| 00:16.53 | *** join/#asterisk DavidR2008 (n=chatzill@71.0.16.133) |
| 00:21.04 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:22.27 | phix | Morning |
| 00:22.29 | phix | What did I Miss? |
| 00:22.54 | DavidR2008 | phix, Morning? most of the day ;-) |
| 00:23.06 | DavidR2008 | course I assume it is morning for you :-D |
| 00:23.14 | phix | hehe, 11:23 :) |
| 00:23.40 | DavidR2008 | 11:23 am or pm |
| 00:23.40 | phix | I will make up for it by staying up until 2:00 the next day :) |
| 00:24.00 | phix | I would of said 23:23 if I meant PM :) |
| 00:24.37 | DavidR2008 | gmt then? or did I miss count? |
| 00:24.51 | phix | gmt +11:00 |
| 00:25.04 | phix | so I said that in local time |
| 00:25.10 | phix | (my local time) |
| 00:25.25 | phix | you want that in GMT / UTC / Zulu time? |
| 00:25.31 | DavidR2008 | no :-) |
| 00:26.00 | phix | :D |
| 00:26.20 | DavidR2008 | I'm gmt -5 I just can't calculate correctly for some reason; I recommend you don't ask me any asterisk questions ;-) |
| 00:26.35 | phix | lol |
| 00:26.56 | phix | no plus 16hrs to your time to get mine? |
| 00:27.01 | phix | no = so |
| 00:27.20 | phix | or minus 4 and invert AM/PM? :) |
| 00:28.07 | phix | -5 == east coast US? |
| 00:28.19 | DavidR2008 | yep |
| 00:28.28 | DavidR2008 | where's +11? |
| 00:28.54 | phix | Australia (Sydney daylight saving time) |
| 00:29.13 | korcan | anyone know why there are a ton of asterisk processes running on my box... |
| 00:29.20 | shmaltz | DavidR2008, what you trying to do? |
| 00:29.25 | phix | korcan: are they zombied? |
| 00:29.31 | korcan | how can I tell? |
| 00:29.33 | DavidR2008 | shmaltz? |
| 00:29.38 | phix | korcan: ps aux |
| 00:29.43 | shmaltz | what you trying to calculate with gmt -5? |
| 00:29.53 | phix | korcan: see any Z symbols in the status column |
| 00:30.06 | DavidR2008 | i'm not calculating, i'm stating my location based on time zone |
| 00:30.13 | phix | :) |
| 00:30.36 | shmaltz | I'm gmt -5 I just can't calculate correctly for some reason; I recommend you don't ask me any asterisk questions ;-) |
| 00:30.39 | korcan | phix, doesn't look like it... |
| 00:30.48 | shmaltz | DavidR2008, what was that? |
| 00:30.54 | phix | korcan: hmmm |
| 00:31.04 | phix | korcan: you have alot of phone calls? :) |
| 00:31.05 | korcan | ps aux only show one asterisk process, but it has a ton of threads |
| 00:31.22 | korcan | right now 0 people on the phones, after hours |
| 00:31.31 | shmaltz | listening to this: http://www.youtube.com/watch?v=ZK0GmiSMNGI |
| 00:31.41 | DavidR2008 | shmaltz, I couldn't calculate phix's gmt offset from his time reference, the rest of the line was just being funny |
| 00:31.47 | phix | korcan: and what does top say? is it eating your resources? |
| 00:31.48 | shmaltz | ok |
| 00:31.50 | shmaltz | sorry |
| 00:31.53 | phix | what is your load avg? |
| 00:31.53 | korcan | phix, also my box is eating ram |
| 00:31.54 | shmaltz | thougth I can help |
| 00:32.20 | shmaltz | is frustrated with symantec backupexec 11d upgrade process |
| 00:32.29 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 00:32.29 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:32.45 | DavidR2008 | shmaltz, thanks for being willing :-D |
| 00:32.58 | shmaltz | np :D |
| 00:33.13 | phix | shmaltz: I havn't used symantec products since Norton Antivirus 2003 |
| 00:33.23 | korcan | phix, almost 0... I am more concerned about Ram eating as asterisk was rejecting calls in the middle of the day (only have 30 phones) |
| 00:33.45 | phix | korcan: It could be rejecting for another reason than lack of resources |
| 00:33.47 | shmaltz | phix, was that antivirus solution so good? :P |
| 00:33.50 | phix | korcan: what does your logs say? |
| 00:33.50 | *** join/#asterisk chendy (n=chatzill@113.91.37.203) |
| 00:34.17 | shmaltz | phix, what os you running? for personal machine that is |
| 00:34.18 | phix | shmaltz: norton's AV is a POS! It didn't even pick up half the shit that a free alternative like AVG or clamav did |
| 00:34.32 | DavidR2008 | phix, I agree totally |
| 00:34.42 | phix | shmaltz: Vista / Ubuntu 9.04 (soon to be 9.10 :D ) |
| 00:34.45 | shmaltz | phix, I never used norton av, I use AVG and my eyes |
| 00:34.55 | phix | yup |
| 00:34.59 | shmaltz | phix, how do you hate vista? |
| 00:35.05 | korcan | phix, what other reason would it reject? |
| 00:35.06 | phix | let me count the ways |
| 00:35.12 | korcan | I have 8GB of Ram in the box... |
| 00:35.28 | DavidR2008 | eyes are good, common sense can be the best AV |
| 00:35.34 | phix | korcan: 64bit OS obviously :) |
| 00:35.47 | korcan | phix, Yes |
| 00:35.52 | phix | DavidR2008: yup, and an intensive audit :) |
| 00:36.01 | shmaltz | to himself: because everytime we touch .... singing |
| 00:36.30 | phix | I am moving more to Ubuntu now though, Running Windows XP in a VM if wine doesn't cut it |
| 00:36.32 | shmaltz | latele I found running rootkit revealers is a must |
| 00:36.51 | korcan | phix, have you tried karmic beta? |
| 00:37.00 | phix | korcan: no, I am going to wait :) |
| 00:37.06 | phix | it should be out today or tomorrow |
| 00:37.13 | korcan | yep |
| 00:37.21 | korcan | it is beautiful :-D |
| 00:38.06 | phix | Tomorrow probably -5 GMT just to fuck with me :( |
| 00:38.18 | phix | which means I Have to wait an extra day |
| 00:38.35 | korcan | phix, who needs sleep? |
| 00:38.37 | phix | The US is lagged :) |
| 00:38.55 | phix | heh, well actually I don't for Friday, I have no jobs booked |
| 00:39.22 | korcan | phix, ubuntu site says 2 days... |
| 00:39.33 | phix | no it doesn't |
| 00:39.41 | korcan | phix, any ideas what would eat 8GB of ram? |
| 00:39.53 | phix | Cache |
| 00:40.02 | phix | but you want your 8GB of RAM being utilised |
| 00:40.11 | phix | otherwise it is pointless having 8GB of RAM in there |
| 00:40.28 | korcan | but asterisk stopped due to lack of ram |
| 00:40.39 | phix | the OS should handle memory management for you |
| 00:40.43 | phix | you have a SWAP? |
| 00:41.06 | korcan | yes, 0 used |
| 00:41.12 | phix | of 8GB / (8 * 1.5)GB minimum? |
| 00:41.13 | *** join/#asterisk KavanS (n=KavanS@c-71-236-202-177.hsd1.or.comcast.net) |
| 00:41.28 | phix | korcan: and your hard disc / filesystem free space? |
| 00:41.46 | korcan | phix, 3 GB Swap |
| 00:42.04 | korcan | 20 GB hdd space free |
| 00:42.31 | phix | hmmmm |
| 00:42.31 | korcan | dual quad core |
| 00:42.38 | korcan | we record every call |
| 00:42.46 | korcan | mixmonitor |
| 00:42.51 | phix | so what is the exact error message and log from asterisk? |
| 00:42.59 | phix | pastebin |
| 00:43.04 | korcan | let me go hunting for it... |
| 00:43.20 | phix | and is Asterisk 32bit? or 64bit? |
| 00:43.38 | phix | ldd should tell you where it is linked |
| 00:44.07 | korcan | 64 |
| 00:44.08 | phix | <PROTECTED> |
| 00:44.13 | phix | ldd /usr/sbin/asterisk |
| 00:46.24 | *** join/#asterisk QaDeS (n=mklaus@p4FC72C9D.dip0.t-ipconnect.de) |
| 00:47.17 | korcan | phix, still hunting... |
| 00:48.44 | phix | ok |
| 00:48.57 | phix | pastebin me when you get some error messages to show me :) |
| 00:49.23 | phix | also include OS / ditro, asterisk version, whether you compiled it yourself or using binary package, etc |
| 00:50.43 | korcan | <PROTECTED> |
| 00:50.51 | korcan | chan_sip.c: Failed to start PBX (call limit reached) |
| 00:51.15 | korcan | phix, ubuntu 9.04 x64 - compiled |
| 00:51.42 | phix | hmmmm |
| 00:51.50 | loather-work | sounds like you have very little memory on your system. |
| 00:51.52 | phix | Ubuntu 9.04!! high five! |
| 00:52.04 | korcan | :) |
| 00:52.10 | ChannelZ | is niiice |
| 00:52.18 | korcan | loather-work, is 8GB little? |
| 00:52.21 | phix | although i am running asterisk under Debian Lenny :P |
| 00:54.13 | korcan | dual quad core xeon 2.5 Ghz (E5420) |
| 00:54.20 | korcan | so it is a really slow machine for 30 users ;) |
| 00:54.46 | korcan | phix, any other ideas? |
| 00:56.36 | korcan | anyone have any ideas as why my machine eats memory? |
| 00:57.01 | Chainsaw | Because it's hungry? |
| 00:57.30 | korcan | Chainsaw, how can I put it on a diet... |
| 00:57.42 | Chainsaw | korcan: What's the problem? Is the machine using swap? |
| 00:57.52 | korcan | Chainsaw, no swap usage |
| 00:58.02 | Chainsaw | korcan: Then you have an appropriate amount of RAM. |
| 00:58.46 | Chainsaw | korcan: The linux kernel is using the RAM as a read cache. Since RAM is about 10000 times faster then a harddrive, this can accelerate read performance quite spectacularly. |
| 01:00.37 | korcan | when mixmonitor is used, is that the cause of all my ram being eaten? |
| 01:00.59 | korcan | because it is available it is being used? |
| 01:01.53 | phix | hmmm, I need to find a windows registery analyser / cleaning tool :/ |
| 01:02.00 | phix | korcan: no sorry |
| 01:02.06 | korcan | phix, its called ubuntu... |
| 01:02.23 | Chainsaw | korcan: Your RAM is *supposed* to be in use. |
| 01:02.29 | phix | korcan: hehe, yes but this isn't for me :) it is for clients that don't like change |
| 01:02.42 | korcan | Chainsaw, I will remove the waterfall setting and pray for tomorrow |
| 01:03.05 | Chainsaw | korcan: You are attempting a fix something that is not broken. |
| 01:03.06 | korcan | phix, use your the force, tell them they like change |
| 01:03.24 | korcan | Chainsaw, I hope your right... know anyone that is hiring ? ;) |
| 01:03.25 | phix | they will just look at me strangely |
| 01:03.29 | Chainsaw | korcan: You're used to Windows, which has such atrocious memory management that it is usual for half or more of your memory to sit there with zeroes in it. |
| 01:04.12 | Chainsaw | korcan: The linux kernel will use the RAM that is available. I have 32GB RAM in my home workstation, and after a healthy compile session it will be using all of that for read cache. This is *normal*. |
| 01:04.23 | korcan | Chainsaw, I am actually running a company with about 5% of the machines with windows... the rest are LTSP! |
| 01:04.39 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 01:04.46 | phix | LTSP? |
| 01:04.49 | korcan | Chainsaw, thanks - everythng you said makes a lot of sense |
| 01:04.54 | Chainsaw | korcan: You don't have a 95% linux mindset yet. If you hit swap, there is undue memory pressure. |
| 01:04.56 | korcan | phix, thin clients |
| 01:05.02 | phix | korcan: wtf, I told you that before lol |
| 01:05.04 | phix | korcan: ok |
| 01:05.19 | Chainsaw | korcan: If the swap sits there at 0, the world is a happy place. All is good. Have a Martini. |
| 01:05.33 | korcan | phix, Chainsaw put it in stupid person language, hence why I understood him :) |
| 01:05.34 | phix | Chainsaw: no, scotch |
| 01:05.38 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:05.44 | phix | korcan: lol ok :/ |
| 01:06.36 | korcan | later peeps, off to writing more code :-D |
| 01:06.48 | korcan | phix, Chainsaw thanks again |
| 01:07.22 | Chainsaw | Quite welcome :) |
| 01:07.55 | phix | np |
| 01:07.59 | phix | *grumble* |
| 01:11.58 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 01:19.33 | shmaltz | ~helo |
| 01:19.33 | infobot | hmm... helo is the first command issued during smtp |
| 01:20.52 | korcan | phix, why the grumble? |
| 01:25.01 | *** join/#asterisk chendy (n=chatzill@113.91.39.149) |
| 01:39.06 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 01:40.07 | DavidR2008 | everyone go to bed? |
| 01:40.21 | IBC_jkenney | no |
| 01:40.29 | IBC_jkenney | i don't wanna go |
| 01:40.33 | Chainsaw | DavidR2008: Pretty much. Open the liquor cabinet. |
| 01:41.40 | DavidR2008 | takes one down and passes it around |
| 01:41.44 | DavidR2008 | ;-) |
| 01:50.19 | *** join/#asterisk syuroff (n=syuroff@h66-222-51-78.mdsnwi.broadband.dynamic.tds.net) |
| 01:55.35 | Katty | wooooooooottttttt!!!!!!! |
| 01:55.42 | Katty | finishes rest of bottle |
| 01:58.40 | jblack | I want booze |
| 02:01.04 | Deeewayne | offers Katty a shot of tequila |
| 02:01.14 | jaytee | I want a big fatty of the dankest kronik, a plate of brownies, some Pringles salt and vinegar chips, a pizza........a bottle of iced cold birch beer |
| 02:01.42 | jblack | I want pot too |
| 02:01.43 | Katty | i want a hug. |
| 02:01.49 | Deeewayne | I want to hang w/ jaytee as he lands that motherload |
| 02:01.52 | jblack | hugs katty |
| 02:01.55 | jaytee | hugs Katty |
| 02:02.05 | Deeewayne | also hugs Katty |
| 02:02.12 | Katty | squees!!! |
| 02:02.27 | Katty | hugs jblack |
| 02:02.29 | Katty | hugs jaytee |
| 02:02.31 | Katty | hugs Deeewayne |
| 02:02.50 | Katty | Dear Universe, thank you for good friends. Love, Katty |
| 02:03.46 | jaytee | I had two cats, now I have one. I miss the other one......he wasn't a picky eater and never whined for something else while ignoring what you provided. He was fat. |
| 02:03.56 | Katty | :< |
| 02:04.04 | Katty | i am sorry for your loss, jaytee |
| 02:04.08 | Katty | kitty was a good friend. |
| 02:04.10 | jaytee | She's always been skinny and fussy. |
| 02:04.18 | loather-work | :( poor kity |
| 02:04.22 | Katty | Dear Universe, thank you for cats. Love, Katty |
| 02:05.47 | Katty | Dear Universe, also thank you for dogs and ferrets. |
| 02:07.52 | jaytee | well, looks like she did eat most of her dinner somewhere in the last hour. sneaky too |
| 02:07.53 | jaytee | I was starting to worry she'd developed anorexia |
| 02:08.18 | jaytee | dogs are awesome |
| 02:08.31 | Katty | riddick is a great joy. |
| 02:09.23 | jblack | what is it with you and riddick? |
| 02:09.32 | jaytee | animals are a gift from god |
| 02:09.33 | jblack | Do you have a thing for big beefy blind guys with silver eyes? |
| 02:09.44 | Katty | riddick is my dog |
| 02:10.22 | jaytee | one of my friends in Florida has a dog named Frell |
| 02:10.22 | Katty | would you like a photo? |
| 02:10.49 | jblack | are you in it? |
| 02:11.27 | Katty | hmm. no, but i have a video of me with the dog. |
| 02:11.30 | jaytee | careful, jblack. this woman used half of YouTube's server storage just for the movie shorts of him as a puppy |
| 02:11.37 | Katty | *hee* |
| 02:11.46 | *** join/#asterisk rommer (n=none@wl103.netrom.com) |
| 02:11.53 | jblack | I'd like a _few_ pics of your dog. |
| 02:12.05 | jblack | Please don't torture me with your 500 most favorite. :) |
| 02:12.30 | jaytee | "oh, but you just have to see this one!!!!!" |
| 02:12.31 | Katty | :P |
| 02:12.33 | Katty | -> |
| 02:13.01 | jaytee | Riddick is a handsome dog, that be true |
| 02:13.20 | jaytee | the German Shepherd is a great breed of dog. |
| 02:13.28 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 02:13.43 | [TK]D-Fender | Prone to genetic defects esp the hips |
| 02:14.00 | jaytee | yes, dysplasia is common |
| 02:14.12 | jblack | my god. She really does have doggie videos? |
| 02:14.15 | jaytee | I had to put my girl down at 12. |
| 02:14.37 | jaytee | that ripped a big ragged hole in my aorta that day |
| 02:14.41 | jblack | baby sheppards are so cute |
| 02:14.50 | Katty | tons |
| 02:15.04 | Katty | tho perhaps not quite half of the youtube server. |
| 02:16.41 | jblack | I miss having a dog |
| 02:18.26 | jblack | jaytee: You weren't kidding... |
| 02:18.30 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:19.29 | jaytee | jblack, somewhere out there in some harsh, fluorescent cinderblock building in a rack of chain link cages a small puppy who desperately wants someone to love him and keep him warm is just waiting for you to find him. |
| 02:19.40 | jblack | cool! Ferrets! |
| 02:19.46 | jblack | I had a ferret when I was a kid |
| 02:19.59 | jaytee | I've never had a ferret |
| 02:20.03 | jaytee | but they're cute |
| 02:20.06 | Katty | 5 ferrets now ;) |
| 02:20.32 | jblack | jaytee: THey're cute as long as you keep them well fed and give 'em lots of attention |
| 02:20.57 | DavidR2008 | ferrets can be conniving and vicious |
| 02:21.15 | jaytee | I've had dogs, cats, hamsters and I even got 2 parakeets and eventually bred them to produce 9 babies. |
| 02:21.37 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 02:21.42 | jaytee | never had a ferret. don't think they were legal where I lived growing up. |
| 02:21.45 | jblack | Yeah. If you don't take good care of 'em, they'll let you know. |
| 02:22.06 | Katty | they're like birds. |
| 02:22.13 | Katty | the less attention they get, the meaner they get |
| 02:22.49 | jblack | I'm sad. |
| 02:22.51 | Katty | anywho, i'm off to bed. |
| 02:22.58 | jblack | I wanted to play borderlands tonight, but the download still has 2 1/2 hours to go. |
| 02:23.01 | jaytee | niters!!! |
| 02:23.03 | jblack | Sleep good. =) |
| 02:24.01 | DavidR2008 | what's borderlands |
| 02:24.03 | jblack | katty's webserver is broken |
| 02:24.22 | jblack | DavidR2008: as I understand it, a mix between fallout3 and diablo. |
| 02:24.33 | jaytee | which webserver? |
| 02:25.14 | jblack | Her listed homepage doesn't exist. |
| 02:25.35 | jblack | but I see that it's actually a dns issue, and a domain that's not katty's in the first place. |
| 02:27.27 | jaytee | she had some geekchick site and a blog at one point, her FB has her recipe site. mmmmm, some nice looking food on that one. |
| 02:28.07 | jblack | I'll have to ask her for that stuff tomorrow |
| 02:29.36 | jaytee | she had a blog aways back dealing with installing FOP on Asterisk that was pretty good |
| 02:36.12 | TJNII | If you get a VPRI you can set the outgoing callerID, correct? |
| 02:36.55 | jblack | most voip providers let you set CID, yes. |
| 02:37.19 | TJNII | Any recommendations for VPRI carriers? |
| 02:38.54 | jblack | No. YOu'll have to test several yourself, to find the ones that are logically closest to you. |
| 02:39.20 | jblack | as a general rule of thumb, provides that seem much cheaper than others are usually cutting corners. |
| 02:52.49 | *** join/#asterisk OrNix (n=ornix@91.151.249.47) |
| 03:06.23 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 03:06.39 | hesco | I am working to add a new DID to my existing * server. iax2 show registry | peers, plus the registration statements are posted at: http://pastebin.com/d37e57075. My existing DID is routing incoming calls appropriately, but the new DID is failing to make its presence known in my logs, much less by ringing my desk. Can anyone advise me what I may be missing, please? |
| 03:13.03 | *** join/#asterisk nsgn (n=brandonb@cpe-24-27-49-209.austin.res.rr.com) |
| 03:13.40 | nsgn | so i'm attempting to narrow down my choice of sip provider down. i'm eyeing voicepulse and future-nine. any experience or input from someone here for either? |
| 03:13.59 | nsgn | *-down |
| 03:17.00 | hesco | Never used future-nine. Had a $1.00 complimentary test account with voicepulse long enough to test a server I built a couple of months ago. It worked fine for those tests, but I've never used either in production. Sorry. |
| 03:17.43 | trogs | voicepulse have some odd registration procedures, at leats they did when i registerd a while ago, they wanted me to fax copies of my drivers license and passport and crap |
| 03:25.58 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-136-66.dsl.stlsmo.sbcglobal.net) |
| 03:26.12 | LemensTS | am i misunderstanding how $agi->hangup(); works in phpagi? http://pastebin.com/m44d4310d |
| 03:26.38 | LemensTS | i figured once it seen that it would kill the phpagi script and return to the next line in the dialplan |
| 03:27.14 | LemensTS | (the phpagi script is called from the dialplan) |
| 03:28.53 | hesco | where is your $agi object defined? |
| 03:29.47 | LemensTS | before anything else at the beggining of the file |
| 03:29.58 | LemensTS | $agi = new AGI(); |
| 03:30.26 | LemensTS | its doing other agi cmds just fine |
| 03:31.14 | hesco | I have used the Asterisk::AGI perl module some and found it buggy in the sense that it did not always perform as the documentation suggested it might |
| 03:31.57 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:c4c6:d4ba:1f33:6923) |
| 03:31.57 | hesco | I would up using `print STDOUT ` statements to interact more directly with the asterisk apps() |
| 03:32.11 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 03:36.02 | LemensTS | yea i get different result with $agi->exec("Hangup"); lol |
| 03:36.29 | LemensTS | it keeps the audio playing instead of it being mute |
| 03:38.46 | nsgn | trogs: really..?! they seem to imply i can sign up and get access instantly. were you in an odd location or something? |
| 03:38.55 | nsgn | i've not signed up yet cause they require prepayment with signup |
| 03:38.57 | trogs | yeah |
| 03:39.00 | nsgn | but i'm about to. i can't find a better deal than them |
| 03:39.04 | nsgn | and they are decently recommended by most |
| 03:39.08 | trogs | not in the US |
| 03:39.15 | trogs | maybe the process is different |
| 03:39.17 | nsgn | the free incoming and 4 channels is what gets me |
| 03:39.27 | nsgn | ah, i'm in the US. hopefully ittl go smooth. i'm fixin to bite the bullet here |
| 03:39.47 | nsgn | feels weird to go back to paying per minute for phone service, but my projections say ittl be cheaper than the monthly sip services |
| 03:39.51 | nsgn | for my usage |
| 03:41.14 | loather-work | heh, i remember the day when free incoming was the norm |
| 03:42.12 | LemensTS | hesco: instead of hangup(), i just did the php cmd exit; it quit the agi, and returned to the dialplan...i dont see any channel open still so i guess its good |
| 03:42.38 | hesco | TIMTOWTDI |
| 03:44.36 | hesco | anyone have any ideas on my iax registration issue? |
| 03:44.52 | hesco | I am working to add a new DID to my existing * server. iax2 show registry | peers, plus the registration statements are posted at: http://pastebin.com/d37e57075. My existing DID is routing incoming calls appropriately, but the new DID is failing to make its presence known in my logs, much less by ringing my desk. Can anyone advise me what I may be missing, please? |
| 03:47.59 | p3nguin_ | nsgn: Isn't Flowroute or VoIP.ms cheaper? |
| 03:49.05 | p3nguin_ | nsgn: Flowroute has less than 1 cent per minute termination. How's that compare? |
| 03:49.46 | p3nguin_ | nsgn: For around $7 per month, I think you can get unliminted inbound, too. |
| 03:51.39 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-136-66.dsl.stlsmo.sbcglobal.net) |
| 04:08.52 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:20.23 | nsgn | p3nguin_: hmm, let me look |
| 04:20.26 | nsgn | you have personal experience with them? |
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| 04:21.54 | nsgn | p3nguin_: awh, flowroute doesn't have DIDs in my area |
| 04:22.57 | *** join/#asterisk shinao1 (n=shinao1@41.219.247.166) |
| 04:23.24 | nsgn | voip.ms claims unlimited channels. is it really..? |
| 04:24.08 | nsgn | if i had unlimited channels on a DID why would i want/need more than one DID unless i needed "direct lines"? |
| 04:25.24 | nsgn | blah, nevermind. they don't have central texas (512) available either :( |
| 04:26.33 | nsgn | voicepulse is the only decently priced service with 512 DIDs |
| 04:26.41 | nsgn | unless someone can open my eyes to another |
| 04:27.14 | p3nguin_ | nsgn: If you have a number, you could port it. Or VoIP.ms can do a carrier order on many areas. |
| 04:27.37 | nsgn | don't wish to port my #. keeping my main line with the provider i trust and forwarding it to a new DID as i experiment with SIP |
| 04:27.53 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
| 04:27.56 | nsgn | i've been in porting nightmares in the past. gonna avoid that for now. austin shows as "backordered" with voip.ms |
| 04:28.00 | nsgn | so someone already tried to order |
| 04:28.56 | nsgn | anyone have a voicepulse referal code? will benefit both you and I as i sign up here |
| 04:30.47 | p3nguin_ | I know there are a few people here that use voicepulse, but not sure when they are active. |
| 04:31.55 | nsgn | this will be pretty nice. if i use 1000 outgoing minutes i'll pay just $15/mo. incoming is free unlimited. i'm excited to have more than one stupid analog line in my little business |
| 04:32.12 | nsgn | but seriously, if anyone wants to get a referral bonus at voicepulse speak up |
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| 04:35.44 | mchou | nsgn: tell us more about your porting horrors |
| 04:35.55 | mchou | inquiring minds want to know |
| 04:36.09 | nsgn | mchou: particularly related to verizon hating vonage's guts |
| 04:36.27 | loather-work | porting sucks |
| 04:36.36 | loather-work | those who have the numbers never want to release them |
| 04:36.42 | nsgn | verizon will NOT port numbers to vonage in areas for some reason. i actually used to work for vonage as an authorized installer and this really gave us heck with half of my local coverage area, where verizon is the monopoly |
| 04:36.54 | loather-work | those who want the numbers don't want to go through the pain to get them |
| 04:37.12 | loather-work | well, verizon will, you just have to threaten them with legal action |
| 04:37.16 | nsgn | no longer work much for vonage (still in their system though, hehe) but man verizon people were always having a horrid time porting |
| 04:37.21 | mchou | nsgn: what's the reason verizon gave? |
| 04:37.44 | nsgn | vonage's system couldn't complete it, verizon would not respond to contact about it or say it simply wasnt possible whenever a user tried to contact them |
| 04:37.55 | nsgn | this was when verizon was simultaniously suing vonage nearly out of existence |
| 04:37.55 | mchou | nsgn: I mean if they were really uncooperative couldnt you just report them to the FCC? |
| 04:38.12 | nsgn | yes. they're paying them off or something. i've had some customers really try to take it places and got nowhere |
| 04:38.37 | nsgn | i have no clue, but it sucked cause it made our customers just go to time warner cable's "digital phone", which verizon would port to in the very same areas |
| 04:38.47 | mchou | hah |
| 04:38.51 | nsgn | twc's phone service is horrid, by the way |
| 04:38.59 | nsgn | the customers we got on vonage in this area were very pleased |
| 04:39.14 | nsgn | not really that useful from the asterisk perspective, but it's a really great single line solution for the home |
| 04:39.30 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 04:39.43 | nsgn | that's where my current DID is. i'll be fortwarding it to whatever SIP provider i do. i get employee rate with vonage so it's dirt cheap security for my number to just leave it there and forward it around |
| 04:40.03 | mchou | I dunno |
| 04:40.18 | nsgn | dunno what? |
| 04:40.22 | mchou | "vonage and security" sounds like oxymoron |
| 04:40.30 | DND | guys what if the telco doesnt know what disconnect supervision is? any alternatives? |
| 04:40.56 | nsgn | despite what the news says they've been around without interruptions in customer services for many many years |
| 04:41.06 | mchou | it aint about the news |
| 04:41.07 | DND | because our VM is full of beep tones messages |
| 04:41.24 | nsgn | they aren't the best price across the board, and voice quality is average, but they excel in a few areas and really push those few areas to be profitable |
| 04:41.25 | mchou | it's about how $$ they must be spending adverising |
| 04:41.37 | mchou | advertising* |
| 04:41.44 | nsgn | we did an absolutely rediculous number of installs for middle eastern folks who moved to america |
| 04:41.52 | nsgn | cause apparently vonage was THE way to call back home cheap |
| 04:42.04 | nsgn | i think 3 out of 4 of our residential installs were people calling out of the country |
| 04:42.59 | nsgn | right now it costs me less to keep my number with vonage than a DID with voicepulse costs |
| 04:44.02 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:44.11 | mchou | so did verizon and vonage ever settle? or is that still ongoing? |
| 04:44.18 | nsgn | i'm surprised how hard it is to google up someone posting their voicepulse referral code. can't find anyone. surprised nobody here has one either. i was previously given high word for voicepulse in this channel |
| 04:44.33 | nsgn | mchou: yeah they settled back in 2007 |
| 04:44.51 | nsgn | vonage paid verizon $120mill for some god awful reason |
| 04:45.11 | nsgn | it was horribly horrible patent rape, but vonage just paid up to get away from it cause the press about the case was harming customer image |
| 04:45.25 | nsgn | and they've rebounded decently sense, though installs were slowwww for a while after |
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| 04:47.24 | p3nguin_ | If you can wait, more people might be active in the daytime and someone might offer a referral to you. |
| 04:47.30 | Pikoro | how about broadvoice? any word on them? |
| 04:47.52 | mchou | broadvoice is teh suck |
| 04:48.05 | mchou | look them up in the boston BBB |
| 04:48.10 | mchou | grade: F |
| 04:48.36 | mchou | they've been kicked out by the BBB |
| 04:48.48 | Pikoro | really? we send people their way (we supply the voip adapter) since they support international addresses, unlike vonage |
| 04:48.55 | mchou | you have to be a major CF for that to happen |
| 04:50.27 | nsgn | broadvoice wants to charge $35/mo per channel/DID |
| 04:50.46 | nsgn | i'd need 3. while with voicepulse i get $11/mo DIDs, each including 4 channels |
| 04:51.01 | nsgn | so voicepulse is cheaper until a really large amount of minutes |
| 04:51.16 | Pikoro | hmm.. |
| 04:51.22 | nsgn | and since it's per minute it's dynamically cheaper on light months |
| 04:52.03 | nsgn | november, where i take a week of vacation, and december where i take another week will be nice and light on the phone bill, while with BV i'd be paying the same amount all the way through |
| 04:52.12 | nsgn | nice for a small business where usage can vary quite a bit per month |
| 04:52.24 | nsgn | they also get stinkin spotless reviews everywhere i search |
| 04:52.27 | Pikoro | i'm looking for a DID provider so I can offer stateside phone numbers to the overseas expat community here where I am |
| 04:56.46 | p3nguin_ | I pay $1/mo for my DID. |
| 04:57.54 | p3nguin_ | If you're just going to forward your number to your VoIP account, does the DID really need to be in that area code? |
| 04:58.59 | nsgn | p3nguin_: hmm, fair point you make...so long as i can force the callerid to show up to my clients as my local number. if it shows up out of area it is not acceptable to me. that workable? |
| 04:59.48 | p3nguin_ | With VoIP.ms, you are allowed to set your own CallerID on a per-call basis. I assume that Flowroute allows the same. |
| 05:00.06 | nsgn | which of the two do you like better? do you use one or the other? |
| 05:00.58 | p3nguin_ | I currently use VoIP.ms and I know a few people that use Flowroute. |
| 05:01.25 | nsgn | p3nguin_: and what do they mean when they say "unlimited channels DID numbers"? |
| 05:01.30 | nsgn | at voip.ms |
| 05:01.56 | p3nguin_ | It means you pay per minute per channel, so you can have as many channels as you need. |
| 05:02.18 | nsgn | p3nguin_: nice. i didnt understand why voicepulse limited my channels when i'm just gonna pay by the min anyway |
| 05:02.21 | nsgn | how is voice quality? |
| 05:02.25 | p3nguin_ | If you pay the unlimited/$6.95 or whatever, you're limited in the number of channels. |
| 05:02.43 | p3nguin_ | I have no complaints in quality. |
| 05:03.20 | nsgn | p3nguin_: but incoming isnt free. a ton of our calling is incoming, customers calling us |
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| 05:06.13 | p3nguin_ | If you go with the unlimited plan, looks like you'll pay $4.95/month for 2 channels with up to 3500 minutes (limited unlimited). |
| 05:06.49 | mchou | that's just lame |
| 05:07.02 | mchou | false advertising |
| 05:07.20 | mchou | either it's limited or it isnt |
| 05:07.21 | nsgn | yeah, that's pretty nice advertising there |
| 05:07.25 | p3nguin_ | How is it false when they are explicitly stating the restriction? |
| 05:07.32 | nsgn | not false, just stupid |
| 05:07.37 | nsgn | "limited unlimited" |
| 05:07.40 | p3nguin_ | If you pay per minute, you pay $0.99 per month for the DID and 1 cent per minute per channel. |
| 05:07.59 | *** join/#asterisk makafre (n=makafre@206-248-165-90.dsl.teksavvy.com) |
| 05:08.14 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/session) |
| 05:08.16 | nsgn | so if i get two of the per month ones i'd have 4 channels and 7000 minutes for $10/mo? |
| 05:08.52 | p3nguin_ | That's the way I calculate it. |
| 05:09.06 | p3nguin_ | 2 channels per DID |
| 05:09.07 | nsgn | and where in the heck on their site does it say this? i cant find it |
| 05:09.16 | mchou | lol |
| 05:09.21 | makafre | hey guys, does permit=204.11.192.0/255.255.255.0 will allow 204.11.192.* hosts to connect ?? |
| 05:09.26 | mchou | that's exactly right |
| 05:09.28 | nsgn | i see the per minute pricing but no plans |
| 05:09.51 | p3nguin_ | Well, I looked it up in my DID manager. I didn't see it on the main site. |
| 05:10.03 | p3nguin_ | I priced a DID in Apollo. |
| 05:10.19 | p3nguin_ | 832-415-xxxx |
| 05:10.30 | nsgn | where the heck is that? |
| 05:10.42 | nsgn | also...how can i trust them if nowhere on their site does it say what i should pay? |
| 05:10.45 | p3nguin_ | Some place in TX. |
| 05:10.56 | nsgn | thats the other thing i liked about voicepulse..it is up front |
| 05:10.58 | p3nguin_ | Let me look on the main site. It has to be there. |
| 05:11.04 | nsgn | and doesnt have a horrible website :) |
| 05:11.15 | nsgn | i'd think, i just couldnt find a obvious location for it |
| 05:11.25 | p3nguin_ | http://voip.ms/specifications.php |
| 05:11.32 | p3nguin_ | Call Origination (DID Numbers) |
| 05:11.34 | nsgn | oh wait it's under DIDs |
| 05:11.39 | nsgn | you gotta act like you want a DID |
| 05:12.04 | nsgn | now that's free inbound but per minute outbound though, right? |
| 05:12.58 | p3nguin_ | The pricing we have been discussing is DID rates. DID is inbound. |
| 05:13.06 | p3nguin_ | Termination rates are per minute, no matter what. |
| 05:13.22 | nsgn | yeah, so i'm just clarifying that outbound is per minute no matter what |
| 05:13.36 | nsgn | was confused for a moment cause channels are bundled with DIDs here |
| 05:13.38 | p3nguin_ | I only know of one provider that offers an unlimited outbound package, and that's CallCentric. |
| 05:13.41 | nsgn | which isnt particularly the case with some others |
| 05:13.59 | nsgn | and what's the reason not to use callcentric? |
| 05:14.02 | nsgn | never heard of them |
| 05:14.12 | p3nguin_ | I think their service sucks. |
| 05:14.35 | nsgn | cant find channel info on their site |
| 05:14.36 | nsgn | scratch that |
| 05:14.41 | nsgn | back to voip.ms |
| 05:14.47 | nsgn | so it seems they are all around cheaper than voicepulse |
| 05:14.54 | nsgn | and you have no issues with their quality? |
| 05:15.01 | nsgn | and don't have horribly low standards? ;) |
| 05:15.19 | p3nguin_ | CallCentric has an unlimited outbound plan which allows unlimited outbound calling on one channel, but if you add a second outbound call you pay per minute for it. |
| 05:15.37 | p3nguin_ | I have no issues with VoIP.ms. |
| 05:16.03 | p3nguin_ | But again, Flowroute is slightly cheaper than VoIP.ms. |
| 05:16.18 | nsgn | i'm more willing to trust voip.ms if you use and trust them |
| 05:16.31 | makafre | talking of callcentric, you guys now how to receive their inbound DID calls without using allowguest=yes ? :- ) |
| 05:16.35 | nsgn | and vouch for a critical feature i need; the ability to set my caller id to ALWAYS show as my main number |
| 05:17.12 | p3nguin_ | There's one single caveat with that... |
| 05:17.25 | p3nguin_ | When you use the value route, CID isn't guaranteed to always pass. |
| 05:17.48 | nsgn | that's a problem for me |
| 05:17.51 | nsgn | it must always pass |
| 05:18.11 | nsgn | i will not have a customer see a confusing number for us |
| 05:18.29 | nsgn | i've kept it simple for them for years and wont change that. so do i need to do premium for that to be a sure thing? |
| 05:18.30 | p3nguin_ | You can use the premium route. |
| 05:18.34 | nsgn | and the cost? |
| 05:18.54 | p3nguin_ | 1.25 cents per minute |
| 05:19.05 | nsgn | hmmm |
| 05:19.16 | nsgn | approaching voicepulse pricing there |
| 05:19.24 | nsgn | though the DIDs are still cheaper |
| 05:19.43 | nsgn | when i set up a DID with voip.ms how long does it take to come into effect? can i start testing/calling right away? |
| 05:19.52 | p3nguin_ | I've never had CID fail, to my knowledge, but there is always a possibility that it could show up as the provider's number when using the value route. |
| 05:20.13 | p3nguin_ | Within seconds it will be active. |
| 05:20.35 | p3nguin_ | Like literally by the time you are ready to make your call, it will be active. |
| 05:20.38 | nsgn | p3nguin_: oh, nice. they had some deal about 1 business day that made me wonder |
| 05:20.47 | nsgn | but that may be their safety net |
| 05:21.08 | p3nguin_ | " Listed numbers are processed in real time and effective immediatly after your order. |
| 05:21.15 | p3nguin_ | (close quote) |
| 05:21.16 | nsgn | OOH, nice |
| 05:21.44 | nsgn | ok, thanks. clears that up. can i go value for my testing and seamlessly switch to premium if i dislike CID stuff? |
| 05:21.58 | nsgn | or is that a choice made at the start, or difficult to modify? it a dialing pattern thing in * perhaps? |
| 05:22.16 | mchou | damn logitech |
| 05:22.19 | p3nguin_ | Switching between value and premium is done with three clicks of the mouse in the web portal. |
| 05:22.33 | nsgn | ok, great |
| 05:22.48 | p3nguin_ | One to pull down the menu, one to click on premium (or value), one to apply. |
| 05:22.59 | nsgn | so if i get one DID and a call forwards to it from my vonage DID, then another call comes in and is also forwarded to it from vonage will it hop on a second channel like i'd expect/need it to? |
| 05:23.50 | nsgn | essentially i'm asking if i need more than one DID for my purposes if i go per minute plan |
| 05:23.53 | p3nguin_ | As long as Vonage doesn't have any type of forwarding restrictions that would limit a second forward to work before the first one finishes, each call will create an inbound channel to Asterisk. |
| 05:24.11 | nsgn | hmm...i'm unsure about that. if that's the case i guess the DIDs are freakin cheap :D |
| 05:24.32 | nsgn | i've not had issue with vonage forwarding. i have enough contacts there i could probably get it tweaked for me if i needed |
| 05:24.51 | p3nguin_ | You can set your CID in the web portal, but you can also set it per call with Set(CALLERID(num)). |
| 05:25.19 | nsgn | ok. i'd probably do it both to be safe. it will not change |
| 05:25.45 | p3nguin_ | I leave it blank in the portal because I don't know if one overrides the other or whatever. |
| 05:25.57 | p3nguin_ | Sometimes I need to change mine, though. |
| 05:26.31 | nsgn | so now i have to choose a plan. i kindof want to go with per minute so i can not worry about bouncing to a second DID if 2 channels are in use. if i got two DIDs on the monthly plan would their 2 channels each merge into 4 available for 1 did? |
| 05:27.15 | p3nguin_ | Another thing about VoIP.ms... you have to deposit a minimum of $25 to get started, but if you are not satified with the services, they will issue you a refund for your unused funds if you just request it and cancel the service. |
| 05:27.32 | DND | guys what if the telco doesnt know what disconnect supervision is? any alternatives? |
| 05:27.35 | DND | because our VM is full of beep tones messages |
| 05:27.39 | DND | it doesnt hang up |
| 05:28.14 | p3nguin_ | So you'll only be out a few dollars if you don't like what they have. |
| 05:29.03 | p3nguin_ | I doubt that you'll get four channels on one DID by paying for two DIDs. |
| 05:29.25 | nsgn | p3nguin_: so i'd have to get fancy with vonage forwarding or just go per minute |
| 05:29.42 | p3nguin_ | You could email or do the web chat thing and ask them about the channels. |
| 05:30.04 | nsgn | at the cheaper rate i can go per minute without loosing my rear though |
| 05:30.07 | p3nguin_ | It's possible that you could pay more for extra channels on the unlimited plan. |
| 05:30.09 | nsgn | so perhaps i'll just do that |
| 05:30.52 | nsgn | our current usage projects i'd pay $20 to $40 a month |
| 05:32.03 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 05:32.56 | p3nguin_ | At 1 cent per minute per channel, go for the pay-per-minute for a month and see how it works out. |
| 05:33.11 | nsgn | p3nguin_: i guess i am missing the fact that this is in no way a big long term decision |
| 05:33.19 | nsgn | all i loose is my time configuring it |
| 05:33.29 | nsgn | ok, lets set me up voip.ms |
| 05:33.38 | nsgn | you get some fantastic referral bonus for your great sales efforts? :) |
| 05:34.15 | p3nguin_ | You'll spend more time writing the dialplan for your incoming number than you'll spend creating and activating your account and DID. |
| 05:34.22 | nsgn | hehe |
| 05:35.36 | nsgn | they have any referral deal? |
| 05:35.41 | p3nguin_ | I don't see any. |
| 05:36.23 | nsgn | when they say "callcenters" are forbidden will they be upset that asterisk answers automatically and directs calls to queues for employees? |
| 05:36.32 | p3nguin_ | nope |
| 05:36.34 | nsgn | set up like a call center, but on an absolutely rediculously small scale |
| 05:37.07 | p3nguin_ | If you have a company that does work, but needs to take calls, that's not considered a call center. |
| 05:37.30 | nsgn | account active |
| 05:37.30 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 05:37.37 | p3nguin_ | They just don't want telemarketers and stuff like that. |
| 05:37.54 | nsgn | now just make a DID? |
| 05:38.19 | p3nguin_ | Did you sign in and get the user portal? |
| 05:38.40 | p3nguin_ | On the bar, go to DID numbers, Order DIDs. |
| 05:38.59 | p3nguin_ | Under local numbers, click United States. |
| 05:39.32 | p3nguin_ | Choose your state or area code (area code is buggy as hell -- gives me other area codes in my state). |
| 05:39.50 | nsgn | yup i'm choosing one |
| 05:40.42 | p3nguin_ | Dallas is cheaper than Houston. |
| 05:40.48 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 05:40.49 | nsgn | oh the prices vary? |
| 05:40.51 | p3nguin_ | Just in case you wanted to choose Houston. |
| 05:41.05 | p3nguin_ | yeah |
| 05:41.12 | nsgn | apollo is 99c mo and 1c min |
| 05:41.18 | nsgn | that cheaper/same as houston? |
| 05:41.43 | p3nguin_ | Dallas is the same as Apollo. |
| 05:41.58 | p3nguin_ | Houston is $1.49 on DIDs. |
| 05:42.09 | nsgn | any reason to choose dallas over apollo? |
| 05:42.32 | p3nguin_ | Maybe more numbers to choose from, maybe more popular area code, etc. |
| 05:43.00 | nsgn | ok. i'm gonna go dallas cause they have a DID POP in dallas. perhaps slightly more efficient |
| 05:43.01 | *** join/#asterisk Nest0r (n=nestor@201.230.99.198) |
| 05:43.01 | Nest0r | hi |
| 05:43.41 | nsgn | hi |
| 05:44.10 | Nest0r | which should be the line to validate a null callerid? |
| 05:44.12 | nsgn | p3nguin_: "impossible to process order". they want $$$ first :) |
| 05:44.23 | Nest0r | exten => 219xxxx,n,GotoIf($[${CALLERID(num)}<1]?colgar) |
| 05:44.45 | nsgn | p3nguin_: can they be configured to automatically top off the account or must i log in every time i fear i'm running out? |
| 05:44.46 | p3nguin_ | (0027.14) <p3nguin_> Another thing about VoIP.ms... you have to deposit a minimum of $25 to get started, but if you are not satified with the services, they will issue you a refund |
| 05:45.12 | nsgn | yeah no problem with that. i want to know if i have to manually refill |
| 05:45.17 | nsgn | i dont see an auto option |
| 05:45.49 | p3nguin_ | I don't see auto, but they do have the threshold email thingy. |
| 05:46.01 | nsgn | bleh, ok |
| 05:46.04 | nsgn | that's kindof lame |
| 05:46.08 | Pikoro | I have 4 fxs lines, is there a way to display what line an inbound call is coming in on instead of the inbound caller id? |
| 05:46.21 | Pikoro | inbound from a tdm410p |
| 05:46.22 | nsgn | even my stupid toll tag refills itself on my credit card |
| 05:46.31 | p3nguin_ | Sounds like another question for an email or live chat. |
| 05:46.46 | nsgn | isnt a show stopper so i'll add my $25 and keep things moving |
| 05:47.46 | p3nguin_ | I'm kinda surprised that there isn't an auto-refill option, now that you have brought it to my attention. |
| 05:48.06 | nsgn | yeah |
| 05:48.12 | p3nguin_ | Vitelity offers it and VoIP.ms is a reseller for them. |
| 05:48.30 | nsgn | i cant use vitelity direct cheaper can i? |
| 05:48.37 | p3nguin_ | nope |
| 05:49.54 | nsgn | ok, paid them their bucks |
| 05:49.57 | nsgn | now to try and turn up my did |
| 05:50.26 | p3nguin_ | Termination is more, unlimited plan is more, DID is more, and per minute inbound is more... they are just higher all the way around. |
| 05:50.26 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:51.15 | nsgn | ok, DID is set up |
| 05:51.17 | p3nguin_ | Toll-free DID is more per month, but cheaper per minute. That's not enough to get me to move to Vitelity. |
| 05:52.02 | nsgn | i have asteriskNOW standing by on a little VM to take the sip info |
| 05:52.15 | nsgn | any config tips? i'm new to voip that doesnt involve a vonage ata :P |
| 05:52.27 | *** join/#asterisk trebaum (n=trebaum@207-67-92-30.static.twtelecom.net) |
| 05:52.33 | trebaum | Hello folks. |
| 05:52.57 | p3nguin_ | nsgn: They have both IAX2 and SIP configs for you in the portal. |
| 05:53.07 | nsgn | ah, trying |
| 05:53.30 | p3nguin_ | Watch for typos, though. I have seen a couple mistakes in the sample configs. |
| 05:53.32 | nsgn | they have any free call destinations, or should i stick a sock in it cause it's just one cent :P |
| 05:53.50 | trebaum | Does anyone know how to dial out a t1 from a specific number if I have a block of 20? |
| 05:54.03 | trebaum | I must be missing something. |
| 05:54.05 | p3nguin_ | You can call toll-free numbers for free as long as you use the value toll free route. |
| 05:54.28 | nsgn | hmm, ok. how can i be sure i'm using that? |
| 05:54.40 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 05:54.49 | p3nguin_ | Main menu, account settings |
| 05:55.01 | p3nguin_ | USA48/Canada Routing ... Toll-Free Routing |
| 05:55.27 | p3nguin_ | That's also the page where you can set the CID. |
| 05:56.36 | nsgn | ok |
| 05:56.43 | trebaum | i'm just trying to figure out how to dial from a specific number instead of from the next open line. |
| 05:57.03 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 05:57.28 | p3nguin_ | trebaum: I think you chose a bad time to come asking questions, because it seems like most everyone is gone/sleeping right now. |
| 05:57.51 | trebaum | p3nguin: I appreciate the response. I will try back during normal hours. :) |
| 05:58.00 | p3nguin_ | If you can wait about 10 hours, people should be very talkative. |
| 05:58.12 | p3nguin_ | 8-10 hours |
| 05:58.20 | trebaum | p3nguin_: yea, this is a pet project. Thanks for the FYI. |
| 05:58.45 | p3nguin_ | Lots of the regulars have PRIs/T-1s, so you'll probably get some answers in the daytime. |
| 05:58.52 | nsgn | p3nguin_: great, got that all set. too bad my new polycoms are sitting at my receiving office 20 minutes away |
| 05:59.00 | florz | trebaum: you are mixing up "numbers" and "lines" and "channels" |
| 05:59.13 | nsgn | p3nguin_: i'll have to try testing via a softphone |
| 05:59.18 | trebaum | florz: that is a very good posibility. |
| 05:59.32 | p3nguin_ | nsgn: I like zoiper, if you need a suggestion for one. |
| 06:00.06 | trebaum | florz: you know much about PRIs/t-1s? |
| 06:00.18 | florz | trebaum: what is "much"? |
| 06:00.27 | p3nguin_ | anything? :) |
| 06:00.31 | trebaum | lol. |
| 06:00.32 | nsgn | p3nguin_: i'm on ubuntu. also do i need to configure inbound routes or does the trunk handle that? |
| 06:00.32 | *** part/#asterisk Nest0r (n=nestor@201.230.99.198) |
| 06:00.34 | florz | *g* |
| 06:01.09 | trebaum | well maybe I can explain what I have, and you can give me an idea of what i don't understand. |
| 06:01.29 | trebaum | at least point me in a direction. |
| 06:01.38 | p3nguin_ | nsgn: You'll need to pass the ports through any firewall, but the register statement tells the provider where you are. |
| 06:02.02 | trebaum | I have an asterisk server w/ a t-1/pri from XO... digium te420b. |
| 06:02.05 | florz | trebaum: I kindof pointed you in a direction, but sure, asking specific questions seldom hurts ;-) |
| 06:02.30 | trebaum | I have a block of 20 DID lines (numbers?) |
| 06:02.32 | nsgn | p3nguin_: ah, firewall ports. which do i ned? |
| 06:02.32 | p3nguin_ | nsgn: If Ubuntu doesn't have zoiper in aptitude/synaptic, it probably has twinkle and/or ekiga. |
| 06:02.33 | nsgn | *need |
| 06:02.46 | p3nguin_ | nsgn: Will you use IAX2 or SIP? |
| 06:02.48 | nsgn | it has ekiga but it is busted for some reason on my system. using kphone. it suuuucks |
| 06:02.51 | nsgn | p3nguin_: SIP |
| 06:03.09 | trebaum | I want to be able to make calls from a specific number in that block. |
| 06:03.13 | p3nguin_ | nsgn: Inbound ports 5060 and a range from 10000-20000 is standard for SIP/RTP. |
| 06:03.19 | p3nguin_ | nsgn: All UDP. |
| 06:03.38 | trebaum | do I need separate to setup separate (groups?) for each (channel?)? |
| 06:03.51 | trebaum | or do I just need to go back and read the asterisk book again? |
| 06:04.01 | florz | trebaum: well, the distinction is quite intuitive: lines are those things that stick out of the wall, channels are those things that are capable of transporting a voice connection, and numbers are those things you dial in order to reach something on the telephone network |
| 06:04.01 | trebaum | trebaum: probably. :p |
| 06:04.41 | trebaum | ok so I have 1 line (t1) |
| 06:04.49 | trebaum | I have 23 channels. |
| 06:04.52 | trebaum | and 20 numbers. |
| 06:05.17 | trebaum | am I lost here... do I just need to set the CID on the outgoing call to make sure it comes from a specific one? |
| 06:05.27 | florz | exactly :-) |
| 06:05.49 | p3nguin_ | nsgn: On the other hand, if you go with IAX2, you only need udp:4569 forwarded. |
| 06:05.54 | trebaum | florz: thanks... you answered my question. |
| 06:06.36 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 06:09.53 | nsgn | ok, firewall reconfigured |
| 06:09.57 | nsgn | having stupid kphone issues |
| 06:09.58 | nsgn | hate it |
| 06:10.01 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:10.18 | nsgn | getting registration failed when i know dang well the proper access to my server |
| 06:10.33 | p3nguin_ | nsgn: Never heard of it. |
| 06:10.56 | nsgn | gonna try reinstalling ekiga |
| 06:11.12 | p3nguin_ | Or skip it and use twinkle or zoiper. |
| 06:11.24 | nsgn | which is better? |
| 06:11.51 | nsgn | ah, found twinkle as an ubuntu package. trying |
| 06:11.55 | p3nguin_ | I've never used zoiper on Linux, but I prefer it over most other free ones for Windows. |
| 06:12.18 | p3nguin_ | When you say found as a package, you are using the package manager, right? |
| 06:12.25 | p3nguin_ | aptitude install twinkle |
| 06:12.44 | p3nguin_ | sudo aptitude ... |
| 06:12.46 | nsgn | yeah i've got it now. configuring |
| 06:13.07 | *** part/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 06:13.25 | nsgn | registration failed service unavailable |
| 06:13.30 | nsgn | oook maybe i do have an issue |
| 06:13.45 | p3nguin_ | sip set debug |
| 06:15.47 | nsgn | i'm getting 503'd by asterisknow |
| 06:16.16 | nsgn | which is a bit odd cause i know it's at that IP. i am on it's web interface that way. somehow sip isn't serving |
| 06:16.36 | p3nguin_ | sudo asterisk -r |
| 06:16.40 | p3nguin_ | sip set debug |
| 06:16.50 | p3nguin_ | register the phone. |
| 06:16.54 | p3nguin_ | See what it says. |
| 06:17.40 | nsgn | threw sip set debug back in my face |
| 06:17.42 | nsgn | no such command |
| 06:18.05 | p3nguin_ | eh... |
| 06:18.18 | p3nguin_ | I think you skipped the sudo asterisk -r part. |
| 06:18.23 | nsgn | nope |
| 06:18.25 | nsgn | i'm at the CLI |
| 06:18.29 | p3nguin_ | hmm |
| 06:18.36 | p3nguin_ | wtf |
| 06:18.46 | p3nguin_ | How can asterisk console not have sip set? |
| 06:18.54 | nsgn | tried "help sip set" as it suggested and it says no such command |
| 06:18.56 | nsgn | wwwwtf |
| 06:19.16 | nsgn | when i installed it went off without a hitch |
| 06:19.23 | nsgn | and the webgui has no issue loading configs into * |
| 06:19.32 | nsgn | and the CLI calls up |
| 06:19.47 | p3nguin_ | Without some debug, it's hard to say what's going on. |
| 06:19.49 | nsgn | but i'm getting "no such command 'sip'" |
| 06:19.58 | nsgn | how the heck am i missing sip in *? |
| 06:20.20 | p3nguin_ | Maybe it's a simple twinkle configuration mistake that you can fix without needing to debug. |
| 06:20.32 | nsgn | i got 503'd in kphone too |
| 06:20.35 | nsgn | its an * issue :/ |
| 06:20.45 | p3nguin_ | oh |
| 06:21.10 | nsgn | i say it's time to reboot the * server :) |
| 06:21.12 | p3nguin_ | ~asterisknow |
| 06:21.13 | infobot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 06:21.54 | nsgn | rebooting the sucker. we'll see how it feels |
| 06:22.04 | p3nguin_ | I predict no change. |
| 06:22.27 | p3nguin_ | It's in a VM? |
| 06:23.17 | nsgn | p3nguin_: yes it is |
| 06:23.21 | p3nguin_ | VMware, VirtualBox, something else? |
| 06:24.40 | nsgn | vmware server |
| 06:24.49 | nsgn | on reboot i'm getting "asterisk is not running; this is a critical service" |
| 06:24.50 | p3nguin_ | Are you using NAT or bridged networking? |
| 06:25.09 | nsgn | i'm networking to it fine. asterisk is screwing up. my vmware setup is solid. i do it for a living |
| 06:25.12 | nsgn | this is * |
| 06:25.33 | p3nguin_ | Okay. I'll have to trust you on this. |
| 06:25.55 | nsgn | i've done a few things with * but never seen an install come up without the actual asterisk service running |
| 06:25.58 | nsgn | what the heck gives |
| 06:26.04 | p3nguin_ | *shrug* |
| 06:26.22 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:26.32 | p3nguin_ | I've never touched asteriskNOW. |
| 06:29.06 | p3nguin_ | Hmm, I did try PiaF once. Hated it! |
| 06:29.24 | p3nguin_ | That was a waste of disk space. |
| 06:31.06 | *** join/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net) |
| 06:31.47 | gilevy | i'm getting this error: sip_call: No audio format found to offer. Cancelling call |
| 06:31.59 | gilevy | anybody run into this? |
| 06:32.34 | gilevy | i believe i have all the codecs |
| 06:35.10 | nsgn | ok for whatever reason after a second reboot i can register and * is ok...<_< |
| 06:35.35 | nsgn | now to set up outbound and try to call on voipms |
| 06:35.56 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 06:37.07 | nsgn | "all circuits are busy now"! |
| 06:39.22 | p3nguin_ | Hmm. I've heard that before, but I do not remember why. |
| 06:40.13 | nsgn | it's when * can't find a way to get your call out. either the trunk isnt registering or i can't get the outbound routing right |
| 06:40.31 | nsgn | the outbound routing is dead simple..so how might i verify the trunk to voipms is indeed registered? |
| 06:40.40 | p3nguin_ | sip show peers |
| 06:41.13 | p3nguin_ | But even if it's not registered, it's still possible to make calls. |
| 06:41.32 | nsgn | displays only my sip phone |
| 06:41.52 | nsgn | guess that means its not working :) |
| 06:42.09 | p3nguin_ | voipms isn't even on the list? |
| 06:42.10 | nsgn | wait, that may not be where it is |
| 06:42.15 | nsgn | because its not in the list |
| 06:42.17 | nsgn | this is only extensions |
| 06:42.28 | p3nguin_ | sip show registry |
| 06:43.24 | nsgn | sip.us2.voip.ms:5060 unregistered |
| 06:43.56 | nsgn | so there's the issue |
| 06:44.04 | nsgn | however i am reviewing my settings and they all appear correct...? |
| 06:46.09 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 06:48.50 | gilevy | i'm getting a no audio format found to offer when i try to dial to a local extension after answering a sipgate incoming sip |
| 06:49.03 | gilevy | anybody have any suggestions or tips? |
| 06:49.17 | nsgn | p3nguin_: found dns issue, resolving |
| 06:51.04 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 06:53.48 | nsgn | registered! calling a 1800 |
| 06:54.14 | p3nguin_ | :) |
| 06:54.19 | nsgn | works |
| 06:54.24 | p3nguin_ | cool |
| 06:54.29 | nsgn | now gotta configure and test incoming |
| 06:54.38 | nsgn | any tips there? |
| 06:55.02 | DND | guys anyone know how to setup physical fax machine |
| 06:55.02 | DND | ? |
| 06:55.19 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 06:55.25 | p3nguin_ | Well, make sure the DID is configured for your POP and that it's set to SIP. That's about it. |
| 06:56.02 | p3nguin_ | Make sure the SIP peer has the right information and that it has your valid context for incoming calls. |
| 06:56.18 | nsgn | sooo let me ask this |
| 06:56.31 | nsgn | when i call the DID from my cell phone i get an AT&T "your call can not be completed as dialed" |
| 06:56.45 | nsgn | did i get a screwey DID or why can't AT&T recognize it? |
| 06:57.42 | *** join/#asterisk s14ck (n=jtorres@190.198.118.119) |
| 06:57.45 | p3nguin_ | Check sip show peers again. If voipms isn't listed, that would explain why. |
| 06:58.19 | nsgn | it's listed and has a status of "unmonitored" |
| 06:58.31 | nsgn | hang on lemme change the NAT flag |
| 06:58.59 | p3nguin_ | host is unspecified or 74.54.54.178? |
| 06:59.09 | nsgn | host is 74.54.54.178 |
| 06:59.12 | nsgn | status is unmonitored |
| 06:59.33 | p3nguin_ | There's probably no matching context for the incoming call, then. |
| 06:59.38 | p3nguin_ | sip set debug |
| 06:59.42 | p3nguin_ | try the call again. |
| 06:59.51 | nsgn | i dont think asterisk is giving me that message. it's identical to AT&T giving me the message (heard in past for other reasons) |
| 07:00.00 | nsgn | ok, debugging enabled. calling |
| 07:00.24 | nsgn | oh wow ok i'm deff hitting * |
| 07:00.28 | nsgn | i gotta fix my contexts :) |
| 07:00.51 | nsgn | so how do i do that in a simple manner? i'm sorely rusty |
| 07:00.58 | nsgn | i've done this once before in * about a year ago |
| 07:01.19 | p3nguin_ | Do you have a context in extensions.conf that matches your DID? |
| 07:02.12 | nsgn | that i do now |
| 07:02.13 | nsgn | *not |
| 07:02.22 | p3nguin_ | Let me give you the concept real quick, just in case you didn't already know... |
| 07:02.56 | p3nguin_ | The context= line for your peer defines where the call will go looking for a matching extension in extensions.conf. |
| 07:03.25 | nsgn | the context is "from-trunk", as per voipms' example |
| 07:03.53 | p3nguin_ | Okay, then in extensions.conf, you must have a [from-trunk] context with matching extens. |
| 07:04.07 | p3nguin_ | That should also have been in the configuration sample. |
| 07:06.16 | p3nguin_ | Maybe something like this: [from-trunk] exten => 8505202233,1,Dial(SIP/2001,30,k) |
| 07:06.29 | p3nguin_ | It might be one of those things that is typoed in the sample. |
| 07:06.54 | nsgn | yeah, i dont have that |
| 07:07.02 | p3nguin_ | Let me look real quick. |
| 07:07.08 | nsgn | so either freepbx screwed it up or their example was really poor |
| 07:08.03 | nsgn | ahha well i got it by having a generic blank incoming context |
| 07:08.04 | nsgn | a catch all |
| 07:08.08 | nsgn | which is actually OK for my situation |
| 07:08.18 | s14ck | hello everybody |
| 07:08.26 | nsgn | left context blank and told it to go to the softphone i'm testing with. rang right away |
| 07:08.49 | nsgn | hello |
| 07:09.01 | p3nguin_ | I looked at the sample, and it's very poor, but it doesn't match what you said you found on there. |
| 07:09.21 | p3nguin_ | https://www.voip.ms/m/samples/sipconfauth.php |
| 07:10.05 | nsgn | p3nguin_: here's the next weird thing. i call it and it goes through. i hang up and call it right back and i get "your call cannot be completed at this time" |
| 07:10.19 | nsgn | is it how asterisk handles the one extension being busy? |
| 07:10.31 | nsgn | if i had it in a ring group with voicemail as the second option perhaps it'd work? |
| 07:10.33 | p3nguin_ | maybe. What does the debug say? |
| 07:12.19 | p3nguin_ | I have asterisk answer my calls for me, so my being on the phone never causes a busy condition. |
| 07:13.04 | s14ck | somebody know how can I set a mfcr2 trunk with one E1 in asterisk and a panasonic PBX? |
| 07:13.09 | nsgn | yeah, thats the eventual plan. answer, play something nice introducing us, drop them in a queue till i'm off the phone |
| 07:13.22 | p3nguin_ | That's super simple. |
| 07:13.39 | p3nguin_ | And if it can wait until another time, I would be happy to help you do it. |
| 07:13.49 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 07:14.14 | nsgn | p3nguin_: argh, no, it has to be something else |
| 07:14.26 | nsgn | i just set up a ring group with the option to play a sound file if extensions are busy |
| 07:14.36 | nsgn | and two quick calls in a row get me "try your call again later" |
| 07:14.51 | p3nguin_ | Again, the debug will tell you what's going on. |
| 07:15.13 | *** join/#asterisk sergey (n=sergey@91.189.232.114) |
| 07:16.25 | s14ck | nsgn, do you know something about R2 in asterisk and panasonic pbx? |
| 07:16.27 | nsgn | p3nguin_: unless i'm mistaken the second time the call never hits * |
| 07:16.28 | p3nguin_ | Are you calling, hanging up, then calling again from the same phone? |
| 07:16.44 | nsgn | yes |
| 07:16.58 | nsgn | though that phone's battery just died :), but that's what i was doing yes |
| 07:17.47 | nsgn | s14ck: nope. i'm new here too |
| 07:17.55 | nsgn | i know everything about vonage, but that's about it ;) |
| 07:18.24 | *** join/#asterisk robinreekers_ (n=robin@213.144.242.211) |
| 07:18.41 | p3nguin_ | http://paste-bin.com/view/60de6937 |
| 07:18.52 | p3nguin_ | No problems. |
| 07:19.08 | nsgn | you tested the same? |
| 07:19.25 | p3nguin_ | I just did what you said you were doing. |
| 07:19.38 | p3nguin_ | From my cell, call, hang up, call, hang up, etc. |
| 07:19.47 | nsgn | then what's up with my setup? |
| 07:20.01 | p3nguin_ | no clue until you paste debugs and your configs. |
| 07:20.38 | s14ck | I need some help with R2 here |
| 07:21.04 | p3nguin_ | According to my verbose output, my calls were between 6 and 9 seconds apart. Was that too long in between? |
| 07:22.01 | nsgn | p3nguin_: i just did testing from my vonage phone after the att cell died |
| 07:22.01 | p3nguin_ | I waited until I heard playback, then hang up, then repeat. |
| 07:22.24 | nsgn | the cell dials faster than the analog phone i have on vonage. took me a few tries but i could still recreate |
| 07:22.42 | nsgn | and it's definately not hitting * |
| 07:22.49 | nsgn | because the message i was played is different on vonage |
| 07:22.59 | p3nguin_ | That doesn't mean anything. |
| 07:23.14 | nsgn | basically it's acting like i've only got one channel and it's busy |
| 07:23.21 | p3nguin_ | The debug shows all this. |
| 07:23.35 | nsgn | the debug scrolls lines of junk for the first call, and nothing for the second |
| 07:24.13 | p3nguin_ | thinks |
| 07:24.34 | nsgn | its like voip.ms is treating it as single channel, and throwing back to the incoming call's carrier that the line is busy |
| 07:25.16 | p3nguin_ | I wonder if there's congestion on the origination side. |
| 07:25.57 | nsgn | would that be voip.ms sucking it up with their peering? |
| 07:26.17 | ChannelZ | When the fuck is Apple going to figure out that not everyone has an iPod or iPhone, nor wants either of those devices, and stops running 5 services in the background to support them? |
| 07:26.20 | p3nguin_ | I don't really know how calls go from PSTN into the VoIP network, but maybe there's a traffic jam. |
| 07:26.22 | nsgn | would it be because i'm using the value termination rather than premium? |
| 07:26.52 | nsgn | ChannelZ: when you learn how to use one of the many free utilities to tweak the services :) |
| 07:27.04 | p3nguin_ | There shouldn't be a value or premium route on DIDs. |
| 07:27.44 | kaldemar | ChannelZ: go buy one and the services don't run in vein anymore. :) |
| 07:28.02 | ChannelZ | I disable them, iTunes just re-enables them. I wrote a stub service for the iPod thing and replaced the binary |
| 07:28.07 | p3nguin_ | I'm even using us2 just like you are and I didn't have the problem that you described. |
| 07:28.24 | *** join/#asterisk QaDeS (n=mklaus@p4FC72C23.dip0.t-ipconnect.de) |
| 07:28.51 | p3nguin_ | How quickly do I need to redial? Is 6-9 seconds fast enough? |
| 07:28.53 | nsgn | p3nguin_: well what are some stabs at what it could be in *? |
| 07:29.14 | nsgn | cause i've setup a ring group that resolves incoming congestion |
| 07:29.25 | p3nguin_ | I'm not sure, since you said there was no debug output on the second call. |
| 07:29.28 | nsgn | by playing sounds for testing purposes if my softphone is busy |
| 07:29.41 | nsgn | and it works, tested by denying a call from my softphone. i get dropped to sounds |
| 07:30.18 | *** join/#asterisk xrmx__ (n=rm@host22-183-dynamic.56-82-r.retail.telecomitalia.it) |
| 07:30.33 | p3nguin_ | Even when you get the failed call announcement from your phone company, the call can still be hitting * and be rejected. |
| 07:30.54 | nsgn | p3nguin_: is there a better way to study the debug than watching it fly by and missing it all? :) |
| 07:30.57 | p3nguin_ | Might be a single line in the debug. |
| 07:31.08 | nsgn | i think the issue is the ocean of it on my tiny remote console |
| 07:31.22 | nsgn | which i can't scroll up courtesy of vmware |
| 07:31.29 | p3nguin_ | Well, maybe... I just increase my scrollback lines to 20000 and then scroll up. |
| 07:31.45 | nsgn | i need to log it to file somewhere |
| 07:32.02 | nsgn | that way i can paste it to ya |
| 07:32.12 | nsgn | wait, i'm a moron at 2:30am |
| 07:32.14 | nsgn | i'll SSH in |
| 07:32.15 | nsgn | geezzzz |
| 07:32.17 | p3nguin_ | /var/log/asterisk/messages? |
| 07:33.15 | nsgn | ok, debug enabled. let me power up cell phone again since it can call way faster |
| 07:33.35 | p3nguin_ | Let me know how many seconds between calls will produce the problem. |
| 07:33.51 | nsgn | p3nguin_: i can re-initeate a call less than 2 seconds after terminating one on my iphone |
| 07:33.58 | nsgn | and that's kindof what i was doing |
| 07:34.16 | p3nguin_ | Oh, my verbose output was from the start to start of each call. |
| 07:34.25 | p3nguin_ | I can hang up and redial in .5 seconds. |
| 07:34.51 | nsgn | blehh, phone is in "i'm really dead tired" mode. we can wait or you can do it if you'd like |
| 07:35.43 | nsgn | if you care to try i'll PM you the did |
| 07:35.49 | p3nguin_ | I can't reproduce your problem. I ended and redialed just as quickly as possible and it went through every time. |
| 07:36.07 | nsgn | p3nguin_: i mean do it on my DID |
| 07:36.08 | p3nguin_ | Better send in a notice; PMs are blocked. |
| 07:36.10 | nsgn | obviously yours works |
| 07:36.33 | nsgn | since my cell phone is dead and wont power up till it charges for a bit |
| 07:36.53 | p3nguin_ | I'll do it on yours if you notice it to me. |
| 07:36.55 | nsgn | and how the heck do i do a note. /msg? |
| 07:37.12 | p3nguin_ | /notice p3nguin_ |
| 07:38.02 | nsgn | went into the abyss, so i hope you got it |
| 07:38.02 | nsgn | should i answer when you call or just deny it? |
| 07:38.02 | p3nguin_ | I did. |
| 07:38.02 | p3nguin_ | Just ignore it. |
| 07:38.05 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:38.08 | nsgn | k |
| 07:38.18 | nsgn | ringing |
| 07:38.25 | nsgn | stopped ringing |
| 07:38.28 | nsgn | ringing |
| 07:38.38 | nsgn | stopped ringing |
| 07:38.43 | nsgn | ringing |
| 07:38.50 | nsgn | wtf apparently AT&T just sucksss |
| 07:39.03 | p3nguin_ | Is that how you were testing it? |
| 07:39.05 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:39.15 | nsgn | pretty much |
| 07:39.25 | nsgn | phone just woke up from it's dead slumber. let me see if i can recreate on log |
| 07:39.58 | p3nguin_ | I was just waiting until I knew * was taking the call (the audio changed slightly), then hangup/redial. |
| 07:40.18 | nsgn | there we go |
| 07:40.20 | nsgn | first try i got it |
| 07:40.26 | nsgn | i answered. perhaps that's required |
| 07:40.30 | nsgn | here comes a paste of the log |
| 07:40.35 | p3nguin_ | not here |
| 07:40.45 | p3nguin_ | pastebin! |
| 07:41.20 | p3nguin_ | If you want me to try it again and answer it each time, I'll do it again just as soon as you say you're ready. |
| 07:42.06 | nsgn | hang on, i gotta do it again |
| 07:42.10 | nsgn | the stupid log didnt log long enough |
| 07:42.21 | nsgn | and my softphone is spewing stuff in the log every 10 seconds |
| 07:42.59 | p3nguin_ | You can reduce some of that by debugging by IP address or peer name. |
| 07:43.01 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 07:43.23 | p3nguin_ | sip set debug ip 74.54.54.178 maybe |
| 07:44.05 | nsgn | ok dude it is NOT hitting * |
| 07:44.30 | nsgn | i just tried from the AT&T cell phone about 2 minutes after the last dry and got the "dun dun dun dun happy song your call can not go through" |
| 07:44.31 | p3nguin_ | I'll do it again. Just to make sure I got the plan right, once you answer, disconnect and call back, right? |
| 07:44.38 | nsgn | yeah |
| 07:44.52 | p3nguin_ | Okay, you answer and I'll hang up on you. :) |
| 07:45.26 | nsgn | ha, it did it to you didnt it |
| 07:45.30 | p3nguin_ | yep |
| 07:45.37 | nsgn | i got NOTHING in the log after the first call |
| 07:46.14 | p3nguin_ | I've never encountered such a thing. |
| 07:46.17 | nsgn | so i got a sucky trunk provider, huh? ;P |
| 07:46.34 | p3nguin_ | It's the same one I'm on. |
| 07:46.41 | p3nguin_ | us2 |
| 07:46.48 | nsgn | yeah. question is what is wrong with my DID that it can't take two calls in quick succession? |
| 07:46.52 | nsgn | they never make it to * |
| 07:47.14 | nsgn | wanna know what blows my mind? |
| 07:47.26 | nsgn | they don't hit the CDR on the panel |
| 07:47.31 | nsgn | voipms' panel |
| 07:47.40 | nsgn | the call i answered is there, the call you attempted is not |
| 07:48.05 | p3nguin_ | I gotta check the ones I did on my own. |
| 07:48.24 | p3nguin_ | All 9 are listed. |
| 07:48.48 | p3nguin_ | Calls: 9 | Duration: 00:00:40 | Total: $0.0261 |
| 07:48.59 | nsgn | yeah but yours all hit * |
| 07:49.12 | p3nguin_ | Yeah, that's the weird part. |
| 07:50.31 | nsgn | they don't have live chat support do they? |
| 07:50.39 | p3nguin_ | If I could reproduce it on my end, I could try to figure out if there's any tiny possibility that it's configuration related. |
| 07:50.56 | p3nguin_ | Yeah, they do, but I don't know if anyone answers at this time of night. |
| 07:50.59 | nsgn | p3nguin_: when it doesnt hit * it's out of my config's hands. it's voipms |
| 07:51.04 | nsgn | i can't find it on the site |
| 07:51.37 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 07:52.20 | *** join/#asterisk loather-work (n=khudson@internal-nat.djnetworks.net) [NETSPLIT VICTIM] |
| 07:52.22 | nsgn | p3nguin_: it wouldnt have something to do with NAT/firewall, would it? |
| 07:52.39 | nsgn | voipms literally not being able to hit me with a second call until the first one completely dies off? |
| 07:52.42 | nsgn | that would make sense here |
| 07:52.59 | p3nguin_ | It's hard to say, really. I would troubleshoot it from all angles. |
| 07:53.13 | p3nguin_ | Live chat is offline. |
| 07:53.48 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:54.25 | p3nguin_ | On the main page (non-https), it's on the left column. Got a chick's face on the button for it. |
| 07:54.45 | nsgn | ok |
| 07:54.57 | nsgn | you sure you gave me proper firewall stuff? |
| 07:54.59 | p3nguin_ | I bet they'll be on at 8 or 9. |
| 07:55.10 | nsgn | 5060 UDP and 10000-20000 UDP |
| 07:56.05 | p3nguin_ | Those are the normal ones to forward, but there's still the NAT thing involved. |
| 07:56.38 | nsgn | well NAT is not an issue with static forwards, from an inbound perspective |
| 07:56.51 | p3nguin_ | I'm half tempted to switch to SIP and test mine again. |
| 07:57.09 | nsgn | think it's a SIP issue? |
| 07:57.12 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 07:57.15 | p3nguin_ | No. |
| 07:57.28 | p3nguin_ | But it's a difference between our setups. |
| 07:57.32 | nsgn | any reason for me to try it IAX? |
| 07:57.41 | nsgn | seems really stinkin easy to change |
| 07:58.01 | p3nguin_ | Might give it a try. Port udp:4569 |
| 07:58.47 | p3nguin_ | It can't really hurt anything, and it could be another piece in the troubleshooting puzzle if it works. |
| 08:00.29 | p3nguin_ | channel 10: open failed: administratively prohibited: open failed |
| 08:00.40 | p3nguin_ | whoops, pastefail |
| 08:01.41 | *** join/#asterisk TommyBotten (n=noosjent@services.nbi.freecode.no) |
| 08:02.34 | p3nguin_ | What the heck is eating up 5 gigs of RAM? |
| 08:02.44 | nsgn | p3nguin_: ok, confirmed |
| 08:02.48 | nsgn | issue happens still |
| 08:03.15 | nsgn | yay crappers |
| 08:03.17 | p3nguin_ | Then there's no need for me to test SIP. |
| 08:03.50 | nsgn | yup |
| 08:03.58 | nsgn | it's a voip.ms issue |
| 08:04.07 | nsgn | which is really kindof ticking me off cause i can't use this until it's fixed |
| 08:05.11 | nsgn | and it has to be so solidly fixed that it doesn't happen in 100 tests. i've experienced customers getting bad phone messages on my number in the past. it really screws things up |
| 08:05.12 | p3nguin_ | My concern is why it happens on yours but not mine. |
| 08:05.49 | nsgn | bad DID? |
| 08:05.58 | nsgn | there such a thing? |
| 08:06.22 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 08:06.23 | p3nguin_ | I can't imagine there is. |
| 08:06.45 | nsgn | i've had ONE before |
| 08:06.53 | p3nguin_ | One bad one? |
| 08:06.58 | nsgn | was a DID generated by a regional wireless carrier, then ported to AT&T |
| 08:07.31 | nsgn | some carriers couldnt call it because somehow their system tried to handle the call with the regional carrier though my number was no longer with them |
| 08:07.37 | trogs | heh, one of my suppliers has a dodgy channel on their e1, every now and again i'll get a call from them and it will be crackly. |
| 08:08.08 | nsgn | i had to hunt down high level techs at two major T1 providers here in town after discovering and putting together the mental puzzle that certain customers of mine couldnt call me |
| 08:08.11 | nsgn | that SUCKED |
| 08:08.19 | p3nguin_ | trogs: At least you're still getting the call, though. |
| 08:08.48 | nsgn | p3nguin_: so is their support good? if they can't fix this i can't use it |
| 08:09.08 | trogs | true, true. |
| 08:09.32 | p3nguin_ | nsgn: Since I've never had any issues, I can't say what support will be like. |
| 08:09.42 | trogs | i dunno why they don't just send calls to us over the SIP trunk we have to them. |
| 08:10.03 | trogs | must be too hard for their cisco call manager engineers to figure out. |
| 08:10.33 | nsgn | i dont know why i cant freaking get two calls in a row |
| 08:10.38 | p3nguin_ | nsgn: I've had some questions in the past and the guy seemed pretty ignorant, but I've never had a _real_ support issue. |
| 08:10.44 | *** join/#asterisk oilinki (n=oil@208.109.22.97) |
| 08:10.46 | nsgn | lovely |
| 08:11.01 | p3nguin_ | support@voip.ms |
| 08:11.15 | p3nguin_ | Email him now so he gets it when he comes in. |
| 08:11.27 | p3nguin_ | Screw live chat. |
| 08:12.30 | p3nguin_ | If they can't solve it and you're 300% sure there is no way it is a configuration problem with your side, ask for a refund. |
| 08:13.30 | p3nguin_ | Bring it up later when more people are active here, and maybe someone knows something. |
| 08:13.55 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:14.12 | p3nguin_ | I hate to think they can't explain this. |
| 08:15.31 | p3nguin_ | I just had another idea. |
| 08:16.07 | p3nguin_ | Maybe. |
| 08:16.08 | nsgn | p3nguin_: there we go, emailed them |
| 08:16.12 | nsgn | what's on your mind? |
| 08:16.29 | p3nguin_ | I'm wondering if there's anyway to put your DID on my box for testing. |
| 08:16.38 | p3nguin_ | Without taking mine offline, that is. |
| 08:16.52 | nsgn | i'm 100% sure it's not in *, unless something in * doesn't log...because i can see every detail of the working call and absolutely nothing of the nonworking call |
| 08:17.08 | p3nguin_ | core set verbose 100 |
| 08:17.19 | nsgn | let's try this |
| 08:17.23 | nsgn | you and i both ring it at the same moment |
| 08:17.29 | p3nguin_ | okay |
| 08:17.37 | nsgn | so we can see if it's congestion of their peering to other carriers |
| 08:17.40 | nsgn | or congestion to my * box |
| 08:17.52 | nsgn | but possible at the NAT/firewall level, since it isnt hitting * |
| 08:17.59 | nsgn | standing by? |
| 08:18.12 | p3nguin_ | yep |
| 08:18.25 | nsgn | go |
| 08:18.38 | nsgn | boom, got denied |
| 08:18.43 | p3nguin_ | fast busy |
| 08:18.49 | nsgn | yeah....ok. WTF |
| 08:18.57 | nsgn | asterisk's log showed nothing |
| 08:19.02 | nsgn | know what this sounds like? |
| 08:19.09 | nsgn | NAT ruining my day |
| 08:19.27 | p3nguin_ | sadface |
| 08:19.31 | nsgn | except i'm on IAX now |
| 08:20.52 | p3nguin_ | You also got the fast busy? |
| 08:20.52 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 08:20.57 | nsgn | so why in the heck can i not get some incoming calls? i'm starting to think it's more by chance now. that my registration is having issues |
| 08:21.16 | nsgn | i dont get fast busy. i get the same happy att song followed by "you call can not be completed as dialed" |
| 08:21.24 | nsgn | on vonage i get fast busy |
| 08:21.44 | p3nguin_ | Check ping times, check qualify times, core set verbose 100. |
| 08:22.00 | nsgn | "verbosity was 3 and is now 100" |
| 08:22.08 | nsgn | how do i check qualify times? |
| 08:22.21 | p3nguin_ | qualify=yes in the peer. |
| 08:22.52 | p3nguin_ | Then iax show peers. |
| 08:23.03 | p3nguin_ | (after iax reload, of course) |
| 08:23.23 | nsgn | added qualify=yes to my iax details |
| 08:23.26 | nsgn | reloaded config |
| 08:23.30 | nsgn | tons of crud scrolls by |
| 08:23.44 | nsgn | what now? |
| 08:23.52 | p3nguin_ | iax2 show peers |
| 08:24.41 | nsgn | voipms/XXXXXX 74.54.54.178 (S) 255.255.255.255 4569 OK (39 ms) |
| 08:24.52 | p3nguin_ | 39 is low enough. |
| 08:25.19 | p3nguin_ | ping times are probably less than that. |
| 08:25.25 | *** join/#asterisk Snoogan (n=asdf@115.69.179.5) |
| 08:25.42 | p3nguin_ | I don't think it's your registration. |
| 08:25.57 | *** join/#asterisk baijum (n=baiju@203.123.174.162) |
| 08:26.23 | nsgn | wtf else do i have to chase though? |
| 08:27.10 | p3nguin_ | With verbose set really high, call twice and hope it messes up. |
| 08:27.14 | nsgn | ok let's both hit it at the same moment again |
| 08:27.16 | nsgn | standing by? |
| 08:27.22 | p3nguin_ | ready |
| 08:27.25 | nsgn | go |
| 08:27.55 | nsgn | yours got it but not mine |
| 08:28.00 | p3nguin_ | I guess I won the race. |
| 08:28.05 | nsgn | nope |
| 08:28.09 | nsgn | i did |
| 08:28.18 | nsgn | the att message was playing before your call hit the logs |
| 08:28.27 | nsgn | and not a thing hit the log while i was getting my denied message |
| 08:28.29 | p3nguin_ | how's THAT happen? |
| 08:28.33 | nsgn | it seriously is NOT touching * |
| 08:28.49 | p3nguin_ | Okay, I'm convinced. |
| 08:28.52 | nsgn | i think we can be 100% certain now |
| 08:29.04 | nsgn | of interest |
| 08:29.09 | nsgn | i can't freaking call it at all from AT&T now |
| 08:29.17 | p3nguin_ | See what support has to say about it in the morning. |
| 08:29.19 | nsgn | trying several times in a row here and getting nothing |
| 08:29.29 | nsgn | and now i call again and get it |
| 08:29.49 | nsgn | i think i've just got a randomly on/off DID |
| 08:30.01 | nsgn | or, more likely, an issue with incoming calls getting to my * box |
| 08:31.37 | p3nguin_ | While I do not wish to have worse service, I wish this problem would surface more often so we can have a definitive answer as to what and why. |
| 08:32.17 | nsgn | have you ever had it, even a few times? |
| 08:32.56 | p3nguin_ | I've never before tested it in the manner that you were. |
| 08:33.36 | nsgn | google tells me to play with "register expires" |
| 08:33.54 | nsgn | default is 3600 and someone got success setting it to 300 |
| 08:34.00 | nsgn | i cant find that setting in * but it makes sense |
| 08:34.09 | nsgn | tells it to keep alive more often, basically |
| 08:34.51 | p3nguin_ | I did find out yesterday that my DID was on failover for an unknown reason. Couldn't see anything wrong, so I reloaded iax2 and the problem was solved, back to primary again. |
| 08:35.02 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 08:35.18 | p3nguin_ | I guess I lost registration at some point. |
| 08:35.33 | nsgn | where can i set the register expires setting? i can't find it on * or on their panel |
| 08:36.06 | kaldemar | nsgn: still on IAX2? |
| 08:36.42 | nsgn | kaldemar: yes |
| 08:36.45 | nsgn | issue happens on both |
| 08:37.02 | nsgn | some calls, sometimes up to 50% of them, are not hitting * at all from the DID |
| 08:37.19 | nsgn | debug verbosity is all the way up. they arent touching * one bit |
| 08:37.31 | kaldemar | minregexpire and maxregexpire are the settings, but those should only affect incoming registrations. |
| 08:39.10 | nsgn | now i'm failing to get in more often than not. multiple calls in a row fail |
| 08:39.12 | p3nguin_ | I don't see any place to configure that on the server side. |
| 08:39.38 | nsgn | i dont either |
| 08:39.44 | nsgn | i swear i saw something about it in * |
| 08:40.00 | nsgn | if i do iax2 show registry it has a "refresh" of 60 |
| 08:40.03 | nsgn | fwit |
| 08:40.05 | nsgn | *fwiw |
| 08:40.28 | nsgn | what is that, and can it possibly be shortened to assist? |
| 08:42.24 | p3nguin_ | That's how often the server allows you to refresh the registration. |
| 08:43.14 | p3nguin_ | I don't think you can change it. |
| 08:43.22 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:43.25 | p3nguin_ | I don't see any setting for it as a client. |
| 08:43.51 | nsgn | i'm just looking for some way to hang onto the server better. hit it more often. i've got my firewall set how it should be but i'm curious if somehow the hole is closing |
| 08:44.06 | nsgn | i've got many other realtime services forwarding through this without issue...but its what it feels like it's doing |
| 08:45.38 | p3nguin_ | We could eliminate that by putting the DID on my box. |
| 08:45.46 | nsgn | ok now i cant frickin call it at ALL |
| 08:45.48 | nsgn | try me again |
| 08:46.07 | nsgn | yeah it works from you |
| 08:46.12 | p3nguin_ | Got a horn in the ear. |
| 08:46.14 | nsgn | cant call at all from AT&T cell now |
| 08:46.54 | p3nguin_ | Was that horn when you answered? |
| 08:47.35 | *** join/#asterisk robinreekers_ (n=robinree@213.144.242.211) |
| 08:47.49 | nsgn | i didnt hear anything |
| 08:47.52 | nsgn | i answereed |
| 08:47.55 | nsgn | then you hung up |
| 08:48.05 | p3nguin_ | It was too loud, so I had to. |
| 08:48.11 | nsgn | wow |
| 08:48.16 | nsgn | well...i have no clue |
| 08:48.22 | nsgn | so you wanna put my DID on your box, huh |
| 08:48.25 | *** part/#asterisk robinreekers_ (n=robinree@213.144.242.211) |
| 08:48.32 | p3nguin_ | I'd like to try it. |
| 08:49.05 | nsgn | ok, what do we gotta do |
| 08:49.11 | nsgn | you just need acct # and secret i guess |
| 08:49.44 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 08:49.44 | p3nguin_ | Yep. You can create a sub account if you prefer that. |
| 08:49.53 | p3nguin_ | Just make sure you route the DID to the sub if you do. |
| 08:49.54 | nsgn | i'm just gonna change the secret to something else |
| 08:50.08 | nsgn | that works the same for secure purposes, correct? |
| 08:50.38 | nsgn | its actually probably what we should do to accurate re-create |
| 08:50.40 | p3nguin_ | Yeah, just change the IAX/SIP secret. |
| 08:50.44 | nsgn | i'm gonna un-register my box |
| 08:51.09 | nsgn | ok |
| 08:51.22 | p3nguin_ | works |
| 08:51.40 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 08:52.38 | nsgn | i am now entirely un-registered and disabled |
| 08:53.36 | p3nguin_ | Configuring now. |
| 08:55.20 | *** join/#asterisk robstar_ (n=robstar_@213.144.242.211) |
| 08:56.30 | p3nguin_ | Hammer away on it if you want. |
| 08:56.50 | nsgn | first try. call can not be completed |
| 08:56.57 | nsgn | I GOT A F'D UP DID. god shoot me |
| 08:57.11 | robstar_ | does anyone have experience with setting up astribanks? |
| 08:57.26 | nsgn | thanks for the great recommendation for voip.ms... ;) |
| 08:58.28 | p3nguin_ | Well hell. |
| 08:58.32 | nsgn | yeah |
| 08:58.36 | nsgn | w t f |
| 08:58.55 | p3nguin_ | I'm having trouble actually comprehending this. |
| 08:59.05 | p3nguin_ | Where does the problem exist? |
| 08:59.30 | nsgn | it's where the call goes from the pstn into voipms |
| 08:59.35 | nsgn | or |
| 08:59.36 | p3nguin_ | In the delivery of the DID to voipms? |
| 08:59.38 | nsgn | in voipms' own setup |
| 08:59.42 | nsgn | one of the two. it has to be |
| 08:59.52 | nsgn | dallas can go screw themselves |
| 08:59.54 | p3nguin_ | I can't imagine they have a bad configuration. |
| 09:00.13 | p3nguin_ | I use us2 for my own DID. |
| 09:00.25 | nsgn | well...imagine it. this is proof |
| 09:00.28 | nsgn | it is clearly not on my end |
| 09:00.35 | nsgn | it's them or their delivery |
| 09:00.56 | p3nguin_ | Maybe they can give you another number and it could be fixed? |
| 09:01.12 | robstar_ | can i just fire away questions here? |
| 09:01.21 | p3nguin_ | ~ask |
| 09:01.22 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:01.34 | nsgn | just a wonderful experience to start with |
| 09:01.50 | robstar_ | ah, ok, got an astribank hooked up to a server trough usb |
| 09:01.57 | p3nguin_ | Yeah, this is the shit that happens to me with EVERYTHING I do. |
| 09:02.10 | robstar_ | lsusb | grep e4e4 gives me 1152, so driver is loaded |
| 09:02.31 | tzafrir__laptop | robstar_, so the firmware is loaded |
| 09:02.34 | nsgn | p3nguin_: done testing? |
| 09:02.39 | robstar_ | i can flash lights trough my console and i see the off-the-hook light when i pick up the phone |
| 09:02.39 | tzafrir__laptop | try: dahdi_hardware -v |
| 09:02.55 | p3nguin_ | nsgn: I'm removing the peer now. |
| 09:02.58 | nsgn | ok |
| 09:03.11 | tzafrir__laptop | ah, so the driver is loaded, indeed |
| 09:03.26 | nsgn | and to think i was considering just getting my vonage buddies to bring me in some lines on an ATA and go into asterisk with another ATA |
| 09:03.30 | robstar_ | the firmware is loaded, but zaptel_hardware gives me nothing... |
| 09:03.34 | nsgn | sounded silly then. would have actually WORKED |
| 09:03.39 | TheOpenSourcerer | Which GUI/Front-end for Asterisk would you recommend for "n00bs" ? |
| 09:03.50 | tzafrir__laptop | robstar_, zaptel or dahdi? |
| 09:04.05 | nsgn | TheOpenSourcerer: asterisknow |
| 09:04.07 | robstar_ | zaptel |
| 09:04.11 | tzafrir__laptop | TheOpenSourcerer, sysadmin or user? |
| 09:04.19 | tzafrir__laptop | robstar_, what version? |
| 09:04.19 | TheOpenSourcerer | mainly user |
| 09:04.26 | tzafrir__laptop | Also: what linux distro? |
| 09:04.37 | TheOpenSourcerer | Ubuntu :-) |
| 09:04.39 | TheOpenSourcerer | Server |
| 09:04.54 | robstar_ | current version, on ubuntu server, also current |
| 09:05.32 | nsgn | p3nguin_: at least the outgoing caller ID works perfectly |
| 09:05.47 | nsgn | i was worried about that |
| 09:05.52 | nsgn | too bad i cant call in |
| 09:05.58 | nsgn | seriously, at all now |
| 09:06.00 | p3nguin_ | I hope that they will offer you a solution that you accept, such as a new number that actually doesn't have this problem. |
| 09:06.04 | nsgn | no matter how many times i try |
| 09:06.04 | robstar_ | everything appears to be working, but no dialtone. and no channels, wich explains the no dialtone i guess |
| 09:07.07 | nsgn | p3nguin_: whoa...just found a weiiird issue on their website |
| 09:07.08 | robstar_ | zt_registration gives me: XBUS-00 usb:1235usb-0000:00:1d.7-4 |
| 09:07.09 | nsgn | the panel |
| 09:07.41 | robstar_ | but should give me 4 spans i think |
| 09:07.42 | nsgn | despite the pop showing as dallas in the overview, and it being what i chose when i registered...editing the DID had it going to houston |
| 09:07.59 | nsgn | waiting for the change to push and testing |
| 09:09.12 | nsgn | makes no difference... |
| 09:09.13 | p3nguin_ | So it was routed to Houston and we were registered to Dallas? |
| 09:09.19 | nsgn | yes...but it didnt help |
| 09:09.23 | nsgn | i waited the 1 minute it said it'd take |
| 09:09.29 | nsgn | also doesnt explain why some calls worked earlier |
| 09:09.36 | nsgn | the issue has progressively worsened throught our chat |
| 09:09.37 | p3nguin_ | yeah |
| 09:09.55 | p3nguin_ | It's GOT to be call routing on the PSTN. |
| 09:10.20 | p3nguin_ | It has to be. |
| 09:11.04 | *** join/#asterisk baijum (n=baiju@203.123.174.162) |
| 09:11.27 | robstar_ | genzaptelconf -sdvM gives me no errors either |
| 09:11.33 | nsgn | if anything it worked better to SIP |
| 09:11.47 | p3nguin_ | I wish we could buy/release/trade DIDs at no additional charge. |
| 09:12.18 | p3nguin_ | Start going through a handful until you find one that works right. |
| 09:12.27 | *** join/#asterisk bala (i=dce39c91@gateway/web/freenode/x-fcxtnrzqvanzutfl) |
| 09:14.24 | robstar_ | ok, genzaptelconf -sdvM gives me 0 channels. version = 1.4.12.1 btw |
| 09:14.32 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 09:15.21 | *** join/#asterisk icyValk77 (n=icyValk7@host81-131-170-159.range81-131.btcentralplus.com) |
| 09:15.25 | nsgn | switched back to sip |
| 09:15.43 | nsgn | i get alternations between "call cannot be completed" and "all circuits are busy now" |
| 09:15.57 | bala | hi, i have an alsa channel connected to a sip channel. how can i make the audio coming in from both legs to play out through alsa channel? |
| 09:16.11 | *** join/#asterisk af_ (n=getsmart@88.149.240.72) |
| 09:17.40 | p3nguin_ | nsgn: Will you be able to describe this problem here later when more people are active? I'm interested to know their thoughts. |
| 09:18.00 | p3nguin_ | I'm going to have to head off in a minute. |
| 09:18.46 | *** join/#asterisk Fister (n=poon@users.757.org) |
| 09:19.31 | nsgn | i'm working wayyy later than i should and when i leave in an hour or so i wont be here for at least 12 hours |
| 09:19.34 | nsgn | but i'll be back sometime |
| 09:19.40 | nsgn | thanks for all your help |
| 09:20.03 | p3nguin_ | I may try to share the experience with the guys later. |
| 09:20.24 | Fister | Does anyone have sip calls working with the slin16 (or ulaw/alaw @ 16kHz) codecs? It looks like 1.6.2.0 may be ready to support this. |
| 09:20.50 | p3nguin_ | I'm also interested in what support replies. |
| 09:21.55 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:22.51 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:26.32 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 09:28.21 | p3nguin_ | nsgn: I gotta run. Keep me posted with what support says and if you can wait around for a solution or not. |
| 09:29.29 | *** join/#asterisk ishchatul (i=ishchatu@bzq-80-22-127.red.bezeqint.net) |
| 09:29.54 | robstar_ | anybody on astribanks/zaptel issues? |
| 09:34.40 | nsgn | p3nguin_: thanks again. night |
| 09:38.15 | ruyo | robstar_, what kind of issues? |
| 09:39.07 | robstar_ | ruyo, no dialtone on my phones. firmware is loaded, but zaptel_hardware gives me nothing. no channels |
| 09:40.05 | ruyo | Do you have power feeding the card? |
| 09:40.32 | robstar_ | yes, all lights work, as well as the off-hook light |
| 09:40.56 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 09:40.58 | ruyo | Are the right modules being loaded? |
| 09:41.58 | robstar_ | i don't know, how can i find that out? |
| 09:42.16 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 09:43.27 | ruyo | The card's manufacturer manual should tell you the right ones. |
| 09:43.52 | *** join/#asterisk icyValk77 (n=icyValk7@host86-162-17-81.range86-162.btcentralplus.com) |
| 09:44.06 | *** join/#asterisk wam_ (i=wam@unaffiliated/wam) |
| 09:44.34 | ruyo | For Openvox's (and others) 4 port cards, wctdm is the module to be loaded. |
| 09:45.12 | robstar_ | i followed xorcom's advice and i know xpp_usb should be loaded, which is. could it be i mis the FXS module? |
| 09:46.09 | tzafrir__laptop | robstar_, if you see the LED lit when you pick up the phone, the module is loaded |
| 09:46.31 | tzafrir__laptop | what version of zaptel? |
| 09:46.50 | robstar_ | ok. so the led is lit when offhook. zaptel version = 1.4.12.1 |
| 09:47.03 | ruyo | Good to know that one. :> |
| 09:50.55 | robstar_ | i had to patch it before i could compile it. could that have anything to do with it? |
| 09:50.55 | ruyo | robstar_, did you generate the conf file with genzaptelconf -sdvM? |
| 09:51.03 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4d5b:fc0b:cb4f:159f) |
| 09:51.12 | robstar_ | yes i did, it tells me it has 0 channels to configure |
| 09:51.36 | ruyo | Though the dial tone shouldn't have anything to do with the conf, I think. |
| 09:52.43 | robstar_ | firtst i thought my bank was broken, but that is not the case, cause a xorcom live-cd does give me a dialtone |
| 09:53.09 | ruyo | I'm out of ideas. The worst problem I had with Zaptel was solved with genzaptelconf. |
| 09:53.38 | ruyo | Could it be the phone you're using for testing? |
| 09:53.59 | *** join/#asterisk lftsy (n=lftsy@88.191.80.8) |
| 09:55.06 | lftsy | Hello, when I remove a peer from my sip.conf and just do a reload, the peer is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk 1.4.21 and Asterisk 1.4.27-rc2. Is it normal? Thanks |
| 09:55.29 | robstar_ | nope, cause the phone works on another bank. i see no lights at all anymore btw... just the on light, but grep does give me a 1152?? |
| 09:55.47 | robstar_ | this really confuses me... |
| 09:56.14 | AhmedElGamil | I am using VMAuthenticate to allow users to check their voice mail |
| 09:56.57 | AhmedElGamil | i need to catch the voice mail number and password in variables to be used within the dial plan .. |
| 09:57.03 | AhmedElGamil | How can i achive that ? |
| 09:57.41 | ruyo | robstar_, are they all FXS modules or can you test it with a FXO as well? |
| 09:58.37 | robstar_ | seriously, i'm not sure... (yes, i'm an idiot) |
| 09:59.47 | tzafrir__laptop | robstar_, what's the output of lszaptel ? |
| 10:01.10 | robstar_ | ### Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: HRtimer) 1" (MASTER) |
| 10:07.11 | *** join/#asterisk henkoegema (n=henkoege@78-21-68-204.access.telenet.be) |
| 10:10.13 | *** part/#asterisk icyValk77 (n=icyValk7@host86-162-17-81.range86-162.btcentralplus.com) |
| 10:13.49 | robstar_ | ok, broke it... tried to reload firmware, but got a 1151 now... |
| 10:23.48 | robstar_ | ok, rebuild zaptel, rebooted, reset the firmware. now i have 86 channels on my astribank. but still no dialtone... |
| 10:24.46 | tzafrir__laptop | what version of zaptel? 1.4.12.1 ? |
| 10:25.04 | robstar_ | yes, 1.4.12.1 |
| 10:26.04 | ruyo | So it detects the channels now? |
| 10:27.20 | robstar_ | well, lszaptel shows me all my spans (5) with 60 E1_TE's and 24 FXS's. but zap show channels gives me zit |
| 10:30.46 | tzafrir__laptop | robstar_, I would generally recommend either DAHDI or "latest SVN" of zaptel 1.4 (which is because Zaptel wasn't really released for quite some time) |
| 10:32.35 | robstar_ | ok, i'll give that a shot. still, i had it working with zaptel 1.2.x. but this new server... man! |
| 10:33.28 | tzafrir__laptop | E.g. http://updates.xorcom.com/astribank/src/zaptel-1.4.12.9.svn.r4653.tar.gz , which is a snapshot of almost latest SVN |
| 10:33.58 | tzafrir__laptop | An SVN checkout should work just as well (or slightly better) |
| 10:35.17 | Fister | If I write a patch, how do I pass it on? Do I create a bug and attach the patch? |
| 10:36.29 | tzafrir__laptop | yes, please |
| 10:36.54 | *** join/#asterisk albertoandrade (n=albertoa@189.4.48.141) |
| 10:42.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:48.23 | Fister | :) :) :) got an slin16 call from my aastra phone :) :) :) |
| 10:55.50 | robstar_ | ok, intalled the svn, but now it get "Can't call method "is_twinstar" on unblessed" at zaptel_hardware -v |
| 10:57.07 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
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| 11:01.48 | DigitalFlux | Can we set a family/key pair using the DB cmd to 1 directly |
| 11:02.21 | DigitalFlux | without assigning 1 to another variable and setting it to the first variable ? |
| 11:20.07 | *** join/#asterisk shinao1 (n=shinao1@41.219.240.243) |
| 11:25.02 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
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| 11:26.12 | *** join/#asterisk mhekman (n=mhekman@94.142.209.198) |
| 11:26.45 | mhekman | Hey all |
| 11:28.26 | kaldemar | DigitalFlux: Set(DB(family/key)=1) <-- is that what you mean? |
| 11:29.06 | robstar_ | tzafrir, zaptel_hardware -v now shows me 5 spans. but still no channels in asterisk -r zap show channels |
| 11:30.12 | tzafrir__laptop | robstar_, are they registered? |
| 11:33.39 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 11:39.20 | robstar_ | tzafrir, how do i check that? |
| 11:39.57 | tzafrir | zaptel_hardware -v |
| 11:40.09 | tzafrir | "unregistered" or "span N"? |
| 11:41.04 | robstar_ | line one: XBUS-00/XPD-00: E1 (31) Span 1 ZAPTEL-SYNC |
| 11:41.12 | robstar_ | line two: XBUS-00/XPD-00: E1 (31) Span 1 ZAPTEL-SYNC |
| 11:41.27 | robstar_ | line three: XBUS-00/XPD-10: FXS (8) Span 3 |
| 11:41.45 | robstar_ | line 4 and 5 are teh same as line three, but with span 4 and 5 |
| 11:45.11 | robstar_ | zap show channels gives me a red on the first E1 and UNCONFIGUR on the other spans |
| 11:46.42 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 11:48.00 | DigitalFlux | kaldemar: Yes, tried that and it works |
| 11:48.05 | jkroon | Hi guys, I was hoping someone here has some advice. I've got some SNOM phones where they seem to lose their secrets, resulting in asterisk reporting chan_sip.c:19958 handle_request_register: Registration from '"Charmaine" <sip:7874@10.10.1.253>' failed for '10.10.1.115' - Wrong password |
| 11:48.16 | jkroon | this is with asterisk 1.6.1.6 |
| 11:48.18 | DigitalFlux | kaldemar: what i need to do know is retrieve the value in a GoToIf Statement |
| 11:48.30 | DigitalFlux | kaldemar: directly from the database |
| 11:48.55 | DigitalFlux | kaldemar: and compare it with 1 ..then go to some extension if success of fail |
| 11:48.56 | kaldemar | DigitalFlux: ${DB(family/key)} will get you the value |
| 11:49.24 | DigitalFlux | kaldemar: Tried it but didn't work .. will get you the full statement |
| 11:49.45 | kaldemar | DigitalFlux: show it. |
| 11:50.56 | DigitalFlux | kaldemar: exten => 30,n(CheckLogin),GotoIf($["$DB(${AUTH_MAILBOX}/LoggedIn" = "1"]?AgentLoggedIn:ContinueAddingAgent |
| 11:51.26 | kaldemar | DigitalFlux: you have a syntax error. do it like i showed you. |
| 11:51.48 | DigitalFlux | kaldemar: Sorry, do what like u told me ? |
| 11:51.49 | kaldemar | $DB(${AUTH_MAILBOX}/LoggedIn <-- that is complete garbage. |
| 11:52.01 | kaldemar | DigitalFlux: like this: ${DB(family/key)} |
| 11:52.43 | kaldemar | ${DB(${AUTH_MAILBOX}/LoggedIn)} <-- it should look like that |
| 11:52.54 | DigitalFlux | kaldemar: I see .. |
| 11:53.04 | DigitalFlux | kaldemar: Will try that now |
| 11:55.16 | mhekman | Did someone ever had the problem that after running a asterisk server for a long time, without changing any settings, the SIP command is unknown? No such command it states |
| 11:55.32 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 11:55.49 | nsgn | mhekman: i had that issue |
| 11:55.52 | nsgn | but it was right after installing |
| 11:56.02 | nsgn | i shut down entirely and booted back up and all was well. never knew the cause |
| 11:56.13 | nsgn | and havent had it since |
| 11:56.17 | kaldemar | that happens when chan_sip.so is not loaded |
| 11:56.48 | mhekman | now we have to find out why it's not loaded, we put a backup back when it still worked, without any progress |
| 11:57.51 | mhekman | Thanks for the information, i'll report back when we have some improvement what soever |
| 11:59.04 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 12:07.55 | mhekman | Strangest is, that when we check for the loaded modules, the chan_sip.so is loaded, but it just randomly chooses to be found or not, even after a full restart. Sometimes it doesn't even give any response to for example: sip show peers. How frustrating is that |
| 12:09.50 | *** join/#asterisk w-m- (i=w@freenode/staff/wikipedia.wimt) |
| 12:09.50 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) [NETSPLIT VICTIM] |
| 12:09.50 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) [NETSPLIT VICTIM] |
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| 12:11.02 | mhekman | even with sip show peers, it doesn't show anything |
| 12:11.29 | mhekman | like it's not responding (ignoring) it |
| 12:12.45 | nsgn | mhekman: did this problem just develop out of the blue on a stable system? |
| 12:12.55 | DND | guys i just need to be pointed to the right direction/file but i wanted to add a tone zone/signalling for my country and indications.conf keeps overwriting it |
| 12:13.23 | mhekman | nsgn: yes that is the frustrating part |
| 12:14.00 | nsgn | is a fresh and up to date install out of the question if you backed up settings? |
| 12:14.11 | *** join/#asterisk cjk (n=cjk@ip-88-207-233-219.dyn.luxdsl.pt.lu) |
| 12:14.36 | mhekman | we're trying to safe this one first, the last option will be a fresh install yes, but we want to know why this is happening |
| 12:14.50 | nsgn | you in vmware or standalone? |
| 12:15.11 | mhekman | no, hyper v |
| 12:15.31 | nsgn | i've heard a few references to this issue in a virtual setup |
| 12:15.41 | nsgn | not sure why, but i'm in one and saw it right off the bat, though only once |
| 12:15.51 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.05 | mhekman | our sip registration is handled by a external party, we are thinking that there must be wrong somewhere there, but when we use a softphone and do it all locally, it still does nothing |
| 12:16.18 | nsgn | yeah, i dont think its an external issue |
| 12:16.18 | mhekman | hmm we're trying our best to find out what it is :) |
| 12:16.25 | nsgn | i had the issue before i configured any trunks |
| 12:16.29 | nsgn | fresh out of the box, second boot |
| 12:17.57 | *** join/#asterisk radovan (i=radovan@uber.1337.sk) |
| 12:18.04 | mhekman | yeah thats really fresh :) |
| 12:18.15 | radovan | hi |
| 12:18.35 | radovan | asterisk[4432]: rc_avpair_new: unknown attribute 1490026597 <- anybody knows what is this? |
| 12:22.02 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 12:25.43 | robstar_ | ok, almost there, just this error: Framing(T1)/Signalling(E1) must be one of 'd4', 'esf', 'cas' or 'ccs', not '' |
| 12:25.48 | *** join/#asterisk syuroff (n=syuroff@h66-222-51-78.mdsnwi.broadband.dynamic.tds.net) |
| 12:27.46 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 12:28.28 | [TK]D-Fender | robstar_: Clearly you have not put a parameter in place properly in your SPAN configs |
| 12:29.59 | jkroon | Hi guys, I was hoping someone here has some advice. I've got some SNOM phones where they seem to lose their secrets, resulting in asterisk reporting chan_sip.c:19958 handle_request_register: Registration from '"Charmaine" <sip:7874@10.10.1.253>' failed for '10.10.1.115' - Wrong password |
| 12:29.59 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 12:30.23 | robstar_ | when i do put the parameter in there, it still shows me the same error |
| 12:36.01 | [TK]D-Fender | robstar_: And you've still done something wrong. |
| 12:36.13 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 12:40.16 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 12:42.00 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 12:42.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:44.56 | radovan | asterisk[4432]: rc_avpair_new: unknown attribute 1490026597 <- anybody knows what is this? |
| 12:46.22 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 12:46.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:46.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:46.25 | tzafrir | robstar_, just generate the config with zapconf ... |
| 12:47.33 | leifmadsen | Anyone have issues and a solution for when Asterisk is behind NAT, and a client is behind symmetric NAT? |
| 12:47.41 | [TK]D-Fender | tzafrir: c'mon... I'm sitting patiently awaiting the formation of a useful thought from his side! |
| 12:48.08 | leifmadsen | I don't have this issue right now; I'm just researching to see if anyone else has this issue, as I have someone who tells me this could potentially be a problem for Asterisk |
| 12:48.15 | beek | mornin' everyone |
| 12:49.01 | tzafrir | [TK]D-Fender, the "pri" config is overly complex because with T1 you have so many options |
| 12:49.39 | tzafrir | With E1 things are very simple, and zaptel.conf / system.conf is normally too complicated |
| 12:49.54 | tzafrir | which is why I prefer to just generate config |
| 12:49.59 | [TK]D-Fender | tzafrir: Not really... and this comes down to a simply matter of counting. And then there is the expectation that we're psychic. |
| 12:50.46 | tzafrir | nah. Why would you need to guess if you have d4 or esf? |
| 12:52.07 | [TK]D-Fender | tzafrir: Shouldn't be a guess... I fail to understand how people don't have the specs of their telco lines of what is set in some other hardware they also control |
| 12:53.00 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:53.20 | tzafrir | Not a guess? Wizdom? |
| 12:53.45 | robstar_ | D_Fender, you should tell that to my telco who gives me jack info on the lines... |
| 12:53.56 | [TK]D-Fender | tzafrir: No, if he's dealing with a telco line, he should have a spec sheet from them stating the signalling on it. |
| 12:53.58 | tzafrir | Why is it, then, that I had to write a script to generate config that works well for most cases? |
| 12:54.10 | [TK]D-Fender | robstar_: Did you call them and ask? |
| 12:54.23 | tzafrir | (and Sangoma has similar scriptology with different functionality) |
| 12:55.04 | tzafrir | No need to guess. E1/ISDN in europe are almost all the same |
| 12:55.04 | [TK]D-Fender | tzafrir: Never had to, but those who can't read the clear documentation on that span line get annoying after a while, and we thank you for silencing at least a few of them :) |
| 12:55.33 | robstar_ | D-Fender, yep, then you get some student who knows nothing and shoves you off with unstatisfiyng answers... service with the telcos here is not very good |
| 12:56.48 | *** join/#asterisk ska (n=ska@cpe-70-124-88-58.austin.res.rr.com) |
| 12:57.01 | telnettech | dont blame the telco.......a telecom tech should always have his system documented and passed on to others that come after. |
| 12:57.12 | radovan | anybody knows what "rc_avpair_new: unknown attribute 1490026597" is? i have no zap chan, just iax and sip |
| 12:57.27 | [TK]D-Fender | robstar_: So if you ask them "Please tell me the framing and signalling for the line I paying you for" a tech won't be able to answer you? |
| 12:57.39 | telnettech | what we have here is a merging of voice and data and the data side of the house is the people in charge and dont know anything about telecom standards |
| 12:57.40 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 12:58.03 | robstar_ | telnettech, i don't blame anyone but myself for not getting this stuff to work |
| 12:58.03 | Nugget | telnet is eeeeeeevil! |
| 12:58.10 | jaytee | I swear that's a script or a bot |
| 12:58.20 | telnettech | everytime |
| 12:58.21 | [TK]D-Fender | robstar_: Well you also need to look at your approach to getting help on it... |
| 12:58.30 | [TK]D-Fender | jaytee: ... it IS a script, duh :p |
| 12:58.54 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:58.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:58.56 | robstar_ | D-Fender, it is hard to get a tech on the phone... but i can ofcourse give it another try |
| 12:59.06 | tzafrir | ~telnet |
| 12:59.07 | infobot | [telnet] An old protocol for remotely logging in to another computer. It is one of the basic utilities for TCP/IP. And also an evil trigger for Nugget |
| 12:59.12 | [TK]D-Fender | [08:25]<robstar_>ok, almost there, just this error: Framing(T1)/Signalling(E1) must be one of 'd4', 'esf', 'cas' or 'ccs', not â <- You didn't put the value in the right place in your own config. Not the telco's fault |
| 12:59.48 | anonymouz666 | which one ejabberd or jabberd to use with asterisk? |
| 12:59.52 | [TK]D-Fender | [08:30]<robstar_>when i do put the parameter in there, it still shows me the same error <-- clearly you still aren't putting the parameters in the right place in the config. Still not the telco's fault |
| 13:00.12 | [TK]D-Fender | robstar_: And you know what you failed to do to help debug this issue? |
| 13:00.23 | robstar_ | D-Fender, I do not blame the telco, actually I don't blame anyone but myself as I said before |
| 13:00.35 | robstar_ | D-Fender, tell me what I fialed to debug |
| 13:00.37 | tzafrir | robstar_, so how about just using zapconf ? |
| 13:00.47 | [TK]D-Fender | tzafrir: no no no .... we're almost there! |
| 13:01.15 | robstar_ | D-Fender, now I'm really getting serious... did zapconf |
| 13:01.38 | [TK]D-Fender | robstar_: No... you just put your faith in some other script because someone told you it'd be "easy". |
| 13:01.49 | robstar_ | D-Fender, aha! css, hdb3, crc4 |
| 13:01.57 | [TK]D-Fender | robstar_: This isn't the essential bit you could have gleaned from all of this. |
| 13:02.39 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 13:02.46 | robstar_ | D-Fender, I know it is NOT easy. And I'm willing to learn and listen to you guys here. Just trying to get my box up, that's all |
| 13:03.27 | [TK]D-Fender | robstar_: Ok.. you don't seem to be evolving much on this so I'm just going to say it : next time show us the damned span line so we can TELL YOU what's wrong with it |
| 13:03.38 | robstar_ | D-Fender, i bought books, studiyng hard to get to know asterisk. I do the best i can, ok... |
| 13:03.42 | [TK]D-Fender | robstar_: You clearly couldn't count off which parameter is which and didn't think to SHOW US |
| 13:03.58 | [TK]D-Fender | robstar_: You just kept say "it doesn't work". |
| 13:03.59 | tzafrir | robstar_, what's now the output of lszaptel ? |
| 13:04.31 | [TK]D-Fender | robstar_: So were you jsut here for a pep talk, or were you here to actually figure it out? |
| 13:04.36 | robstar_ | tzafir,Span 2: XBUS-00/XPD-01 "Xorcom XPD #00/01: E1" HDB3/CCS/CRC4 ClockSource |
| 13:04.48 | robstar_ | tzafir, and all fxs channels |
| 13:04.49 | [TK]D-Fender | tzafrir: AH, he's already one of yours :p |
| 13:05.06 | radovan | anonymouz666: that doens't really matter what jabber daemon you use. i preffer ejabberd |
| 13:05.11 | angryuser | Xorcom clan yeah! |
| 13:05.15 | tzafrir | robstar_, could you please PB the complete output? |
| 13:05.15 | robstar_ | D-Fender, man, I just try to figure things out ok |
| 13:05.18 | tzafrir | ~pb |
| 13:05.19 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:05.20 | anonymouz666 | radovan: what client? |
| 13:05.44 | [TK]D-Fender | robstar_: guess you weren't really looking for help and that this is the next best thing to blogging about your errors? |
| 13:06.00 | radovan | anonymouz666: adium, why? |
| 13:06.13 | [TK]D-Fender | robstar_: For the next time... SHOW US what you're having a problem with if you expect suggestions on how to correct it |
| 13:06.44 | anonymouz666 | radovan: I am installing for the first time... so I am open to the suggestion about good clients also |
| 13:06.44 | robstar_ | D-Fender, will do so.... sjees, sorry about this man |
| 13:06.52 | [TK]D-Fender | robstar_: :P |
| 13:07.11 | [TK]D-Fender | robstar_: Just trying to inspire or impart some common sense into you is all... |
| 13:08.10 | robstar_ | tzafrir, http://pastebin.com/d40baa87c |
| 13:08.56 | robstar_ | D-Fender, cool, tnx for your great efforts to educate me. |
| 13:09.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.05 | tzafrir | you have just one port connected to the provider, and that is port 2? |
| 13:09.17 | [TK]D-Fender | robstar_: Bad = "My VCR doesn't work, what could be wrong with it?" (answer = ANYTHING). Good = "Here, look at my VCR, see this big flashing error light down there, and hear that grinding sound? What could be causing that?". Show us the broken bits, the symptoms it has and then maybe we can help |
| 13:09.31 | robstar_ | tzafir, yes, this is a test box, so just have one isdn modem here |
| 13:09.31 | radovan | anonymouz666: that depends... on windows I was used to miranda, osx adium, on linux pidgin, jabbim or psi |
| 13:09.37 | [TK]D-Fender | robstar_: This approach will pay of big and fast in places like this |
| 13:09.42 | tzafrir | Next thing to do would be to configure Asterisk |
| 13:10.19 | tzafrir | zapconf created a configuration snippet in /etc/asterisk/zapata-channels.conf |
| 13:10.44 | robstar_ | D-Fender, i understand that, i'm not an idiot... i've been hanging out here for some time and have posted a bit more info then the info you are referring to, but will try to be a good boi and post more next time, k? |
| 13:10.53 | tzafrir | either copy it to the end of /etc/asterisk/zapata.conf or add to the end of that file: #include zapata-channels.conf |
| 13:10.57 | radovan | anonymouz666: http://xmpp.org/software/clients.shtml |
| 13:10.58 | tzafrir | and restart asterisk |
| 13:11.09 | [TK]D-Fender | robstar_: Excellent :) |
| 13:11.11 | tzafrir | radovan, is jabberd actively developed? |
| 13:11.43 | radovan | tzafrir: of course |
| 13:11.52 | tzafrir | 1? 2? |
| 13:12.03 | radovan | tzafrir: all the google wave and realtime web is based on jabber/xmpp |
| 13:12.34 | tzafrir | I'm currently using 1 for my pet server as ejabberd is too bloated and jabberd2 is not yet in Debian Stable |
| 13:12.34 | radovan | oh, jabberd, i don't know, i'm an ejabberd fan :) |
| 13:12.43 | robstar_ | tzafir, the include was already in there |
| 13:12.51 | tzafrir | But I didn't test any Jingle |
| 13:13.09 | tzafrir | what version of asterisk is it? |
| 13:13.18 | tzafrir | Do you use freepbx or the like? |
| 13:14.10 | robstar_ | tzafir, 1.4-current |
| 13:16.00 | tzafrir | try, from the asterisk CLI: dahdi restart |
| 13:16.10 | tzafrir | do you get any error? |
| 13:16.52 | *** join/#asterisk gandhijee (i=akp@66.202.34.165) |
| 13:17.43 | robstar_ | tzafir, yes, this: http://pastebin.com/m7516be29 |
| 13:18.38 | radovan | robstar_: touch zapata-channels.conf |
| 13:19.56 | radovan | anybody knows what "rc_avpair_new: unknown attribute 1490026597" is? i have no zap chan, just iax and sip, and scrambled voice since this begun to appear in log |
| 13:19.59 | robstar_ | radovan, did that, same error |
| 13:21.38 | radovan | robstar_: but this isn't critical error, and one line above there's "just" warning. it should work |
| 13:21.57 | radovan | robstar_: perhapse you touched in on wrong place |
| 13:22.45 | robstar_ | radovan touched /etc/asterisk/zapata-channels.conf. the file is there |
| 13:24.16 | tzafrir | robstar_, hmm... it means that this file appears before any 'category', that is, it: you miss '[channels]' on the top of zapata.conf |
| 13:25.58 | robstar_ | tzafrir, yes! dialtone! thank you so much! |
| 13:26.43 | robstar_ | tzafrir, sjees... what a stupid problem... :/ thank you very much for your patience and help |
| 13:27.39 | *** join/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 13:27.55 | ManxPower-work | ~answers |
| 13:27.56 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:28.49 | tzafrir | hmmm.... process_text_line: parse error: No category context for line 11 of /etc/asterisk/zapata-channels.conf |
| 13:29.30 | tzafrir | I wonder how to make that message more obvious. "No category context ([])"? |
| 13:30.08 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 13:30.29 | [TK]D-Fender | tzafrir: "Hey buddy, put the [channels] header BEFore setting parameters, kthxbibi" |
| 13:30.41 | robstar_ | tzafrir, well, now you said it, it is actually pretty obvious... |
| 13:30.59 | tzafrir | But the error is from the general config parsing code |
| 13:31.41 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 13:33.09 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 13:33.40 | robstar_ | D-Fender, if you ever have problems with girls, i'm more then willing to help you out. |
| 13:35.51 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4d5b:fc0b:cb4f:159f) |
| 13:40.07 | *** join/#asterisk slinksh0t (n=slinksh0@adsl-9-177-182.mia.bellsouth.net) |
| 13:45.44 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 13:45.51 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 13:46.32 | *** join/#asterisk Katty (n=Angela@adsl-70-253-172-140.dsl.stlsmo.swbell.net) |
| 13:49.53 | jaytee | morning Katty |
| 13:50.44 | Katty | hi jaytee |
| 13:55.45 | leifmadsen | Katty: yo! |
| 13:55.56 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 13:57.55 | Katty | hi leifmadsen |
| 13:59.07 | *** join/#asterisk syuroff (n=syuroff@216.47.253.194) |
| 14:01.04 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 14:02.55 | Katty | sister has aquired the oinkflus. |
| 14:03.50 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:03.59 | Katty | hi ariel_ |
| 14:04.06 | raden_work | morning Katty |
| 14:04.16 | ariel_ | Morning everyone |
| 14:04.22 | Katty | hi raden_work |
| 14:04.36 | ariel_ | Morning Katty .....hugs.... |
| 14:04.37 | eppigy | hello |
| 14:04.43 | eppigy | raden_work: how is the networkign thing coming? |
| 14:04.49 | Katty | hi eppigy |
| 14:04.53 | eppigy | hi :> |
| 14:04.59 | eppigy | takes Katty in his arms |
| 14:04.59 | jaytee | dave |
| 14:05.03 | eppigy | HELLO jaytee |
| 14:05.06 | eppigy | caploc |
| 14:07.37 | raden_work | eppigy, working on it, were you the one helping me with my ASA the one night ? |
| 14:07.42 | eppigy | yes indeed |
| 14:08.56 | raden_work | Gah i been looking for you dude LOL |
| 14:09.06 | eppigy | ol |
| 14:12.46 | *** join/#asterisk wideruled (n=UserNick@216-110-94-245.static.twtelecom.net) |
| 14:13.04 | *** join/#asterisk chendy (n=chatzill@116.25.81.155) |
| 14:13.16 | wideruled | hey guys, if my IVR isn't receiving any input all of a sudden from people on cell phones inputting extensions.. is it likely my drunk or a carrier issue? |
| 14:13.26 | wideruled | trunk, wish it were drunk |
| 14:14.52 | Katty | i have a sick ferret on my hands today :< |
| 14:18.56 | eppigy | :< |
| 14:19.04 | eppigy | poor weasel |
| 14:21.28 | Katty | yes. his fur is falling out |
| 14:21.58 | eppigy | D: |
| 14:22.51 | wideruled | that might make him smell better than your average ferret |
| 14:23.12 | beek | Mornin' Katty... are you going to take him to the vet? |
| 14:23.24 | *** join/#asterisk Tim_Toady (n=moi@adsl54-170.kln.forthnet.gr) |
| 14:23.49 | Katty | he's already been. |
| 14:24.14 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:24.39 | beek | Any meds? What does the vet say? |
| 14:25.20 | Fister | wideruled: you mean DTMF? |
| 14:25.31 | Katty | well....it could be one of two things. |
| 14:25.46 | wideruled | probably. |
| 14:25.47 | Katty | firstly, it could be Rat Tail where they shed their fur and regrow it back |
| 14:25.54 | Katty | or, secondly, it could be Adrenal Disease |
| 14:26.54 | Fister | wideruled: there are three ways to do DTMF in SIP. You can set them per peer in sip.conf. |
| 14:26.59 | beek | I've never had a ferret before so I have no idea what the care and feeding of such critters are like. |
| 14:27.39 | Katty | well, if it's adrenal disease, he will have to be on lupron injections the rest of his life. |
| 14:27.41 | wideruled | Fister, i know. like i said this happened last night, nothings changed, so i'm guessing its the itsp |
| 14:28.07 | Katty | that or risky surgery |
| 14:28.56 | Fister | who is interested in wideband codecs here |
| 14:29.03 | eppigy | yeah ferrets are prone to many illnesses |
| 14:29.12 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 14:29.18 | beek | They're on the short list of animals I've NOT had has pets. |
| 14:29.29 | beek | s/has/as/ |
| 14:29.54 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:29.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:30.13 | *** join/#asterisk af_ (n=getsmart@88.149.240.182) |
| 14:32.45 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:36.42 | beek | Katty: http://www.ferretnews.com/issues/ferret-news-95.html about 2/3 the way down. |
| 14:39.16 | *** join/#asterisk trebaum (n=trebaum@207-67-92-30.static.twtelecom.net) |
| 14:42.07 | Katty | (= |
| 14:43.56 | Kobaz | mmm |
| 14:43.58 | Kobaz | nice and warm |
| 14:54.59 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.179.22) |
| 14:55.02 | Katty | grins |
| 14:55.02 | Katty | pippin has claimed a plastic cup as Valued Toy |
| 19:24.46 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 19:24.46 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.8 (2009/10/26), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 19:26.24 | *** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 19:31.15 | [TK]D-Fender | ~botsnack |
| 19:31.15 | infobot | :), [TK]D-Fender |
| 19:31.20 | [TK]D-Fender | infobot: Good boy! |
| 19:31.21 | infobot | thanks, [TK]D-Fender |
| 19:32.13 | beek | [TK]D-Fender: You have WAY TOO MUCH fun with that thing. |
| 19:32.25 | [TK]D-Fender | beek: MY BITCH!@ |
| 19:32.31 | [TK]D-Fender | ~infobot |
| 19:32.32 | infobot | ACTION *uNF* *uNF* *uNF* *uNF* *uNF* *uNF* *uNF* |
| 19:32.39 | wcselby | haha |
| 19:32.42 | [TK]D-Fender | OMG! Who dared! |
| 19:32.44 | *** join/#asterisk fainsys (n=fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 19:33.17 | fainsys | I have a new install using a Digium TE121 T1 card (with EC onboard). The customer notices that there is an occasional echo on the their side of the conversation. Sometimes it is more obvious than others. |
| 19:34.37 | [TK]D-Fender | ~infobot |
| 19:34.37 | infobot | i heard infobot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 19:34.59 | [TK]D-Fender | NOBODY FUCKS WITH MY BITCH, K!? |
| 19:35.05 | [TK]D-Fender | pulls a shiv |
| 19:35.16 | loather-work | BITCH I WILL CUT YOU |
| 19:37.04 | wcselby | fainsys - is there a question coming or are you just sharing? |
| 19:37.47 | eppigy | lets all share |
| 19:37.55 | ManxPower-work | fainsys: if you get echo on a card with hardware echo canceling, then you should contact Digium support. |
| 19:38.34 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 19:39.27 | *** part/#asterisk bluequijote (n=bluequij@189.136.250.131) |
| 19:39.53 | fainsys | sorry I forgot the question part :o ok, i'll contact digium |
| 19:40.35 | aiksa[LV] | any idea why ooh323 ignores calling pres ettings? |
| 19:41.09 | aiksa[LV] | as faras i understand it should work nicely with that |
| 19:41.23 | aiksa[LV] | or am I getting something wrong here |
| 19:41.40 | path | calling pres ettings? |
| 19:41.48 | ManxPower-work | aiksa[LV]: very few people use H323 with Asterisk. You might try searching the mailinglist archives |
| 19:41.49 | ManxPower-work | ~mailinglist |
| 19:41.51 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 19:42.01 | path | why use h323 anyway? |
| 19:43.26 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 19:44.18 | doolittlework | hi there anyone worked with tormenta2 quad pri before? |
| 19:47.11 | ManxPower-work | doolittlework: those have not been made in at least 8 years |
| 19:47.31 | aiksa[LV] | path: because provider does not offer callingpres service for SIP |
| 19:47.32 | aiksa[LV] | :P |
| 19:47.48 | ariel_ | has anyone heard of anyone install asterisk on edian firewall OS ??? |
| 19:48.00 | ariel_ | edian/endian |
| 19:48.12 | aiksa[LV] | ManxPower-work: google back an forth. for now the best idea seems injecting appropriate bits in source and recompile |
| 19:48.25 | aiksa[LV] | kinda doesnt seem reliable for me |
| 19:48.40 | ManxPower-work | http://www.thestandard.com/news/2009/10/27/internet-phone-systems-become-fraudsters-tool |
| 19:54.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:55.58 | *** join/#asterisk netpro25_ (n=mmanning@64.238.176.248) |
| 19:56.37 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:56.55 | netpro25_ | [TK]D-Fender: How's it going. Hey I fixed that NAT problem, turned out to be a problem with the multiple virtual network interfaces that the server had. Problem cleared up when I disabled the virtual interfaces. |
| 19:59.30 | aiksa[LV] | netpro25_: or you could specify the exact IP for the service instead of 0.0.0.0 |
| 19:59.53 | netpro25_ | aiksa[LV]: did that too |
| 20:00.07 | netpro25_ | aiksa[LV]: specified the bind address |
| 20:03.47 | doolittlework | ManxPower-work: can one still get this card to work with dahdi? |
| 20:05.48 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 20:09.19 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 20:15.32 | ecrane | Heard of anyone using Asterisk to initiate a modem call over voip? Like.. I don't even have a real modem wired into the asterisk box. I want to use software to have asterisk 'dial' over VoIP into a real modem someplace else and set up a PPP session, TCP/IP, etc.... |
| 20:16.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.54 | aiksa[LV] | ecrane: sounds like disaster |
| 20:18.41 | aiksa[LV] | with all of the jitter and losses introduced over VOIP line, you would not get reliable service |
| 20:18.43 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:19.08 | aiksa[LV] | the very same reason why faxes over standart voip methodologies often fail |
| 20:19.35 | aiksa[LV] | my $00.2 |
| 20:22.43 | ecrane | aiksa[LV]: yeah does sound like a disaster. But even at slow speeds, it would be useful. I guess what I need to try it is a software modem in the literal sense. |
| 20:25.48 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 20:26.21 | aiksa[LV] | ecrane: take a look at spandsp, i guess it had some additional modem standarts appart from fax protocols. but I might be off by a mile |
| 20:27.45 | *** join/#asterisk btsteve (n=tstevens@12.44.147.4) |
| 20:29.10 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:30.08 | [TK]D-Fender | checkout time, BBL |
| 20:30.11 | ecrane | Thanks, that's someplace to start. I appreciate it. |
| 20:30.53 | doolittlework | does anyone know if he tormenta2 qual pri works with dahdi? |
| 20:32.08 | *** join/#asterisk gsiener (n=gsiener@d-63-245-118-26.batelnet.bs) |
| 20:33.46 | *** join/#asterisk abradley (n=Data@72.12.1.38) |
| 20:34.07 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:38.14 | torrancew | how difficult is it to get faxing up under asterisk? |
| 20:38.47 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 20:39.49 | Anth8708 | Hi guys. where is the best place for me to find information on setting up BLF for Asterisk 1.6.1.1? |
| 20:42.28 | voipmonk | your dev box, Anth8708 :) |
| 20:44.03 | Anth8708 | voipmonk: in the source then? |
| 20:44.50 | voipmonk | no |
| 20:45.36 | voipmonk | google , then try out what you've read in your dev environment |
| 20:45.46 | voipmonk | the source might help , too :) |
| 20:46.36 | Anth8708 | voipmonk: thanks. I'm having a not so easy time on google. Will the setup for *1.4 be similar? Lots of hits for that. Not so much for 1.6 |
| 20:49.08 | voipmonk | hopefully u have the basics setup - |
| 20:49.13 | voipmonk | one phone calling another |
| 20:49.18 | voipmonk | yes? |
| 20:49.57 | Anth8708 | voipmonk: yes. i have buddy watching working as well. |
| 20:50.45 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 20:50.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:51.15 | Anth8708 | voipmonk: what i would like is when a buddy key is pressed, it does a pickup. that's a phone limitation from what i understand? |
| 20:51.18 | wcselby | Anth8708 - what type of phone are you trying to get BLF to work on? |
| 20:51.36 | wcselby | what you're asking for works on newer polycom's |
| 20:51.43 | wcselby | not so much on newer ciscos |
| 20:51.51 | wcselby | and I have no clue about the other brands |
| 20:52.01 | *** part/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 20:52.08 | Anth8708 | voipmonk: polycom 560. so you can make the buddy key do something other than dial the direct number it's watching? |
| 20:52.18 | Anth8708 | er, that was for wcselby, sorry |
| 20:52.54 | wcselby | Anth8708 - read through the sip admiin guide for that phone, it will explain the extended blf stuff for you |
| 20:53.06 | wcselby | make sure you're running the latest firmware release for the phone (3.2.x) |
| 20:53.53 | Anth8708 | wcselby: rgr. i have a PRI incoming, and to do this, i won't have to map an extension to a particular trunk, correct? |
| 20:54.13 | wcselby | Anth8708 ... not as far as I know |
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| 20:54.44 | Anth8708 | wcselby: thanks. i'll see what I can find |
| 20:55.04 | *** join/#asterisk kazaa_lite (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 20:55.19 | kazaa_lite | how to run asterisk as user mod onubuntu? |
| 20:56.52 | aiksa[LV] | exit |
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| 21:00.13 | *** join/#asterisk [TK]D-Fender (n=joeblow@161.216.185.63) |
| 21:00.45 | abradley | how much space does asterisk take up? |
| 21:00.59 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 21:01.13 | abradley | in other words, whats the smallest fixe hdd i could use? |
| 21:01.18 | [TK]D-Fender | abradley : not much,why? |
| 21:01.22 | abradley | fixe = size |
| 21:01.55 | citywok | Hey guys, currently i use the AMI to redirect a call, but i am looking for a way to do this, but specify the channel as a variable for asterisk to look up. I have a database variable that is set to a channelid. |
| 21:02.15 | [TK]D-Fender | abradley: it can run off OpenWRT if that's a clue |
| 21:02.16 | citywok | is this possible, or wishful thinking? |
| 21:02.28 | abradley | interesting |
| 21:02.35 | kazaa_lite | is there is there someone running asterisk as non-root on ubuntu? |
| 21:02.36 | abradley | so 1mb then |
| 21:03.00 | [TK]D-Fender | ~asterisk-non-root |
| 21:03.01 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 21:03.03 | wcselby | kazaa_lite - yes, multiple people |
| 21:03.11 | [TK]D-Fender | kazaa_lite: ^^^^ |
| 21:03.33 | citywok | [TK]D-Fender: is this possible? http://pastebin.com/d73cbe04f |
| 21:03.54 | kazaa_lite | when i try to run it as non-root user it never starts..... and any howtos i find on internet are for centos/rh |
| 21:03.56 | citywok | it doesnt work obviously, or i wouldne be asking, but i'm curious if there is some trick to doing it. |
| 21:04.26 | [TK]D-Fender | citywok: ask me in 30mins when I'm home |
| 21:04.40 | citywok | will do |
| 21:04.44 | wcselby | [TK]D-Fender - do you remember who gave you that last link? |
| 21:04.53 | wcselby | the thinkdebian.org link |
| 21:05.07 | [TK]D-Fender | wcselby: nope |
| 21:05.31 | torrancew | we just added 2 fxo modules to our digium card (4 fxo total now), what config files other than chan_dahdi do i need to modify so that change can be reflected? |
| 21:05.46 | Qwell | torrancew: stuff in /etc/dahdi/ |
| 21:05.57 | [TK]D-Fender | wcselby: only that it was someone I'm not used to hearing useful info from.... |
| 21:05.58 | torrancew | kk |
| 21:06.07 | *** join/#asterisk robstar_ (n=robstar_@ip9135d4b2.speed.planet.nl) |
| 21:06.10 | torrancew | thought so, just wasn't sure |
| 21:06.26 | wcselby | [TK]D-Fender - i remember thinking the same thing, I was trying to find that link recently |
| 21:06.38 | kazaa_lite | ??? |
| 21:06.58 | kazaa_lite | is there any tutorial for running asterisk on ubuntu as non-root user? |
| 21:07.06 | wcselby | kazaa_lite - http://www.thinkdebian.org/archives/828 |
| 21:07.28 | [TK]D-Fender | kazaa_lite: nothing that needs to ge Ubuntu specific |
| 21:07.38 | kazaa_lite | thnx |
| 21:08.09 | [TK]D-Fender | kazaa_lite: linux user concepts are the same regardless |
| 21:14.37 | kazaa_lite | cool |
| 21:15.04 | *** join/#asterisk look2 (n=look2@90-231-196-10-no112.tbcn.telia.com) |
| 21:15.42 | look2 | hi, anyone have any experiense with alarmreceiver? I can't get it to work and looking for some help... |
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| 21:24.15 | kazaa_lite | there is no /etc/init.d/asterisk file in my ubuntu machine |
| 21:25.02 | wcselby | did you do a "make config" ? |
| 21:25.30 | kazaa_lite | nopes |
| 21:25.37 | kazaa_lite | is it created by make config? |
| 21:25.49 | ManxPower-work | kazaa_lite: that's the entire reason for make config |
| 21:27.01 | kazaa_lite | so this is the minimal set of commands to issue: ./bootstrap.sh, ./configure, make config, make, make install, make samples |
| 21:27.22 | Anth8708 | wcselby: OK, I've found some material on SLA for the * side and have set it up for testing. I don't suppose you have a working cfg for a polycom 560 you would share? |
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| 21:29.12 | ManxPower-work | Anth8708: almost nobody uses SLA with Asterisk, just BLF |
| 21:30.56 | Athahira | hello * pros... |
| 21:31.25 | Anth8708 | ManxPower-work: I know. But I'm trying. I've hit a wall with the execs wanting their phones to work EXACTLY as they do now. Single users are easily pleased, but the bosses want their extension to appear on the secretaries phone, to ring and be answerable by pushing that line's button. |
| 21:31.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:32.29 | Athahira | anyone installed libss7 can help with some questions ? |
| 21:32.59 | torrancew | Anth8708: I can relate to that problem |
| 21:33.10 | torrancew | (the execs, not the specific situation) |
| 21:33.31 | Athahira | does asterisk have any console commands for the staus of ss7 link ? |
| 21:33.53 | ManxPower-work | Anth8708: Best of luck with that. |
| 21:34.30 | Anth8708 | torrancew: hehe:). I have blf functional, but I need the secretaries appearance to: 1) the line to "ring", not just flash and 2) the be able to be picked up by pressing the line button |
| 21:34.46 | wcselby | Anth8708 - it's the enhanced blf or whatever |
| 21:35.00 | wcselby | I think it was on page 117 of the sip admin guide |
| 21:35.12 | ManxPower-work | Anth8708: You'll be one of the few people in the world that get it working then. |
| 21:35.14 | wcselby | spip_ssip_vvx_Admin_Guide_SIP_3_2.pdf - page 117 |
| 21:35.30 | Anth8708 | wcselby: rgr. You are the man. I'm looking for it now. |
| 21:35.32 | wcselby | ManxPower-work - I've successfully tested it with a polycom ip 650 |
| 21:35.46 | [TK]D-Fender | Anth8708: That can be accomplished by sending the call to both devices anyway |
| 21:35.52 | ManxPower-work | wcselby: tested BLF or real, actual SLA? |
| 21:36.09 | wcselby | I don't know about SLA, I'm talking about what polycom calls their enhanced blf |
| 21:36.19 | ManxPower-work | wcselby: do you know what a Key System is? |
| 21:37.11 | Anth8708 | [TK]D-Fender: yes, those two items can, but then the line doesn't show "in-use" via hints, from what I understand. I need both the in-use indicator and the other two items |
| 21:37.28 | wcselby | http://pastebin.com/m44e1b8c2 |
| 21:37.48 | wcselby | i don't have it in production, just tested it |
| 21:38.10 | wcselby | and no, ManxPower-work - I've not use a Key System or know what specifically you are referring to |
| 21:38.58 | [TK]D-Fender | Anth8708: The latter is simply a BLF |
| 21:39.05 | Anth8708 | wcselby: this should do exactly what I need, and actually will add a "feature" as well i think, disallowing the secretary from making a call "from" the boss's extension. |
| 21:39.34 | Anth8708 | [TK]D-Fender: OK. cool. Sorry for the ignorance. I'm not a phone guy, but I am becoming one:). |
| 21:39.38 | [TK]D-Fender | Anth8708: And *'s "SLA" (cough) does not have you press the BLF key you use to grab a RINGING line |
| 21:39.43 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
| 21:40.00 | wcselby | so anyways, I'm heading out |
| 21:40.03 | wcselby | o/ |
| 21:40.10 | Anth8708 | wcselby thanks a million! |
| 21:40.11 | [TK]D-Fender | Anth8708: the kludge factor in attempting this alone is kinda sad |
| 21:40.55 | Anth8708 | [TK]D-Fender: it's ok. as long as I can get there. My old system is dying and I don't want to maintain it. I want to get off copper between buildings and have the million other features that * has to offer. |
| 21:42.08 | citywok | [TK]D-Fender: is this possible? http://pastebin.com/d73cbe04f |
| 21:42.51 | citywok | i tried using DBGet which is an AMI command, but it wouldnt do that either. i'm not sure if there is any way to do something like this. |
| 21:43.00 | thehar | http://saltlakecity.craigslist.org/eng/1441915476.html << Need a Job? |
| 21:43.32 | [TK]D-Fender | citywok: Channel: ${DB(VMCALLID/8888551)} <- WTF? |
| 21:43.38 | [TK]D-Fender | citywok: this does not get PARSED |
| 21:44.07 | citywok | yea, i'm aware haha. |
| 21:44.20 | [TK]D-Fender | citywok: and that is not an "AMI DBget". That is attempt to shove dialplan function calls into a place that isn't dialplqan |
| 21:44.31 | citywok | i'm curious if there is a way to do it natively, or if i need to do a database get for the variable, find it, and then write it back. |
| 21:44.51 | citywok | yea, i did another attempt with channel: dbget(vmcalid/2154) |
| 21:45.46 | [TK]D-Fender | citywok: You can't shove shit to be PROCESSED into "channel" |
| 21:46.17 | [TK]D-Fender | citywok: this is not extensions.conf. this has to be the FIXED VALUE you use as the tech , peer & number to use |
| 21:46.20 | citywok | i figured, thats why earlier i asked if it was just wishful thinking |
| 21:46.38 | citywok | lol, stupid wishful thinking can cause trouble. |
| 21:46.43 | [TK]D-Fender | citywok: Use another AMI call to retrieve the value then do your other call |
| 21:47.20 | torrancew | when setting up the firewall, what files can i use to control *'s ports |
| 21:47.22 | torrancew | ?* |
| 21:47.41 | [TK]D-Fender | torrancew: sip.conf, rtp.conf |
| 21:51.11 | torrancew | [TK]D-Fender: thanks. all are udp, correct? |
| 21:53.04 | [TK]D-Fender | torrancew: torrancew yes |
| 21:54.57 | torrancew | and lastly, I see that rtp uses a range, does sip as well? or is 5060 (the only specified one) all i need? |
| 21:56.36 | [TK]D-Fender | torrancew: No range for sip |
| 21:56.46 | torrancew | cool |
| 22:01.07 | abradley | after installing asteriskNOW, should one be able to access some sort of webmin or must some setup be done before that can happen? -or can some sort of admin gui be accessed locally? |
| 22:04.03 | *** join/#asterisk izo (n=izo@cpc1-lutn8-0-0-cust299.lutn.cable.ntl.com) |
| 22:04.37 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:14.12 | [TK]D-Fender | abradley: AsteriskNOW includes FreePBX GUI. |
| 22:14.19 | [TK]D-Fender | abradley: And GUI's are NOT suported in this channel |
| 22:14.30 | [TK]D-Fender | abradley: Visit #freepbx or #asterisknow |
| 22:15.17 | *** part/#asterisk izo (n=izo@cpc1-lutn8-0-0-cust299.lutn.cable.ntl.com) |
| 22:19.07 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:19.07 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:21.00 | Katty | hi |
| 22:23.06 | ryduh | hello |
| 22:23.11 | Chainsaw | Hi there. |
| 22:26.32 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
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| 23:09.37 | *** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:15.25 | ecrane | Wow Ekiga is messing w. me too. Apparently it automatically does STUN even when it's not necessary. And the gui feature to disable stun was removed in version 3.X so you gotta run a gconf. Grrrrrrr.......... |
| 23:16.00 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 23:22.33 | *** join/#asterisk Wollie (n=tamar@190.106.64.22) |
| 23:22.43 | Wollie | good evening! |
| 23:23.01 | Wollie | I'm having a strange problem that I can't find the solution for. Anybody in for some advice? |
| 23:23.18 | Wollie | I have 2 phones, 1 Linksys and one brand unknown, both connected to the Asterisk |
| 23:23.26 | Wollie | phonenumber 201 and 301. |
| 23:23.42 | Wollie | if 301 calls 201 I can only hear the other side, but not the response |
| 23:23.54 | Wollie | if 201 calls 301 everything goes well. |
| 23:24.06 | Wollie | in the logfiles I noticed a difference between the calls |
| 23:24.21 | Wollie | the failed call: |
| 23:24.28 | Wollie | <PROTECTED> |
| 23:24.28 | Wollie | <PROTECTED> |
| 23:24.28 | Wollie | <PROTECTED> |
| 23:24.28 | Wollie | <PROTECTED> |
| 23:24.33 | Wollie | and the good call: |
| 23:24.47 | Wollie | -- Called 201 |
| 23:24.47 | Wollie | <PROTECTED> |
| 23:24.47 | Wollie | <PROTECTED> |
| 23:24.52 | Wollie | the difference is the Packet2Packet line |
| 23:25.29 | Wollie | can anybody tell me why that line is there and how do I make sure no call is packet2packet? |
| 23:26.26 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
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| 23:41.17 | AlmightyOatmeal | can i have more than one localnetwork specification in sip.conf? |
| 23:43.56 | AlmightyOatmeal | friggin sweet |
| 23:43.57 | *** part/#asterisk AlmightyOatmeal (n=jamie@76-255-16-163.lightspeed.mdsnwi.sbcglobal.net) |
| 23:48.03 | ecrane | Not asterisk question, but if anyone knows this they're in this channel.... for Cisco 7960 phones, is there any way I can reconfigure them from the handset? Or do I have to set up a TFTP server or BootP or something? |
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