a status. |
| 15:03.24 | asterboy | you'll get grey pubics by the time your done |
| 15:05.08 | techman97_andy | I'm back - catching up 1 sec |
| 15:05.14 | mut | thinking the easiest way would be a ignorance check against the astdb, something like |
| 15:05.16 | mut | User/PhoneNumber/BlockNumber = 'yes' |
| 15:05.19 | mut | and check if it exists |
| 15:05.26 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:05.27 | mut | other ideas? |
| 15:05.45 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 15:05.48 | mut | for call blocking.. |
| 15:06.48 | dpryo | Is it possible to execute an AGI, and not wait for it to return? |
| 15:07.28 | mut | have the agi exec another script in the background |
| 15:07.33 | mut | then have it return itself |
| 15:07.39 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:07.49 | dpryo | Yeah, I'm doing that.. But it isn't as pretty as I want it :) |
| 15:07.51 | mut | i dunno if there is a switch to do it directly tho |
| 15:07.55 | dpryo | Just wondering if theres another way |
| 15:08.02 | nextime | dpryo : you *must* wait for agi to return, *but* you can do a double fork to daemonize the agi so it return with 0 before to execute what you must execute |
| 15:08.26 | nextime | so, you have to manage and daemonize the agi script from itself |
| 15:09.10 | stack_ | does anyone have a good way of remotely starting call recording... for example, our call center trainer would like to record a series of calls that our agents do and would like to just dial something to start recording on a certain extension... |
| 15:09.10 | dpryo | It doesn't really needs to be AGI, could be executed from system(), I do stuff on callerid. |
| 15:09.12 | nextime | or you can do some similar from fastagi, returning before to execute your work |
| 15:09.36 | nextime | stack_ : use AMI |
| 15:10.27 | *** join/#asterisk bon (i=bon@shadow.radiolan.sk) |
| 15:11.21 | bon | exten => _0.,2,SetCallerID(${CALLERIDNUM}) < should set cid to my number, not the number i am calling, or? |
| 15:11.47 | Blackthorn | I am trying to figure out why when I call from spa through my local pri It rings, then on the second reng it starts to double ring about 3/4 through the ring) it was meniotned to set priinband=never but I am not able to find any such setting on the wiki any help? |
| 15:13.31 | [TK]D-Fender | bon : that line is WORTHLESS. |
| 15:13.48 | [TK]D-Fender | bon : like "X=X" |
| 15:14.08 | techman97_andy | asterboy: man Grandstream's support guy is difficult. Not even before he had my capture, he was telling me that I was wrong and it's a problem with the server, not their phone |
| 15:14.27 | bon | [TK]D-Fender: well i use 123${CALLERIDNUM} |
| 15:14.36 | bon | i need to add some prefix in front of my number |
| 15:14.52 | bon | but anyway, $CALLERIDNUM seems to contain the numbe i am calling to, not the originating number |
| 15:14.57 | bon | is that correct behaviour? |
| 15:15.10 | MrChimpy | ahh |
| 15:15.13 | brif8 | [TK]D-Fender: is it possible the phones fail when nat = yes is set, because the firewall / router routes all 5060 back to the * box on the same local subnet ? |
| 15:15.17 | techman97_andy | asterboy: here's a thought - I purchased 10 channels of G729 from Digium - could those be possibly conflicting? |
| 15:15.21 | MrChimpy | I think it may be the way i'm calling exec |
| 15:18.01 | [TK]D-Fender | brif8 : Local traffic should be point-point within your lan, and * should be sending SIP & RTP pactets with the LOCAL IP of the machine and work 100%. I might suggest adding sip=no into the PHONE definition just to be extra safe. |
| 15:18.11 | kay2 | how would I add a CDR variable for knowing how long a user has been hold in a queue and how long he was in communication with an agent |
| 15:19.32 | [TK]D-Fender | kay2 : you don't. Thats what the QUEUE LOG is for. |
| 15:19.46 | brif8 | [TK]D-Fender: within sip.conf If I add the nat = yes and externip and localnet inside [teliax] it will only affect teliax not all other right ? and where do I put sip = no ? |
| 15:20.09 | [TK]D-Fender | brif8 : No, those params are NOT for peers, only [general] |
| 15:20.41 | [TK]D-Fender | brif8 : you put nat=no in your SIP PHONE's definition, not your ITSP's |
| 15:20.53 | MrChimpy | in AGI, how do you pass multiple parameters to EXEC? EXEC MONITOR "wav" "monitor_test" "m" doesn't work (only 1st option picked up) and EXEC MONITOR "wav,monitor_test,m" picks up all the parms as being the first |
| 15:20.57 | brif8 | ok nat = no you had sip = no |
| 15:22.19 | MrChimpy | ah, message here says use pipe instead. how obvious |
| 15:22.55 | [TK]D-Fender | brif8 : I'm entitled to typos and minor aneurisms too! |
| 15:22.55 | MrChimpy | there's some serious insanity going on in dialplan language and AGI |
| 15:24.05 | bon | any hint on my $CALLERIDNUM trouble? |
| 15:26.50 | asterboy | techman97_andy, not sure why you bought those licenses in the first place...unless you have a need to enable them on some fxs ports. |
| 15:27.10 | brif8 | [TK]D-Fender: what a pro like you lol I do it all the time, I'm just trying to find something before I go mad |
| 15:29.19 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
| 15:29.26 | *** join/#asterisk ghost99 (n=neville@222-153-96-3.jetstream.xtra.co.nz) |
| 15:29.26 | salviadud | !seen _paulo_ |
| 15:33.19 | trelane_ | has anyone seen this before? |
| 15:33.22 | trelane_ | May 4 15:15:40 WARNING[25654] channel.c: Thread 1079597408 Blocking 'Zap/pseudo-1160181049', already blocked by thread 1077733728 in procedure ast_waitfor_nandfds |
| 15:33.26 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:33.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:33.40 | MrChimpy | dammit |
| 15:34.07 | MrChimpy | now I've found a sample of multiple parameters to exec in agi and my script hangs at that line! |
| 15:34.16 | MrChimpy | <PROTECTED> |
| 15:34.19 | MrChimpy | (perl) |
| 15:34.20 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:34.29 | MrChimpy | AGI debug doesn't show an RX for it |
| 15:36.35 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 15:37.18 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 15:40.10 | MrChimpy | guess what that hanging problem was? |
| 15:40.19 | MrChimpy | i'd missed the \n from the exec line |
| 15:40.32 | mut | yesssssssssss |
| 15:40.56 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 15:40.58 | faberk | hi folks |
| 15:41.01 | mut | wanna fix my problem? |
| 15:41.09 | faberk | just a quick question |
| 15:41.30 | nettie | hello guys, I'm connecting a voip service usign 711u codec. Voice is fine but telephone events are not working. So if I call a cellphone and the called party hangups instead of pickup I keep wait forever and asterisk never figure out if the phone is still ringing or not. Of course I dont hear the ringback.. nothing. Sip debug shows this: Non-codec capabilities: us - 0x1 (telephone-event) which seems that the codecs are compatible only for |
| 15:41.35 | nettie | Anyone know what could be wrong please? |
| 15:41.56 | faberk | I need to setup an * box with BRI |
| 15:42.20 | faberk | suggestion for a card to use? also, what driver? |
| 15:43.28 | faberk | do you allow just ulaw? |
| 15:43.49 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 15:44.14 | faberk | nettie: do you allow just ulaw? |
| 15:44.32 | nettie | yes |
| 15:44.54 | nettie | I just bought g729 from digium to test with it as well |
| 15:44.58 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:45.17 | faberk | during the handshake, what codec the other party request? |
| 15:45.50 | faberk | try to allow more than just one code |
| 15:45.53 | faberk | codec |
| 15:46.04 | MrChimpy | feck it, now mixes and takes the options fine but only records a second or so |
| 15:46.15 | tamp4x | .anyone here have problems with vega pbx's interactign witha sterisk? |
| 15:47.09 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 15:47.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:47.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:48.12 | trelane_ | follow the bouncing denon |
| 15:48.28 | *** join/#asterisk [Airwolf] (n=airwolf@ip144-57.dsl.introweb.nl) |
| 15:51.57 | tzafrir_home | faberk, which ISDN channel do you use with asterisk? misdn? chan_capi? bristuff? (zaphfc or capi?) visdn? |
| 15:52.23 | nettie | whatever codec I use I still get: |
| 15:52.24 | nettie | Found description format GSM |
| 15:52.24 | nettie | Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) |
| 15:52.24 | nettie | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) |
| 15:54.17 | froguz | just a linux question... is there a parameter that when passed to ifconfig it shows me only my eth0 inet addr |
| 15:54.18 | froguz | ? |
| 15:54.50 | tzafrir_home | froguz, no. ip may produce a nicer output |
| 15:55.03 | *** join/#asterisk zerotrace (i=synack@fury.csh.rit.edu) |
| 15:55.25 | zerotrace | is anybody else having problems with connecting to FWD using IAX? |
| 15:56.18 | tzafrir_home | in rapid I used: /sbin/ifconfig | perl -n -e 'BEGIN{$/="";} s/\s+/ /gs; s/^(\S+).*inet addr:(\S+).*/$1\t$2/; print "$_\n"' |
| 15:56.57 | tzafrir_home | you can use a much simpler filter if you just want the IP address of a single interface |
| 15:57.48 | froguz | thanks tzafrir_home |
| 15:58.18 | *** part/#asterisk stina (n=fa@CPE000c29ba6c70-CM000f9fac90ec.cpe.net.cable.rogers.com) |
| 15:59.33 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 16:00.00 | *** part/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 16:00.04 | *** join/#asterisk icall_andy (n=andym@67.106.157.244.ptr.us.xo.net) |
| 16:00.26 | icall_andy | Anyone have experience with large scale deployments? I'm looking for some numbers on users vs. concurrent call counts. |
| 16:00.59 | asterboy | Interesting, GenTek is *not* the best price on Sangoma |
| 16:01.10 | asterboy | iFax is cheaper than GenTek. |
| 16:03.44 | asterboy | icall |
| 16:03.50 | asterboy | Internal testing with dual Intel® Xeon 1.8GHz processors allowed 60 concurrent G.729 calls. Dual Xeon 2.8GHz processors allowed 80 concurrent G.729 calls. |
| 16:04.03 | asterboy | that is of course specific |
| 16:04.19 | asterboy | you are *not* that specific now are you? :P |
| 16:05.20 | icall_andy | Yeah, we've done a lot of dimensioning. |
| 16:05.48 | icall_andy | I'm saying, for example, you have 1,000 extensions. What's a general rule of how many concurrent calls you'll have active? |
| 16:06.03 | icall_andy | Obviously it's somewhat specific to the purpose, but I've seen numbers before. |
| 16:07.03 | froguz | asterboy, how much RAM? |
| 16:07.26 | faberk | tzafrir_home: none yet. I have to setup one, but I have no experience with BRI. I've just used PRI. |
| 16:07.57 | [TK]D-Fender | asterboy : No way.... |
| 16:08.05 | faberk | nettie: try to allow ulaw, alaw, gsm |
| 16:08.16 | faberk | all those three |
| 16:08.40 | tzafrir_home | faberk, IIRC some of them actually use alaw. Part of the bristuff patch is to (allw? default?) alaw as the ISDN channel encoding |
| 16:09.02 | tzafrir_home | faberk, what hardware do you have? |
| 16:10.02 | faberk | my reply about codecs, was for NETTIE |
| 16:10.05 | faberk | :o) |
| 16:10.31 | faberk | about BRI, I have to learn |
| 16:10.34 | asterboy | well, upto 15 is 3GHz, 1Gb Ram recommended...so for 1000 your going to need a distributed architech |
| 16:12.26 | asterboy | icall, sounds like you want to know what % of a 1000 exten environment actually gets used. |
| 16:12.53 | asterboy | that is a real hard one to answer. |
| 16:13.05 | asterboy | If there is a Katrina knocking at the door, I'd say 100% |
| 16:13.19 | snitt | :D |
| 16:13.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 16:14.06 | *** join/#asterisk gezick (n=gezick@c-68-50-25-85.hsd1.dc.comcast.net) |
| 16:15.29 | *** join/#asterisk Lucas| (n=mmcguire@193.111.227.220) |
| 16:16.32 | *** join/#asterisk budmang (i=smdepot@12.206.134.162) |
| 16:17.14 | wunderkin | who said icall is using g729 |
| 16:18.08 | icall_andy | asterboy: Yeah, we have a distributed architecture :) We're just doing out beta right now of about 500 users, and I don't think that sampling is enough to make judgements off of. I was just wondering if anyone has an experience with larger scaled environments. |
| 16:18.30 | *** part/#asterisk ghost99 (n=neville@222-153-96-3.jetstream.xtra.co.nz) |
| 16:18.35 | wunderkin | hasnt even said if using voip |
| 16:18.58 | asterboy | ah |
| 16:19.04 | *** join/#asterisk lorinc (n=ang@caracas-0321.adsl.interware.hu) |
| 16:19.41 | asterboy | I'm climbing the ladder, not to that call volume ... yet |
| 16:20.08 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:20.28 | generalhan | whats up all ? |
| 16:22.00 | asterboy | my sleave |
| 16:22.14 | generalhan | asterboy: why's that ? |
| 16:22.21 | asterboy | (that's a joke) you ask what's up |
| 16:22.28 | asterboy | I reply "my sleave" |
| 16:22.35 | asterboy | "what's up my sleave" |
| 16:22.35 | generalhan | yea but why would your sleave be up ? lol |
| 16:22.39 | generalhan | i see |
| 16:22.39 | asterboy | ah forget it.. |
| 16:22.42 | asterboy | :P |
| 16:22.48 | generalhan | more so to finish the statement ... not a reply to it |
| 16:23.00 | asterboy | yep |
| 16:23.02 | generalhan | ha |
| 16:23.17 | generalhan | whats going on in here today ... seems quite this morning |
| 16:23.18 | asterboy | grandpa gave me that one. |
| 16:23.33 | asterboy | he also gave a good check list before you leave the house. |
| 16:23.55 | generalhan | [TK]D-Fender: you around ?? |
| 16:23.59 | ioerror | If I have an asterisk box on the net and a grandstream device behind a NAT, is it possible to make them play nice without an stun server? |
| 16:24.13 | asterboy | Eyes, Specticles, Testicles, Watch and Wallet. Have those and you can go anywhere in the world. |
| 16:24.39 | generalhan | lol |
| 16:24.41 | generalhan | oh yea ? |
| 16:25.14 | asterboy | ioerror, better to ask a more specific question or describe your problem. |
| 16:25.29 | asterboy | GXP and * both work with NAT and STUN. |
| 16:25.43 | asterboy | But the GXP is a piece of crap so expect to experience frustration |
| 16:26.14 | ioerror | asterboy, I have a grandstream 386 and an asterisk box. I want the granstream to send/recv calls. The grandstream is behind a linksys nat device. |
| 16:26.33 | [TK]D-Fender | generalhan : no. |
| 16:26.35 | ioerror | It's a port restricted cone nat I believe. |
| 16:26.38 | [TK]D-Fender | :D |
| 16:26.46 | Blackthorn | When I call ata to ata, ata to ld provider rining works fine. when calling ata to local pri i get a double ring. Any idea how to fix? |
| 16:26.51 | techman97_andy | Specticles, Testicles, Wallet, and Watch - I was taught that once in school for the sign of the cross |
| 16:26.52 | generalhan | [TK]D-Fender LOL thats how i feel |
| 16:26.52 | techman97_andy | hehehehe |
| 16:27.13 | ioerror | asterboy, my question i suppose is this: do i need to do anything other than enabling nat on asterisk and telling the grandstream it's behind a nat? do I need to setup and install an stun server? |
| 16:27.31 | nettie | I tried other codecs, I keep getting no codec compatibility for telephone-events |
| 16:27.41 | generalhan | [TK]D-Fender: i have a quick question for you ... you told me not to edit callerid(number) and to use (name instead ... why is that ?? |
| 16:28.37 | [TK]D-Fender | generalhan : becauseI can't trust that the number can hold ALPHA, and the uniqueid holds a "." in it |
| 16:28.44 | [TK]D-Fender | generalhan : Always a reason :) |
| 16:29.57 | generalhan | [TK]D-Fender: ok i understand that, but i can set it to CAllerid(number) once it comes out of the DB right? becuaes i have 3 different queues that this works for now, and usually i have it come up on the phone with the name of the queue the call is coming from, and of course now its gone and my reps are bitching again ! |
| 16:30.12 | asterboy | ioerror, I have setup a Polycom with * and no stun server. |
| 16:30.23 | asterboy | so you may be able to do it. |
| 16:31.17 | ioerror | ok, so stun isn't required per se, the device might suck and require it but asterisk itself can deal with knowing the device is natted (if i tell it) and respond accordingly |
| 16:31.22 | ioerror | thanks for your help asterboy |
| 16:31.29 | asterboy | yep |
| 16:31.34 | ioerror | this may also be helpful: http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
| 16:31.54 | ioerror | as it seems to answer my question in great detail. Though nothing is as nice as an authoratative response :-) |
| 16:33.08 | asterboy | ya that is a nice description |
| 16:34.18 | asterboy | ioerror, and *DO* upgrade the firmware on the 386 to save some headaches. |
| 16:35.14 | nettie | [TK]D-Fender Hey TK! how's going? at the wanst the voip carrier not sending signalling but seems to be a codec compatibility issue. |
| 16:35.31 | nettie | ehehe.. at the end wasnt I emant :) |
| 16:35.33 | nettie | meant |
| 16:35.44 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:36.40 | asterboy | So GenTek gave me a better price and free shipping of $1000 |
| 16:36.55 | asterboy | Was expecting more from Wholesale. |
| 16:37.26 | asterboy | The echo module went from $400 to $175 though. |
| 16:37.36 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 16:37.39 | asterboy | So I'm going to get that now that it's more reasonable. |
| 16:38.16 | asterboy | Anyone else have good prices from VOIP vendors |
| 16:39.15 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:39.20 | brif8 | If my firewall handles VoIP eg SIP transformations should this be enabled or not ? |
| 16:39.49 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 16:40.03 | bkw_ | yes enable it |
| 16:40.17 | *** join/#asterisk saftsack (n=saftsack@p54A7D14C.dip.t-dialin.net) |
| 16:40.22 | brif8 | bkw_ ok what about Permit non-SIP packets on signaling port |
| 16:41.46 | brif8 | bkw_ also is I enable SIP Transformations how will that effect nat = yes is sip.conf |
| 16:43.37 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 16:43.39 | Blackthorn | When I call ata to ata, ata to ld provider rining works fine. when calling ata to local pri i get a double ring. Any idea how to fix? |
| 16:43.56 | *** join/#asterisk Olobola (i=Olobola@216.218.221.165) |
| 16:45.17 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:45.38 | brif8 | bkw_: any ideas ? |
| 16:45.55 | Qwell[] | bkw_: hey, got a sec? |
| 16:49.10 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 16:50.36 | asterboy | Just read the the TDM04B does *not* need the 12V connector for FXO ports |
| 16:51.06 | asterboy | Interesting, since I have it plugged in...maybe should remove that if not needed |
| 16:53.55 | Qwell[] | asterboy: it won't hurt it, but yeah, it's only needed for fxs |
| 16:54.18 | zerotrace | is anybody else having rouble with FWD's IAX server? |
| 16:54.35 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 16:55.18 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:55.37 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:56.00 | kaz0358 | i found something strange and i don't know if it is a bug or not.. if i do a Set(CALLERID(name)="KSU") and then later do a Dial(SIP/xxx), then there is no audio when the SIP connection gets going.. when I do a Set(CALLERID(name)=KSU) and then Dial i get audio... |
| 16:56.16 | Netgeeks | quick question for anyone. In the latest versions of asterisk is there a sip.conf nat= argument that works with most common UA's regardless whether they are behind a nat or not? |
| 16:57.21 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 16:58.13 | *** join/#asterisk arley (n=alex@209-78-110-242.ded.pacbell.net) |
| 16:58.26 | arley | hello? |
| 16:59.03 | Druken | hello, hello hello hello hell hell hel he he he he h h h h h |
| 16:59.13 | kaz0358 | what is odd about setting the callerid is that http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set indicates to use the quotes |
| 16:59.43 | kaz0358 | and asterisk does not give an error whether you put them on or not |
| 17:00.29 | sevard | god damn virtual hosts and me not understanding them |
| 17:00.31 | arley | any one have problems with digium TE110P, Dell SC430 with T1 card in no playback audio although Voip Audio works |
| 17:00.46 | Netgeeks | sevard: http virtual hosts? |
| 17:00.49 | sevard | yeah. |
| 17:00.52 | Deep6 | anyone here doing stuff with LDAP and asterisk? |
| 17:00.55 | sevard | i get why they work i don't get how they work |
| 17:01.11 | Netgeeks | I was admitted to an insane asylum after working with them for a while... but I got better |
| 17:01.24 | kaz0358 | arley, i'm using the TE210P and voip and not having issues.. except for the strangeness that i reported above. |
| 17:01.26 | sevard | Netgeeks: in my router's walled garden setup I can't add host names I can only add IP addresses |
| 17:02.01 | arley | are you running an SC430? the only thing i can figure is no sound card on the system (adding one didnt help) |
| 17:02.01 | sevard | I need to add this domain.tld but the provider is obviously using virtual hosts because if i go to the ip address that resolves at domain.tld i get a different freaking website |
| 17:02.08 | arley | i remove the T1 card audio returns |
| 17:02.20 | sevard | Netgeeks: what to do? :( |
| 17:02.35 | kaz0358 | arley, no.. we just bought components and put it together ourselves |
| 17:02.36 | arley | sevard - what are you tryign to do i just joined part way through |
| 17:02.48 | Netgeeks | sevard: in the end I admitted I didn't want to become a http expert, so I payed a consultant |
| 17:02.58 | sevard | arley: go to a website that is using virtual hosts via the host's ip |
| 17:03.12 | sevard | arley: i'd repost my question but the guys here get pissy :) |
| 17:03.12 | Druken | sevard: you can't |
| 17:03.13 | arley | kaz - i am about to try another server, sucks as the SC430 is a good price system |
| 17:03.14 | [TK]D-Fender | arley : There are know iddues with several Dell servers and Digium cards |
| 17:03.20 | kaz0358 | arley, i don't know about that specfic dell.. but i have seen documentation about particular Dell's not working. i'm guessing it had something to do with IRQs, but i have no idea. i didn't read to closely |
| 17:03.25 | Qwell[] | sevard: It will go to the first site on the vhost list with that IP |
| 17:03.38 | Druken | namevirtualhost the browser basically requests the site |
| 17:03.42 | sevard | Druken: yeah, i found that out |
| 17:04.01 | arley | sevard - you need to manipulate the header of the HTTP request otherwise you will only ever see the default web site on the server by using the IP only |
| 17:04.08 | sevard | Druken: and tricks liek ip/domain.tld isn't working either |
| 17:04.25 | sevard | arley: by injecting a different URI? |
| 17:04.40 | Qwell[] | The "default" site can be changed... |
| 17:04.42 | arley | yeah my fear is that the DELL wont work (works with my analogue installs) this is my first T1 install |
| 17:04.45 | Qwell[] | Just reorder your vhost list |
| 17:04.54 | Druken | Qwell: i don't think it's his server |
| 17:04.55 | Druken | hehe |
| 17:04.56 | kaz0358 | out of curiosity, has anyone looked at ISN? http://www.internet2.edu/sip.edu/isn/ .. anyone think it is better in the long run than pointer to sipproker.com for gatewaying between voip providers |
| 17:05.01 | sevard | Qwell[]: I'd have done it ages ago but it's not my server :P |
| 17:05.11 | Qwell[] | then...what's the problem? :P |
| 17:05.19 | arley | servard - yeah your HTTP request should carry the name of the requested site or you get default only |
| 17:05.23 | sevard | i need to point my walled garden |
| 17:05.42 | sevard | at an ip of a website that's stuck behind a virtual host |
| 17:05.54 | Qwell[] | /etc/hosts |
| 17:05.57 | Druken | so move the site to a properly hosted place |
| 17:05.58 | Netgeeks | kaz0358: Yes, I even have an ITAD number! |
| 17:06.14 | sevard | jesus man i can't touch the website, it's not mine. |
| 17:06.22 | kaz0358 | netgeeks, really? that's cool. i just requested one for k-state. how long did it take you to get a number? |
| 17:06.29 | Qwell[] | sevard: Do you know what domain it should be called by? |
| 17:06.32 | Druken | sevard: then your SOL |
| 17:06.38 | Qwell[] | Just add a simple hosts entry for it. |
| 17:06.55 | asterboy | Does Digium have a EC hardware? |
| 17:06.55 | Qwell[] | and why can't you call it by hostname directly? |
| 17:07.01 | Qwell[] | asterboy: on the quad t1 cards |
| 17:07.04 | Qwell[] | and the tdm2400p |
| 17:07.05 | sevard | Qwell[]: I do know what domain it has to be called by but within my router's walled garden a I can only add IP addresses for it to work |
| 17:07.11 | arley | sevard -you want to hit a site that was hosted but DNS is no longer pointed at it? |
| 17:07.12 | asterboy | ah, that is what I thought. |
| 17:07.23 | asterboy | *not* available on the TDM400P |
| 17:07.34 | Qwell[] | why would it be? |
| 17:07.37 | Qwell[] | It's only 4 lines :p |
| 17:07.38 | sevard | arley: dns is pointed at it, just this router's walled garden doesn't work with host names. |
| 17:07.40 | Cresl1n | asterboy: nope |
| 17:07.47 | Qwell[] | not worth it |
| 17:08.02 | asterboy | I guess not, since your CPU can do it. |
| 17:08.07 | Cresl1n | asterboy: are you having echo problems? |
| 17:08.13 | asterboy | Although, my home system runs on a 233Mhx |
| 17:08.14 | Netgeeks | kaz: about 6 weeks or so I think... it was a while back... http://www.iana.org/assignments/trip-parameters (I'm 257 - the second assigned itad) |
| 17:08.24 | arley | sevard - this isn't a routign issue if the IP works - is a resolution issue |
| 17:08.25 | asterboy | Cresl1n, yes |
| 17:08.31 | asterboy | on a 800Mhz |
| 17:08.43 | Cresl1n | asterboy: what kind of lines are they? |
| 17:08.46 | asterboy | Done the recompile with options |
| 17:08.48 | asterboy | PSTN |
| 17:08.50 | Cresl1n | FXO or FXS? |
| 17:08.52 | Cresl1n | ok |
| 17:08.53 | kaz0358 | netgeeks, yeah.. that is a low number :) |
| 17:08.55 | Cresl1n | so FXO |
| 17:08.57 | asterboy | yep |
| 17:09.05 | Cresl1n | asterboy: did you run fxotune on the lines? |
| 17:09.26 | asterboy | not yet |
| 17:09.32 | Cresl1n | asterboy: do that first |
| 17:09.39 | asterboy | ok, I'll try that |
| 17:09.44 | sevard | arley: dude. i can't use the domain name. the domain name points at a virtual host. if i use the ip address i get a different website, a different virtual host. |
| 17:09.49 | Cresl1n | asterboy: it's likely that that will fix it |
| 17:09.51 | Qwell[] | Cresl1n == king of zaptel |
| 17:09.59 | file | Cresl1n r0x0rz |
| 17:10.06 | asterboy | ya, done everything else but the fxotune |
| 17:10.47 | Netgeeks | kaz: I actually use it quite a bot for calling frieds at some of the other itad number holders. However, I still think it's a little broken, singe it's not something someone who uses a normal phone would think to dial |
| 17:10.57 | Cresl1n | I'm totally going to make a page on how to debug echo problems |
| 17:11.02 | asterboy | /dev/zap/1 absent: Device or resource busy |
| 17:11.07 | sevard | arley: I could change the meth0d |
| 17:11.16 | Qwell[] | Cresl1n: You should totally make an app that does it... |
| 17:11.22 | Cresl1n | Matt F's thoughts and suggestions on getting rid of echo |
| 17:11.23 | asterboy | hmmm...I'm guessing you need to be in single user mode? |
| 17:11.26 | Qwell[] | Cresl1n: ./fix-my-echo-problems -please |
| 17:11.28 | Netgeeks | and so once again we see a break in the merging of voip and legacy telecommunications instead of moving closer together |
| 17:11.45 | Cresl1n | asterboy: is asterisk running? |
| 17:11.48 | asterboy | yes |
| 17:11.59 | Cresl1n | asterboy: you can't run fxotune while asterisk is running |
| 17:12.10 | asterboy | that would explain the error |
| 17:12.11 | asterboy | :P |
| 17:12.24 | arley | sevard - yeah this is resolution thats the issue - you need to create a host file on your computer and point the domain you want at the IP you want (reboot after you creat the host if on Win) or if you have access to a DNS server add the entry there and point your computer at that DNS server |
| 17:12.57 | asterboy | dialing <number> to clear dialtone |
| 17:13.17 | asterboy | so give it any number to get rid of dialtone? |
| 17:13.21 | Cresl1n | yep |
| 17:13.24 | asterboy | k |
| 17:14.04 | Cresl1n | it takes a couple of minutes per module |
| 17:14.19 | brif8 | can someone please shed light on what this is all about http://pastebin.com/700375 |
| 17:14.21 | asterboy | Tuning module 1Skipping non-TDM / non-FXO |
| 17:14.21 | asterboy | Failure! |
| 17:14.37 | asterboy | hmmm..on my system I have the X100P clone cards |
| 17:14.48 | asterboy | so it does not like those. |
| 17:14.50 | dlynes | asterboy: fxotune doesn't work with x100p anything |
| 17:15.08 | asterboy | man, just can't get a break. |
| 17:15.16 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 17:15.17 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 17:15.21 | asterboy | I'll do this at the client's site which does have the TDM |
| 17:15.59 | Cresl1n | asterboy: oh, that's your problem |
| 17:16.05 | Cresl1n | x100ps suck |
| 17:16.05 | asterboy | lol, yep |
| 17:16.09 | asterboy | and yep |
| 17:16.15 | *** join/#asterisk TheoC (n=theochao@c-67-166-183-193.hsd1.tx.comcast.net) |
| 17:16.26 | asterboy | I'm replacing those with Sangoma |
| 17:16.36 | Cresl1n | lame..... |
| 17:16.37 | asterboy | and EC hardware module |
| 17:16.44 | asterboy | Sangoma lame? |
| 17:16.50 | mut | shite nekka |
| 17:16.51 | Cresl1n | no, I bet we could fix it |
| 17:16.53 | mut | my sangoma kicks ass |
| 17:16.57 | *** join/#asterisk gmaruz1 (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 17:17.04 | mut | only thing other than my cisco box to get rid of the echo |
| 17:17.09 | dlynes | asterboy: cause it's not digium....Cres is a digium employee :) |
| 17:17.14 | asterboy | ah |
| 17:17.16 | asterboy | lol |
| 17:17.24 | mut | Cresl1n: care to wager anything on that statement? |
| 17:17.33 | mut | i'm lookin for a downpayment on a house |
| 17:17.36 | Cresl1n | mut: fixing it? |
| 17:17.39 | mut | yes |
| 17:17.46 | asterboy | X100P fix? |
| 17:17.47 | mut | actaully mine is a pri tho |
| 17:17.49 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 17:18.07 | TheoC | I'm trying to set up a pri line into a TE210 card are we set it up - but on ztcfg - we get "wct2xxp: Clearing yellow alarm" and then it freezes. Also our provider says that our equipment isn't respnonding. Anyone have any ideas about what could be done to fix it? |
| 17:18.15 | Cresl1n | mut: we can have a go at it |
| 17:18.18 | dlynes | mog_work: Just thought I'd let you know that putting the card onto a different slot didn't fix the problem |
| 17:18.38 | dlynes | mog_work: It got rid of the interrupt sharing problem, but the zttest problem is still there |
| 17:18.42 | Cresl1n | mut: what have you tried so far? |
| 17:19.13 | dlynes | mog_work: I'm going to see if I can grab the proper cable for the hard drive to see if maybe the cable that's on there is an 80C ribbon cable, or not |
| 17:19.15 | mut | changed thru 4 diff echo cans, with each different configuration of taps/gains from 8 to -8 on rx and tx |
| 17:19.19 | mut | steps of 2db |
| 17:19.26 | dlynes | mog_work: apparently dma can't be enabled if you're using an 80C ribbon cable on some setups |
| 17:19.43 | mog_work | weird |
| 17:19.45 | mog_work | never heard of that |
| 17:19.50 | mog_work | but /me could be a nubb |
| 17:19.50 | Cresl1n | mut: do you get echo on all calls, or just on certain numbers? |
| 17:19.56 | mut | all calls |
| 17:20.02 | mut | well |
| 17:20.03 | mut | kinda |
| 17:20.11 | mut | all calls out the pri to the pstn |
| 17:20.14 | brif8 | I'm trying to find the root causes to Destroying call and scheduling destruction of call for calls that aren't real ? |
| 17:20.17 | mut | internal pri's have no echo |
| 17:20.23 | dlynes | mog_work: That's something straight out of the dma documentation for the kernel |
| 17:20.29 | Cresl1n | mut: which card is this on? |
| 17:20.38 | mut | te405 |
| 17:20.50 | [TK]D-Fender | mut : Been there with my 2 old TE405P's. |
| 17:21.00 | dlynes | mog_work: btw...the motherboard it's on is a tyan motherboard |
| 17:21.00 | mut | rev 2 |
| 17:21.10 | mut | yeh tk |
| 17:21.11 | dlynes | mog_work: that should not be a problem, right? |
| 17:21.14 | mut | this sangoma card works great |
| 17:21.23 | [TK]D-Fender | mut : I had a rev1 & rev2 card. You on an A104d now? |
| 17:21.23 | mut | other than faxing is 'broken' now |
| 17:21.28 | mut | yea |
| 17:21.41 | [TK]D-Fender | mut : faxing seems fine on mine... |
| 17:21.44 | Cresl1n | mut: so it's not even setup to debug it? |
| 17:21.45 | mut | i used to have a lot of customers that could fax with my old cisco configuration |
| 17:21.47 | [TK]D-Fender | (spanDSP anyways). |
| 17:21.49 | *** join/#asterisk KaBewM (n=DA-MAN@pulse.landshark.net) |
| 17:21.52 | mut | but none of those customers can fax anymore |
| 17:21.54 | russellb | mut: have you contacted DIgium support to debug your problem? |
| 17:22.00 | mut | yea russellb |
| 17:22.24 | mut | Cresl1n: nope, sent the one card back, i still have one in use in another town |
| 17:22.38 | dlynes | mog_work: anyways...I'll let you know what happens after I've replaced the ribbon cable...hopefully that's the only problem |
| 17:22.42 | mut | but thats doing fxs chan bank stuff |
| 17:22.49 | Cresl1n | mut: is it having echo problems? |
| 17:22.55 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 17:22.56 | mut | sometimes it does |
| 17:22.59 | mut | but it's tolerable |
| 17:23.13 | dlynes | mog_work: right now though, out of two runs, I only had one run that gave me a 94% return on one of the tests, but patlooptest is still bugging out like crazy |
| 17:23.23 | mut | ~90% of the time it works fine |
| 17:23.50 | dlynes | mog_work: thanks again for the help last night |
| 17:23.55 | Cresl1n | mut: between the phones on the channel bank you have echo? |
| 17:24.21 | mut | yea.. what else would it be between? |
| 17:24.31 | Cresl1n | mut: that's really weird |
| 17:24.37 | vader-- | say in the features.conf you enable one touch record and it's *1 |
| 17:24.41 | vader-- | how do you activate it? |
| 17:24.41 | mut | it's not common |
| 17:24.46 | mut | but it still exists from time to time |
| 17:24.53 | mut | nothing major to need to replace it tho |
| 17:24.56 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 17:25.10 | mut | but the main pri to the world it was horrid |
| 17:25.19 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:25.19 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 17:25.31 | twisted[mobile] | honk honk |
| 17:25.52 | Qwell[] | twisted[mobile]: y0 |
| 17:26.18 | mut | i'm still confused why going from my cisco setup to the sangoma card my faxing is broken now |
| 17:26.19 | file | meeeeeeeeeeeep |
| 17:26.34 | mut | with the cisco setup it was even double voip hop instead of single like it is with the sangoma car |
| 17:26.34 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 17:26.35 | mut | d |
| 17:26.48 | twisted[mobile] | hey Qwell |
| 17:26.51 | twisted[mobile] | packet data.... heh... |
| 17:33.49 | thogueATredfone | wow ... thats the port interesting quit message I think ive seen |
| 17:37.50 | arley | CRES - know of any issues with TE110P and playback audio not working - not working on a DELL SC430 or another "homemade" system that we've tested so far - if I pull the card audio returns |
| 17:37.53 | Netgeeks | okay, one more quick question.... What would cause Playback to sound like a 33rpm record played at 78rpm? |
| 17:38.20 | *** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com) |
| 17:39.33 | Qwell[] | Netgeeks: 33rpm wav? |
| 17:39.41 | Netgeeks | lol |
| 17:40.06 | Druken | 33.wav.rpm |
| 17:40.23 | Netgeeks | basically I've got a system on an ubuntu dist that plays all the standard asterisk prompts as if it was on speed |
| 17:40.31 | *** join/#asterisk espn (n=DA-MAN@pulse.landshark.net) |
| 17:40.34 | Netgeeks | any thoughts? |
| 17:42.15 | Netgeeks | I take the blank silence as a no to the previous query! |
| 17:42.26 | arley | anyone - any probs with a TE110P and playback? pulling the card restores playback (unfortunately just found the prob and had a 3pm cutover today so no can do now) |
| 17:43.32 | Netgeeks | playback doesn't work with the card installed? |
| 17:43.44 | arley | no |
| 17:44.01 | arley | if i pull the card it works - when i re-install the card it stops |
| 17:44.01 | Netgeeks | yeah.. how old is your te11op card? |
| 17:44.06 | arley | brand new |
| 17:44.07 | Netgeeks | te110p even |
| 17:44.29 | Netgeeks | I had the exact same issue with a te410p card on a system two weeks ago |
| 17:44.42 | generalhan | Qwell: could you do me a favor and take a look at my queues.conf file... some on told me there was a lot of lines in there that i didnt need and i would like to clean it up. but im not really sure what to get rid of. http://generalhan.pastebin.ca/53748 |
| 17:44.48 | arley | netgeeks - hmm any luck fixing it? |
| 17:45.05 | Netgeeks | I configed the te410p card and put a loopback jack in one of the ports to see if a circuit would come up, it was stuck in yellow alarm |
| 17:45.30 | Netgeeks | we contacted digium, they got on the system, looked at the card, and told us to send it to them, they sent us one back and it works fine now |
| 17:45.31 | vader-- | does anyone know if asterisk has the ability or is able to record a greeting from a phone? |
| 17:45.39 | Qwell[] | vader--: Record() |
| 17:45.46 | vader-- | like say we want a greeting when people call our main number |
| 17:46.01 | vader-- | is there a way i can give an office clerk the ability from their phone to record the message |
| 17:46.08 | vader-- | and asterisk will play that message? |
| 17:46.12 | kaz0358 | vader, yes is the answer |
| 17:46.14 | arley | netgeeks -so sounds like a bad card? |
| 17:46.29 | Netgeeks | I'm working on writing a new asterisk app called DoWhatIWantNotWhatIWrote(). It's expected to massively simplify dialplans |
| 17:46.44 | generalhan | vader--: Qwell[]: said it all .. look up Record() and the correct syntax to use and you can do exactly what youre talking about |
| 17:46.45 | Netgeeks | arley, yes, try playback with ztdummy as a driver |
| 17:46.53 | vader-- | ok cool |
| 17:47.21 | Netgeeks | if playback works with ztdummy then I would definately run a check on the card and have digium take a look at the problem |
| 17:47.43 | arley | i'll try that out |
| 17:48.07 | arley | need to re-read up on ztdummy, been a while since i looked at that |
| 17:48.39 | arley | just foudn out timewarner cant delay my T1 cutover - yikes - 4 hours to make this work |
| 17:48.40 | Netgeeks | so... it appears that using nat=yes in a sip.conf entry for a UA works when the UA is both behind a nat or on a public ip. At least for a Sipura 841, which I just tested. Any thoughts as to where this might not work? |
| 17:48.56 | Netgeeks | arley: where are you physically located? |
| 17:49.10 | generalhan | can some one answer a queues.conf question for me? i dont want it to announce any of the default stuff like hold time or queue position, but i want it to play a custom message ever 60 seconds... the only way i could figure out how to do it is to set all the default sounds = to silence: http://generalhan.pastebin.ca/53748 but i would like to clean it up a bit, if anyone has any suggestions |
| 17:49.48 | arley | pasadena, ca |
| 17:50.39 | Netgeeks | hrm, first thing I would do if I were you is call digium right away and get them to log into your box and look at the card |
| 17:50.56 | Netgeeks | be sure to tell them that playback works without the card installed, but not with it installed |
| 17:51.16 | generalhan | becareful with that though they messed me up prety baddly when they went into my server |
| 17:51.31 | generalhan | well as long as youre not on a production server you'll be ok ! lol |
| 17:51.40 | arley | netgeeks - yeah i may do that, running AAH not their business version so hope they will. Its a clean install if they can diagnose I will reinstall. |
| 17:52.02 | Netgeeks | second I'm sure there is someone within a couple hours of you that has either digium or sangoma cards in hand... you could chase one down |
| 17:52.14 | Qwell[] | if I had one...I'd loan it to you |
| 17:52.38 | Qwell[] | I've got an x100p in El Monte, if you want it :P |
| 17:52.45 | Netgeeks | arley: digium supports thier cards as well as thier business version. If you have a te110p card that digium built, you will get support |
| 17:53.01 | arley | gotcha |
| 17:53.02 | Netgeeks | I've got about 12 sangoma cards, but I'm 12 hour drive north of you |
| 17:53.28 | arley | a tad far |
| 17:53.34 | Netgeeks | yeah :( |
| 17:53.52 | Qwell[] | charter a small plane |
| 17:54.12 | arley | hmm now that is always an option - well maybe a paper plane |
| 17:54.23 | [TK]D-Fender | generalhan : Set the announce-position=no <- |
| 17:54.58 | Netgeeks | Qwell, you know thats funny, but I have used a Bonanza out of the local airport for some semi-local travel for work... |
| 17:55.08 | Qwell[] | heh |
| 17:55.14 | Netgeeks | If it was still here, I could make it to pasadena in 4 hours |
| 17:55.26 | Netgeeks | alas, it's in Las Vegas at the moment |
| 17:55.30 | Qwell[] | road trip! |
| 17:55.51 | arley | ah - i was in vegas until yesterday |
| 17:56.00 | arley | Interop |
| 17:56.18 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 17:56.19 | Netgeeks | the cost of the flight would be a bit more than the value of the card |
| 17:56.29 | Qwell[] | money is no object :P |
| 17:56.29 | Netgeeks | especially with fuel costs right now |
| 17:56.32 | generalhan | [TK]D-Fender: but then how do i get it to play MY message ever 60 seconds ? |
| 17:56.35 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 17:56.47 | generalhan | ever = every |
| 17:56.49 | Netgeeks | if arley tells me money is no object, I can have a replacement card to him in under 4 hours |
| 17:56.53 | Qwell[] | heh |
| 17:56.54 | arley | just found a posting that also mentiones the ztdummy issue and the exact problem i am seeing |
| 17:57.04 | [TK]D-Fender | generalhan : You aren't reading the WIKI page right.. its all there.... |
| 17:57.22 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:57.33 | [TK]D-Fender | generalhan : "config queues.conf". that'll tell you exactly what to do for each kind of element |
| 17:57.38 | arley | sadly money is always an object - well not in my wallet anyway, no objects today |
| 17:58.16 | brif8 | can someone please shed light on what this is all about http://pastebin.com/700375 I'm trying to find the root causes to Destroying call and scheduling destruction of call for calls that aren't real ? |
| 17:58.30 | generalhan | [TK]D-Fender: ive played with all the options ... the only thing i can do inorder to get that message to play on a cycle is to leave anounce on |
| 17:58.32 | file | registrations... options... |
| 17:58.55 | Netgeeks | rats!!! there is a guy that has a cessna biz jet here at the local airport, I really wanted to call him up and say, I need to get to pasadena CA in less than 4 hours, help me! |
| 17:58.59 | file | brif8: why are you so concerned? |
| 17:59.20 | Qwell[] | Netgeeks: rofl |
| 17:59.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:59.32 | [TK]D-Fender | generalhan : was this not clear enough? periodic-announce = thank-you-message |
| 17:59.32 | [TK]D-Fender | <PROTECTED> |
| 17:59.48 | Netgeeks | I'm guessing he'd do it for probably $1500 to $2500 |
| 17:59.51 | [TK]D-Fender | generalhan : and THIS... ; How often to announce queue position and/or estimated holdtime to caller (0=off) |
| 17:59.51 | [TK]D-Fender | <PROTECTED> |
| 17:59.51 | [TK]D-Fender | <PROTECTED> |
| 18:00.06 | brif8 | file: because I'm trying to hunt down the mysterous cause of dropped calls and echo that happen randomly. |
| 18:00.10 | [TK]D-Fender | generalhan : so set announce-frequency=0 <- |
| 18:00.18 | Qwell[] | Netgeeks: plus your fees, plus fuel, plus airport fees, plus the cost of the card... |
| 18:00.23 | Qwell[] | Netgeeks: yeah, that's be worth it :P |
| 18:00.31 | [TK]D-Fender | generalhan : and periodic-announce-frequency=[whatever] |
| 18:00.37 | brif8 | file: I have searched everything I can think of and now I'm grabbing at straws to find out why |
| 18:00.43 | file | brif8: uh ok! |
| 18:01.03 | froguz | does Asterisk 1.2.7.1 allready support the 'i' and 's' extensions in RealTime mode? |
| 18:01.13 | Qwell[] | should |
| 18:01.19 | Netgeeks | I think the 1.5k to 2.5k would cover his charge... add $700 for the card, and my time would be free (I just wanna fly his jet in the co-pilot seat) |
| 18:01.23 | Qwell[] | it's just a quick query |
| 18:01.26 | Qwell[] | Netgeeks: heh |
| 18:01.33 | brif8 | file: it is very hard to work on the phone and every now and again the call either drops or has a real bad echo. Now yes echo I'm hearing it so it is caused by the other side but that isn't hepling me much |
| 18:01.51 | file | brif8: what's the two sides... zap and SIP? |
| 18:02.18 | brif8 | file: for me IP Phone (SIP) <-> * <-> Internet <-> VoIp Provider <- |
| 18:02.21 | brif8 | file: for me IP Phone (SIP) <-> * <-> Internet <-> VoIp Provider <-> PSTN |
| 18:02.34 | file | brif8: yeah there's nothing Asterisk can do about echo in that arrangement |
| 18:02.56 | file | 'tis just passing audio back and forth |
| 18:03.11 | file | and as for call drops, use sip debug to determine which side is sending it |
| 18:03.22 | brif8 | I agree, but if I can find proof that it's not my side then it have to be the VoIP provider, so I'm trying to rule out all my mistakes |
| 18:03.47 | brif8 | I'm running sip debug but haven't seen anything yet |
| 18:04.10 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 18:04.22 | *** join/#asterisk BadPacket_ (n=BadPacke@unaffiliated/badpacket) |
| 18:04.45 | brif8 | like I said it is very random |
| 18:04.48 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 18:04.52 | tainted- | SIP phone <-> pony <-> asterisk <-> ?? <-> intarweb |
| 18:05.14 | cekc | i want a pony |
| 18:05.39 | mut | *poof* PONY *poof* |
| 18:06.03 | cekc | did my pony just appear and then disappear? |
| 18:06.23 | mut | no |
| 18:06.27 | cekc | sweet |
| 18:06.30 | mut | i just threw 2 smoke balls on the ground |
| 18:06.31 | thogueATredfone | lol |
| 18:06.35 | mut | cause i'm a ninja! |
| 18:06.42 | cekc | ninja > pirate |
| 18:06.42 | brif8 | file: do you understand the pastebin information ? |
| 18:06.46 | mut | NINJA VANISH! |
| 18:06.49 | mut | *poof* |
| 18:06.54 | cekc | NINJA MAGIC! |
| 18:06.55 | file | what pastebin information? |
| 18:07.16 | brif8 | file: http://pastebin.com/700375 |
| 18:07.37 | file | yes, I know what those are |
| 18:07.46 | brif8 | care to share what |
| 18:08.05 | file | could be registrations |
| 18:08.08 | vader-- | is there anyway to adjust the voice of festival? |
| 18:08.12 | file | or OPTIONs |
| 18:09.48 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 18:10.32 | brif8 | ok OPTIONS I'll go with since on the firewall I get "SIP OPTIONS (404 Not Found) (asterisk@207.174.202.4 to s@192.168.25.22:5060)" where 207.174.202.4 is my VoIP provider |
| 18:10.44 | brif8 | but why or what are these |
| 18:11.00 | g__ | call for volunteers: if you have a digium pci card, can you please take a minute and send mail the make and model pci-id? I'm working with the guy who maintains the database, and he thinks I'm on crack. |
| 18:11.22 | froguz | it's possible to include contexts in RealTime? how? |
| 18:11.23 | file | they happen if you have qualify turned on... it sends a SIP packet to the device to make sure it's alive, also used to keep NAT mapping rules open |
| 18:11.55 | *** join/#asterisk simon_ca (n=sedgett@h209-17-142-193.gtcust.grouptelecom.net) |
| 18:13.43 | brif8 | file: aaah Thank you at last yes I have qualify = yes |
| 18:13.46 | brif8 | I'll change it |
| 18:14.02 | file | but that shouldn't affect it... |
| 18:14.24 | g__ | So please.. take 1 minute and run 'lspci -v -n' on your favouriate PBX and send me the result! |
| 18:14.50 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 18:16.20 | *** join/#asterisk hydride (n=nathan@d57-4-109.home.cgocable.net) |
| 18:16.27 | DrPete | If anyone is interested, cisco allow you to download the sip image for the 7960 without a contract, you just use anonymous as the user and your email address as the password. |
| 18:16.33 | hydride | do I need a 2.6 kernel to use asterisk/ |
| 18:16.35 | hydride | ? |
| 18:16.45 | file | to use asterisk? no |
| 18:17.23 | hydride | I've been trying to compile the latest asterisk on a 2.4 box but zaptel doesn't build and nither does asterisk |
| 18:17.36 | file | pastebin what you get. |
| 18:18.27 | Qwell[] | DrPete: eh? |
| 18:18.52 | DrPete | Qwell? |
| 18:19.06 | puzzled | DrPete: you still have to agree with the license... |
| 18:19.41 | DrPete | whats the matter with the licence |
| 18:22.19 | DrPete | ahh nevermind |
| 18:22.23 | DrPete | have fun :) |
| 18:22.34 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
| 18:25.12 | vader-- | hmm kinda sucks festival's voice sounds like crap |
| 18:25.16 | vader-- | is there anyway to adjust it? |
| 18:25.31 | vader-- | i know if other text to speech programs you can adjust pitch and speed, etc. to change the voice |
| 18:26.37 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 18:28.46 | froguz | it seems like nomoby has been interested in realtime.. everytime i had asked something about it i get no reply at all |
| 18:29.37 | budmang | Where is the best freebsd how to install site? |
| 18:30.10 | budmang | for asterisk of course. |
| 18:30.35 | hydride | here's what I get when I try to install zaptel, it's not working :( |
| 18:31.21 | hydride | budmang, I'd imagine Current freebsd release would be the best |
| 18:32.07 | hydride | here's what I get when I try to install zaptel |
| 18:32.11 | hydride | http://pastebin.com/700524 |
| 18:32.11 | budmang | mail# make clean |
| 18:32.11 | budmang | "Makefile", line 28: Missing dependency operator |
| 18:32.11 | budmang | "Makefile", line 31: Need an operator |
| 18:32.26 | wunderkin | budmang, gmake |
| 18:33.30 | budmang | Is there an easy IAX config site? |
| 18:33.48 | budmang | Im just using softphones and calling eachother locally. |
| 18:33.57 | budmang | I have done this before. |
| 18:36.17 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 18:37.58 | Hmmhesays | iax.conf is pretty easy |
| 18:39.12 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:39.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:41.19 | hydride | can anyone give me any clues as to why zaptel won't compile on my box? |
| 18:41.57 | generalhan | can some one help me out with this syntax plz :: GotoIf(${ATCOUNT} < 2?3:6) :: how would i really write that ? IF ATCOUNT is LESS THAN 2 then go to 3 otherwise go to 6 |
| 18:42.15 | Qwell[] | generalhan: surround it in $[] |
| 18:42.29 | generalhan | ohh |
| 18:42.42 | generalhan | just like the callerid example |
| 18:43.39 | Qwell[] | hydride: pastebin the entire error |
| 18:43.46 | Qwell[] | hydride: it looked like that was only a portion of it |
| 18:43.47 | Hmmhesays | Gotoif($[${ATCOUNT} < 2]?3:6) |
| 18:43.54 | Qwell[] | sure |
| 18:44.05 | hydride | Qwell, the rest went off the screen :( |
| 18:44.17 | Qwell[] | hydride: well, we need it...scroll up |
| 18:45.30 | *** join/#asterisk Lino` (n=Lino@i577BD208.versanet.de) |
| 18:45.50 | generalhan | ok i got that part in now ... how about the Set(ATCOUNT=${ATCOUNT}+1) : thats not working for me either how would that look ? |
| 18:45.57 | hydride | how do I get putty to increase the number of lines it retains? |
| 18:46.32 | Qwell[] | generalhan: same |
| 18:47.13 | generalhan | Qwell[]: put in the $[] |
| 18:47.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:50.26 | Blackthorn | I have double rings when calling out our pri, does anyone know what I need to adjust? I've been searching wiki/websites for hours now. |
| 18:50.35 | hydride | http://pastebin.com/700550 that's all of it |
| 18:50.58 | CunningPike | Say I want to do failover dialing: s,n,Dial will fail to s,n+101,Dial - what does the next fail look like? s,n+202?? |
| 18:52.41 | CunningPike | nm - figured it our |
| 18:52.48 | CunningPike | s/our/out/ |
| 18:53.58 | Qwell[] | CunningPike: n+101 is deprecated. |
| 18:54.04 | Qwell[] | Nothing will jump in 1.4 |
| 18:54.19 | CunningPike | Qwell[]: Already? I just started using it :( |
| 18:54.33 | Qwell[] | use things like ${DIALSTATUS} |
| 18:54.57 | CunningPike | Qwell[]: Oh, OK - may as well do it now while I'm re-writing anyway- thanks for the tip' |
| 18:57.30 | ckwall | am i misunderstanding how to forward an extension to voicemail on busy and no answer? exten => 3001,1,Dial(SIP/10.0.0.178/3001) |
| 18:57.31 | ckwall | exten => 3001,2,Voicemail(b3001@10.0.0.178) |
| 18:57.31 | ckwall | exten => 3001,3,Voicemail(u3001@10.0.0.178) |
| 18:57.55 | Qwell[] | ckwall: yes. look at the stdexten macro in extensions.conf.sample |
| 18:58.08 | ckwall | ok... thanks |
| 18:59.50 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:00.13 | *** join/#asterisk suma (n=suma@cm145.gamma29.maxonline.com.sg) |
| 19:00.29 | suma | is there is a pcless asterisk with fxo ? |
| 19:00.37 | Qwell[] | pcless? |
| 19:00.49 | russellb | that would be interesting |
| 19:00.49 | suma | everything in a box not a PC |
| 19:01.04 | russellb | a PC is a box |
| 19:01.07 | Qwell[] | get a netgear router, and an ata |
| 19:01.29 | suma | netgear router ? |
| 19:01.31 | Qwell[] | or, some kind of funky pci-minipci adapter |
| 19:01.35 | suma | which model ? |
| 19:01.37 | Qwell[] | which I'd bet exist |
| 19:01.57 | Qwell[] | dunno |
| 19:02.15 | suma | can i load asterisk on all netgear models ? |
| 19:02.16 | Qwell[] | the wireless one that runs linux |
| 19:02.17 | Qwell[] | no |
| 19:03.55 | MikeJ[Laptop] | betcha it takes a long time to compile if you do it on the netgear ;D |
| 19:04.32 | Qwell[] | no longer than on my sparcstation |
| 19:04.42 | MikeJ[Laptop] | heh |
| 19:04.54 | dlynes_ | CunningPike: That's not quite true....in Asterisk 1.2, you can use the 'j' option to allow 1.2 to use 1.0 behaviour on the jumps |
| 19:05.20 | dlynes_ | CunningPike: But the ${DIALSTATUS} is a better way to go, imho |
| 19:05.35 | dlynes_ | CunningPike: Much more flexible |
| 19:05.57 | Olobola | are t1 pri's dedicated or can you use them as a regular internet connection too? I'm not even sure if this question makes sense. |
| 19:07.07 | dlynes_ | CunningPike: If you want an example, I can email you one |
| 19:07.25 | docelmo | Olobola if its a clear channel T1 you can use it for voice or data |
| 19:07.47 | docelmo | Olobola but only one at a time. If you want to combine them then you will need special hardware for this |
| 19:08.02 | Blackthorn | PRI's are 23 data channels and one used for singnaling information. An isp would use a pri to connect up to 23 modems dial-up users on one. |
| 19:08.26 | Blackthorn | You can also use a pri for voice, I have 23 voice channels hooked into my * box. |
| 19:09.11 | Olobola | I see. So what is your per minute charge for origination from your provider? |
| 19:09.13 | Blackthorn | With certain equipment you can setup any combination of channles/data provided that your provider can do the same. ie 10 data/13 voice channels. etc etc |
| 19:09.38 | Blackthorn | I have flat rate incoming/outgoing (local) |
| 19:10.17 | *** part/#asterisk BadPacket_ (n=BadPacke@unaffiliated/badpacket) |
| 19:11.53 | kaz0358 | is there a way to reset the TE210P card without rebooting the machine? i've tried running ztcfg -f -vvvv and it doesn't make any difference. i'm get weird zhan_zap warnings |
| 19:12.06 | kaz0358 | err chan_zap |
| 19:12.37 | dlynes_ | kaz0358: did you try doing that while asterisk is running? |
| 19:12.49 | kaz0358 | no, i exited from asterisk first |
| 19:13.32 | Olobola | so you can easily order toll-free #'s for a descent per minute charge using a t1 pri (2 or 3 cents)? |
| 19:13.32 | kaz0358 | its very weird.. when any call comes in. it starts out on span 2 and decides to move the call to span 1. but then it says it can't' because the channel is already in use.. and then it hangs up |
| 19:14.00 | dlynes_ | kaz0358: modprobe -r wct4xxp ; sleep 5 ; modprobe wct4xxp ; sleep 10 ; ztcfg |
| 19:14.14 | dlynes_ | kaz0358: no idea about that problem |
| 19:14.37 | kaz0358 | i've gotten into a similiar situation before when asterisk seg faulted in the middle of a call over the te210p. i think it has something to do with this weird Set(CALLERID(name)="something" bug |
| 19:15.36 | kaz0358 | dlynes, nope that didn't work either. i guess i'll reboot |
| 19:15.55 | docelmo | I use PlainVoip the costs for almost all 1+ is 1.1c |
| 19:16.06 | Blackthorn | Ok guess it's time to ask again :P Who knows how to fix a double ring when you dial-out on a pri? |
| 19:16.23 | docelmo | Blackthorn what hardware? |
| 19:17.28 | dlynes_ | kaz0358: which version of the zaptel driver are you using? |
| 19:17.32 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:18.11 | kaz0358 | dlynes, 1.2.5 |
| 19:18.30 | dlynes_ | kaz0358: weird |
| 19:18.37 | *** part/#asterisk opus_ (n=opus@68.216.187.60) |
| 19:18.44 | dlynes_ | but mind you |
| 19:18.51 | dlynes_ | i didn't use zaptel 1.2.5 for long on my pri |
| 19:19.07 | dlynes_ | so i wouldn't know if it's got any major issues or not |
| 19:19.33 | kaz0358 | dlynes, yeah.. i'm really not for sure what the deal is.. i haven't really stressed tested it either to make sure that both spans do indeed work. |
| 19:19.39 | dlynes_ | well, besides the issue i encountered, but that issue might be directly attributable to my hardware, too |
| 19:19.48 | Hmmhesays | someone want to help me with a regexp? |
| 19:19.54 | dlynes_ | Hmmhesays: shoot |
| 19:20.24 | Hmmhesays | say i have ajsadljsflogin:as;ldkfjas;lf and I just want to match login: |
| 19:20.26 | dlynes_ | kaz0358: have you run zttest and patlooptest? |
| 19:20.36 | Blackthorn | spa-2000 -->* ---digium card-->pri's |
| 19:20.41 | dlynes_ | Hmmhesays: Login at the beginning of the line? |
| 19:20.54 | dlynes_ | Blackthorn: don't use ',r' in your dial command |
| 19:21.06 | Hmmhesays | no its nestled in in a string of characters |
| 19:21.24 | hydride | here's my zaptel compile, can someone tell me why it won't compile? http://pastebin.com/700608 |
| 19:21.28 | kaz0358 | docelmo, plainvoip charges to terminate toll free us numbers? you can do that for free... as far as having cheap toll free incomming.. i don't know Olobola |
| 19:21.32 | dlynes_ | Hmmhesays: Yeah, but do you want it to only match login at the beginning of the line, or anywhere on the line? |
| 19:21.39 | Hmmhesays | anywhere in the line |
| 19:21.42 | Hmmhesays | if it exists in that line |
| 19:21.48 | dlynes_ | Hmmhesays: in which utility/language? |
| 19:21.54 | Hmmhesays | perl |
| 19:21.56 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-96.claranet.co.uk) |
| 19:22.21 | dlynes_ | Hmmhesays: if( $string =~ /.*login:.*/ ) { |
| 19:22.35 | Hmmhesays | danke |
| 19:22.40 | Hmmhesays | i'll give that a shot |
| 19:22.41 | Blackthorn | ok just checked there are no ",r" in any of the dialplan. |
| 19:22.50 | dlynes_ | Blackthorn: are you using Bristuff? |
| 19:23.25 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 19:23.37 | Hmmhesays | hrm |
| 19:23.41 | dlynes_ | Blackthorn: if you are, make sure you're not using a ',R' in your dialplan, either |
| 19:23.57 | Hmmhesays | ok, i'm using waitfor() from net::telnet |
| 19:23.57 | Nugget | telnet is eeeeeeevil! |
| 19:24.02 | Blackthorn | I do not know what bristuff is. |
| 19:24.10 | Blackthorn | so i'm probibly not using it |
| 19:24.14 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:24.25 | dlynes_ | Blackthorn: Just check your dial statements then, and make sure they don't have a ',R' at the ends of them |
| 19:24.32 | Blackthorn | calling from ata to ata, ata ---ld provider all works fine. |
| 19:24.50 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 19:24.52 | dlynes_ | ld provider meaning voip provider? |
| 19:24.57 | Blackthorn | the local pri handles only the local traffic. and it's a strait dial command. |
| 19:25.14 | Blackthorn | ld = long distance = sure.. the voip provider. |
| 19:25.35 | brif8 | ok got this when a call dropped, finally got data that relates to a dropped call any ideas ? http://pastebin.com/700622 |
| 19:25.39 | dlynes_ | Blackthorn: any chance of pb'ing a log of the call where you get a double ring? |
| 19:25.53 | dlynes_ | Blackthorn: without sip debug enabled or pri debugging enabled, to start |
| 19:26.06 | dlynes_ | Blackthorn: just high verbosity level |
| 19:27.18 | brif8 | any ideas what would cause those messages ? |
| 19:27.28 | kaz0358 | dlynes_, restarting fixed the te210p problem.. |
| 19:27.30 | Blackthorn | yes i can do that, I need to wait for this call to drop |
| 19:27.52 | dlynes_ | brif8: is that 2 minute delay real, or did you just filter out 2 minutes worth of logs? |
| 19:28.10 | dlynes_ | kaz0358: weird |
| 19:28.20 | dlynes_ | kaz0358: Do you have a crapload of drivers loaded by any chance? |
| 19:28.26 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 19:28.33 | brif8 | dlynes_: no thoses lines follow one after the other in the log file BTW sorry I dropped off last night I got called away |
| 19:28.33 | *** join/#asterisk gmaruz1 (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 19:28.50 | dlynes_ | brif8: I don't even remember |
| 19:28.58 | Olobola | so the only way to avoid going through a flaky provider is to get a t1 pri (800 origination)? |
| 19:28.58 | brif8 | ok |
| 19:29.01 | dlynes_ | brif8: I had to go attend to my server...the kernel update didn't take |
| 19:29.09 | brif8 | ooops |
| 19:29.42 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 19:29.45 | dlynes_ | brif8: the pri card driver is really picky about what order it gets loaded in, and if chan_zap.so is loaded in asterisk, and it can't talk to the pri card, it dominates the machine so much that I can't even ssh in |
| 19:30.01 | kaz0358 | dlynes_, drivers? as in what kind of drivers? kerenel level stuff? |
| 19:30.13 | brif8 | dlynes_: man that is not good to hear |
| 19:30.36 | dlynes_ | brif8: anyways...problems fixed now...I just had to do some major modifications to my startup scripts |
| 19:30.53 | dlynes_ | brif8: and force driver priorities, instead of letting the scripts autohandle it |
| 19:30.59 | dlynes_ | kaz0358: pardon? |
| 19:31.02 | brif8 | dlynes_: glad you got it fixed |
| 19:31.12 | asterboy | <PROTECTED> |
| 19:31.26 | asterboy | that should improve my X100 clone setup |
| 19:31.52 | *** join/#asterisk darby_t (n=tom@abdd92.neoplus.adsl.tpnet.pl) |
| 19:31.53 | dlynes_ | asterboy: what kinda sangoma cards? |
| 19:31.57 | kaz0358 | dlynes_, what kind of drivers are you referring to? i don't have much loaded in.. SATA driver, nic driver, and maybe a couple of other things |
| 19:32.22 | dlynes_ | kaz0358: ah...like sound card drivers, and other such drivers that you're not using |
| 19:32.37 | dlynes_ | kaz0358: also, make sure you disable all hardware in the cmos that you're not using, too |
| 19:32.50 | dlynes_ | kaz0358: a lot of that stuff can steal interrupts, and conflict with telephony hardware |
| 19:33.19 | kaz0358 | dlynes_, i have disabled all hardware that isn't being used.. but linux still detects some stuff, such as the secondary intel nic |
| 19:33.27 | dlynes_ | kaz0358: also if you do an lspci -v | grep IRQ |
| 19:33.34 | hydride | here's my zaptel compile, can someone tell me why it won't compile? http://pastebin.com/700608 |
| 19:33.38 | dlynes_ | kaz0358: make sure you're not sharing any interrupts with the digium card |
| 19:34.01 | kaz0358 | dlynes_, actually i am sharing one.. but it is supposably sharing with the "unused" nic card. maybe that is the problem. |
| 19:34.57 | Hmmhesays | how do i match a string that either ends in # or * |
| 19:35.06 | Hmmhesays | perl regexp |
| 19:35.21 | Qwell[] | #$ ? |
| 19:35.23 | kaz0358 | Olobola, teliax.com has incomming toll free service. i didn't use them for that, but i had no problems with the single DID from them. |
| 19:35.34 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 19:35.56 | hardwire | shiiiiiiit |
| 19:35.59 | dlynes_ | Hmmhesays: \#$, \*$ |
| 19:36.06 | kaz0358 | dlynes_, i guess before we really get this box in production i'll have to move the card to a different slot and see if i can get it to an unused IRQ |
| 19:36.19 | trixter | got a new sangoma a101u (single span t1/e1 card) and a new digium tdm410b (1 fxs module) for sale if anyone is interested msg me |
| 19:36.36 | Hmmhesays | thats right $ is ending in isn't it |
| 19:36.37 | dlynes_ | kaz0358: yeah...don't let it share an interrupt with a nic |
| 19:36.46 | dlynes_ | kaz0358: nics use a lot of interrupts, too |
| 19:36.53 | kaz0358 | hmmhesay or /(\#|\*)$/ |
| 19:36.58 | trixter | device polling is better :P |
| 19:37.14 | dlynes_ | Hmmhesays: there ya go...looks like kaz knows perl a lot better than me |
| 19:37.27 | Olobola | kaz0358: I would really like to avoid going through third party provider if possible. |
| 19:37.31 | kaz0358 | dlynes, shh.. don't tell anyone that :) |
| 19:37.35 | Hmmhesays | haha |
| 19:37.38 | brif8 | dlynes_: any ideas what a SIP debug looks like if the call got dropped |
| 19:37.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:38.06 | dlynes_ | brif8: not offhand, no |
| 19:38.07 | kaz0358 | dlynes, okay.. we'll see if it makes any difference. like i said. it is the unused card. the interface isn't even up according to ifconfig, but who knows. it certainly couldn't hurt to move it. |
| 19:38.18 | dlynes_ | brif8: i've never had a call that got dropped by sip |
| 19:38.24 | dlynes_ | brif8: only zaptel |
| 19:38.30 | brif8 | I wish I could say that |
| 19:38.45 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 19:39.09 | dlynes_ | Hmmhesays: I take it you and buu don't get along? |
| 19:39.12 | gbodemantv | does anyone know this error? |
| 19:39.25 | gbodemantv | WARNING[25573]: format_wav.c:247 update_header: Unable to find our position |
| 19:39.29 | Hmmhesays | me and who? |
| 19:39.39 | gbodemantv | its just keeps looping on my 1.2.4 system |
| 19:39.43 | dlynes_ | Hmmhesays: oh...thought maybe you were avoiding #perl because of buu :) |
| 19:39.53 | Hmmhesays | oh no, i just happened to be in here |
| 19:40.15 | dlynes_ | He can tend to be a bit ornery sometimes |
| 19:40.48 | dlynes_ | kaz0358: also, enable dma on your harddrives |
| 19:41.02 | *** join/#asterisk hydride (n=nathan@d57-4-109.home.cgocable.net) |
| 19:41.03 | gbodemantv | anyone...can't even get to my CLI |
| 19:41.08 | hydride | friggin network :( |
| 19:41.11 | hydride | here's my zaptel compile, can someone tell me why it won't compile? http://pastebin.com/700608 |
| 19:41.15 | dlynes_ | kaz0358: if you're using ide you should be able to do hdparm /dev/hda to see if it's enabled |
| 19:41.23 | dlynes_ | hydride: Which kernel version are you using? |
| 19:41.29 | dlynes_ | hydride: 2.4? |
| 19:41.34 | hydride | yeah |
| 19:42.13 | dlynes_ | your own compiled kernel, or stock kernel from your distribution? |
| 19:42.28 | hydride | I believe it's a stock kernel, but I'm not sure |
| 19:42.30 | hydride | it's on a VPS |
| 19:42.48 | dlynes_ | hydride: why are you trying to run asterisk on a vps? |
| 19:43.00 | trixter | cause they are cheap :P |
| 19:43.04 | hydride | cause I just need sip |
| 19:43.06 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.232) |
| 19:43.09 | hydride | and it's cheap |
| 19:43.15 | trixter | a friend has one and its not bad |
| 19:43.15 | tomcontr3 | how can I clear Asterisk Cache? |
| 19:43.20 | kaz0358 | dlynes_, okay. i'll look into that too. i did read something somewhere that mentioned you should try and avoid SATA with asterisk, but i should be fine if zttest never dips below the magic number even when thrashing the disk |
| 19:43.28 | dlynes_ | tomcontr3: asterisk doesn't cache afaik |
| 19:43.54 | kaz0358 | dlynes, nope.. sata |
| 19:43.56 | dlynes_ | kaz0358: what's the magic number? |
| 19:44.02 | tomcontr3 | but, why could it be that some times I can make calls using one trunk and some times I get 403 error? |
| 19:44.14 | kaz0358 | dlynes, i don't rember off hand.. its something like 99.978 or some such |
| 19:44.15 | hydride | so can someone help me with that error? |
| 19:44.16 | trixter | it caches in /var/ something |
| 19:44.23 | trixter | its a sqlite file iirc |
| 19:44.29 | dlynes_ | tomcontr3: ask your sip provider...they're returning you an error |
| 19:44.37 | Blackthorn | I started the server in verbors mode with -vvvv and watched the call and it looked pretty simple with nothing looking strage. just showed the dial command, ringing, and hangup. |
| 19:44.38 | trixter | find /var -name astdb\* |
| 19:44.49 | trixter | forget the full name, maybe that is it, I just dont remember right now |
| 19:45.17 | gbodemantv | WARNING[25573]: format_wav.c:247 update_header: Unable to find our position ??? |
| 19:45.37 | gbodemantv | anybody have a clue on this one |
| 19:45.46 | gbodemantv | repeats over and over |
| 19:45.48 | tomcontr3 | that ?/var/cache/apt/archives |
| 19:46.10 | trixter | no that would be for debian packages |
| 19:46.19 | trixter | the file has astdb in its name that is all I recall at this time |
| 19:46.24 | dlynes_ | Blackthorn: do you have notice,warning,error, and verbose enabled for the console? |
| 19:46.29 | trixter | it really only caches things like sip registrations and such |
| 19:46.48 | trixter | I had to rm it at one point because of stale info that was breaking stuff |
| 19:47.01 | dlynes_ | kaz0358: do you have any of those numbers dipping below 98%? |
| 19:47.10 | dlynes_ | kaz0358: like even one? |
| 19:47.22 | gbodemantv | qwell: any help on this one? |
| 19:47.28 | Qwell[] | no |
| 19:47.41 | gbodemantv | only found one post about it |
| 19:47.44 | kaz0358 | dlynes_, let me rerun it |
| 19:47.59 | gbodemantv | and that person was asking what I am? |
| 19:48.50 | *** join/#asterisk jmacz (n=jmacz@201.244.243.43) |
| 19:49.04 | kaz0358 | dlynes_, once in a great while it gives 99.987793% |
| 19:49.55 | dlynes_ | kaz0358: and the rest of the time? |
| 19:51.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:51.13 | dlynes_ | gbodemantv: would be nice if they told you what the real error was :) |
| 19:51.15 | kaz0358 | dlynes_, the rest of the time it says 100% |
| 19:51.21 | dlynes_ | kaz0358: ah, ok |
| 19:51.31 | dlynes_ | kaz0358: try patlooptest on each span then, too |
| 19:51.41 | *** join/#asterisk |cleric| (n=dacleric@p54821B99.dip0.t-ipconnect.de) |
| 19:51.43 | dlynes_ | kaz0358: your zttest isn't showing any interrupt issues |
| 19:51.52 | kaz0358 | dlynes, i'm not familiar with that. is that something you do within asterisk? |
| 19:52.02 | dlynes_ | kaz0358: patlooptest? |
| 19:52.11 | Blackthorn | dlynes: I just started asterisk -vvvv |
| 19:52.12 | kaz0358 | dlynes, yeah.. the patlooptest |
| 19:52.24 | dlynes_ | Blackthorn: check your logger.conf...the configuration is in there |
| 19:52.51 | dlynes_ | kaz0358: cd /usr/local/src/.../zaptel-1.2.5/ ; gcc -o patlooptest patlooptest.c ; ./patlooptest /dev/zap/1 |
| 19:53.09 | *** join/#asterisk trbldwine (n=trbldwin@vpn163182.vpn.northwestern.edu) |
| 19:53.16 | *** join/#asterisk incontwin (n=FreePBX9@69.44.163.21) |
| 19:53.36 | dlynes_ | gbodemantv: I've got a solution for you, if you don't mind rolling up your sleeves, getting out a text editor, and recompiling asterisk |
| 19:54.14 | kaz0358 | dlynes, okay.. it is spitting out lots of something '(Error 81553): Unexpected result, 255 != 0, 1 bytes since last error.' |
| 19:54.37 | dlynes_ | kaz0358: yeah...as I understand it that means there's problems with the pri d channel or something |
| 19:54.51 | dlynes_ | kaz0358: I don't know for sure though...that's just what i've been told |
| 19:55.06 | kaz0358 | dlynes, meaning that i have something configured wrong? |
| 19:55.07 | dlynes_ | kaz0358: I get that crap spitting out left, right and center, too |
| 19:55.14 | dlynes_ | kaz0358: not necessarily |
| 19:55.29 | dlynes_ | kaz0358: I'm trying to resolve that issue with mog right now, myself |
| 19:55.47 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 19:55.48 | dlynes_ | kaz0358: I'm going to replace a ribbon cable with a non-80C ribbon cable to see if that solves my problem |
| 19:56.01 | dlynes_ | kaz0358: type hdparm /dev/hda (if you're using ide) |
| 19:56.07 | dlynes_ | kaz0358: and see if dma is enabled |
| 19:56.25 | dlynes_ | kaz0358: dma enabled on your hard drive can cut down on system load, too |
| 19:56.50 | dlynes_ | gbodemantv: did you see my message? |
| 19:56.51 | asterboy | dma no greater the udma 3 though |
| 19:56.57 | kaz0358 | dlynes, okay. i'm doing something pretty straight forward. a single trunk is spread over two T1 spans. there is one primary and backup d channel.. 24 and 48 |
| 19:56.58 | asterboy | no less than 2 |
| 19:57.24 | dlynes_ | kaz0358: yeah...same as me, four spans, only one in use, but other than that, the same |
| 19:57.45 | dlynes_ | asterboy: how do you specify which dma you want to use? |
| 19:57.55 | gbodemantv | why would I need to recompile? |
| 19:57.58 | kaz0358 | dlynes, i'm use a SATA drive and it doesn't really say anything in reguards to DMA.. just that IO support = 0 or 16bit.. and read ahead is turned on |
| 19:58.07 | asterboy | hdparm |
| 19:58.10 | dlynes_ | gbodemantv: because i'm going to give you some changes to make to the code before you recompile |
| 19:58.21 | dlynes_ | kaz0358: i have no idea...never used sata before |
| 19:58.28 | dlynes_ | kaz0358: but maybe asterboy might know |
| 19:58.35 | asterboy | example: |
| 19:58.36 | asterboy | hdparm -d 1 -X udma2 -c 3 /dev/hda |
| 19:59.07 | asterboy | sata is an issue with digium and slower systems |
| 19:59.11 | kaz0358 | asterboy, this is safe to set on a SATA drive?? |
| 19:59.12 | asterboy | ide is preffered |
| 19:59.23 | asterboy | u can't change a sata drive |
| 19:59.27 | asterboy | only ide |
| 19:59.30 | kaz0358 | asterboy, yeah.. i read ide was preffered after we had everything setup... |
| 19:59.41 | asterboy | get an ide and dd |
| 19:59.44 | Qwell[] | why ide? |
| 19:59.49 | Qwell[] | who claims it's preferred? |
| 19:59.55 | asterboy | The Bible |
| 20:00.29 | afrosheen | that's hard to argue |
| 20:00.40 | kaz0358 | asterboy, i havent had any problems until i decided to set the callerid and put quotes around it. somehow that causes things to go very awry. |
| 20:00.47 | Hmmhesays | =~('/^#/') will that match anything starting with a # perl regexp? |
| 20:00.54 | dlynes_ | asterboy: yeah...still doesn't work for me |
| 20:01.14 | dlynes_ | Hmmhesays: =~ /^\#/ |
| 20:01.29 | dlynes_ | Hmmhesays: why are you using parentheses and quotation marks? |
| 20:01.48 | Hmmhesays | oh, that was for net::telnet |
| 20:01.55 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 20:02.17 | kaz0358 | asterboy, strangely not using quotes and everything is all happy when setting callerid. the te210p card has no problems. i have no idea if it is some kind of bug or whether it is an interrupt issue as dlynes suspected. |
| 20:02.36 | dlynes_ | kaz0358: i'm not suspecting anything |
| 20:02.44 | dlynes_ | kaz0358: I was just helping you rule out possibilities |
| 20:02.50 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 20:02.56 | dlynes_ | kaz0358: I don't know enough about the pri cards to suspect anything |
| 20:03.01 | kaz0358 | dlynes, my mistake.. sound trouble shooting |
| 20:03.57 | dlynes_ | kaz0358: as i understand it though, cheap sata's are just pata's inside, and expensive sata's are just scsi inside |
| 20:04.00 | *** join/#asterisk darby_t (n=tom@abdd92.neoplus.adsl.tpnet.pl) |
| 20:04.43 | asterboy | sata sucks your i/o |
| 20:05.02 | kaz0358 | dlynes, really? i never heard that. i usually don't follow hardware too often. |
| 20:05.16 | dlynes_ | kaz0358: yeah...that's just something i've learned recently |
| 20:05.18 | asterboy | Here is the info I based those suggestions on: |
| 20:05.20 | asterboy | http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 20:05.25 | dlynes_ | kaz0358: It kinda surprised me, too |
| 20:05.36 | gbodemantv | dlynes : any idea why this happens |
| 20:05.40 | asterboy | Paragraph 2 |
| 20:05.40 | hydride | make[1]: *** [cdr_tds.o] Error 1 |
| 20:05.41 | hydride | make[1]: Leaving directory `/usr/src/asterisk-1.2.7.1/cdr' |
| 20:05.43 | hydride | any ideas? |
| 20:06.00 | dlynes_ | gbodemantv: well, if you'd let me tell you some changes to make and recompile asterisk |
| 20:06.09 | dlynes_ | gbodemantv: we can find out exactly why you're having that issue |
| 20:06.21 | asterboy | IRQ misses with your Digium hardware can be due to I/O problems on your |
| 20:06.21 | asterboy | system. You may test if you are having I/O problems on your system by |
| 20:06.22 | asterboy | running "hdparm -t /dev/Hard Drive Device". |
| 20:06.23 | dlynes_ | gbodemantv: but without extra information, it's really just a shot in the dark |
| 20:06.36 | kaz0358 | dlynes, when you first start asterisk up, do all your channels restart? is that normall? |
| 20:06.47 | asterboy | This is why I've gone to Sangoma |
| 20:06.47 | dlynes_ | kaz0358: yeah, that's normal |
| 20:07.00 | asterboy | supposedly, they don't have these issues |
| 20:07.01 | kaz0358 | dlynes, more specifically.. i mean my channels restart the first time a call is placed over any of the channels |
| 20:07.08 | dlynes_ | kaz0358: asterisk autorestarts all unused channels periodically |
| 20:07.25 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:07.28 | dlynes_ | kaz0358: and it's not all channels...only channels currently unused |
| 20:07.35 | kaz0358 | dlynes, yeah.. i see that and you can turn it off. but the first call that goes through the te210p gets dropped after asterisk is restarted |
| 20:07.53 | asterboy | bbl...time to collect money so I can buy beer. |
| 20:08.26 | dlynes_ | kaz0358: perhaps it's a race condition that's not being checked for because the call is accepted and, the code that resets the pri channels isn't checking for calls that early in asterisk execution |
| 20:08.40 | dlynes_ | kaz0358: You might want to check bugs.digium.com to see if someone else has reported the issue |
| 20:08.49 | dlynes_ | kaz0358: if they have, you might want to try a follow up to it |
| 20:08.51 | kaz0358 | dlynes, maybe.. it happens even if i wait 30 seconds after asterisk is started. |
| 20:09.08 | dlynes_ | kaz0358: yeah...that's definitely not the way it's supposed to work |
| 20:09.19 | kaz0358 | dlynes, okay.. i'll do some checking. |
| 20:09.29 | dlynes_ | kaz0358: but yes, it is supposed to reset on startup...but only unused channels |
| 20:09.42 | dlynes_ | kaz0358: and it does it during startup |
| 20:09.42 | kaz0358 | i specifically get chan_zap.c:9013 pri_dchannel: Got restart ack on channel 0/1 span 1 with owner |
| 20:09.47 | dlynes_ | kaz0358: not 10 seconds later |
| 20:10.24 | kaz0358 | dlynes, yeah.. that's what i thought. it isn't too critical as i don't restart asterisk all that often |
| 20:10.51 | dlynes_ | kaz0358: yeah, but you should still report the bug, if you can reproduce it every time |
| 20:10.59 | dlynes_ | kaz0358: and let them know it's reproducable |
| 20:11.11 | dlynes_ | kaz0358: and tell them the steps they need to take to reproduce it |
| 20:11.22 | dlynes_ | kaz0358: including what card you're using |
| 20:11.35 | dlynes_ | kaz0358: and preferably a full lspci -v dump |
| 20:11.42 | kaz0358 | dlynes, okay.. sounds reasonable. maybe it has something to do with what is on the other side of the T1? i don't know. it wouldn't surprise me if avaya wasn't going by the specification 100% |
| 20:11.55 | dlynes_ | kaz0358: the more information you can give them about the environment you're running in, the better |
| 20:12.18 | dlynes_ | kaz0358: could be...is this not hooked up to a pri then? |
| 20:12.24 | kaz0358 | dlynes, okay. after i have time to get the card on a different irq, i'll report it if it is still happening |
| 20:12.39 | dlynes_ | kaz0358: also, is it a digium card? |
| 20:12.47 | kaz0358 | dlynes, yeah it is a digium card |
| 20:12.49 | dlynes_ | kaz0358: if it is, they do offer free support |
| 20:13.00 | froguz | somebody knows how to send a new feature/improvement to bugs.digium.com? |
| 20:13.11 | dlynes_ | kaz0358: just phone them up, let them know what the problem is you're having, and they'll take a look at it, and offer suggestions and/or fix it for you |
| 20:13.28 | kaz0358 | dlynes, okay.. i'll give them a call. i'm not really all that busy today :) |
| 20:13.42 | dlynes_ | kaz0358: 1-888-LINUX-ME, I think it was |
| 20:13.48 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 20:14.33 | dlynes_ | kaz0358: but yeah...get your interrupt problem fixed before you call them |
| 20:14.42 | dlynes_ | kaz0358: that way that's ruled out ahead of time |
| 20:15.12 | kaz0358 | yeah.. |
| 20:16.44 | kaz0358 | random question.. asterisk supports speex as a wideband codec, but it downsamples often times because the code base is written around 8khz |
| 20:16.52 | kaz0358 | ? |
| 20:16.52 | dlynes_ | gbodemantv: i guess you're not all that concerned with your problem, after all |
| 20:17.14 | dlynes_ | kaz0358: everything in asterisk is 8Khz, pretty much |
| 20:17.45 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 20:17.55 | russellb | dlynes_: there is no "pretty much", it absolutely is :) |
| 20:18.04 | dlynes_ | heh |
| 20:18.24 | Strom_C | i wondered why this channel seemed so quiet... |
| 20:18.25 | kaz0358 | so it would be more than a tiny tweaking to make it 32khz? :) |
| 20:18.27 | dlynes_ | russellb: well, no |
| 20:18.34 | dlynes_ | russellb: the interrupts are 1KHz :) |
| 20:18.45 | russellb | dlynes_: that's zaptel |
| 20:18.49 | russellb | :D |
| 20:18.51 | dlynes_ | kaz0358: it would require a complete rewrite :) |
| 20:20.02 | afrosheen | I thought asterisk 2.x was supposed to include 16 bit audio support...read it somewhere |
| 20:20.19 | dlynes_ | afrosheen: there is no asterisk 2.0 |
| 20:20.42 | dlynes_ | afrosheen: asterisk 1.4 hasn't even come out yet |
| 20:20.44 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 20:21.21 | afrosheen | I realize that, but I think it was in a digium dev log for the future of * |
| 20:21.33 | dlynes_ | ah |
| 20:22.15 | dlynes_ | 16-bit audio is a bit less of a recode than changing the frequency though, too |
| 20:22.36 | afrosheen | anyone have a link to the digium blogs? |
| 20:22.49 | afrosheen | nm just found it |
| 20:23.53 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 20:27.15 | dlynes_ | eh? |
| 20:27.31 | dlynes_ | I thought APIC was supposed to help with zaptel problems? |
| 20:27.49 | dlynes_ | According to that wiki, it exascerbates the problems |
| 20:29.23 | kaz0358 | minus some skype hardware, i'm not for sure there is any sip widband codec hardware |
| 20:30.05 | kaz0358 | so it doesn't matter too much that asterisk doesn't native do wideband codecs.. |
| 20:30.21 | justinu|laptop | supposed the gxp2000 supports wideband g722 |
| 20:30.35 | hardwire | file, poke |
| 20:30.44 | file | prod |
| 20:31.02 | *** join/#asterisk espn (n=DA-MAN@pulse.landshark.net) |
| 20:32.06 | kaz0358 | justin, hmm.. i guess i'm wrong |
| 20:32.43 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 20:32.53 | justinu|laptop | i dunno any other device that does, kaz |
| 20:33.00 | justinu|laptop | just that one crappy phone |
| 20:35.16 | kaz0358 | are there any hardware phones that do speex of any kind? |
| 20:36.14 | Nivex | zwelch: and a pony? |
| 20:36.33 | zwelch | Nivex: well, while i'm conjouring up whimsical dreams, why not? :) |
| 20:36.56 | dlynes_ | what's so great about speex, anyways? |
| 20:37.07 | dlynes_ | I thought it was crappy quality? |
| 20:37.15 | zwelch | freedom is never crappy |
| 20:37.48 | zwelch | i think that sums up my personal anser to your first question ;) |
| 20:37.53 | tainted- | SIP phone <-> pony <-> asterisk <-> ?? <-> intarweb |
| 20:37.55 | *** join/#asterisk bewmIES (n=rwm@orion.guardiandigital.com) |
| 20:37.58 | dlynes_ | not really |
| 20:38.26 | Nivex | tainted-: IP over Pony Express? |
| 20:38.34 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 20:38.39 | boch | hi |
| 20:38.59 | dlynes_ | tainted-: got any bites on your nugget pranks yet? |
| 20:39.06 | tainted- | Nivex it does Rocky Mountain traversal |
| 20:39.20 | zwelch | dlynes_: okay. by that i meant, "it's part of a royalty-free media encoding and encasupation frameworks with GPL implementations, and that makes it pretty great to me" |
| 20:39.25 | tainted- | dlynes i'm working on it |
| 20:39.53 | dlynes_ | zwelch: ah...but aren't most of the codecs used by asterisk royalty free? |
| 20:40.44 | dlynes_ | except for of course g.723 and g.729 |
| 20:40.47 | zwelch | it's true that there are alternatives, some of which may be better quality; however, none of them fit as nicely into a technology architecture like speex does with OGG |
| 20:41.05 | zwelch | (at least, from what i have ascertained in my assessment of the relevant technologies) |
| 20:41.13 | boch | is this ok? Goto(IF(${CALLERIDNUM:0:6}!=114311?[8])) |
| 20:41.54 | dlynes_ | i c |
| 20:42.25 | russellb | boch: not quite |
| 20:42.26 | Qwell[] | boch: Use GotoIf |
| 20:42.32 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 20:42.47 | boch | Qwell[]: oh didnt know that, thanks |
| 20:43.45 | elvisthedj | anybody here using chan-sccp that could perhaps decipher this error message http://pastebin.com/700771 ? |
| 20:44.25 | dlynes_ | There ya go, Qwell...there's your chance :) |
| 20:44.53 | elvisthedj | i'm guessing the answer will be "your firmware is outdated" .. ahh .. if only I could update it :) |
| 20:44.56 | *** join/#asterisk sebirc^ (n=seb@86.72.21.13) |
| 20:47.23 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 20:47.38 | trixter | got a new sangoma a101u (single span t1/e1 card) and a new digium tdm410b (1 fxs module) for sale if anyone is interested msg me |
| 20:47.41 | dlynes_ | elvisthedj: you mean chan_skinny? |
| 20:47.57 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:49.00 | elvisthedj | dlynes_: nope. chan_sccp |
| 20:49.01 | dlynes_ | elvisthedj: sccp_socket.c doesn't seem to be part of the asterisk distribution |
| 20:49.11 | dlynes_ | elvisthedj: nor does chan_sccp |
| 20:49.12 | elvisthedj | dlynes_: it's a 3rd party app |
| 20:49.19 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 20:49.33 | dlynes_ | elvisthedj: have you tried chan_skinny? |
| 20:49.47 | dlynes_ | elvisthedj: The author of that is on here, regularly |
| 20:49.48 | generalhan | can some one tell me if the n+101 priority is no good anymore ... where does ChanIsAvail go if no Chans are Available ?? |
| 20:50.05 | dlynes_ | generalhan: n+101 is still good |
| 20:50.07 | elvisthedj | dlynes_: It's supposedly an improved chan_skinny .. i can't get either to work, but i can't upgrade my 7940 (at least not now) so i'm trying to use what's currently loaded (a very old cm) |
| 20:50.12 | dlynes_ | generalhan: but you need to enable priority jumping |
| 20:50.28 | dlynes_ | generalhan: or use the 'j' option to your dial command |
| 20:50.29 | generalhan | dlynes_: what do you mean ? |
| 20:50.32 | generalhan | ohh ok |
| 20:50.42 | generalhan | so as long as i use the j option n+101 is ok ! |
| 20:50.43 | generalhan | ? |
| 20:50.48 | dlynes_ | generalhan: priorityjumping=yes in your global context |
| 20:50.58 | dlynes_ | generalhan: or the 'j' option to your dial command |
| 20:51.07 | dlynes_ | generalhan: if you don't wnat it to be the default on every call |
| 20:51.22 | generalhan | dlynes_: cool thanks ill just use the j option |
| 20:51.28 | dlynes_ | generalhan: or you can check ${DIALSTATUS} |
| 20:51.40 | dlynes_ | generalhan: ${DIALSTATUS} is much more flexible |
| 20:53.21 | generalhan | dlynes_: please explain |
| 20:53.28 | *** join/#asterisk catch23[work] (n=catch23@hosta.sixcontinentshotels.com) |
| 20:54.11 | generalhan | i want to set up my macro to check to see if that channel is avail. before i dial that exten, if not straight to VM no 1 ring or anything like htat |
| 20:56.22 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 20:56.25 | mutilator | wheres a good place to take a vacation for like 3-4 days? |
| 20:57.56 | dpryo | The basement |
| 20:58.04 | generalhan | mutilator: take a 3 day mexican cruise ! |
| 20:58.04 | generalhan | haha |
| 20:58.13 | mutilator | lookin for something a lil cheaper |
| 20:58.16 | generalhan | lol |
| 20:58.27 | mutilator | only got like $1000 for 2 ppl plus a lil extra |
| 20:58.34 | generalhan | the cruise itself is CHEAP i just went on that same one ... but our bar bill was $1500 |
| 20:58.43 | dlynes_ | generalhan: exten => x,x,ChanIsAvail(...) ; exten => x,x,GotoIf($["${AVAILSTATUS}"="blehbleh"]?x:x) ; |
| 20:59.18 | dlynes_ | mutilator: Canada! |
| 20:59.19 | generalhan | dlynes_: i know how to do that ... i was asking you to explain how i would use ${DIALSTATUS} instead of ChanIsAVail |
| 20:59.51 | mutilator | just wanted to suprise the gf with a trip, dunno what to do yet |
| 20:59.51 | *** join/#asterisk daguerro (i=dag@lss-67-102.ee.itb.ac.id) |
| 21:00.01 | mutilator | take a week off around july 4th |
| 21:00.02 | dlynes_ | generalhan: exten => x,x,Dial(...) ; exten => x,x,GotoIf($["${DIALSTATUS}"="blehbleh"]?x:x) ; |
| 21:00.08 | generalhan | hahaha |
| 21:00.15 | generalhan | dlynes_: haha thanks ! |
| 21:00.22 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:00.47 | generalhan | what is the output of AVAILSTATUS ? 1 for avail 0 for not ? |
| 21:02.24 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-53.claranet.co.uk) |
| 21:03.14 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 21:03.22 | CunningPike | Can someone give me a quick example of a regex in GotoIf - here is what I have: exten => s,n,GotoIf($[${ARG1:-4:4} : \(80[0-1][0-9]\)]?:DNV), but the parser doens't like it |
| 21:03.41 | Qwell[] | need to use the REGEX function, or something |
| 21:03.44 | *** join/#asterisk Derkommissar (n=Derkommi@209.168.221.130) |
| 21:04.12 | CunningPike | Qwell[]: So The Book is wrong then? |
| 21:04.16 | mutilator | so no ideas? |
| 21:04.22 | Qwell[] | oh, that isn't regex |
| 21:05.17 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 21:06.06 | generalhan | Qwell[]: what is the returned values of ${AVAILSTATUS} ? |
| 21:06.38 | CunningPike | Qwell[]: What isn't? |
| 21:07.48 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 21:09.37 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 21:10.18 | CunningPike | nm - fixed it myself |
| 21:10.21 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 21:10.53 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:10.56 | robin_sz | meep? |
| 21:11.27 | robin_sz | I've got a GXP2000 right? and .. get this .. the display has writing on it!!! |
| 21:12.07 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 21:12.07 | elvisthedj | Well, for the first time ever (about 4 months or so) I got my 7940 to register with asterisk using chan_skinny.. It cleared the 7940's screen and put a time/date stamp.. then after about 10 seconds, the phone reboots. If someone can take a look at this and point me to some info, i'd be very greatful http://pastebin.com/700820 |
| 21:12.33 | robin_sz | s/greatful/grateful/ |
| 21:12.47 | elvisthedj | :p |
| 21:13.23 | robin_sz | you should be grateful, your atleast had a display .. mines been blank for months, but now it has writing! :) |
| 21:13.37 | riddlebox | does anyone know how to access the sipura ata devices? |
| 21:13.54 | elvisthedj | robin_sz: A display is a step in the right direction, but a phonecall would be nice ;) |
| 21:13.54 | robin_sz | ok, so mines lost the ability to connect to things, but hey. I can see what its doing now |
| 21:14.15 | Olobola | I wonder if people were stealing minutes from nufone |
| 21:14.24 | robin_sz | well, it sounds like we both hove equally useful devices then :) |
| 21:14.26 | elvisthedj | i'm sure my problem is this outtdated firmware i can't seem to update :( |
| 21:15.00 | Zodiacal | anyone know why the play command stoped working? it was working fine yesterday... |
| 21:15.03 | robin_sz | im sure my probles started when I updated |
| 21:15.17 | Derkommissar | Sup peoples! |
| 21:15.17 | Zodiacal | it just sits their, i have to press ctrl+c |
| 21:15.27 | elvisthedj | robin_sz: which type are you using? did u go for the sip upgrade? |
| 21:15.28 | robin_sz | s/their/there/ |
| 21:15.37 | Zodiacal | there |
| 21:15.51 | elvisthedj | robin_sz: i'd settle for any type that is actually current enough to talk to asterisk |
| 21:15.59 | robin_sz | yeah, I did the 1.1.x series update to my gxp |
| 21:16.38 | elvisthedj | App load = P00306000400 |
| 21:16.44 | robin_sz | anyway, now it has writing |
| 21:16.51 | robin_sz | im sooooooo happy |
| 21:19.09 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:19.21 | *** join/#asterisk pigpen2 (n=mark@65-194-27-53.block4.gvtc.com) |
| 21:21.48 | dlynes_ | generalhan: it's a string |
| 21:22.11 | dlynes_ | generalhan: chekc the chanisavail.c file to find out what all the possible strings are |
| 21:22.28 | Derkommissar | anyone here know how to use the # awnser confirmation using SIP instead of ZAP ? |
| 21:22.40 | generalhan | dlynes_: i figured it out LOL i just did 2 test calls an NoOP'd the results ! lol |
| 21:22.55 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 21:23.03 | generalhan | dlynes_: so ive got it now ! |
| 21:24.56 | Zodiacal | asterisk can use the sound card, but the play command stoped werkin... |
| 21:25.05 | Zodiacal | anyone know how i can diagnose that? |
| 21:25.10 | Zodiacal | centos 4.2 |
| 21:27.54 | Zodiacal | hrmm.. |
| 21:28.01 | Zodiacal | but ztdummy was running allways |
| 21:31.57 | dlynes_ | generalhan: heh |
| 21:32.19 | *** join/#asterisk ToTo (n=ToTo@host191-138.pool872.interbusiness.it) |
| 21:35.52 | *** part/#asterisk ckwall (n=ckwall@63.149.122.94) |
| 21:36.30 | generalhan | dlynes_: well i thought i had it but apperently i do not. http://generalhan.pastebin.ca/53769 this is how i have it setup but it doesnt work. it takes the first call into the phone ... then any others while the phone is busy and nothing happens you just sit there to silence |
| 21:38.17 | generalhan | so i obviusly have something wrong im just not sure what ... cause it looks good to me ! |
| 21:38.43 | Olobola | I wonder if hurriane electric has PRI's |
| 21:39.59 | [TK]D-Fender | generalhan : What is the point of chanisagain in there in the first place? If it isn't availble during the dial it will just continut on to voicemail ANYWAYS. |
| 21:40.06 | dlynes_ | generalhan: damn...you're such a problem child you have your own private pastebin |
| 21:40.30 | generalhan | dlynes_: hahaha its not my fault the first time i set it up i thought that i HAD to create an account ! |
| 21:40.55 | generalhan | [TK]D-Fender: so if i have the chanisavail and its NOT avail it will just go to the next one automagically ? |
| 21:41.45 | [TK]D-Fender | generalhan : no magic, thats jsut what dial does |
| 21:44.12 | riddlebox | does anyone have a sipura 2100? |
| 21:46.42 | *** join/#asterisk adorah (n=FreePBX5@87.69.73.175) |
| 21:48.07 | ioerror | if I want to dial two different channels with dial() is there anything wrong with doing: dial(${SIPPHONE}&${CELLPHONEVIAIAX2}) |
| 21:48.25 | ioerror | I've seen that example with Zap interfaces but not with say, sip and iax2 channels |
| 21:48.27 | *** join/#asterisk cr_0 (n=y@toronto-HSE-ppp4334596.sympatico.ca) |
| 21:48.47 | ioerror | and it only seems to ring the sip device and it never tries to even dial the iax2 string |
| 21:48.51 | qdk | ioerror: the nature of the channel is not important. |
| 21:48.57 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:49.02 | *** part/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 21:49.04 | [TK]D-Fender | &*#$$##$($#$ |
| 21:49.18 | [TK]D-Fender | \f'n electrical failures |
| 21:49.27 | robin_sz | [TK]D-Fender, umm, that a perl regex, right? |
| 21:49.28 | ioerror | qdk, i assumed that but then I was unable to make the SIP phone ring at the same time as the iax2 line |
| 21:49.31 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 21:49.41 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 21:49.56 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:49.57 | cr_0 | gentoo has 1.2.x masked as unstable. does the asterisk community consider it stable (for home use anyway)? should i move to it, or should i expect crashes. |
| 21:50.07 | [TK]D-Fender | robin_sz : Yeah it says "kill my fucking landlord" |
| 21:50.09 | qdk | ioerror: is the IAX trunk up? |
| 21:50.14 | cr_0 | i'm insterested in realtime asterisk, which seems to be only in 1.2.x which is why i ask. |
| 21:50.25 | ioerror | qdk, yes, i can get and send calls with it |
| 21:50.33 | ioerror | i just can't ring it at the same time as the sip phone |
| 21:50.49 | ioerror | my sip string is expanded to: SIP/grandstream|20|r |
| 21:50.50 | [TK]D-Fender | generalhan : So yeah, no need for Chanisavail in your case. And you should be using a STDEXTEN varient to handle ringing your phones 95% of the time |
| 21:51.04 | Altair256 | any familiar with troubleshooting incoming DID issues with a T1 (E&M Wink)? |
| 21:51.05 | qdk | ioerror: have you tried placing the IAX first? |
| 21:51.22 | cr_0 | maybe i could do a quick poll. who i using 1.0.x and who is on 1.2.x? |
| 21:51.33 | qdk | ioerror: i never tried calling more than 1 SIP, but i works perfectly with 2 IAXs. |
| 21:52.09 | qdk | cr_0: waste of time backporting the feature. |
| 21:52.47 | cr_0 | qdk: i agree. i'm wouldn't backport either. just wondering if 1.2.x is stable enough to move to, or should i wait? |
| 21:53.01 | adorah | Does anyone here use Asterisk CDR for billing? |
| 21:53.09 | Qwell[] | cr_0: 1.4 will be released soon...time to upgrade |
| 21:53.11 | ioerror | qdk, no, i hadn't done that, i can try, but it seems that this is a problem one way or another, perhaps with my SIP string? |
| 21:53.40 | cr_0 | Qwell: heh, i should stay one step behind you are saying :-) |
| 21:53.49 | cr_0 | Qwell: is there a roadmap published? |
| 21:54.09 | qdk | cr_0: well its released as stable, so I should think so... i have both 1.2.5 and 1.2.7.1 in production, and they seem quite stable... im having a few issues with chan_SS7 though- |
| 21:54.25 | Qwell[] | cr_0: yes |
| 21:55.42 | Druken | any wine drinkers here? |
| 21:55.49 | hardwire | sorta |
| 21:55.51 | hardwire | atleast for me |
| 21:55.56 | qdk | ioerror: well timeout and ring should work fine... perhaps leaving them out when dialing more than 1 and add them like this chan1&chan2&chan3,timeout,ring... doesnt make much sense having different timeouts for the channels. |
| 21:55.58 | hardwire | I seem to have a vondness for raspberry wines |
| 21:56.01 | hardwire | vond/fond |
| 21:56.13 | trixter | depends on the wine, my father has made some really good stuff but he also made some stuff that tastes like kerosene |
| 21:56.16 | hardwire | also referred to as fuck me up juice |
| 21:56.18 | Druken | i'm on the hunt for a good cheap wine :) |
| 21:56.23 | adorah | If we were sober had we have our heads sunken into asterisk? |
| 21:56.34 | hardwire | there is an $11 bottle of this raspberry wine that gets about everybody Iknow tippy |
| 21:56.43 | hardwire | its all skinny |
| 21:56.48 | ioerror | qdk, well, the timeout for my cell phone is different than my sip phone but that makes sense |
| 21:57.07 | [TK]D-Fender | Druken : Plenty of newspapers have wine reviews for those on a budget and you can get 4 * stuff for about $20 |
| 21:57.52 | adorah | I can get a decent bottle locally for under 2.5US$ |
| 21:58.51 | Druken | i dun want a 5 bottle... i know i said cheap.. but please :) |
| 21:59.44 | generalhan | [TK]D-Fender: it doesnt work ... when i call that extension and they are on the phone, * does nothing ... it just sits there. doesnt go to the next priority or anything |
| 21:59.56 | Altair256 | Druken, to my knowledge there is not a single channel on freenode that involves wine (the drink), possibly trying another IRC server would help? >.> |
| 22:00.27 | Druken | are you trying to be a smart ass? cause it ain't worken... |
| 22:01.18 | [TK]D-Fender | generalhan : repastebin |
| 22:02.02 | budmang | when i have all the right settings in my idefisk i get nothing on my asterisk CLI as in errors but it times out. If I set something incorrect in IDEfisk i see the error on asterisk. any ideas? |
| 22:02.05 | [TK]D-Fender | Druken : Try asking Hmmhesays' Ex ;) |
| 22:02.24 | Druken | hehehe |
| 22:02.43 | Druken | and that would be? |
| 22:02.52 | [TK]D-Fender | I am *so* bad.... it simply AMAZING |
| 22:02.58 | qdk | ioerror: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial <- the timeout and such are only present once. |
| 22:03.02 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/53775 |
| 22:03.37 | [TK]D-Fender | generalhan : oh god that is so wrong... you made it WORSE |
| 22:03.38 | budmang | any ideas? |
| 22:03.42 | generalhan | HAHAHA |
| 22:03.48 | generalhan | [TK]D-Fender: i ALWAYS make it worse ! |
| 22:03.48 | Qwell[] | anybody running rh/fedora/centos? |
| 22:03.53 | ioerror | qdk, if i place iax2 in front, it rings both and it automatically passes CID |
| 22:04.00 | generalhan | Qwell[]: i use FC4 |
| 22:04.06 | Qwell[] | generalhan: Could you test something for me? |
| 22:04.11 | generalhan | whats that ? |
| 22:04.14 | [TK]D-Fender | generalhan : http://generalhan.pastebin.ca/53776 |
| 22:04.16 | Qwell[] | nevermind |
| 22:04.16 | qdk | ioerror: does the IAX string have timeout and such? |
| 22:04.25 | generalhan | Qwell[]: ?? ok ? |
| 22:04.38 | Qwell[] | I think I just need centos, from somebody using trunk |
| 22:04.45 | ioerror | qdk, nope, you've got my bug. thanks for finding it! |
| 22:04.52 | qdk | ioerror: np |
| 22:05.10 | generalhan | [TK]D-Fender: what is that ? that is just how to dial a number ... i need to see if they are avail first ... if they are THEN dial ... if they ARENT then go STRAIGHT to VM, no ringing the phone even once |
| 22:05.34 | generalhan | I feel bad now ... people here dont even think i can make a Dial() statement correctly ! lol |
| 22:05.53 | qdk | generalhan: use chanisavail |
| 22:06.02 | generalhan | qdk: thats what im doing |
| 22:06.05 | generalhan | well trying to do |
| 22:06.13 | [TK]D-Fender | generalhan : if they aren't available Dial aborts instant anyways! |
| 22:06.19 | generalhan | no |
| 22:06.23 | [TK]D-Fender | generalhan : AH, I see. |
| 22:06.26 | [TK]D-Fender | gen.... |
| 22:06.27 | *** join/#asterisk [Airwolf] (n=airwolf@ip147-245.dsl.introweb.nl) |
| 22:06.31 | [TK]D-Fender | generalhan : 1 sec |
| 22:06.32 | generalhan | [TK]D-Fender: these people have callwaiting on their phones and i dont want it to beep at them EVER |
| 22:06.36 | generalhan | [TK]D-Fender: k |
| 22:07.00 | qdk | generalhan: exten => _XX.,n(cc),ChanIsAvail(${IAXTRUNK01-RES}) |
| 22:07.29 | generalhan | qdk: i have the chanisavail part in there im working on the GotoIf so that if they arent avail it goes straight to VM |
| 22:08.08 | [TK]D-Fender | generalhan : Here http://generalhan.pastebin.ca/53777 |
| 22:08.12 | Druken | beep beep :) |
| 22:08.20 | qdk | generalhan: then the next jumping point will be -> exten => _XX.,cc+101,voicemail()... |
| 22:08.48 | generalhan | [TK]D-Fender: ok thats almost the same thing that i had in the GotoIf i wonder why it wouldnt work that way ? |
| 22:08.52 | generalhan | ill try that one though |
| 22:09.49 | qdk | generalhan: use NoOp() for debugging... to print varables and "locations". |
| 22:09.51 | Druken | i'd like to have someone that doesn't have voicemail, then i'd use the voicemail as a playback "this person obviously isn't home or doesn't want to talk to you, hang up the god damn phone!" |
| 22:10.39 | qdk | dr0ck: playback(fuck_off_sound) instead of the voicemail part. |
| 22:10.43 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 22:11.01 | Druken | qdk: exactly :) hehe |
| 22:11.02 | qdk | Druken: that was for you. |
| 22:11.05 | generalhan | [TK]D-Fender: you know what ... something else is wrong then .. cause thats not working either ... in fact it is doing the EXACT same thing it was when i had the GotoIf statment in there |
| 22:11.13 | Druken | qdk: i figured as much |
| 22:11.21 | qdk | Druken: :-D |
| 22:11.47 | hydride | I keep getting this error: May 5 18:10:21 NOTICE[11070]: chan_sip.c:7708 handle_request: Registration from 'Nathan <sip:1000@64.20.44.10>' failed for '24.57.7.96' |
| 22:11.51 | Druken | i'd monitor it, just to hear the reactions :) |
| 22:11.52 | hydride | any ideas? |
| 22:12.14 | qdk | hydride: turn of the VoIP unit. |
| 22:12.15 | elvisthedj | jeez.. i lost my cisco support contract. i had it right here, but I think my dog ate it. anybody got an sccp image newer than PC0303010100 :D |
| 22:12.31 | hydride | qdk, you mean x-lite? |
| 22:12.48 | qdk | hydride: SoftVoIP unit. :-D |
| 22:13.11 | hydride | I turned it off, do I just turn it back on? |
| 22:13.16 | [TK]D-Fender | generalhan : my bad... typo |
| 22:13.22 | [TK]D-Fender | generalhan : http://generalhan.pastebin.ca/53779 |
| 22:13.40 | [TK]D-Fender | generalhan : Had 7000 & 7007 mixed in the previous one |
| 22:13.41 | qdk | hydride: something is trying to register on your *, but it doesnt allow it... for some really. |
| 22:13.57 | hydride | qdk, how do I find that reason? |
| 22:14.04 | ioerror | qdk, do you know how to test music on hold funcionality easily? |
| 22:14.10 | generalhan | [TK]D-Fender: i didnt just copy and paste hat you had ... i used what you had with my numbers so that typo wasnt the reason |
| 22:14.21 | ioerror | I have an mp3 in my moh directory but asterisk never seems to play it or even have an error |
| 22:14.24 | [TK]D-Fender | generalhan : pastebin what you've got back. |
| 22:14.29 | qdk | [TK]D-Fender: and you spelled default wrong. :-P |
| 22:14.35 | generalhan | [TK]D-Fender: ok 1 sec |
| 22:14.35 | *** join/#asterisk ghost99 (n=neville@222-153-96-3.jetstream.xtra.co.nz) |
| 22:14.35 | ioerror | manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 22:14.41 | [TK]D-Fender | qdk : SHUP YUO! |
| 22:14.41 | ioerror | that's my musiconhold.conf |
| 22:14.44 | simon_ca | Anyone tested sending CNAM using FACILITY_IE as opposed to DISPLAY_IE? Connect to a 5E via PRI. |
| 22:14.48 | Druken | ioerror: exten => 7,1,musiconhold() |
| 22:14.51 | qdk | ioerror: dial in? :-D |
| 22:15.32 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:15.37 | qdk | hydride: add the correct entry in you sip.conf and login with that info, if thats what you want? |
| 22:15.43 | qdk | [TK]D-Fender: :-) |
| 22:16.10 | *** join/#asterisk ToTo (n=ToTo@host191-138.pool872.interbusiness.it) |
| 22:16.42 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/53780 |
| 22:17.12 | qdk | ioerror: hmm... do you have mp3 support? do the files work? |
| 22:17.14 | ioerror | Druken, i have it in a dial context |
| 22:17.31 | ioerror | qdk, the mp3s play fine in xmms and i assume mpg123 can play them as well |
| 22:17.38 | Druken | so it plays moh instead of rings? |
| 22:18.02 | qdk | generalhan: goto is 103 |
| 22:18.26 | qdk | generalhan: priority(2)+101 = 103 |
| 22:18.39 | ioerror | drray, yes |
| 22:18.40 | [TK]D-Fender | generalhan : You mangled the order.... |
| 22:18.42 | ioerror | Druken, yes |
| 22:18.46 | qdk | ioerror: you using x64? |
| 22:18.53 | [TK]D-Fender | generalhan : 102 now need to be 103 <- |
| 22:18.53 | ioerror | qdk, no, i686 |
| 22:19.07 | Druken | ioerror: either call yourself, or get someone to call you |
| 22:19.09 | qdk | generalhan: use the n priority with labels. |
| 22:19.10 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 22:19.10 | ioerror | ah, i finally got an error |
| 22:19.11 | ioerror | May 5 15:18:36 WARNING[4179]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class '') on channel SIP/grandstream-8fee |
| 22:19.21 | ioerror | so at least i'm getting somewhere |
| 22:19.46 | carrar | randstream ever get tftp config IP from the DHCP server working yet? |
| 22:19.46 | qdk | ioerror: oh, you lack to add a class to you sip setup? |
| 22:19.49 | *** join/#asterisk espn (n=DA-MAN@pulse.landshark.net) |
| 22:19.49 | carrar | +g |
| 22:20.21 | ioerror | qdk, hrm, i'm unfamiliar with this concept, can you link me ? |
| 22:20.41 | qdk | ioerror: when your sip has registered you can just dial 7 if you use "Druken ioerror: exten => 7,1,musiconhold()" an it will execute that extension. |
| 22:20.53 | ioerror | qdk, yeah, that's how i got the above error |
| 22:21.02 | riddlebox | man my sipura 2100 keeps reverting back to broadvoice's setup |
| 22:21.25 | qdk | ioerror: Druken ioerror: exten => 7,1,musiconhold() |
| 22:21.55 | qdk | ioerror: ok, but thats because you lack the class, i think... havent used moh yet. |
| 22:22.15 | iDunno | ow. |
| 22:22.16 | Druken | put default in the () |
| 22:22.52 | Druken | oh... |
| 22:22.59 | Druken | put manual rather |
| 22:23.37 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 22:24.04 | qdk | Druken: just about 2 say that. :-) |
| 22:24.50 | [TK]D-Fender | generalhan : well? |
| 22:25.28 | ioerror | qdk, i lack class huh :-p |
| 22:26.03 | qdk | ioerror: ;-) |
| 22:26.25 | r_evolution | ok wtf. |
| 22:27.52 | generalhan | [TK]D-Fender: sorry man ... stupid employees are KILLING ME. "i dont understand .. how do i know what phone extensions im sitting at?" |
| 22:28.19 | generalhan | ........ how bout ITS THE LARGEST TEXT ON YOUR PHONE IT EVEN SAYS EXTENSION=XXX |
| 22:28.26 | generalhan | ohh i just wanna hurt people today |
| 22:28.30 | Druken | assign them each a phone :) tell them to take it with them |
| 22:28.33 | Druken | hehehe |
| 22:29.11 | *** join/#asterisk marcelbarbulescu (n=mbarb@38-118-35-20.rapiddsl.net) |
| 22:29.26 | riddlebox | is there any IP phones that you can assign buttons on, like say a particular button is the park button or voicemail? |
| 22:29.53 | Druken | yeah it's called speed dial :) |
| 22:29.55 | qdk | riddlebox: yes, just about any. |
| 22:30.07 | generalhan | [TK]D-Fender: ok i just tested it and THAT is why ... i thought it was n+101 not n+102. |
| 22:30.26 | riddlebox | gdk, you just program the buttons then from the phone? |
| 22:30.26 | dlynes_ | riddlebox: Aastra, Polycom, and Sipura 841/941, and GXP-2000 I think |
| 22:30.29 | qdk | generalhan: you though correctly... the first time. |
| 22:30.43 | budmang | does idefisk have a channel:-) |
| 22:30.43 | [TK]D-Fender | generalhan : So all working now? |
| 22:30.47 | generalhan | OHHHHHHH DAMNIT ... i had the dial on priority 2 |
| 22:30.48 | ioerror | qdk, can you explain that? lack of class? |
| 22:30.54 | qdk | generalhan: you just forgot the math you learned in first grade. |
| 22:31.24 | generalhan | qdk: WOAH easy there ,,, im pretty sure that was REMEDIAL math, but no 1st grader is tought how to add 101 to anything ! |
| 22:31.25 | generalhan | lol |
| 22:31.31 | Druken | you mean 1+1=3 isn't right? |
| 22:31.33 | generalhan | they tend to stick to the 10s and 20s ! lol |
| 22:31.38 | qdk | ioerror: well you need a class and in musiconhold($MY_CLASS}) you use it... i think. |
| 22:32.03 | hydride | anyone have a link to a default very basic config for sip phone to sip phone calling? |
| 22:32.14 | hydride | I got one soft phone and 2 zyxels |
| 22:32.24 | qdk | generalhan: correct again, but they do learn to count to 10... you lost it at 2. |
| 22:33.04 | [TK]D-Fender | Druken : 1 + 1 = 10 :) |
| 22:33.15 | Druken | i thought so |
| 22:33.19 | qdk | hydride: isnt there like 10 different in the default sip.conf? |
| 22:33.22 | trixter | got a new sangoma a101u (single span t1/e1 card - asking about $400) and a new digium tdm410b (1 fxs module - asking about $100) for sale if anyone is interested msg me - both are brand new never used |
| 22:33.34 | Qwell[] | trixter: asterisk-biz |
| 22:33.37 | [TK]D-Fender | trixter : Why are you selling? |
| 22:33.39 | trixter | I did already |
| 22:33.52 | trixter | beucase they are surplus and I dont use them nor do I have any plans to anytime soon |
| 22:34.05 | ioerror | qdk, hrm, i need to go RTFM, thanks1 |
| 22:34.06 | hydride | qdk, maybe |
| 22:34.41 | hydride | but I don't know what I do and don't need |
| 22:34.59 | qdk | ioerror: the link i posted + the link for cmd musiconhold in the bottom on the page, should help you a lot. |
| 22:35.53 | qdk | hydride: callerid, username, secret + default setting might be enough. |
| 22:36.15 | hydride | so I just need to add the individual users right? no extras? |
| 22:36.32 | robin_sz | [TK]D-Fender, there are 10 kinds of people. those who understand binary, and those who don't |
| 22:36.38 | trixter | I will sell both cards to the same person for $400 which isnt a bad deal just to be done with it |
| 22:36.57 | qdk | hydride: not sure what you mean by users? |
| 22:37.14 | hydride | extensions |
| 22:37.21 | hydride | or phone numbers |
| 22:37.24 | trixter | robin_sz: I always thought it was ... there are 10 kinds of people, those who undertand trinary, those who dont and those that think becuase its 1's and 0's it must be binary |
| 22:37.27 | qdk | hydride: you dont have extentions in sip.conf |
| 22:38.19 | robin_sz | trixter, theres always 1 .... |
| 22:38.26 | [TK]D-Fender | robin_sz : There are 3 kinds on people, those that know math, and those that don't :) |
| 22:38.38 | Qwell[] | [TK]D-Fender: (for large values of 1) |
| 22:38.44 | robin_sz | s/math/maths/ |
| 22:38.48 | tainted- | binary = base 2 |
| 22:38.49 | qdk | hydride: the only relations between sip and extensions is the context and the sip id... sip id is what you put in [] and context is where you sip will start in your dialplan. |
| 22:39.03 | tainted- | base 2 = 1's and 0's trixster.. what kind of crack u smokin |
| 22:39.27 | qdk | hydride: (so you need context and probably also type in your sip.conf + the stuff i already mentioned) |
| 22:39.31 | trixter | read what I said again |
| 22:39.40 | trixter | you are aparently classifying yourself as not knowing trinary |
| 22:39.43 | snitt | trinary :P |
| 22:39.49 | robin_sz | ternary |
| 22:40.11 | Qwell[] | robin_sz: indeed |
| 22:41.01 | robin_sz | ok, the exhaustion and the Baileys are finally getting to me .. I must away to bed |
| 22:41.16 | *** join/#asterisk arley (n=alex@209-78-110-242.ded.pacbell.net) |
| 22:41.45 | robin_sz | tomorow I have lots of Perl and java to do .. and about 4 hours of welding |
| 22:42.05 | dlynes_ | perl! damnit! |
| 22:42.09 | dlynes_ | not Perl! |
| 22:42.16 | robin_sz | ummm. no. |
| 22:42.20 | riddlebox | why does my sipura 2100 keep defaulting to broadvoice's setup and erasing my asterisk stuff? |
| 22:42.22 | robin_sz | trust me. its Perl |
| 22:42.27 | dlynes_ | heh |
| 22:42.27 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 22:42.32 | dlynes_ | not if you talk to #perl :) |
| 22:42.41 | dlynes_ | they insist it's not Perl, it's not PERL, it's just perl :) |
| 22:42.50 | robin_sz | no. they dont |
| 22:43.10 | marcelbarbulescu | anybody know if there is a way to test the codec used from dialplan (to route by codec for example)? SIP_CODEC is used only for trying to set the codec from what I saw from the code... |
| 22:43.15 | dlynes_ | I think it was tainted- or someone that got banned from there the other day fro calling it PERL |
| 22:43.20 | [TK]D-Fender | riddlebox : becuase it is polling their server and reprovisioning itself. Thought you could ursurp their control of it? |
| 22:43.36 | robin_sz | well, yeah, calling it Perl is close to balsphemy |
| 22:43.43 | riddlebox | [TK]D-Fender, I have for the longest time |
| 22:43.53 | robin_sz | I have been seen on #london.pm from time to time |
| 22:43.57 | tainted- | it's a backronym |
| 22:44.04 | tainted- | according to the #perl guys |
| 22:44.09 | dlynes_ | heh |
| 22:44.11 | tainted- | call it PERL/Perl and u die |
| 22:44.17 | dlynes_ | yeah...your runini with them was pretty funny :) |
| 22:44.33 | tainted- | it was waaay longer than i expected |
| 22:44.46 | tainted- | i was thinking 5 mins tops for a kick/ban |
| 22:44.48 | dlynes_ | but you antagonized them quite a bit, too :) |
| 22:44.58 | tainted- | i made a friend |
| 22:45.13 | dlynes_ | yeah...buddy that wanted to get back into his gentoo channel :) |
| 22:45.16 | tainted- | the #otw guys always say hi now |
| 22:45.29 | tainted- | irc adventrues |
| 22:45.56 | dlynes_ | marcelbarbulescu: i've been wanting to do that for a little while, too |
| 22:46.12 | *** part/#asterisk budmang (i=smdepot@12.206.134.162) |
| 22:46.19 | tainted- | u know what was funny |
| 22:46.19 | dlynes_ | marcelbarbulescu: but i've only got one customer with that need, and they're a voip termination company, so they're a pretty special case |
| 22:46.30 | robin_sz | dlynes, http://faq.perl.org/perlfaq1.html#What_s_the_differenc |
| 22:46.33 | robin_sz | see ..?? |
| 22:46.40 | riddlebox | [TK]D-Fender, hrmm I have found a box to turn provisioning off, maybe thats where it is ;) |
| 22:46.42 | tainted- | couple of years ago we got dialout access through this company PBX |
| 22:46.56 | tainted- | started conferencing people in and prank calling |
| 22:47.13 | [TK]D-Fender | riddlebox : Hey... whats this big red button do?! |
| 22:47.19 | tainted- | it was un-fucking-believeable what we could get away with |
| 22:47.19 | robin_sz | dlynes, and http://www.perl.org/ refers to it exclusively as Perl |
| 22:47.57 | dlynes_ | robin_sz: yeah, I think i've seen that before |
| 22:48.08 | marcelbarbulescu | dlynes_: ok, thanks, just confirmed by findings... time for some programming now :) |
| 22:48.18 | dlynes_ | robin_sz: but you'd think the real way to spell it would be PERL, after all, it's an acronym |
| 22:48.34 | dlynes_ | robin_sz: For 'Practical Extraction and Reporting Language' |
| 22:48.36 | tainted- | i remember calling motel 6 (a chain of inexpensive motels) who had the slogan, "we'll leave the light on for you" and complaining that we went back to our rooms but no one had left a light on for us |
| 22:49.05 | tainted- | or calling pizza hut and connecting them to round table pizza |
| 22:49.12 | tainted- | good times |
| 22:49.18 | generalhan | [TK]D-Fender: if i have a macro that calls on ${ARG2} and i dont include an ARG2 what happens ? will the whole thing fail or will it put something in for ARG2 ? |
| 22:49.32 | dlynes_ | tainted-: heh...my ex roommate used to take the police frequency and bridge it in with the transit police frequency :) |
| 22:49.58 | dlynes_ | used to confuse the hell out of the cops and the transit cops :) |
| 22:50.05 | tainted- | woah |
| 22:50.11 | tainted- | that's criminal! |
| 22:50.24 | riddlebox | [TK]D-Fender, now I am back to normal, port 2 works, but cannot get port 1 to connect for some reason |
| 22:50.25 | dlynes_ | yeah, but humorously criminal :) |
| 22:50.35 | tainted- | lol |
| 22:50.47 | tainted- | ever dispatch taxi drivers to remote places? |
| 22:50.48 | dlynes_ | anyways...i need to get back to work...alters |
| 22:50.54 | tainted- | that's a good one too |
| 22:50.59 | tainted- | yea finish your gd billing |
| 22:51.23 | JackEStorm | hey any of you peeps have unlimited incomming SIP/IAX DID's? if so with whom? |
| 22:51.32 | tainted- | have u guys read about IVR phishing? |
| 22:51.45 | tainted- | JackEStorm what do u need it for? |
| 22:51.52 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:52.08 | [TK]D-Fender | generalhan : Any variable that was never set will just return null. |
| 22:52.23 | generalhan | hmm i dont want that |
| 22:52.25 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:52.25 | [TK]D-Fender | generalhan : so as long as you account for that you're ok. |
| 22:52.32 | generalhan | cuase ARG2 is a directory name |
| 22:52.33 | [TK]D-Fender | generalhan : depends on your macro |
| 22:52.35 | JackEStorm | tainted-: biz and home number...I used to use SixTel, but after Katrina and the problems there, and lack of updates, I need to move. |
| 22:52.44 | [TK]D-Fender | generalhan : Show me what you're making. |
| 22:52.45 | qdk | generalhan: then call your macro with something usefull |
| 22:52.47 | r_evolution | Hey TK |
| 22:52.56 | r_evolution | Ever setup anything with SPA941s? |
| 22:53.12 | r_evolution | seems that they're not so happy doing a conference with G729... |
| 22:53.23 | generalhan | [TK]D-Fender: thats ok i just figured out what i could fill ARG2 with so thats all set. now i just need to make this only apply to certain phones. |
| 22:53.24 | r_evolution | as soon as i go to bridge it... it switches over to ulaw |
| 22:53.33 | [TK]D-Fender | r_evolution : I used to own one... I ditched it |
| 22:53.53 | [TK]D-Fender | r_evolution : Nope... you can only run 1 G.729 channel at a time.... |
| 22:54.03 | generalhan | If i do a _71XX,whatever and i also have a 7100 that will be ok right, cuase it will try and match exactly first before it tries the pattern matching |
| 22:54.10 | r_evolution | O_o |
| 22:54.12 | [TK]D-Fender | r_evolution : not a lot of power in its CPU and maybe didn't pay to license it |
| 22:54.21 | r_evolution | ohhhh |
| 22:54.29 | qdk | generalhan: yes, most specific always wins. |
| 22:54.34 | generalhan | yay |
| 22:54.38 | r_evolution | well thats fuckin dirty. |
| 22:55.21 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 22:58.20 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:58.34 | *** join/#asterisk hydride (n=nathan@d57-4-109.home.cgocable.net) |
| 22:58.56 | hydride | I can't get my zyxel to say it's registered, do I need to set up nat trvsl on it/ |
| 22:58.58 | hydride | ? |
| 23:00.25 | qdk | hydride: nat=yes never hurts if you are in doubt. |
| 23:00.42 | hydride | in sip.conf? |
| 23:01.00 | r_evolution | yeah... under each peer behind a NAT |
| 23:01.19 | tainted- | r_evolution meetme = ulaw only |
| 23:01.20 | qdk | hydride: yes |
| 23:01.43 | r_evolution | nah a 3-way isn't ulaw only... |
| 23:01.58 | r_evolution | or else we'd have a bunch of pissed off customers :) b/c I disallow ulaw for most of them |
| 23:02.01 | r_evolution | unless they need to fax |
| 23:02.02 | tainted- | meetme != 3way |
| 23:02.13 | r_evolution | i didn't say i was doing a meetme :) |
| 23:02.15 | tainted- | how is fax over ulaw working out |
| 23:02.15 | r_evolution | just a conference |
| 23:02.21 | r_evolution | decently actually |
| 23:02.26 | r_evolution | not exactly a PERFECT fax |
| 23:02.28 | tainted- | really |
| 23:02.33 | r_evolution | but im sayin... you're going through the internet |
| 23:02.35 | tainted- | ever try t38 |
| 23:02.38 | r_evolution | it's not going to be great quality |
| 23:02.41 | r_evolution | nah not yet |
| 23:02.48 | r_evolution | right now Level3 apparently doesn't support T.38 |
| 23:03.02 | tainted- | i'm still waiting to hear from someone who sucessfully deployed t38 |
| 23:03.03 | r_evolution | which means they = bitches. |
| 23:03.04 | justinu|laptop | they don't even support RTCP... thoe jerks |
| 23:03.10 | r_evolution | chya. |
| 23:03.23 | r_evolution | haha. psst ulaw faxing is working well enough for me... |
| 23:03.29 | r_evolution | if it ain't broke... |
| 23:03.32 | r_evolution | tweak it on a test server ;x |
| 23:03.33 | justinu|laptop | werd |
| 23:03.46 | tainted- | justinu|laptop u fax over ulaw too? |
| 23:03.55 | justinu|laptop | only on ethernet networks |
| 23:04.09 | r_evolution | heh... we're faxing all the way over IP justin ;x |
| 23:04.10 | r_evolution | scary. |
| 23:04.17 | RaYmAn-Bx | say, how would I tell whether asterisk is picking up that I added ztdummy? :P |
| 23:04.25 | tainted- | well the ethernet eventually touches the net |
| 23:04.31 | tainted- | doesn't it? |
| 23:04.36 | r_evolution | not in the sense he's talking about i don't think... |
| 23:04.36 | tainted- | or just fax intra office? |
| 23:04.57 | r_evolution | i think he's meaning in a WAN sense... right? |
| 23:04.59 | justinu|laptop | no, goes out via TDM |
| 23:05.08 | tainted- | ohh |
| 23:05.10 | justinu|laptop | got a PRI for faxes etc |
| 23:05.25 | tainted- | r_evolution u go ulaw to SIP provider right? |
| 23:05.29 | r_evolution | yeah |
| 23:05.36 | r_evolution | ulaw from ATA to * to provider |
| 23:06.47 | qdk | RaYmAn-Bx: and IAX trunk in with trunk=yes will trashtalk you if you havent. :-D |
| 23:07.40 | r_evolution | so im thinkin... it's not going to KILL me to run ulaw all over the entire office... and then 729 it out... |
| 23:07.45 | r_evolution | but it's the principle... ya know? |
| 23:08.13 | trixter | if you do different codecs you have to transcode which takes cpu |
| 23:08.21 | r_evolution | i know :) |
| 23:08.23 | trixter | so depending on load it might kill you, or at least the server |
| 23:08.27 | r_evolution | nah |
| 23:08.30 | justinu|laptop | he;s goit a monster box |
| 23:08.33 | r_evolution | I've got enough to handle it |
| 23:08.38 | RaYmAn-Bx | qdk: heh. Thing is, I just want to see whether asterisk is aware that it was added (I basically just restarted asterisk after modprobing ztdummy..) |
| 23:08.49 | trixter | a monster box still wont handle 10000 concurrent calls ulaw to g.729 :P |
| 23:08.52 | r_evolution | modprobing sounds dirty :-\ |
| 23:09.03 | r_evolution | yeah but the day ASTERISK handles 10,000 concurrent calls |
| 23:09.04 | r_evolution | actually |
| 23:09.06 | RaYmAn-Bx | it is =P |
| 23:09.09 | trixter | I didnt know about load why I brought that up :P |
| 23:09.10 | justinu|laptop | lol |
| 23:09.12 | r_evolution | the day SIP handles 10,000 current calls ;x |
| 23:09.15 | tainted- | who is going to handle 10k calls over one asterisk box |
| 23:09.26 | trixter | tainted-: no one :P |
| 23:09.27 | r_evolution | i think the hard-coded limit for sip is like 900 some isn't it? |
| 23:09.29 | trixter | ever |
| 23:09.33 | tainted- | sip can handle 10k concurrent calls |
| 23:09.35 | r_evolution | i dunnoooo |
| 23:09.41 | r_evolution | really? i thought it had a hard-coded limit? |
| 23:09.45 | tainted- | but not asterisk's implmentation of sip |
| 23:09.55 | r_evolution | anyone wanna build a cluster capable of handling 10K calls? |
| 23:10.05 | r_evolution | i'll donate a dollar |
| 23:10.08 | tainted- | it's called asterisk/SER |
| 23:10.16 | justinu|laptop | it's more the calls per second that gets you than simultaneous calls |
| 23:10.24 | file | there is no hard coded limit in SIP, there is in IAX2 though |
| 23:10.34 | file | due to the way that channel structures are kept around |
| 23:10.49 | tainted- | ast_struct_ass |
| 23:10.52 | justinu|laptop | lol |
| 23:10.54 | tainted- | isn't it? |
| 23:10.55 | r_evolution | well thats good to know :) |
| 23:11.06 | r_evolution | i wonder how many concurrents i can get on the other box then justin... O_O |
| 23:11.18 | tainted- | ass = almost sip standard |
| 23:11.19 | r_evolution | b/c ive taken as much of the CPU load offa that as possible |
| 23:11.20 | trixter | file: did the rewrite that tony did to iax2 stuffs remove that limit? |
| 23:11.49 | file | trixter: what rewrite? and I just meant internally in chan_iax2 |
| 23:12.04 | trixter | ahh, tony took a chainsaw to libiax2 |
| 23:12.09 | trixter | and some other stuff |
| 23:12.16 | tainted- | trixter doesn't mean it committed it ;) |
| 23:12.17 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:12.21 | trixter | because it was bothering him with some of the problems that presented themselves |
| 23:12.22 | tainted- | s/it/he/ |
| 23:12.23 | RaYmAn-Bx | hmm, does anyone happen to remember the name of that linux tool that can tell what programs is accessing a file? |
| 23:12.38 | trixter | tainted-: its committed just not where you would probably want it :) |
| 23:12.50 | tainted- | i think ur right now |
| 23:12.54 | trixter | rayman: lsof? |
| 23:12.55 | tainted- | s/now/tho/ |
| 23:13.21 | tainted- | trixter: http://www.cluecon.com/anthm.html |
| 23:13.28 | trixter | I know who he is |
| 23:13.33 | trixter | I also know where this iax2 code is |
| 23:13.35 | trixter | :D |
| 23:13.49 | tainted- | asterlink production boxes? :D |
| 23:13.50 | RoyK | <PROTECTED> |
| 23:13.51 | trixter | but its not compatible with asterisk afaik not sure, havent read the license on the stuff he wrote for it |
| 23:14.14 | qdk | RaYmAn-Bx: if the module is modprobet correctly * should just be aware of it... dont know any test or status to check it |
| 23:14.19 | trixter | from what I heard he eventually rewrote it from the ground up but that may be rumor, if its a ripped apart libiax2 then its lgpl and is compatible |
| 23:14.28 | trixter | but it may not be funcationally compatible, I really have no idea on that |
| 23:14.41 | trixter | I just know it works with freeswitch :) |
| 23:14.46 | RaYmAn-Bx | qdk: hopefully lsof can tell if asterisk is using any of the zaptel devs |
| 23:14.57 | tainted- | wonder if he'd make a free switch to enable that |
| 23:15.04 | tainted- | hmm |
| 23:15.34 | trixter | most of freeswitch is mpled although not all of it |
| 23:16.00 | tainted- | what does mpl involve anyway |
| 23:16.11 | tainted- | what's different vs the gpl |
| 23:16.15 | trixter | basically 3 clause BSD + you must kick back changes you make to the original authors |
| 23:16.23 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 23:16.41 | tainted- | ah |
| 23:16.48 | trixter | so in many ways its more free than the gpl, while still allowing the original authors to get the updates that people make |
| 23:18.02 | tainted- | so u could include non-gpl'ed libraries etc w/o gpl'in that lib |
| 23:18.07 | tainted- | like that? |
| 23:18.28 | trixter | yes you can turn it into a commercial product, include commercial components, etc |
| 23:18.53 | trixter | but anything that is MPLed (mozilla license) you have to give your changes (even if you dont distribute you are supposed to give your changes back) |
| 23:19.26 | *** join/#asterisk dvorak (n=dvorak@luser.fnord.net) |
| 23:19.54 | trixter | so if you modify a MPL section to add support for some commercial thing (say cepstral support) the MPL parts have to be given back to the authors, but that lets you freely distribute that product with cepstral preinstalled |
| 23:19.58 | trixter | the gpl forbids that |
| 23:20.21 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) |
| 23:20.24 | trixter | technically the cepstral module for asterisk cant be GPLed because it requires non gpl-compliant stuff to work |
| 23:20.37 | trixter | but that is a different discussion |
| 23:20.42 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 23:20.54 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 23:21.45 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-60.cybersurf.com) |
| 23:22.09 | *** part/#asterisk dvorak (n=dvorak@luser.fnord.net) |
| 23:23.11 | *** join/#asterisk saftsack (n=saftsack@p54A7D14C.dip.t-dialin.net) |
| 23:31.45 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 23:36.33 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 23:37.42 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 23:38.53 | *** join/#asterisk MarkMcQ (n=mark@HSE-Kitchener-ppp194888.sympatico.ca) |
| 23:40.38 | MarkMcQ | Hello. I want to thank [TK]D-Fender for the advice about Sangoma cards and the Definity. All of my D-Channel reset problems are gone! Thank you! |
| 23:41.20 | snitt | :P |
| 23:43.01 | Manipura | Anyone ever seen this error |
| 23:43.03 | Manipura | <PROTECTED> |
| 23:43.11 | generalhan | can anyone explain to me how i would put a password prompt to get to a context ? im using ChanSpy but i dont want any of the employees to figure out that they can do it to so i want to put a password to even get to that context. |
| 23:45.31 | [TK]D-Fender | MarkMcQ : You're welcome... and I think remember hearing those names in the same request, but no details :) |
| 23:45.38 | generalhan | ive heard there is a way to prompt the user for a password to even let them get to that context |
| 23:45.39 | [TK]D-Fender | <- brain like a siv |
| 23:47.01 | MarkMcQ | I have upgraded to Fedora Core 5, at the same time installed the Sangoma Card. I now have a "minor" issue. Voicemail is not being recorded properly... |
| 23:47.55 | [TK]D-Fender | how so? |
| 23:48.18 | [TK]D-Fender | generalhan : Show how you get to it now. |
| 23:48.25 | MarkMcQ | Recording seems to stop every 100ms or so. So "one Mississippi" comes out as "wamsspi" |
| 23:49.08 | MarkMcQ | WAV files are about 1/5 the size of the old files...I've tested this against an older system (FC4, WIldcard). |
| 23:49.19 | [TK]D-Fender | So its consistantly jittery? |
| 23:49.53 | MarkMcQ | Ummmm...it doesn't seem to stutter, it just seems to be really speeded up |
| 23:50.15 | generalhan | [TK]D-Fender: exten => _99XXXX,1,Goto(internal_barge,${EXTEN:2},1) |
| 23:51.12 | MarkMcQ | Backing out to * 1.2.5 doesn't help... |
| 23:53.30 | [TK]D-Fender | generalhan : Shove an authenticate in front of it |
| 23:53.49 | generalhan | [TK]D-Fender: where ? |
| 23:53.52 | *** join/#asterisk ckwall (n=ckwall@63.149.122.94) |
| 23:54.00 | ckwall | <PROTECTED> |
| 23:54.14 | [TK]D-Fender | MarkMcQ : Is that SIP -> *, Zap -> *, or both? |
| 23:54.48 | JackEStorm | ckwall: man asterisk, and www.voip-info.org |
| 23:54.49 | MarkMcQ | [TK]D-Fender: Both. |
| 23:54.56 | generalhan | [TK]D-Fender: like the priority before it put down something like Authenticate(password[|options]) |
| 23:56.26 | ckwall | JackEStorm: That is where I found that info. But I dont think I am understanding how it gets to my extension. |
| 23:56.39 | MarkMcQ | ckwall: exten => 6406,1,Dial(sip/3000@10.0.0.177) won't do it? |
| 23:56.55 | CunningPike | ~seen dlynes_ |
| 23:57.05 | jbot | dlynes_ is currently on #asterisk. Has said a total of 491 messages. Is idling for 1h 6m 17s, last said: 'anyways...i need to get back to work...alters'. |
| 23:57.05 | ckwall | it doesnt seem to. |
| 23:57.05 | hydride | I have 3 sip phones authenticated to *, I just need an extension to make them talk something basic so they can call eachother, can someone help me? |
| 23:57.15 | ckwall | well, it sends it to the other extension, but fails there. |
| 23:57.24 | ckwall | do i need an answer command on 3000? |
| 23:57.31 | CunningPike | ~seen dlynes |
| 23:57.33 | jbot | dlynes is currently on #asterisk. Has said a total of 31 messages. Is idling for 6h 33m 43s, last said: 'mog_work: thanks again for the help last night'. |
| 23:57.51 | CunningPike | Thanks, jbot |
| 23:57.55 | dlynes | CunningPike: ? |
| 23:58.05 | CunningPike | dlynes: Hi |
| 23:58.08 | dlynes | hi |
| 23:58.19 | MarkMcQ | ckwall: Maybe. exten => 6406,1,answer exten => 6406,n,Dial... may work. |
| 23:58.30 | ckwall | lemme try that. |
| 23:58.31 | CunningPike | I did a quick calc on your bandwidth and it equates to about 9.5 hours of calls in the month |
| 23:58.38 | dlynes | CunningPike: ah |
| 23:58.56 | CunningPike | dlynes: So it looks reasonable |
| 23:59.02 | dlynes | 2.2GB's of an even split between g.729 and g.711? |
| 23:59.27 | CunningPike | dlynes: I did the calc based on all G711 |
| 23:59.31 | dlynes | ah |
| 23:59.43 | CunningPike | Even so, it's not out of the ballpark |
| 23:59.59 | dlynes | CunningPike: so, iow, we need to find a cheaper provider :) |