00:01.05 | drray | there are lots of reason to curse in here |
00:01.14 | ManxPower | http://pastebin.ca/72584 |
00:02.22 | smackus | ok, on my old asterisk system, I successfully had cdr logging to my mysql database. On my new system, I thought I had everything set up correctly, but it is still going to the csv. I have may database up, my cdr_mysql pointed at it. what am i overlooking. I am trying to find in the wiki if there is a file that has to tell it to use the database... I am lost. I must have done it by accident last time. Where am I missing my setting. |
00:02.28 | smackus | wow... sorry. |
00:02.31 | smackus | i type too much |
00:02.52 | *** join/#asterisk Qb3rt (i=jhgjkgui@216.252.87.8) |
00:02.57 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
00:03.37 | *** join/#asterisk albertito (n=net@host10.201-253-236.telecom.net.ar) |
00:03.46 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
00:03.53 | terrapen | does anybody know of a voice other than The Voice? |
00:03.59 | terrapen | I'm not too keen on her voice |
00:04.08 | rob0 | There are voices in my head ... |
00:04.18 | terrapen | I need a voice that's a little more relaxed and less uptight |
00:04.20 | terrapen | heh rob |
00:04.36 | smackus | you dont like alison? she makes me all tingly |
00:04.42 | drray | no kidding |
00:04.54 | smackus | like the rope in gym class :-D |
00:04.58 | ManxPower | Must. resist. fighting. Asterisk. |
00:05.00 | terrapen | she's a hottie no doubt |
00:05.02 | albertito | Hi! I've just connected two asterisk boxes with an E1, and both alarms are cleared, but when I start asterisk I see (on only one of them) lots of "PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1". Channels seem to get up tho, but I haven't tested they work yet. Is this expected? What should I check? |
00:05.07 | terrapen | (at least the head shot) |
00:05.11 | rob0 | Hmmm, in fact I could swear that she was one of the voices! |
00:05.30 | terrapen | but I need somebody more suitable for this bicycle stores |
00:05.33 | terrapen | err these |
00:05.34 | smackus | is there a way to verify if asterisk-addons is installed? |
00:05.43 | ManxPower | albertito, either there are errors on the E1 or you have a device on the system that is locking interrupts and causing dropped data |
00:06.28 | albertito | ManxPower: so it could be a faulty cable that causes the HDLC errors? |
00:06.30 | rob0 | You're talking about hiring someone? Just find an actor/actress/broadcaster. An ad agency might help. |
00:06.38 | ManxPower | albertito, correct. |
00:06.42 | drray | and you'll need studio time |
00:06.53 | ManxPower | That means "corrupted or lost data" It does not tell you where the problem is happening |
00:06.56 | drray | which depending on where you live could be cheap |
00:07.00 | smackus | i think he is looking for something already completed |
00:07.08 | smackus | recordings already done |
00:07.40 | albertito | ManxPower: given that the box with errors is a decent Pentium D, without significant load, and that the card doesn't share the interrupt, I guess I'll blame on the cable. Thanks a lot! I'll swap the ends just to confirm (they should appear on the other box) |
00:08.00 | terrapen | nope |
00:08.01 | Qb3rt | i am having a very nice problem!!! i have 3 polycom phones behind a firewall and when i try to dial it says url call disabled... can someone tell me what is the problem?? |
00:08.13 | terrapen | i want to e-mail some prompts to someone with a good voice and get some WAVs back |
00:08.22 | terrapen | and i'm looking for pricing similar to The Voice |
00:08.52 | ManxPower | albertito, The problem can also be caused by GigEthernet, RAID, SATA, etc |
00:08.57 | terrapen | http://www.intervoice24.com/ |
00:08.58 | terrapen | niiiice |
00:09.11 | terrapen | thank you google adwords |
00:09.43 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
00:09.57 | rob0 | I googled "professional voice recordings", saw a bunch |
00:09.58 | terrapen | their samples suck tho :) |
00:10.03 | albertito | ManxPower: I've switched the ends and the error remains on the pentium D (from now on, "big box", because the other one is a Pentium II, "small box") |
00:10.35 | albertito | ManxPower: how can SATA (I don't have the other ones) interfer with the digium card, since there's no disk activity going on? |
00:10.52 | ManxPower | albertito, there is logging going on isn't there |
00:11.10 | albertito | ManxPower: it's logging the errors coming out of asterisk, yes |
00:11.33 | ManxPower | albertito, that's all it took on one system I installed. Also voicemail caused disk writes. |
00:12.01 | albertito | ManxPower: but vmstat shows only activity once every five seconds, and about 80k each write... that just can't be it |
00:13.16 | albertito | ManxPower: I have around 1300 interrupts per second, and given that HZ=1000 here, that's about 300 interrupts per second over the base |
00:13.41 | ManxPower | albertito, I have given you my diagnoses. |
00:13.42 | albertito | ManxPower: OTOH the small box is running at 2000 interrupts per second |
00:14.00 | ManxPower | albertito, what is the interrupt latency and interrupt jitter? |
00:14.11 | albertito | ManxPower: Thanks a lot =) I'm just confused because I can't see it fit, but I'll dig into it |
00:14.22 | albertito | ManxPower: I don't know how I find out about those |
00:14.35 | ManxPower | albertito, Neither do I. |
00:14.38 | terrapen | so, what makes for the best on-hold music? |
00:14.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:14.49 | terrapen | i like Americana myself |
00:15.28 | albertito | ManxPower: I understand your diagnosis about the irqs, but they're just two unloaded machines, with no calls going on, which is why I find it strange |
00:16.46 | ManxPower | albertito, Perhaps you do not understand that regardless of how many interrupts are happening per second, if one device or driver, such as a SATA controller locks interrupts for a very long time, the digium card will fail to get data fast enough, since Digium cards do not have onboard buffers. |
00:17.43 | ManxPower | I had a problem with a Dual Xeon motherboard that had the same errors any time there was ANY disk activity. Replaced the motherboard with a different make/model and everything worked. |
00:17.59 | ManxPower | MANY people have reported similar experiences with PCI RAID, SATA, and GigEthernet |
00:19.05 | albertito | ManxPower: mmmmm I don't have RAID or GigEthernet, and SATA is inactive most of the time... but I have one USB device that might be the one, now that I think of it... |
00:19.42 | ManxPower | albertito, All it takes is ONE disk read or write |
00:20.40 | albertito | ManxPower: if there is at least one disk read or write, vmstat "bo" or "bi" column should notice it, and it doesn't (except once every 5 seconds, but these errors appear continuously) |
00:20.53 | albertito | ManxPower: and the usb device wasn't either |
00:21.02 | Qb3rt | I have 3 polycom phones behind a firewall and when i try to dial it says url call disabled... Is it because of th firewall??? how can i fix that? |
00:21.21 | ManxPower | Qb3rt, It is NOT because of the firewall. |
00:21.33 | ManxPower | The phone is not registered to a server and therefore cannot make calls |
00:23.15 | albertito | ManxPower: anyway, thanks a lot for the diagnosis and support, I'll look deeper into it. I'll see if I can get irq latency measures using some kernel instrumentation |
00:23.16 | *** join/#asterisk Samoied (n=Samoied@201.22.215.135.adsl.gvt.net.br) |
00:23.49 | Qb3rt | ManxPower: ok thanks! and when i will call one of these 3 phones how the router will know to wich one send the call? |
00:23.56 | *** join/#asterisk brc__ (n=brc_@pdpc/supporter/basic/brc) |
00:24.19 | ManxPower | Qb3rt, The router modifies the source port of the registration |
00:24.38 | ManxPower | That is what NAT does. |
00:25.09 | Qb3rt | hehe yeah!!! that was a pretty stupid question if i think about it!!! |
00:27.04 | Qb3rt | ManxPower: so basically the polycom phone will register through any kind of firewalling? |
00:27.31 | Qb3rt | i mean on d-link and linksys |
00:27.39 | ManxPower | Qb3rt, Do you understand the difference between NAT and Firewall? |
00:27.47 | Qb3rt | yeah for sure |
00:28.07 | Qb3rt | but on a router you can enable a basic firewall system |
00:29.00 | ManxPower | somegeek, no if the firewall is blocking the packets then obviously the phones won't work. However if the router is just doing NAT everything should work if you have nat=yes and qualify=10000 in sip.conf for each registration |
00:33.38 | mindwarp | Hi guys, I have a question that I have done my best to detail in this forum post: http://forums.digium.com/viewtopic.php?t=7610 ... Not getting very far so far in the forums, so please take a look if you feel so inclined. Thanks a lot! |
00:35.06 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-156-227.dsl.pltn13.sbcglobal.net) |
00:35.58 | ManxPower | http://pastebin.ca/72615 |
00:38.49 | darkgamer20 | an the TDM400P functions like a analog telephone adapter right? |
00:41.50 | ManxPower | darkgamer20, In a way. |
00:42.39 | darkgamer20 | ManxPower: yea so I can connect my phone to the FXS and the phone line to the FXO port and have asterisk handle my calls? |
00:42.52 | ManxPower | darkgamer20, correct |
00:43.04 | darkgamer20 | ManxPower: perfect |
00:43.48 | darkgamer20 | ManxPower: how much dose a TDM400P card cost? also you know if theres any alternatives i can turn to if its too expensive? |
00:44.13 | ManxPower | darkgamer20, did you go to the Digium web site? All similar cards cost about the same. |
00:44.19 | darkgamer20 | oh |
00:45.39 | darkgamer20 | wooooooo the one i need (and the cheapest one) is 241$$ |
00:45.52 | darkgamer20 | isnt there anything under 100? |
00:47.55 | ManxPower | darkgamer20, Expect to pay $80 - $120 per port, depending on the number of ports you need. The fewer the number of ports the more expensive each one is. |
00:48.24 | ManxPower | darkgamer20, Traditional telcom equipment is closer to $500 per port |
00:48.31 | ManxPower | many times much higher. |
00:48.35 | darkgamer20 | man thats expensive ManxPower |
00:48.36 | darkgamer20 | wow |
00:49.44 | darkgamer20 | ManxPower: can you suggest anything cheaper for a hobby project, cause I am not trying this for an office enivironment |
00:50.18 | ManxPower | darkgamer20, Well if you want to spend 10x the amount of time working on it I guess you could get a SIPura |
00:50.43 | drray | I would get a govarion 4 port tor2 card, and a $100 zhone channel bank |
00:51.18 | darkgamer20 | drray: can you tell me the exact name of the product your suggesting? |
00:52.42 | drray | http://govarion.com/home.php and zhone channel bank |
00:54.08 | drray | the idea is to get things that you can use later |
00:54.09 | drray | you'll throw the zhone in the garbage 9 months after you get it |
00:54.48 | darkgamer20 | drray: sorry but where is the zhone again? i cant find in the page you entered |
00:55.29 | darkgamer20 | ohhh |
00:55.52 | darkgamer20 | nevermind they are two different things, i kept thinking that zhone is on govarion |
00:55.57 | drray | http://cgi.ebay.com/Channel-Bank-24FXS-Ports_W0QQitemZ9744803968QQihZ008QQcategoryZ51271QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
00:56.09 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
00:56.20 | drray | you can get 24 fxs ports for $150 or so |
00:56.27 | ManxPower | you'll have trouble finding FXO ports for channel banks |
00:56.29 | drray | or 8 fxo 16 fxs ports |
00:56.38 | drray | ebay |
00:56.50 | ManxPower | Dude! The 4-port card is $700 |
00:57.04 | drray | and that corpsys site sells them on the side |
00:57.26 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
00:57.34 | drray | 4 ports of t1 is a bargain |
00:57.54 | ManxPower | Yes, but if he can't afford $241, he can't affford $70 |
00:57.56 | ManxPower | ..er.. $700 |
00:58.07 | darkgamer20 | lol |
00:58.16 | darkgamer20 | yea ManxPower is right |
00:58.34 | ManxPower | and zhone is crap |
00:58.41 | drray | so is a MTA |
00:58.51 | darkgamer20 | a Mail Transger Agent? |
00:59.03 | drray | there are only two things the zhone can't do |
00:59.07 | drray | pass caller ID |
00:59.16 | drray | and detect remote disconnects |
00:59.29 | darkgamer20 | well i need caller ID |
00:59.31 | drray | for $150 they are great |
00:59.35 | ManxPower | Um, so that would be called "crap" |
01:00.27 | darkgamer20 | ok I dont i have the need for a channel bank, do you guys know if I can get a TDM400P on ebay for less? |
01:01.28 | drray | seeing as you cant/shouldnt run 2 tdm400p in one box |
01:01.41 | Un1x | darkgamer20: yea you could but who knows, if it's going to work or , if it does work for how long will it work, or what will happen with it.. |
01:02.07 | drray | with the t1 card, you can upgrade to a PRI as you expand |
01:02.27 | Un1x | they have some but most cards are the same price as digium anyway |
01:02.30 | darkgamer20 | Unlx: you got a point there, but I what can I do...? |
01:02.30 | Un1x | hardly any difference |
01:02.34 | Un1x | maybe 2-5 dollars |
01:02.44 | Un1x | darkgamer20: what do you need it for |
01:02.47 | darkgamer20 | drray: I am not looking to expand this is a hobby project |
01:02.49 | Un1x | how many phones how many pstn lines |
01:03.00 | RoyK | ka-ding |
01:03.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
01:03.18 | RoyK | ka-ching! i got my tax refund :D |
01:03.23 | Un1x | lmao |
01:03.38 | darkgamer20 | Unlx: 1 PSTN line and 2 phones (1 phone and a fax machine) |
01:03.49 | RoyK | ~nickometer Un1x |
01:03.54 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
01:03.58 | Un1x | so youd need, 2FXS and 1 FXO |
01:04.11 | Un1x | ~fxofxs |
01:04.13 | jbot | well, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
01:04.14 | RoyK | websae: it's wasabi |
01:04.25 | Un1x | yea u need 2fxs and 1fxo |
01:04.35 | websae | haha |
01:04.42 | websae | what's going on? |
01:04.52 | Un1x | http://cgi.ebay.com/Digium-Wildcard-TDM12B-TDM400P-Asterisk-1FXS-2FXO_W0QQitemZ9744374347QQcategoryZ61839QQssPageNameZWD1VQQrdZ1QQcmdZViewItem |
01:05.03 | Un1x | heres one darkgamer20 but no gauruntees on it |
01:05.08 | darkgamer20 | Unlx: before i buy anything, i can have asterisk have the calls go to the right places right? like have a fax go to the fax machine and a phone call to the phone? |
01:05.20 | Un1x | lets check digiums price |
01:05.22 | Un1x | i beleive so |
01:05.25 | Un1x | ask dlynes_home |
01:05.29 | RoyK | lots of voip nerds and one pretty drunk norwegian engineer - that's going on |
01:06.19 | Un1x | hey |
01:06.38 | Un1x | Darkgamer20: this is pretty much what youre looking for http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?category_id=17&product_code=RTDM11B |
01:06.44 | *** join/#asterisk Qwellj2me (n=Qwell@unaffiliated/qwell) |
01:06.46 | Un1x | you could get it for the same price almost from digium |
01:06.53 | Un1x | comes with 5 yr warranty also :) |
01:07.02 | Un1x | and it's same price as on ebay :p |
01:07.16 | darkgamer20 | lol |
01:07.32 | Un1x | just email them and tell them you want one extra FXS port, and theyd do that and change the price to what need be |
01:07.36 | Un1x | it should be under 300 |
01:07.59 | darkgamer20 | it will probably be as much as the ebay one huh? |
01:08.02 | Un1x | yes |
01:08.10 | Un1x | pretty much with gauruntee it wil work |
01:08.13 | Un1x | and with warranty |
01:08.17 | Un1x | and the cables etc |
01:08.26 | Un1x | and the screwdriver and mousepad lmao |
01:08.51 | darkgamer20 | lol |
01:08.57 | Un1x | thats just a bonus |
01:09.04 | darkgamer20 | might as well get it from digium then |
01:09.06 | Un1x | heh at least u get somethings added for same price |
01:09.10 | Un1x | i personaly think it's worth it |
01:09.30 | darkgamer20 | alright whatever i can spend more than a 100$ if theres a benifit |
01:09.50 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
01:09.59 | Un1x | nah |
01:10.01 | Un1x | not even 100$ |
01:10.07 | Un1x | from wjat i can see on ebay |
01:10.09 | Un1x | for the same config |
01:10.17 | Un1x | only around 30$ extra max |
01:10.31 | Un1x | US $293.40 |
01:10.35 | Un1x | is the one on ebay for same config |
01:10.41 | Un1x | it should be around the same price from digium |
01:11.04 | darkgamer20 | oh ok |
01:11.15 | Un1x | anyway i gotta go mann ask the other people in here, gnite maybe' i'll come around later.,, see ya all |
01:11.16 | Un1x | :) |
01:11.17 | darkgamer20 | Un1x Thanks soooo much for your help |
01:11.21 | Un1x | no problem |
01:11.24 | darkgamer20 | see ya |
01:19.53 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
01:20.03 | sorush20 | hi guys |
01:20.08 | sorush20 | vonage anyone? |
01:20.40 | rob0 | TDM S110M FXS Module US$67.5 ... know of any better deals? Preferred vendors? |
01:21.00 | Strom_C | that's a pretty damned good deal right ther |
01:21.09 | websae | lol |
01:21.10 | Strom_C | assuming, of course, that it works ;) |
01:21.11 | websae | vonage |
01:21.12 | websae | ha |
01:21.18 | Strom_C | also, lol vonage |
01:21.23 | sorush20 | !voange |
01:21.47 | websae | what's the word that vonage always makes me think of.... |
01:22.02 | websae | oh yeah it's a phrase |
01:22.04 | mindwarp | ownage? |
01:22.06 | rob0 | I found 2 vendors with the same price: voipstore.atacomm.com, www.voiplink.com |
01:22.10 | websae | soon to file for bankruptcy |
01:22.20 | websae | what are you looking for rob0? |
01:22.25 | rob0 | Voyage :) |
01:22.34 | drray | vonage to the bottom of the sea |
01:22.35 | rob0 | I need one FXS and one FXO. |
01:23.00 | rob0 | in that order ... I have an x101p for FXO. |
01:23.30 | websae | http://www.pbxeq.com/ |
01:23.42 | websae | they are rock solid hardware supplier |
01:24.22 | *** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
01:24.48 | websae | oh so they helped me out a lot |
01:27.10 | rob0 | thanks |
01:29.37 | websae | yeah |
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01:42.03 | skraelings001 | Hi |
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01:44.19 | FuriousGeorge | would everyone agree that bristuff is probably safer than trunk for getting device states with parking working in a production enviornment? |
01:44.28 | Brijn | Good evening all |
01:44.49 | skraelings001 | Good eve. |
01:44.58 | w0rmzw3rth | Is it possible if I buy 5 voip lines from voicepulse.com that I can add those lines to my asterisk box so that 5 users can use it at the same time and place calls from the box |
01:45.00 | [TK]D-Fender | Not if you need PRI which it conflicts with as I'm told |
01:45.43 | file | [TK]D-Fender: My my look who it is! |
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01:49.21 | [TK]D-Fender | OMGZ! |
01:49.32 | [TK]D-Fender | file : In Mississauga now :) Leeching Wi-Fi from... somewhere ;) |
01:49.40 | file | [TK]D-Fender: ooh, why are you there? |
01:49.48 | [TK]D-Fender | w0rmzw3rth : Sure |
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01:51.52 | skraelings001 | does anyone experienced strange behaviour when trying to call to phone numbers that doesn't exit? , i should get a msg from telco but i don't |
01:52.13 | [TK]D-Fender | Vacation.. you knew :) get with the program. |
01:52.24 | file | I know lots of things, doesn't mean I remember them |
01:52.48 | Tholia | Are most of the people in here telecom people by trade? I'm kinda a half-assed telecom person, thrown to the wolves on a couple Rolm systems and learning everything from scratch |
01:52.58 | anonymouz666 | file haha good one |
01:53.11 | file | Tholia: I was a high school student when I got involved in VoIP, been with it since! |
01:53.41 | Tholia | my company is in the middle of a VoIP conversion from the Rolm and Aspect switches we have to an Avaya VoIP solution |
01:53.43 | skraelings001 | i recently got involve with asterisk cause of work |
01:54.11 | anonymouz666 | file? pthread is hard to learn? |
01:54.22 | file | anonymouz666: it's not too bad to learn |
01:54.58 | Tholia | Trying to get into asterisk is like drinking water from a fire hose. I have a spare box, I can afford a simple FXO card, after that trying to find a VoIP provider seems like the dicey part |
01:56.26 | [TK]D-Fender | Tholia : IP office? |
01:56.28 | file | Tholia: cheat, turn the water level down! |
01:57.21 | [TK]D-Fender | Tholia : I almost got forced down that path.... So glad I'm not stuck with their toaster :) |
01:57.24 | Tholia | :) I'm up for any recommendations y'all have on hardware or software for the home tinkerer (that has an Windows Active Directory domain infrastructure) :) |
01:57.57 | [TK]D-Fender | Tholia : Windows is irrelevent as * is basically a *nix platofrm. For hardware it al depends on what you want. |
01:58.20 | [TK]D-Fender | Tholia : What kind/number of lines & phones would you like to use? |
01:58.39 | PakiPenguin | can someone do some testing with me? |
01:58.43 | PakiPenguin | call this number ? 708-547-8653 ? |
01:59.12 | Tholia | right just saying that's my level of home tinkering. initially I'm OK with using a softphone to call wiht (to test), I'll have a dedicated PC. Eventually I'd like to have a POTS line (or VoIP line) coming into the PBX, with autoattendant and conferencing and whatnot |
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01:59.39 | Tholia | for a few softphone extensions initially, then maybe even an ATA or two or some IP phone hardware |
02:00.00 | PakiPenguin | please |
02:00.09 | websae | Tholia: try www.pbxeq.com --- got some nice deals |
02:00.38 | *** join/#asterisk b00gz1 (n=b00gz@d233-124-245.col.wideopenwest.com) |
02:00.43 | Tholia | looking at their site now. I don't really think I need a FXS port quite yet |
02:00.50 | Tholia | don't even think I need an FXO port yet |
02:01.01 | websae | get a VoIP provider and ATA/SIP phone |
02:01.15 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
02:01.34 | Tholia | recommended VoIP providers? I don't need any international calling, and probably spend about 2 hours a month on the phone really, we're mostly a cell house |
02:03.55 | [TK]D-Fender | Tholia : Then get one that bills by the minute. ATA's are great if you want to reuse your wiring, etc. Good first model for you would be an SPA-3102. That'd give you 1 FXS (extensions), and take in your home analog line as well (1 FXO). All for $90USD |
02:04.18 | justinu | 3102 now? any improvements? |
02:04.52 | Tholia | whats a good rate for US only bill by the minute plans? |
02:05.00 | *** join/#asterisk Katty (n=Administ@dialup-4.244.123.58.Dial1.StLouis1.Level3.net) |
02:05.20 | Katty | evening. |
02:05.43 | [TK]D-Fender | justinu : More room for firmware, etc. |
02:05.47 | [TK]D-Fender | Katty: Mew. |
02:05.48 | justinu | hey katty |
02:05.52 | FuriousGeorge | [TK]D-Fender: i dont need pri, sorry for delayed response |
02:06.04 | w0rmzw3rth | TKD-Fender so if I'm looking for just a small pbx for people to dial into and make out going calls I can just get 5 phone lines at voicepulse.com and add them in the AIX config file and I will have 5 lines for 5 people to use to make outbound phone calls correct |
02:06.18 | Katty | [TK]D-Fender: hey you. how's it goin? |
02:06.24 | Katty | hey justinu (= |
02:06.26 | [TK]D-Fender | FuriousGeorge : There is a subset for the "pseudo state" driver that was hoped to be merged for 1.4 last I recall so you don't need to patch |
02:06.27 | FuriousGeorge | till your internet goes down :) |
02:06.45 | justinu | i bought expensive dsl |
02:06.54 | justinu | they actually sorta care |
02:07.20 | [TK]D-Fender | w0rmzw3rth : Yes, you could do that... all depends on whats most econimical. Some places you just pay be the minute and they allow multiple simultaneous calls up to a prescribed limit. |
02:08.15 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-97-202.w86-206.abo.wanadoo.fr) |
02:09.00 | justinu | ~ecfo |
02:09.05 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
02:09.20 | [TK]D-Fender | Katty : Getting by. Life changes to come, personal stuff. On vacation now just taking a load off |
02:09.27 | w0rmzw3rth | ? |
02:10.19 | justinu | weird... getting some ECFO on L3 |
02:10.31 | justinu | sounds just like it does with the zap software EC |
02:11.14 | w0rmzw3rth | I wouldn't need to buy any hardware then correct if I did it that way [TK]D-Fender |
02:11.32 | justinu | you at least want your own ATA |
02:11.41 | FuriousGeorge | [TK]D-Fender: i see now... i had looked into that patch, but i applied it against the wrong version i guess, im gonna try again with 1.2.9.1 |
02:11.57 | [TK]D-Fender | w0rmzw3rth : Depends what you want to use for phones.... |
02:12.05 | w0rmzw3rth | justinu where you talking to me? |
02:12.06 | FuriousGeorge | we are talking about the meetermaid patch that was backported to 1.2.9.1 right? |
02:12.37 | skraelings001 | should PRI manage all the signals and messages from telco? like "this number does not exist", "the number has changed, the new number is ..."?? |
02:12.37 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
02:12.46 | w0rmzw3rth | I don't need any phone I just want someone to call into my pbx have it answer have them login thier mail box and have options to check mail or place a call |
02:12.48 | justinu | w0rmzw3rth: yeah |
02:12.59 | justinu | ok, then you need a pbx |
02:13.12 | justinu | or you need someone else to set you up on theirs |
02:13.21 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
02:13.24 | *** join/#asterisk chumper2342 (n=cj@cpe-70-112-211-200.austin.res.rr.com) |
02:13.26 | *** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
02:13.50 | justinu | skraelings001: that's a good question... typically you provide those messages yourself |
02:14.02 | justinu | but there are ISDN cause codes you can return to the telco to indicate such things |
02:14.45 | chumper2342 | I need a little help with my asterisk 1.2.9.1 server. When i sip reload, it says "..Cannot allow unknown format 'g711'. |
02:14.47 | justinu | asterisk has all the voice prompts you need to play them yourself, and it sounds quite professional |
02:15.00 | justinu | chumper2342: change it to "ulaw" |
02:15.05 | chumper2342 | in my sip.conf i have disallow=all, allow=g711, |
02:15.07 | chumper2342 | ok |
02:15.13 | justinu | or "alaw" if you're not in USA |
02:15.29 | skraelings001 | sometimes i got those messages in a MESSAGE ALERT and they pass the audio before desconnect but usually it doesn't happen |
02:16.01 | justinu | skraelings001: the way it's supposed to work, is you get a "progress" IE telling you inband info is available |
02:16.17 | [TK]D-Fender | chumper2342 : Its ALAW or ULAW, not entered as G711 |
02:16.49 | skraelings001 | justinu: you mean only in the case this is available? |
02:17.17 | justinu | yeah, in theory |
02:17.23 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
02:17.29 | justinu | otherwise, they just send DISCO right away, with the appropriate cause code |
02:17.55 | justinu | afk |
02:18.09 | chumper2342 | ok, we that didn't solve my origional problem lol, when i call a local number it goes out, rings my cell phone then: "No path to translate from SIP/mysoftphone to SIP/Provider-outbound" "Had to drop call because I couldn't make Sip/mysoftphone compantible with SIP/Provider-outbound |
02:18.12 | skraelings001 | yes, i checked Q.931 |
02:19.05 | [TK]D-Fender | chumper2342 : "allow=g711" = no legit codec name |
02:20.27 | skraelings001 | i'm sure that my telco is passing such info but can't get it |
02:23.28 | PakiPenguin | has anyone used these http://www.junghanns.net/en/GSM-PCI_produkt.html ? |
02:23.36 | chumper2342 | ok calls to my cell work but I can't hear anything on either ends. Provider says g729,g711 ulaw. sip.conf = disallow=all, allow=ulaw |
02:24.07 | [TK]D-Fender | chumper2342 : pastebin your sip.conf |
02:24.16 | skraelings001 | chumper2342: probably problems with nat |
02:25.28 | [TK]D-Fender | skraelings001 : Not yet, so far its a codec incompatability issue |
02:25.47 | iq | PakiPenguin: what r u upto ;) |
02:26.42 | PakiPenguin | iq, a lot of things :) |
02:27.17 | iq | PakiPenguin: i can see that |
02:27.28 | PakiPenguin | :) |
02:27.40 | SplasPood | http://newyork.craigslist.org/mnh/mar/166423486.html <-- that job rocks my world :P |
02:27.41 | PakiPenguin | god i miss wifi :'( |
02:27.54 | SplasPood | or rather, the job posting does |
02:28.04 | [TK]D-Fender | ~pb |
02:28.07 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
02:28.11 | chumper2342 | [general] |
02:28.11 | chumper2342 | context=default |
02:28.11 | chumper2342 | srvlookup=yes |
02:28.11 | chumper2342 | register => user:pass@*.com/# |
02:28.11 | chumper2342 | [telasip-outbound] |
02:28.12 | chumper2342 | type=peer |
02:28.14 | chumper2342 | host=*.com |
02:28.17 | chumper2342 | username= |
02:28.18 | chumper2342 | secret= |
02:28.20 | SplasPood | tooooo laaaaaaaate |
02:28.20 | chumper2342 | qualify=yes |
02:28.22 | chumper2342 | disallow=all |
02:28.24 | chumper2342 | allow=ulaw |
02:28.26 | chumper2342 | context=outbound |
02:28.28 | chumper2342 | [linux-softphone] |
02:28.29 | Strom_C | chumper2342: DON'T DO THAT AGAIN |
02:28.30 | chumper2342 | type=friend |
02:28.32 | chumper2342 | secret= |
02:28.34 | chumper2342 | qualify=yes |
02:28.36 | chumper2342 | nat=no |
02:28.36 | SplasPood | he's not done :) |
02:28.38 | chumper2342 | host=dynamic |
02:28.40 | chumper2342 | canreinvite=no |
02:28.41 | Strom_C | oh dear god |
02:28.42 | chumper2342 | context=internal |
02:28.44 | chumper2342 | disallow=all |
02:28.46 | chumper2342 | allow=ulaw |
02:28.46 | iq | chumper2342: please stop ! |
02:28.48 | chumper2342 | [windows-softphone] |
02:28.50 | chumper2342 | type=friend |
02:28.52 | chumper2342 | secret= |
02:28.54 | chumper2342 | qualify=yes |
02:28.54 | SplasPood | I think it's too late for him |
02:28.56 | chumper2342 | nat=no |
02:28.58 | chumper2342 | host=dynamic |
02:29.00 | chumper2342 | canreinvite=no |
02:29.02 | chumper2342 | context=internal |
02:29.04 | chumper2342 | disallow=all |
02:29.06 | chumper2342 | allow=ulaw |
02:29.08 | chumper2342 | very sorry |
02:29.15 | Strom_C | chumper2342: PASTEBIN |
02:29.16 | anonymouz666 | doomed |
02:29.17 | SplasPood | haha |
02:29.18 | anonymouz666 | hehe |
02:29.19 | SplasPood | that ruled |
02:29.20 | Strom_C | idiot |
02:29.20 | SplasPood | do it again |
02:29.30 | PakiPenguin | shit |
02:29.36 | PakiPenguin | idiot! |
02:29.51 | Strom_C | ~pb |
02:29.53 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
02:30.50 | SplasPood | Oh come on guys |
02:30.55 | SplasPood | everyone makes that mistake once |
02:31.14 | Strom_C | even after being told to use pastebin? |
02:31.18 | Strom_C | pfffffft |
02:31.21 | Strom_C | I think not |
02:31.29 | chumper2342 | i was in a hurry, sorrry...... |
02:31.39 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
02:32.03 | SplasPood | Strom_C: I think paste had been initiated before ~pb responded |
02:32.30 | Strom_C | I WIN! |
02:32.46 | Strom_C | :) |
02:33.45 | chumper2342 | http://bzflag.pastebin.ca/72713 |
02:34.28 | Strom_C | chumper2342: now what's the problem, exactly? |
02:34.36 | chumper2342 | my xlite soft phone is using g711 ulaw |
02:34.53 | Strom_C | and that's a problem? |
02:35.00 | chumper2342 | when I call my cell from my soft phone, i hear no sound on either end |
02:35.03 | Brijn | Anyone has a nice set of Visio stencils for ASterisk/VoiP? |
02:35.26 | Strom_C | chumper2342: is the asterisk box and/or the SIP phone behind a NAT? |
02:35.28 | chumper2342 | it works calling computer from 1 soft phone to another (internal) |
02:35.40 | chumper2342 | no |
02:35.53 | Strom_C | the asterisk box has a public IP address? |
02:35.57 | chumper2342 | yes |
02:36.02 | Strom_C | and the SIP phone also has a public IP address? |
02:36.12 | chumper2342 | oops |
02:36.17 | chumper2342 | but i tried nat=yes |
02:36.26 | chumper2342 | no |
02:36.48 | Strom_C | chumper2342: so theres a NAT between the asterisk box and the SIP phone, right? |
02:36.53 | chumper2342 | yes |
02:37.09 | Strom_C | is the SIP provider using ulaw? |
02:37.41 | chumper2342 | yep, their email says exactly: vocoder: g729, g711 ulaw |
02:37.46 | chumper2342 | does that mean it needs both? |
02:37.50 | Strom_C | no |
02:38.14 | NotJohnDavid | anyone used one of the aastra phones and can compare it to the grandstream gxp2000 ? |
02:38.28 | Strom_C | try putting canreinvite=no into the ITSP's entry |
02:39.24 | Brijn | Cool: Nokia 770 panel for < 400US$, better then all this UMPC shit |
02:39.32 | [TK]D-Fender | NotJohnDavid : What do you want out of a phone? |
02:40.52 | NotJohnDavid | tkd: well I have a gxp2000. it seems to work. doesn't seem to shout "quality" to me though. |
02:41.28 | chumper2342 | nope, didn't work |
02:41.53 | Strom_C | chumper2342: you did a sip reload, right? |
02:41.58 | chumper2342 | yep |
02:42.27 | Strom_C | you're sure there's no NAT between your asterisk box and the ITSP, right? |
02:42.32 | NotJohnDavid | There'll be 3 lines coming in. probably 5 extensions. the gxp2000 would work but the aastra's look nicer from the small pics i've seen. didn't know if anyone had used both and would compare the quality of construction/how they are to use |
02:42.44 | Strom_C | aastra has higher quality construction |
02:42.53 | Strom_C | of course, anything has higher quality construction than grandstream :) |
02:42.59 | NotJohnDavid | figures :) |
02:43.01 | chumper2342 | sure, it has a public ip, i can access it from anywhere |
02:43.27 | NotJohnDavid | i liked how this grandstream didn't even come with an instruction manual. or any piece of paper for that matter |
02:43.46 | Strom_C | chumper2342: what happens when you set up a call just between the asterisk box and the PSTN? does the asterisk box play audio? |
02:44.06 | NotJohnDavid | i guess i'd be considering between linksys/polycom/aastra |
02:44.15 | [TK]D-Fender | NotJohnDavid : Where are you located? |
02:44.22 | NotJohnDavid | tkd: eastern tennessee |
02:44.27 | Strom_C | polycom is going to be the highest quality of those three, though I like teh cisco phones, to be honest |
02:44.36 | [TK]D-Fender | NotJohnDavid : Polycom is for you then. Looking at PoE? |
02:44.49 | NotJohnDavid | why is polycom for me? because i'm in the US? |
02:44.53 | skraelings001 | anyone help with these piece of verbose output, http://pastebin.com/732477 |
02:45.14 | [TK]D-Fender | NotJohnDavid : Polycom Or Cisco, but unless you've got a real deal on Cisco Polycom's quality comes at a far better price point. |
02:45.18 | NotJohnDavid | what should I call you... tk.. tkd....fender? PoE is a consideration but I haven't spoken with the client. I think PoE would be the way to go |
02:45.36 | [TK]D-Fender | NotJohnDavid : Well just that Polycom is a fair bit more expensive outside North America, yes. |
02:45.46 | PakiPenguin | night everyone |
02:45.56 | NotJohnDavid | and they have XML displays ? |
02:46.19 | [TK]D-Fender | NotJohnDavid : PoE Will shape your choice of models a bit. I'd suspect IP430's fine for most workers, IP 601 for areceptionist, and 501's for anyone in between. |
02:46.36 | [TK]D-Fender | NotJohnDavid : No, only the IP 601 has XML capabilities. |
02:47.50 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
02:48.18 | TripleFFFF | i know this isnt asterisk related.. well could be. .security wise. .whats the app that checks if any file was modified since last checksum |
02:48.43 | hads|home | TripleFFFF: samhain |
02:48.52 | TripleFFFF | ? |
02:48.52 | NotJohnDavid | are they backlit ? |
02:49.06 | *** part/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
02:49.16 | chumper2342 | nope, don't hear anything either |
02:49.31 | TripleFFFF | is ther somethig for windows ? |
02:50.14 | justinu | hey paki |
02:50.31 | justinu | oh, he left |
02:50.39 | NotJohnDavid | tkd: it's a small company. basically a retail shop that does service calls. currently 3 employees. usually one person at the store answering calls, two people on the road. probably could grow to 5 employees fairly soon. |
02:51.10 | skraelings001 | justinu: can u take a look at it http://pastebin.com/732477 ? |
02:51.26 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
02:51.44 | NotJohnDavid | i've heard good things about polycom. i remember i went to their site. it'd have been nice to have a comparison chart/table |
02:51.55 | *** part/#asterisk Meaw (n=debian@213.6.131.96) |
02:51.59 | justinu | that av-bani guy did a really nice chart |
02:52.35 | FuriousGeorge | anyone using metermaid patch |
02:52.56 | chumper2342 | Strom_C: still don't hear anything when I call them |
02:53.29 | NotJohnDavid | tkd: thanks for the suggestions |
02:53.30 | [TK]D-Fender | ~phones |
02:53.31 | jbot | [phones] at http://bani.anime.net/phones/ |
02:53.59 | Strom_C | chumper2342: what about when you try a different ITSP? |
02:54.13 | NotJohnDavid | i get the impression I asked the same question a lot of people do :P |
02:54.14 | FuriousGeorge | anyone using the metermaid patch on 1.2.9.1? |
02:54.16 | [TK]D-Fender | NotJohnDavid : Polycom are great choices at a great value. Cisco is a great feel, slightly less reliable firmwar, and pricey, but some have great deals. |
02:54.26 | [TK]D-Fender | Most of the rest just don't add up to those 2. |
02:54.47 | NotJohnDavid | wish there were a local store |
02:54.51 | orlock | Hmm.. |
02:54.56 | orlock | 7960 has started locking up hard :-\ |
02:54.58 | justinu | skraelings001: what are we looking for? |
02:55.14 | chumper2342 | i called my company's main number (asterisk also) different itsp and it says connected but my call is silent |
02:55.26 | justinu | skraelings001: "called equipment is non-ISDN" |
02:55.33 | justinu | pretty self explanatory... you won't get cause codes in that case. |
02:56.19 | Strom_C | chumper2342: also SIP? |
02:56.39 | chumper2342 | yep |
02:57.22 | NotJohnDavid | you know $100 a phone is cheap... considering this client would want each phone on a different extension. FXS port is going to cost at least $60-70 right? |
02:57.50 | [TK]D-Fender | NotJohnDavid : Nope, $35/port using SPA-2002's |
02:57.53 | justinu | you can do it on the cheap if you can find a good deal on a used channelbank |
02:57.54 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-18.cybersurf.com) |
02:58.00 | Strom_C | chumper2342: try an IAX2 ITSP. I wonder if your ISP is messing with SIP traffic |
02:58.10 | justinu | used cb $300, te110p/A101u $350 |
02:58.21 | justinu | $650/24 ports |
02:58.42 | justinu | sorry, 450 for that T1 card |
02:58.45 | justinu | so 750/24 |
02:59.03 | [TK]D-Fender | justinu : Assuming it doesn't go flakey on you... and $350 for a 1 port PRI? New? |
02:59.12 | justinu | my bad |
02:59.43 | [TK]D-Fender | NotJohnDavid : For the sizes you're talking about SPA-2002 would probably be the way to go. |
02:59.57 | [TK]D-Fender | NotJohnDavid : But for larger, the CB idea can work. |
03:00.08 | justinu | spa's are great |
03:00.15 | justinu | some guys said they do not like heat tho |
03:00.31 | NotJohnDavid | tkd: ip phones just seem more flexable |
03:00.31 | *** join/#asterisk FoFiN (n=FoFiN@201.153.31.166) |
03:00.39 | chumper2342 | holy crap!, just as a test I set shorewall policy to accept all to all, and it works, something in there is blocking it |
03:00.48 | NotJohnDavid | tkd: not to mention that everywhere a phone is there'll already be a network connection. |
03:00.51 | Strom_C | I need to brush up on my polycom configuration skills |
03:00.53 | NotJohnDavid | (next to computers) |
03:00.53 | riddlebox | hrmm anyone having troubles with broadvoice? |
03:00.57 | *** join/#asterisk Eigh (n=jason@69.63.224.130) |
03:01.00 | [TK]D-Fender | NotJohnDavid : They are. It feels more natural for sure and with PoE saves on wiring. |
03:01.05 | Strom_C | chumper2342: gah, see, why didnt you mention the bloody firewall? |
03:01.41 | skraelings001 | justinu: u saw? i try to call to those non-existing phone numbers. In the first case i was able to her the message from telco but when i tryed with the other i didn't hear it. I checked E1(connecting other equipment from telco- independent of asterisk line was good and that was receiving those messages |
03:01.44 | [TK]D-Fender | NotJohnDavid : Well you'll want to aim for your phones being on their own lan if at all possible... |
03:02.07 | justinu | skraelings001: so in 2nd case, you hear nothing? |
03:02.14 | justinu | 2nd call, that is |
03:02.18 | skraelings001 | justinu: aha |
03:02.25 | Strom_C | justinu: you're in the san fernando valley, right? |
03:02.29 | justinu | Strom_C: yes |
03:02.32 | chumper2342 | lol, sorry, do you know what by chance i need to accept? i have it to accept net-fw and loc-fw 5060 |
03:02.38 | Strom_C | got an extra polycom phone you want to sell? :) |
03:02.55 | NotJohnDavid | tkd: yeah it'd be possibly but unneeded just yet |
03:03.04 | justinu | no, those all sold |
03:03.04 | Strom_C | chumper2342: you also have to open up ports 16384-32768 for RTP transport |
03:03.10 | justinu | i'm stuck with some gxp2000's :P |
03:03.13 | Strom_C | heh |
03:03.39 | [TK]D-Fender | Strom_C : Maybe :) But not at a price you'd probably find worthwhile :) |
03:03.47 | justinu | skraelings001: you may have a legit bug. |
03:03.48 | Strom_C | try me |
03:03.50 | Strom_C | ;) |
03:04.01 | justinu | skraelings001: asterisk's q931 implementation is a bit odd. |
03:04.06 | *** part/#asterisk zwelch (n=chatzill@pdpc/supporter/sustaining/zwelch) |
03:04.30 | justinu | skraelings001: what is the time delta between DISCONNECT and RELEASE? |
03:04.34 | justinu | in the last call |
03:05.02 | skraelings001 | justinu: i couldn't tell |
03:05.28 | justinu | can you refer to your full log maybe? |
03:05.33 | skraelings001 | actually is pretty quickly, i tryed with NoOp and Wait |
03:05.53 | riddlebox | can someone help me resolve this problem http://pastebin.ca/72738 |
03:06.01 | [TK]D-Fender | Strom_C : Located where? |
03:06.07 | Strom_C | Los Angeles |
03:06.17 | [TK]D-Fender | Strom_C : Shipping would suck from Montreal... |
03:06.26 | justinu | damn you canadians |
03:06.31 | justinu | and your french speaking city |
03:06.45 | justinu | fender: did you go to the F1 race? |
03:06.49 | Strom_C | hence why I asked justinu if he was near me ;) |
03:07.14 | [TK]D-Fender | justinu : Nope.. I'm anti-spectator..... |
03:07.17 | justinu | booo |
03:07.33 | justinu | actually, i wanted to ask you about flying to montreal |
03:07.34 | justinu | i want to go |
03:07.37 | justinu | any tips? |
03:08.10 | Katty | take a snack. |
03:09.19 | [TK]D-Fender | justinu : Not much to say... plan your transport to where you'll stay and the rest is gravy... |
03:09.48 | FuriousGeorge | anyone using the metermaid patch against 1.2.9,1? |
03:09.48 | NotJohnDavid | tkd: you know you'd think linksys phoens would be similar to the cisco ones |
03:09.52 | *** join/#asterisk nohope (i=1000@201-13-87-52.dsl.telesp.net.br) |
03:09.56 | [TK]D-Fender | justinu : Pretty easy actually... |
03:10.44 | [TK]D-Fender | NotJohnDavid : I owned an SPA-941. Nothing to write home about. Definately not in Pollycom's class. They shine overseas where their price point strongly defeats Polycom though. |
03:11.27 | NotJohnDavid | well i had heard wonderful thigns about the polycoms. now i just show the client and see what they want :^D with a heavy hand haha |
03:11.37 | [TK]D-Fender | And the speakphone is tinny, the handset only slightly. the pverall phone is too light. It makes horrific use of the LCD (much like th GXP-2000). |
03:12.30 | chumper2342 | Strom_C: thanks. works now |
03:12.32 | justinu | fender: any airline recommendations? air canda seems really expensive out of LAX |
03:13.02 | [TK]D-Fender | Basically in North America you are either trying to be cheap (and getting what you payed for), or going decent with Polycom. Its competitors in its price bracket aren't worth it excepts for very small cases. |
03:13.22 | NotJohnDavid | the gxp-2k seems pretty hard to read unless the backlight is on. |
03:13.26 | Brijn | justinu: See if Air Alaska flies to Montreal as well |
03:13.34 | justinu | i was thinking of flying up to vancouver |
03:13.35 | [TK]D-Fender | justinu : Travel these days is an exercise in bargain hunting. hit the internet sky auctions, etc for deals |
03:13.43 | justinu | then going to montreal from there |
03:13.51 | Brijn | justinu: Air Alaska to YVR was cheap I think |
03:13.59 | Strom_C | justinu: I'm seeing $530ish roundtrip on expedia from LAX to Montreal |
03:14.02 | [TK]D-Fender | justinu : Avoice Air Canada if you can and go with an express charter. |
03:14.20 | [TK]D-Fender | 530$sounds like a great deal. |
03:14.34 | [TK]D-Fender | (to my limited knowledge) |
03:14.36 | Strom_C | $530 on NWA, with a stopover in Detroit |
03:14.51 | justinu | bah, stopping sucks |
03:15.00 | Strom_C | leave on a Tuesday, return the following Tuesday |
03:17.38 | JackEStorm | Well, I think it's official, I've given up on Sixtel |
03:17.38 | justinu | fender: you should have gone to the race, if only to see jaques villeneuve crash out w/ 7 laps to go |
03:17.44 | justinu | laping a backmarker of all things! |
03:18.14 | [TK]D-Fender | justinu : Sorry, not a "spectator". I pplay sports, not watch. |
03:18.28 | riddlebox | is anyone else experiencing problems connecting to broadvoice? |
03:19.23 | justinu | bah, enjoy the moment with me, at least :P |
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03:20.10 | [TK]D-Fender | justinu : Yeah I can "join in", but never on my own... |
03:20.30 | *** part/#asterisk nohope (i=1000@201-13-87-52.dsl.telesp.net.br) |
03:22.46 | *** join/#asterisk albertito (n=net@host178.201-252-23.telecom.net.ar) |
03:23.55 | albertito | Hi! I'm having some problem with a zapata card, I was here earlier. ManxPower helped me and told me it was probably caused by interrupt latency |
03:24.20 | Telamon | Anyone have an opinion on the GXP-2000 1.1 series of firmware? Is it fairly stable or should I stick with 1.0.2.13? |
03:24.24 | justinu | is it sharing interrupts? |
03:24.53 | albertito | As this surprised me, I decided to measure it: I'm using Ingo Molnar latest -rt patch, which includes an "interrupt off latency meter" |
03:25.23 | FuriousGeorge | anyone got any idea what this means: chan_sip.c:9962 handle_response: Notify answer on an owned channel? |
03:25.44 | FuriousGeorge | i suspect that has something to do with why my presence isnt working with this patch and parking |
03:25.44 | albertito | It reports almost no delay in serving irqs, the max in both CPUs (it's a dual core) is 19 and 20, and average is 0 |
03:26.18 | justinu | albertito: cat /proc/interrupts |
03:26.28 | albertito | justinu: it's not sharing an interrupt |
03:26.42 | albertito | justinu: (sorry I didn't replied, I wasn't sure you were talking to me) |
03:27.22 | justinu | so what's the issue you're having? |
03:27.25 | CunningPike | albertito: What does zttest show? |
03:27.43 | justinu | albertito: try switching kernels to non-SMP? |
03:27.59 | albertito | justinu: I'm getting a lot of "PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1" |
03:28.05 | albertito | CunningPike: hold on |
03:28.16 | justinu | albertito: ok, that's bad |
03:28.36 | justinu | otoh, your circuit could be taking errors |
03:28.36 | CunningPike | albertito: Ah - you may have a timing problem on your PRI |
03:28.41 | justinu | or timing |
03:28.54 | riddlebox | can someone tell me what all this means? http://pastebin.ca/72759 |
03:28.59 | CunningPike | albertito: pastebin your zaptel.conf |
03:29.03 | albertito | CunningPike: lot's of 99.987793% and some 100% |
03:29.40 | CunningPike | albertito: That sounds reasonable - it doesn't appear to be an interrupt problem - are you running hyperthreading or anything like that? |
03:29.45 | albertito | CunningPike: (this is a debug kernel and the rt measuring it's supposed to introduce some overhead) |
03:29.54 | albertito | CunningPike: no, but this is a dual core box |
03:30.05 | justinu | albertito: i would advice trying the mainline kernel |
03:30.05 | albertito | justinu: I'll compile one non-smp in a second |
03:30.12 | justinu | and also try non-SMP, just for fun |
03:30.13 | CunningPike | albertito: That's OK - ours are too, but we have noht set |
03:30.17 | albertito | justinu: I run the mainline kernel all the time |
03:30.31 | justinu | so you had problems on both mainline and -rt? |
03:30.49 | albertito | justinu: I compiled this one because somebody earlier said it was possibly an IRQ latency problem, and I wanted to measure it just to be sure |
03:31.12 | Telamon | albertito: Just an FYI, I've had bad luck with any asterisk feature that relies on PRI timing with SMP based systems. |
03:31.40 | albertito | justinu: I have the same problem with mainline with HZ=250 and no preempt (but low latency), HZ=1000 and preempt, and rt with and without the debug code (both HZ=1000 and preempt) |
03:31.55 | justinu | yeah, it's something else |
03:31.58 | albertito | CunningPike: this CPU doesn't have HT |
03:32.22 | albertito | CunningPike: one second and I'll pastebin the ztcfgf |
03:32.30 | CunningPike | albertito: OK |
03:32.34 | riddlebox | can someone tell me what all this means? http://pastebin.ca/72759 |
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03:36.15 | albertito | CunningPike: http://pastebin.ca/72764 |
03:36.47 | CunningPike | albertito: So, running E1? |
03:36.51 | albertito | CunningPike: the other end of the cable is a smaller box (Pentium II) with identical config, but span = 1,0,0,ccs,hdb3,crc4 (difference is the "0" in the second column) |
03:36.56 | albertito | CunningPike: yes |
03:37.12 | albertito | CunningPike: it's a TE110P in E1 mode (I'm in Argentina) |
03:37.18 | justinu | back to back with another * box, obviously |
03:37.32 | albertito | justinu: yes, it's a testbed |
03:37.34 | justinu | your config looks good |
03:37.45 | CunningPike | albertito: OK - that's all good - and it's a crossover cable? |
03:37.56 | CunningPike | albertito: Any d-channel errors? |
03:38.14 | albertito | CunningPike: yes. The errors only show on the big box, so I switched ends and they keep appearing on the big box |
03:38.37 | albertito | CunningPike: not that I can see, but hold on, I'll grep the logs just in case I missed it |
03:39.15 | CunningPike | albertito: Ah - that's instructive. I would do what justinu suggested - switch to a non-SMP kernel and see if that makes it go away |
03:39.33 | skraelings001 | albertito: no deber?as usar crc4 |
03:39.41 | CunningPike | albertito: Failing that, I wonder is it a motherboard compatibility issue |
03:39.51 | justinu | i agree with CunningPike |
03:39.55 | justinu | if you can't solve it, change mobos |
03:40.07 | justinu | or slots |
03:40.31 | albertito | skraelings001: why not? Anyway, while I can get away without using it now, I'll probably have to use it eventually since it's going to be connected to exchanges that use it |
03:40.33 | CunningPike | albertito: It's the simplest of setups, so it should work no problem. I assume you tried another cable...... |
03:41.10 | CunningPike | albertito: I don't mean to be patronizing, but your English is excellent |
03:41.15 | albertito | CunningPike, justinu: I'll build an UP kernel. Any suggestions regarding which one? (mainline, -rt, preempt/no preempt, etc.) |
03:41.21 | justinu | mainline |
03:41.31 | albertito | CunningPike: thanks =) |
03:41.32 | justinu | vanilla, or your distro's patches |
03:41.40 | riddlebox | can someone tell me what all this means? http://pastebin.ca/72759 |
03:41.43 | CunningPike | albertito: Yes, vanilla |
03:42.21 | c | ingo molnar's branch |
03:42.24 | *** part/#asterisk c (i=ix@c-24-60-193-83.hsd1.ma.comcast.net) |
03:42.34 | CunningPike | riddlebox: No - if someone here could, they would - no need to keep asking. You're probably better off posting to the mailing list or trying again in ~12 hours when more people are awake |
03:42.41 | skraelings001 | albertito: i got the same board and configured without crc4, mainly because of compatibility with telco. but i think i also read it somewhere |
03:43.34 | albertito | Ok, I'll be back in a minute with the new kernel |
03:45.39 | albertito | skraelings001: that's strange... if you find the place where you read it, please let me know. I run with crc4 because that's a requirement for the exchanges I use |
03:46.08 | skraelings001 | sure |
03:49.25 | skraelings001 | albertito: hey sorry, it's optional ..i didn't put it cause of telco. |
03:50.11 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
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03:51.39 | mrdigital | can recommend parts for my asterisk box? |
03:52.29 | justinu | a cpu, some ram, possibly a pstn interface board |
03:52.37 | justinu | hard drive might be nice |
03:52.38 | mrdigital | yeah like whats good enough |
03:52.59 | justinu | anything made in the last 2-3 years will do splendidly |
03:53.06 | Greek-Boy | do macros only work with an s extension? |
03:53.13 | justinu | Greek-Boy: yes |
03:53.18 | mrdigital | justinu: what FXO board do you recommend? |
03:53.35 | justinu | mrdigital: either the TDM400p or the SPA-3000 |
03:53.40 | mrdigital | SPA? |
03:53.42 | mrdigital | err |
03:53.45 | justinu | sipura spa-3000 |
03:53.45 | mrdigital | SPA-3000 hmmmm |
03:54.00 | justinu | sangoma A200 card possibly as well |
03:54.04 | [TK]D-Fender | mrdigital : depends how many lines, if you want to use SpanDSP, etc. |
03:54.09 | mrdigital | 1 line, |
03:54.11 | mrdigital | 1 extenstion |
03:54.19 | [TK]D-Fender | And yes, the A200 is a great contender |
03:54.31 | Greek-Boy | i've got like 100 extensions which have the same priority list. if i change something i have to change it for all??? i thought macro would help me with that? |
03:54.32 | [TK]D-Fender | mrdigital : then the SPA-3102 is for you. |
03:54.32 | justinu | each has its pros/cons |
03:54.39 | *** join/#asterisk viking78 (n=chatzill@66-168-98-144.static.jcsn.tn.charter.com) |
03:54.41 | justinu | digium is very well supported here |
03:55.02 | justinu | thats my chief reason for recommending it to newbies |
03:55.11 | mrdigital | [TK]D-Fender: can you est a price range? |
03:55.24 | [TK]D-Fender | $90 |
03:55.32 | justinu | i think the 4 line capable PCI boards run north of 350USD |
03:55.44 | Qwell | justinu: just at, I believe |
03:55.52 | Qwell | depending on fxo vs fxs configs |
03:55.55 | skraelings001 | [TK]D-Fender: i have some PRI problems ,can you take a look at http://pastebin.com/732477 ? |
03:56.07 | mrdigital | justinu p3 500mhz with 256mb and a 20gig hdd sound decent? |
03:56.13 | justinu | yeah, just fine. |
03:56.57 | CunningPike | Greek-Boy: If an extension matches your pattern, it will follow the dialplan for that match |
03:57.03 | mrdigital | how do i connect the 3102 to asterisk? |
03:57.09 | justinu | ethernet |
03:57.24 | [TK]D-Fender | mrdigital : Its a SIP device... just networked to it somehow |
03:57.25 | *** join/#asterisk viking78 (n=aherbert@66-168-98-144.static.jcsn.tn.charter.com) |
03:59.06 | Greek-Boy | CunningPike, so it can only do patterns and s extensions? |
04:00.11 | CunningPike | skraelings001: In your first case, the dialed equipment is non-ISDN. In the second, "Cause: Unallocated (unassigned) number " |
04:00.34 | [TK]D-Fender | skraelings001 : No idea |
04:00.55 | CunningPike | Greek-Boy: Yes - that's enough for most applications - what are you trying to accomplish? |
04:01.26 | mrdigital | so with the 3102 or 3000 i can use a pstn line with a regular analog phone? |
04:02.08 | CunningPike | mrdigital: Yes - it has an FXO and an FXS port. |
04:02.13 | mrdigital | Awesome |
04:02.31 | skraelings001 | CunningPike: I know both transmit messages from telco (this number does not exist) but i'm only capable to hear the first one |
04:02.39 | CunningPike | mrdigital: It provides direct-to-PSTN failover in the event of a network or power failure |
04:02.55 | mrdigital | Awesome so if asterisjk system crashes |
04:03.04 | Greek-Boy | CunningPike, not trying to accomplish much. Just trying to avoid changing a priority list a 100 times (ie, for every phone) |
04:03.04 | mrdigital | ... phone is still usable without reconnecting stuff |
04:03.14 | [TK]D-Fender | mrdigital : yup |
04:03.20 | justinu | that's one plus of the 3000, but have never tried that failsafe myself |
04:03.21 | [TK]D-Fender | mrdigital : useful in so many ways.... |
04:03.24 | CunningPike | mrdigital: Exactly - we use them for 911 - with a red phone plugged into each one |
04:03.34 | [TK]D-Fender | bbiab |
04:03.35 | justinu | the batphone! |
04:03.37 | mrdigital | CunningPike: explain your setup |
04:03.44 | justinu | CunningPike: explain yourself! |
04:03.53 | CunningPike | justinu: lol |
04:03.53 | justinu | what is your purpose? |
04:04.04 | CunningPike | mrdigital: The whole thing? |
04:04.25 | justinu | no one does |
04:04.33 | mrdigital | sur |
04:04.34 | mrdigital | sure |
04:06.04 | CunningPike | mrdigital: Well, at 50,000ft level, we have redundant asterisk servers with a TE410P in each connected to the PSTN via a PRI and to our legacy Nortel via 2 PRIs. We have Polycom phones, and use SPA-3000 for other things like 911, elevator phone and other stuff like that. Any questions? |
04:07.12 | file | yes, what's the meaning of life? |
04:07.36 | justinu | 42 |
04:07.40 | justinu | that answer is well known |
04:08.00 | CunningPike | ~meaningoflife |
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04:08.24 | CunningPike | jbot! You should be ashamed of yourself |
04:08.27 | mrdigital | cool setup CunningPike |
04:08.32 | mrdigital | did you do it yourself? |
04:08.39 | CunningPike | jbot, meaningoflife is 42 |
04:08.41 | jbot | CunningPike: okay |
04:08.46 | CunningPike | sheesh |
04:08.50 | justinu | i have a pretty neat pure SIP setup |
04:09.00 | justinu | when my "coworkers" aren't fuicking it up |
04:09.01 | CunningPike | justinu has SIP envy |
04:09.05 | CunningPike | hee hee |
04:09.19 | justinu | i used to run TDM switches |
04:09.23 | file | SIP... envy... I never thought I'd hear those two words used in the same sentence |
04:09.24 | justinu | we had 160 T1s |
04:09.31 | justinu | i'm so glad to be done with that crap |
04:09.59 | CunningPike | mrdigital: Thanks - yes - it's home grown. We're pretty pleased with it so far. Our top guy has moved onto it now and loves it |
04:10.21 | CunningPike | mrdigital: He also loves the money it saves us |
04:10.52 | CunningPike | justinu: Yes - our telecom guy is just waiting to push our Nortel over a cliff |
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04:11.22 | justinu | PRI works better on asterisk than it did on a $150k switch |
04:11.51 | justinu | dunno if asterisk can do 20 span NFAS groups tho |
04:12.10 | CunningPike | justinu: Not yet ;) |
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04:15.07 | justinu | someone just told me the 1337 color screen cisco phone is SCCP only |
04:15.10 | justinu | true or false? |
04:16.06 | albertito | CunningPike, justinu: I'm using vanilla UP, and it didn't go away |
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04:16.15 | justinu | albertito: :( |
04:16.36 | CunningPike | albertito: Bummer - I can't recall if you told me you'd tried another cable...... |
04:16.39 | albertito | CunningPike, justinu: I'll try to get a different cable tomorrow (this one worked just fine yesterday, but on a different setup...) |
04:16.53 | justinu | chances of it being cable are slim |
04:17.01 | justinu | unless your crimps suck that much |
04:17.28 | albertito | CunningPike: I didn't because I don't have any at hand (I'm at home), but I'll get one tomorrow |
04:17.43 | justinu | albertito: what about different slots? |
04:17.51 | CunningPike | albertito: Maybe skraelings001 was right........ could you try esf,b8zs or whatever it is? |
04:18.09 | albertito | justinu: yeah, or maybe swap the cards to rule out connector problems |
04:18.32 | CunningPike | albertito: I know you need crc4 for your telco, but it might indicate something |
04:18.45 | albertito | CunningPike: sure, it's worth the try |
04:19.24 | justinu | esf is for T1 only |
04:19.37 | skraelings001 | albertito: CunningPike: justinu: not all of telco work with crc4 |
04:19.39 | justinu | as is sf/f4 |
04:19.46 | justinu | s/f4/d4/ |
04:19.48 | CunningPike | justinu: Ah - I didn't know that |
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04:21.03 | albertito | I do get the errors, although less. I maybe not having the crc4 is making some of the errors go unnoticed |
04:22.05 | albertito | hhhmmmmmmm now I got a couple of errors on the other box too (although just 3 of them, quite apart) |
04:22.22 | justinu | albertito: since your testbed is back to back, both sides should use the same line-coding |
04:22.38 | albertito | justinu: yeah, they're both without crc4 |
04:22.56 | skraelings001 | albertito: lately i have realized i was getting almost the same but with indication 6, but they are too few |
04:23.09 | albertito | (s/I maybe/Maybe/ up there) |
04:24.06 | CunningPike | albertito: What is the motherboard anyway? |
04:25.12 | albertito | CunningPike: it's ccs,hdb3 with optional crc4 for E1 PRI, and (IIRC) cas,hdb3 for E1 R2. The "ccs" stands for "common channel signalling" and "cas" is "channel associated signalling". I don't remember what's hdb3 anymore :S |
04:25.35 | albertito | CunningPike: Intel D945G, with a Pentium D 830 |
04:26.20 | CunningPike | albertito: Hmmm - that should work. I don't think any Intel motherboards give many compatibility issues, do they? |
04:26.42 | Qwell | justinu: 7970? false |
04:26.57 | Qwell | 7985 however, I believe is sccp only |
04:27.08 | justinu | cool, thx |
04:27.10 | albertito | CunningPike: I've always used Intel motherboards with digium cards and never had an incompatibility problem before... but it's the first one I try this model |
04:27.39 | albertito | CunningPike: I'll swap the cards as justinu suggested, to rule out connector and slot problems. I'll be back in a minute |
04:27.50 | CunningPike | albertito: OK |
04:28.17 | skraelings001 | night fellows |
04:28.54 | *** part/#asterisk skraelings001 (n=skraelin@201.230.111.182) |
04:29.34 | justinu | albertito: this stuff crops up once and a while, strangely enough it's always affecting you south americans and E1s |
04:29.43 | justinu | (mobo incompatibility) |
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04:34.35 | w0rmzw3rth | What can I use for a Cheap Backbone for a WISP of a 50 mile radius for about 100 customers |
04:37.45 | Qwell | w0rmzw3rth: umm...that's a bit...out of scope |
04:40.08 | w0rmzw3rth | would a tier 1 cut it? |
04:41.13 | inv_Arp | Qwell: heh awol would prob know... |
04:41.54 | Qwell | inv_Arp: indeed, but it's quite complex :p |
04:43.39 | FuriousGeorge | someone tell me if the fhis sounds feasable: |
04:44.43 | FuriousGeorge | i wanna set a button on my snom to call an extension and set a db_variable PAGE_FLAG for each user that, upon the next call, it setssipheader to page the phone |
04:45.08 | Qwell | sure |
04:45.09 | FuriousGeorge | so if they call another snom after setting page_flag (by hitting the corresponding button) it pages that phone |
04:45.44 | FuriousGeorge | Qwell: based on the CID i check for the flag, then after hangup i reset it? |
04:45.49 | Qwell | sure |
04:46.01 | FuriousGeorge | Qwell: thanks |
04:46.08 | Qwell | see above |
04:46.33 | FuriousGeorge | Qwell: how far up? |
04:47.07 | FuriousGeorge | Qwell: what you mean "see above" |
04:47.13 | CunningPike | FuriousGeorge: I think Qwell is humoring you |
04:47.13 | FuriousGeorge | are you just joshing me? |
04:47.34 | CunningPike | Qwell: Is actually just a 'sure' bot |
04:47.38 | FuriousGeorge | Qwell: now i dont know if you were serious about the feasability of my proposal |
04:47.44 | Qwell | CunningPike: something like that |
04:48.33 | Qwell | FuriousGeorge: You and me both |
04:49.33 | FuriousGeorge | Qwell: but you in ernest think ill be able to set a page button as described above on my phone? |
04:49.42 | Qwell | ...sure |
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05:03.14 | albertito | CunningPike, justinu: I swapped the cards between boxes and the problem swapped with them |
05:03.34 | justinu | odd |
05:03.39 | CunningPike | albertito: Interesting - could be a dodgy card then |
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05:03.48 | CunningPike | albertito: Got another one? :) |
05:03.52 | albertito | So I used a third card I had and it has the same problem, but looks worse (more errors per second) |
05:04.02 | justinu | very odd |
05:04.06 | CunningPike | albertito: Are all 3 cards the same type? |
05:04.25 | justinu | do you have a BER tester? |
05:04.27 | drray | I need a better phone than a cisco 7960, for forwarding calls, I have a site that can't forward calls.. any suggestions? |
05:04.44 | albertito | Yes, all 3 are exactly the same TE110p model, and were ordered together about a year ago, but weren't tested until now |
05:05.10 | albertito | justinu: I don't know what that is... is that a regular electric tester? |
05:05.23 | justinu | no, a bit error rate tester |
05:05.34 | justinu | the ones I use are Sunrise Telecom Sunset T1s |
05:05.43 | albertito | justinu: I'm afraid not, I don't have one around |
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05:06.20 | justinu | albertito: i'm out of ideas, very strange indeed |
05:06.24 | albertito | justinu: what could I use it for? |
05:06.31 | justinu | to test the framer on the digium cards |
05:07.04 | justinu | anyways, bedtime |
05:07.12 | albertito | justinu: maybe they're just faulty... or have some connector problem |
05:07.18 | *** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
05:07.19 | albertito | justinu: thanks a lot for all your help! sleep well |
05:07.56 | CunningPike | albertito: I'm out of ideas, too. |
05:08.00 | CunningPike | albertito: Sorry |
05:08.08 | CunningPike | albertito: Where did you get the cards> |
05:08.09 | CunningPike | ? |
05:08.17 | albertito | CunningPike: please, you've given me invaluable help! |
05:08.27 | albertito | CunningPike: Imported from Digium USA |
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05:08.44 | CunningPike | albertito: I would phone Digium in the morning........ |
05:09.05 | CunningPike | albertito: I think you have a couple of bad cards - not very usual, but it sure sounds like it |
05:09.13 | albertito | CunningPike: I will, and I'll retry both cards just in case |
05:09.20 | CunningPike | albertito: OK - good luck! |
05:09.38 | albertito | CunningPike: thanks! I'm off to bed too |
05:09.55 | CunningPike | albertito: OK - good night |
05:09.59 | albertito | (after rebooting to get back my SMP kernel =) |
05:10.01 | albertito | night! |
05:10.03 | CunningPike | Buenos nochas |
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05:29.00 | stephane_ | jour |
05:33.11 | CunningPike | nuit |
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05:46.34 | FuriousGeorge | is there any way nat can mess with presence indications on chan sip and chan local? i just installed the metermaid patch on 12.9.1 and the LED indications for parking spots work flakely, and ringing indications are not working |
05:46.49 | FuriousGeorge | take that back, its working for ringing |
05:47.35 | FuriousGeorge | its the presence on parking patch that i installed (metermaid) that is flakey. could it be a nat thing, and i can expect it to work ok from inside the building |
05:47.36 | FuriousGeorge | ? |
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05:57.22 | X-Rob_ | yes it could be a nat thing. |
05:57.40 | X-Rob_ | _could_ |
05:57.44 | X-Rob_ | dunno if it is tho 8) |
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06:03.05 | X-Rob_ | Woo. |
06:03.12 | X-Rob_ | Call pickup works on the GXP2000's too |
06:06.37 | *** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
06:07.43 | [hC] | X-Rob_: you mean like, *8? |
06:07.54 | X-Rob_ | as in BLF-push-the-flashy-light pickup |
06:08.10 | [hC] | ahh |
06:08.16 | P-NuT | Hey all, I need some help with SIP trunking. As in, how to I build one.. |
06:08.28 | [hC] | on like, a parked call? |
06:08.31 | [hC] | what are you picking it up from/ |
06:09.07 | X-Rob_ | no, like xtn 345 is ringing |
06:09.11 | X-Rob_ | the light for xtn 345 flashes |
06:09.17 | [hC] | ohhh i see. :) |
06:09.17 | X-Rob_ | you push the button next to the light |
06:09.20 | X-Rob_ | and you pick up the call |
06:09.32 | [hC] | gotcha |
06:10.12 | X-Rob_ | omfg |
06:10.17 | X-Rob_ | the ligths just went berko |
06:10.21 | X-Rob_ | they've got a ring-all group |
06:10.27 | X-Rob_ | watching all the lights flash on the gxp is most amusing |
06:10.27 | X-Rob_ | 8) |
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06:15.29 | smackus | i am stuck trying to get cdr_mysql working on my new asterisk server. |
06:15.46 | smackus | i have edited the configs, installed asterisk-addons |
06:15.48 | smackus | can anyone help me? |
06:17.55 | smackus | has everyone gone to be? |
06:17.57 | smackus | bed? |
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06:23.01 | smackus | i have run the asterisk-addons install, and expected to see cdr_addon_mysql.so. any idea as to how to resolve this? |
06:26.50 | qdk | smackus: recompiile asterisk? |
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06:28.14 | smackus | qdk: tried that |
06:28.34 | smackus | what is the link to the svn version of the asterisk-addons. |
06:28.36 | smackus | i will try that |
06:28.53 | qdk | smackus: and you have mysql and its libs installed? |
06:29.42 | smackus | i think so |
06:29.44 | qdk | smackus: svn shouldnt give you any better results. |
06:29.44 | smackus | mysql for sure. |
06:29.53 | smackus | how can i tell for sure that I have all of the libs |
06:29.54 | qdk | smackus: ok, dist? |
06:29.54 | smackus | ? |
06:30.01 | smackus | red hat ent |
06:31.29 | qdk | sad linux, anyway, the libs might be in a seperate packages as most other, and propper, packagessystems have chosen to do. |
06:33.54 | smackus | hmm ok |
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06:35.31 | smackus | as far as the mysql packages that i have installed... |
06:35.56 | smackus | mysql-4.1.20-1.RHEL4.1 |
06:36.14 | smackus | mysql-devel-4.1.20-1.RHEL4.1 |
06:36.42 | smackus | mysqlclient10-3.23.58-4.RHEL4.1 |
06:36.54 | smackus | mysql-server-4.1.20-1.RHEL4.1 |
06:37.07 | smackus | php-mysql-4.3.9-3.12 |
06:37.18 | smackus | am i missing anything important to cdr_mysql? |
06:37.29 | rob0 | The -devel one is probably it. |
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06:37.43 | smackus | the one i am missing, or the one i need? |
06:38.41 | rob0 | The one you need. Were there errors in the "make"? |
06:39.10 | smackus | make of the asterisk-addons? |
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06:39.14 | smackus | i did not notice any |
06:39.33 | smackus | i have mysql-devel installed. |
06:40.22 | qdk | smackus: perhaps -lib or something. |
06:40.43 | rob0 | and "make install" installed it? |
06:40.44 | qdk | smackus: anyway, RPM-linux is hell, so good luck. |
06:41.27 | smackus | qdk:thanks |
06:41.46 | smackus | rob0: is there a way to tell that it is for sure installed? |
06:42.01 | qdk | smackus: rpm -qa? |
06:42.28 | tzafrir | rpm -q packagename |
06:43.14 | tzafrir | Are those the rpm packages from mysql.com? |
06:43.56 | smackus | i was asking is there a way to know for sure if the asterisk-addons was installed |
06:44.14 | tzafrir | mysqlclient10 is obsolete. Unless you need to communicate with an old mysql 3.23 server |
06:44.26 | rob0 | Look in the Makefile, see where the file should have been copied, see if it's there. |
06:45.18 | rob0 | configure it, try to use it :) |
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06:53.26 | locelavi | hi evrybody |
06:53.47 | smackus | ok, so i am starting from scratch |
06:54.02 | locelavi | just a little question... |
06:54.06 | locelavi | ...of dead |
06:54.15 | locelavi | and with iptables |
06:54.20 | smackus | i run "perl -MCPAN -e "install DBD::mysql"" |
06:54.42 | smackus | I get an error... make test had returned bad status, won't install without force |
06:54.49 | locelavi | how can i configure it with sip ? |
06:55.25 | locelavi | i can open 5060 all right |
06:55.27 | smackus | "/usr/bin/make" was ok |
06:55.29 | [hC] | so, any of you guys got hudlite going on a non trixbox pbx? |
06:55.41 | smackus | is there a way to skip "make test"? |
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06:55.59 | locelavi | the phon s ringinig |
06:56.11 | codestr0m | someone willing or know how to help me debug chan_ss7 |
06:56.34 | locelavi | but can't ear somthing |
06:57.15 | locelavi | i don't want to open a rank of port |
07:00.03 | Strom_C | locelavi: the only way to do SIP through a firewall is to either (a) restrict the media ports every single SIP device will ever use (and that can be hairy) or (b) open up ports 16384-32768 |
07:03.05 | locelavi | i thought that when the phones were conected the datagrams didn't pass throught the server ? |
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07:04.10 | Strom_C | locelavi: that can be the case, but you can also set up Asterisk to not allow reinvites |
07:04.26 | Strom_C | also, if asterisk has to do transcoding or needs to stay in the media path for any reason, it typically will |
07:04.38 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
07:04.43 | Strom_C | and usually you want asterisk to stay in the media path for proper call detail record generation |
07:04.52 | codestr0m | Strom_C: Have you ever poked with chan_ss7? |
07:05.10 | Strom_C | codestr0m: no, I havent...though I don't see why you'd be having /audio/ issues, since SS7 is just signaling |
07:05.13 | locelavi | i don t understand "poked". i m french |
07:05.15 | kmilitzer | codestr0m: I am using chan_ss7 ... |
07:05.54 | kmilitzer | codestr0m: But I just came in, so I don't know what your problem is ;) |
07:05.58 | codestr0m | <PROTECTED> |
07:06.13 | Strom_C | locelavi: babelfish tells me "poke" translates to french as "poussé" |
07:06.34 | *** part/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
07:06.51 | locelavi | ok sorry |
07:07.32 | kmilitzer | codestr0m: If I recall right not even developers have a good explanation why this happens from time to time ... |
07:07.35 | locelavi | can i really open a lot of port on Internet , |
07:07.44 | locelavi | i think it s no serious |
07:08.05 | kmilitzer | codestr0m: Can you hear these glitches in your calls, or do you just get the messages? |
07:08.17 | codestr0m | kmilitzer: what versions are you using and is it working in production pretty well for you.. this is my first round trying to debug chan_ss7 |
07:09.14 | kmilitzer | codestr0m: I am using the latest version of chan_ss7. It's working now for a small group of users since early this year (starting with chan_ss7-0.2) |
07:09.44 | kmilitzer | codestr0m: I am going to put more users on it in the next month ... |
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07:11.37 | codestr0m | kmilitzer: I'd really test that before going full on with it... |
07:11.50 | codestr0m | and are you using digium? |
07:11.55 | Strom_C | why are you going with SS7, kmilitzer? |
07:12.07 | kmilitzer | codestr0m: Yes, I am using digium TE205P cards |
07:12.08 | smackus | ok, so is it just me, or has anyone else using redhat enterprise had issues getting asterisk-addons installed. |
07:12.24 | smackus | i have been at this for a couple of days now, and it looks like i cant even get the basics to install |
07:12.38 | kmilitzer | Strom_C: Becase there is no other way to get different numbers on one E1-trunk ... |
07:12.52 | Strom_C | kmilitzer: what do you mean "different numbers" |
07:12.55 | smackus | i have tried to install the "perl -MCPAN -e "install DBD::mysql"" wont go |
07:12.56 | kmilitzer | Strom_C: So if I would stick with an E1, I would not be able to port numbers |
07:13.09 | smackus | tried to connect to the cvs.digium.com |
07:13.14 | smackus | unknown host. |
07:13.17 | smackus | i am stumped |
07:13.20 | smackus | any help? |
07:13.21 | kmilitzer | smackus: cvs is offline |
07:13.22 | Strom_C | smackus: uh, cvs has been deprecated forever |
07:13.25 | kmilitzer | smackus: Use svn |
07:13.26 | Strom_C | use svn |
07:13.35 | Strom_C | echo |
07:13.42 | smackus | ok... mystery solved on cvs |
07:13.52 | Strom_C | kmilitzer: what do you mean "different numbers" |
07:13.54 | smackus | I am just going off of http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons |
07:14.20 | smackus | any idea as to how to get past the perl module for DBD_mysql? |
07:14.24 | kmilitzer | Strom_C: As I said. Ported number, different blocks of DIDs (for example 01234-5678xxx and 012345-89012xxxx) |
07:14.33 | codestr0m | kmilitzer: any ideas on trying to troubleshoot this? |
07:14.40 | Strom_C | kmilitzer: what, and your telephone company won't do PRI? |
07:15.33 | kmilitzer | Strom_C: My TelCo (which is acutally our mother company ;) ) cannot do this on an E1, only on DID block works on E1s |
07:15.59 | Strom_C | that's pretty dumb |
07:16.04 | smackus | i had all of this working just fine on FC5, but digium recommends using RHEL4, so I moved over to it. now i have had issues getting stuff installed. |
07:16.25 | kmilitzer | kmilitzer: The audio-problem? No idea. There were some rumors on the asterisk-ss7 mailinglist, that it might only happen when using SIP and/or IAX and you have a slight packet loss |
07:16.41 | smackus | all i want is to install asterisk-addons (crying) |
07:16.53 | kmilitzer | Strom_C: I thought that this was a problem with every TelCo (at least in germany) |
07:17.13 | Strom_C | kmilitzer: no, in theory, multiple noncontiguous DIDs should be deliverable over PRI |
07:17.33 | Strom_C | kmilitzer: I know I've done installs here in the U.S. where the telco just assigns DIDs all over the place |
07:17.40 | kmilitzer | Strom_C: And what about ported numbers? |
07:17.48 | Strom_C | ported numbers work just fine |
07:18.00 | kmilitzer | Strom_C: Maybe it's only a problem in europe ... or just with the switch of the telco ... |
07:18.04 | nounoursfr | morning all |
07:18.16 | Strom_C | either that or the telco is lazy and doesnt want to learn to provision their stuff correctly |
07:19.03 | kmilitzer | Strom_C: Nevertheless, SS7 scales better, when adding voice slots, because you need only one slot for signaling ... |
07:19.23 | codestr0m | kmilitzer: are you using digium hardware? |
07:19.24 | kmilitzer | Strom_C: Failover works better and so on ... |
07:19.26 | Strom_C | kmilitzer: and with PRI you can have one D-channel for multiple E1s |
07:19.38 | kmilitzer | codestr0m: Yes, I do ... |
07:20.14 | kmilitzer | Strom_C: AFAIK you have for every E1 on signaling channel, that cannot be used by other E1s. |
07:20.26 | codestr0m | kmilitzer: would you maybe be so kind as to post your ss7.conf to pastebin? |
07:20.27 | kmilitzer | s/on/one/ |
07:20.44 | Strom_C | there's a framing channel on all E1s, sure, but thats the case regardless of whether its channelized or PRI |
07:20.48 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-156.lsanca.fios.verizon.net) |
07:21.01 | Gamercjm | Whats the best way to use fax with asterisk? |
07:21.05 | Strom_C | with PRI, say you have 30 B-channels and 1 D-channel on the first E1 |
07:21.25 | Strom_C | you can have that same D-channel handle signaling for 31 B-channels on a second E1 |
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07:21.29 | Strom_C | and on a third |
07:21.30 | Strom_C | and so on |
07:21.47 | Strom_C | Gamercjm: keep it all-TDM |
07:21.52 | kmilitzer | Strom_C: Are you sure? I think this is only possible for T1 and not for E1 |
07:22.20 | Strom_C | kmilitzer: I'm not completely sure, but I believe ITU-T PRI specs allow for it |
07:22.27 | kmilitzer | The only way to use one D-channel for more than one E1 (read: two E1s) is V5.2 |
07:22.44 | Gamercjm | Storm_C: whats TDM? |
07:22.51 | Strom_C | time division multiplexing |
07:23.27 | Strom_C | Gamercjm: don't use fax over voip |
07:23.33 | Strom_C | keep it all time-division |
07:24.19 | Gamercjm | its more of to just learn about asterisk/fax, so doesnt need to be as reliable or anything like that |
07:24.23 | Strom_C | kmilitzer: ah, what do I know. I'm just some idiot American telecom consultant ;) |
07:24.33 | Gamercjm | so i was wondering what the best fax method or program would be |
07:25.20 | kmilitzer | Strom_C: The real differences between the old and the new world can only be seen at a second look, I guess ;) |
07:26.16 | Strom_C | kmilitzer: I'm actually extremely surprised that the telco provisioned SS7 links to your premises |
07:26.24 | Strom_C | nothing like that would ever happen her |
07:26.26 | Strom_C | here |
07:27.17 | kmilitzer | Strom_C: As I said, it is our mother company, so technically it's an "internal" deal ... I don't think it would have been easy to get SS7 as a common customer ;) |
07:27.49 | codestr0m | http://pastebin.ca/72921 That's a sample of what I'm seeing.. I don't know if it's SS7 causing this or maybe something else.. does it give more information? |
07:29.26 | kmilitzer | codestr0m: Post more info about your system, like CPU, shared interrupts, etc. Are you using echo cancelation? |
07:29.49 | codestr0m | kmilitzer: the interrupts looked okej to me.. one sec though |
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07:36.29 | raidenz | Hello |
07:37.05 | codestr0m | kmilitzer: http://pastebin.ca/72927 (I just started working on this box today and tell me if anything seems funny there..) hmm... |
07:37.21 | raidenz | Does anyknow how to get the SIP Response message back from a failed dial. I want to see what the provider has sent back to me [running Latest Asterisk SVN] |
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07:40.28 | kmilitzer | codestr0m: Your config/system setting looks good to me, nothing strange to see. |
07:40.52 | kmilitzer | codestr0m: How do you get your calls to the ss7-system? Via SIP? |
07:41.27 | codestr0m | it's coming in on zap |
07:41.56 | codestr0m | toll free is delievered via zap and then going out over ss7 |
07:42.31 | kmilitzer | codestr0m: Hmm, then the audio-losses cannot be related to packet losses ... |
07:42.57 | kmilitzer | codestr0m: Try to contact sifira.dk (developer of chan_ss7), maybe they have an ide |
07:43.03 | kmilitzer | s/ide/idea/ |
07:43.41 | codestr0m | kmilitzer: is there an irc where sifira.dk dev(s) hang out? |
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07:44.22 | kmilitzer | codestr0m: No, I fear not :( try chan_ss7@sifira.dk, or better look it up at their website |
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07:56.26 | smackus | how can i change the location where mixmonitor stores the recorded calls? |
07:56.45 | Strom_C | thats an argument to mixmonitor, IIRC |
07:56.56 | Strom_C | show application mixmonitor |
07:57.45 | smackus | thank you |
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08:21.52 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
08:22.51 | tengulre11 | HI,all |
08:23.53 | *** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-89.perspektivbredband.net) |
08:28.44 | *** join/#asterisk h3x (i=hex@ip70-189-236-254.lv.lv.cox.net) |
08:28.48 | h3x | asterisk cdrs are so useless |
08:29.01 | Strom_C | how so? |
08:29.07 | h3x | the call durations are all off |
08:29.15 | h3x | how can you have a 0 second ANSWERED call |
08:29.42 | Strom_C | a call where the delta between the answer and hangup messages is less than one second? |
08:29.50 | Strom_C | it can happen, you know |
08:29.58 | h3x | right but what carrier charges nothing for an answered, 0 second call |
08:30.11 | h3x | nobody, except those that dont know and are using asterisk |
08:30.11 | h3x | heh |
08:30.33 | h3x | its also usually off by a second versus the carriers records |
08:30.50 | Strom_C | well, if they've written their billing software correctly, they'll strip out anything less than two seconds |
08:30.53 | Strom_C | oooh, a whole second |
08:30.54 | Strom_C | gasp |
08:30.57 | Strom_C | the horror |
08:31.00 | h3x | I think the solution would be to measure call durations in tenths of second precision and round up |
08:31.17 | h3x | multiply that by a few million calls and you got a big chunk of change |
08:31.39 | h3x | eh, strip out <2 second calls? who does that |
08:31.43 | h3x | it aint a cell phone |
08:31.55 | h3x | intercarrier compensation happens |
08:32.08 | Strom_C | 2-second billing delay has been standard telco practice for decades now :) |
08:32.25 | h3x | 2 seconds the wrong way! heh |
08:36.59 | h3x | <PROTECTED> |
08:36.59 | h3x | <PROTECTED> |
08:37.30 | h3x | that figures |
08:37.31 | h3x | look at this |
08:37.50 | h3x | <PROTECTED> |
08:37.50 | h3x | <PROTECTED> |
08:37.50 | h3x | <PROTECTED> |
08:37.50 | h3x | <PROTECTED> |
08:37.50 | h3x | <PROTECTED> |
08:38.18 | Strom_C | so file a bug report and/or write a patch |
08:38.19 | h3x | and heres the dirty code |
08:38.30 | h3x | <PROTECTED> |
08:38.31 | h3x | <PROTECTED> |
08:38.31 | h3x | <PROTECTED> |
08:38.31 | h3x | <PROTECTED> |
08:39.23 | h3x | so the fix would be to keep all 3 call counters in milliseconds |
08:39.31 | Strom_C | filing a bug report and/or submitting a patch will do a lot more good than kvetching about it in #asterisk |
08:39.34 | h3x | in memory anyway, and subtract those |
08:39.45 | h3x | heh i will do that :P |
08:39.52 | h3x | but it dosent hurt to debate the best way to do it |
08:39.58 | Strom_C | or, hell, kvetch about it in #asterisk-dev |
08:40.04 | Strom_C | they'll care more than me |
08:40.21 | h3x | true. heh |
08:45.48 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:52.20 | *** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-10.perspektivbredband.net) |
08:53.50 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
08:54.07 | QuAtRo[NL] | I have some kind of ghost calls |
08:55.10 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
08:55.15 | QuAtRo[NL] | When a call ends unexpectedly, it seem te keep a ghost call open |
08:55.46 | QuAtRo[NL] | The phone isn't ringing again in the queue |
08:56.14 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
08:56.40 | |oranjia| | i usually get those when i have a softphone making calls |
08:56.45 | |oranjia| | like xten |
08:56.47 | |oranjia| | or wengo |
08:57.14 | h3x | yeah me 3 |
08:57.59 | QuAtRo[NL] | Is that a bug or what? |
08:58.28 | |oranjia| | but me this : http://www.thinkgeek.com/computing/input/8193/zoom/ |
08:58.38 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
08:58.45 | QuAtRo[NL] | Is there some place where i can change the timeout time before a call is killed by Asterisk |
08:59.10 | |oranjia| | buy me that |
08:59.20 | h3x | yeah |
08:59.24 | h3x | show application dial |
08:59.40 | h3x | and i think theres a dialplan variable you can set |
09:00.34 | codestr0m | does svn latest release branch of 1.2 have the jitterbuffer in it already or do I have to the patches? or go pull trunk? |
09:00.35 | RoyK | <PROTECTED> |
09:02.39 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
09:03.09 | kay2 | Is there any Framework in C for AGI/FAGI ? |
09:11.38 | *** join/#asterisk op3r (n=op3r@124.107.26.34) |
09:11.45 | op3r | hi |
09:11.47 | op3r | anyone up? |
09:11.53 | Strom_C | I am! |
09:11.57 | op3r | hi Strom_C |
09:12.01 | Strom_C | hi hi |
09:12.15 | op3r | mahy i ask whats this error |
09:12.17 | op3r | /usr/bin/ld: cannot find -lssl |
09:12.17 | op3r | collect2: ld returned 1 exit status |
09:12.17 | op3r | make: *** [asterisk] Error 1 |
09:12.37 | op3r | Im trying to install asterisk and thats the first time i saw that error |
09:13.04 | QuAtRo[NL] | Asterisk says: 'Agent/3 is ringing' but in phone isn't ringing... |
09:13.06 | Strom_C | install libssl |
09:13.13 | Strom_C | like the instructions tell you to |
09:13.15 | op3r | hmm ok |
09:13.49 | op3r | I thought there was a problem with openssl because it is already installed |
09:13.51 | RoyK | op3r: apt-get install libssl-dev |
09:14.05 | op3r | so no more open-ssldev? |
09:14.13 | RoyK | ldconfig -v |grep libssl |
09:15.02 | op3r | i think open-ssldev solved it |
09:17.00 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
09:17.05 | kmilitzer | h3x: If you are going to implent msecs for CDR-calculation make sure these are also writte into the CDR-record ... |
09:18.46 | *** join/#asterisk r0d3nt|m (n=RatMan@ip68-108-184-243.lv.lv.cox.net) |
09:20.57 | *** join/#asterisk sercz (n=sercz@dslb-084-056-233-056.pools.arcor-ip.net) |
09:23.36 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
09:26.50 | *** join/#asterisk neilbags (n=neilbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
09:31.59 | *** join/#asterisk stringslayer (n=admin@208.4.153.208) |
09:34.13 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-156-227.dsl.pltn13.sbcglobal.net) |
09:34.38 | kay2 | lol |
09:35.19 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
09:35.31 | kay2 | could anyone tell me how I could link two channels from AMI ? |
09:35.42 | darkgamer20 | buying a X100P PCI card and 2 analog phone adapter is the same as buying a TDM400P with 1 FXO module and 2 FXS modules right? just wanted to confirm before I make my purchase |
09:37.28 | kay2 | lol |
09:37.28 | kay2 | no |
09:38.07 | darkgamer20 | kay2: were you answering my question? |
09:39.18 | stringslayer | Effectiv use would be the same -- but quality would be be as good |
09:39.53 | stringslayer | oops -- quality would NOT be as good |
09:40.12 | darkgamer20 | you mean the voice quality would not be as good? |
09:40.20 | stringslayer | correct |
09:40.33 | stringslayer | overall throughput would be less |
09:40.48 | darkgamer20 | damn |
09:40.59 | darkgamer20 | would this be ok for a hobby project/home |
09:41.01 | darkgamer20 | ? |
09:41.44 | stringslayer | Sure- but not for business |
09:42.32 | darkgamer20 | oh ok |
09:42.38 | darkgamer20 | thanks stringslayer |
09:43.17 | stringslayer | np |
09:43.48 | stringslayer | do you already have your ata's |
09:44.01 | darkgamer20 | no I am going to buy that now |
09:44.25 | stringslayer | Seach ebay for dta 310 |
09:44.37 | stringslayer | You can buy them for less than $10 each |
09:44.57 | stringslayer | they are locked with packet8 firmware- so everyone sells them cheap |
09:44.59 | stringslayer | but |
09:45.23 | stringslayer | go here and you can unlock it and they work great -- I have 8 of them lol |
09:45.39 | darkgamer20 | wow |
09:45.45 | darkgamer20 | big talker eh? |
09:45.45 | stringslayer | http://www.stromcarlson.com/projects/dta-310 |
09:46.25 | stringslayer | Nah -- I let friends/family use them to call me ..etc.. |
09:46.39 | darkgamer20 | ohh i see |
09:46.52 | *** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no) |
09:46.56 | darkgamer20 | through the internet i suppose? |
09:46.57 | stringslayer | They have extensions to my box |
09:47.00 | stringslayer | yep |
09:47.30 | darkgamer20 | oh ok so you configure them so that they can connect them to your box through the internet and talk to your for free |
09:47.35 | darkgamer20 | right? |
09:47.36 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
09:48.07 | stringslayer | Yep, its a beautiful thing.. |
09:48.12 | darkgamer20 | man thats sweet |
09:49.11 | stringslayer | Are you keeping your pots line? |
09:49.18 | darkgamer20 | this site http://www.digitnetworks.com/store/product_info.php?products_id=28 says that i can call long distance for free, now do i have to subscribe for long distance from my phone company? my phone service is att |
09:49.35 | darkgamer20 | or can i somehow configure that with asterisk? |
09:49.47 | kay2 | hey |
09:49.50 | stringslayer | Asterisk is the way to go |
09:50.11 | kay2 | how can I put someone in MusicOnHold() until the callee answers his phone ? |
09:50.11 | stringslayer | Try telasip.com or teliax |
09:50.13 | kay2 | is it possible ? |
09:50.47 | *** join/#asterisk blitz[laptop] (n=blitzrag@217.41.237.104) |
09:50.53 | stringslayer | background |
09:51.15 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
09:51.21 | blitz[laptop] | foreground |
09:52.04 | stringslayer | no, use background in the extension config for that extension |
09:52.07 | *** join/#asterisk fulgas (n=fulgas@82.102.2.30) |
09:52.32 | darkgamer20 | stringslayer: so in order for me to have long distance i have to sign up for it, either from a voip company or a phone company right? |
09:52.47 | blitz[laptop] | yes -- asterisk != free calls |
09:52.51 | stringslayer | yea -- |
09:54.16 | darkgamer20 | blitz[laptop]: lol just say not not all of us are programmers |
09:54.41 | stringslayer | If you only have voip-- then you only need to buy one ata |
09:55.16 | QuAtRo[NL] | I have a problem with calling |
09:55.21 | P-NuT | Hey all, |
09:55.29 | P-NuT | just so I have this right in my head, |
09:55.54 | P-NuT | if I have an SPA3000 and I want to dialout pf the PSTN port on it, |
09:56.07 | QuAtRo[NL] | When i call from one phone to another Asterisk says: Calling 1 \n Agent/1 is ringing |
09:56.18 | QuAtRo[NL] | But in fact, the phone isn't ringing at all |
09:56.25 | P-NuT | I just create an extension in asterisk for the spa3000, and I then dial that followed by a number? |
09:56.30 | P-NuT | kinda? |
09:57.27 | darkgamer20 | stringslayer: thanks a lot for your help, that dta 310 helped me cut my costs by big margin |
09:57.29 | darkgamer20 | thanks again |
09:57.33 | darkgamer20 | see you guys |
09:57.51 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
10:01.36 | kay2 | how can I put someone in MusicOnHold() until the callee answers his phone ? |
10:02.19 | *** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR) |
10:02.30 | nXOR | hello ppl, i have a problem with my asterisk setup |
10:02.43 | stringslayer | what say you |
10:02.48 | nXOR | locally it works fine, however when i try to make a call outside via isdn |
10:02.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
10:02.53 | *** mode/#asterisk [+o denon] by ChanServ |
10:02.58 | nXOR | it starts then 503's me |
10:03.18 | nXOR | i see that visdn interface is being raised in asterisk console |
10:03.25 | sshadow | kay2: use dial(SIP/xxx|m) |
10:03.26 | nXOR | but then it dies |
10:03.27 | stringslayer | never messed with asterisk and isdn,, sorry |
10:03.34 | nXOR | hm |
10:03.51 | nXOR | do you perhaps have any good resources on setting up visdn and asterisk |
10:04.03 | nXOR | ive roamed the wastelands of internet searching for a quality tutorial |
10:04.18 | stringslayer | try google search for trixbox without tears |
10:05.51 | *** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
10:07.31 | P-NuT | if I want to dial out of the SPA300 do I use Dial(SIP/${EXTEN:1},70,Tt) ????? |
10:07.32 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-156.lsanca.fios.verizon.net) |
10:07.49 | P-NuT | and do I need to CREATE a channel in sip.conf to dial out of? |
10:09.53 | kay2 | sshadow: what does the |m does ? |
10:10.29 | sshadow | kay2: m([class]) - Provide hold music to the calling party until a requested |
10:10.29 | sshadow | <PROTECTED> |
10:10.29 | sshadow | <PROTECTED> |
10:11.05 | kay2 | sshadow: thx |
10:11.29 | sshadow | kay2: you're welcome |
10:11.38 | kay2 | sshadow: so if the callee doesnt pick up |
10:11.48 | kay2 | I could put in the next extension MusicOnHold() |
10:12.01 | Joe__11 | can anybody help me to understand how chan_local.c works? what does the function 'ast_channel_masquerade' do? |
10:12.20 | blitz[laptop] | Joe__11: thats probably more of a #asterisk-dev question... |
10:12.38 | Joe__11 | ok |
10:12.43 | blitz[laptop] | Joe__11: not sure if anyone in here will be able to answer that -- although none of the main devs seem to be up yet |
10:15.46 | sshadow | kay2: no. just do this: exten=> 100,1,Dial(SIP/mysip||m) |
10:16.03 | sshadow | kay2: the m option will start MOH |
10:24.00 | blitz[laptop] | jeeebuz -- my ssh tunnel worked first try and I can connect to the DB via pgAdmin :) |
10:26.04 | lilalinux | When I place a Wait(15) before an Answer(), the CAPI sends a DISCONNECT_IND after 10 seconds. The Telephones continue ringing, but asterisk doesn't handle it anymore. |
10:27.19 | kay2 | sshadow: that has a lil probleme |
10:27.53 | kay2 | let say that I want to dial someone that is in an other asterisk, would it work ? |
10:27.53 | *** join/#asterisk tardisx (n=justin@ppp167-251-29.static.internode.on.net) |
10:31.14 | tardisx | hi, can you use + signs in dialplans? My handset is passing numbers like +61555555555 to asterisk, and I don't seem to be able to match it (I really just want to strip the plus off) |
10:31.46 | lilalinux | tardisx: then use ${EXTEN:1} |
10:32.03 | *** join/#asterisk mrtwister (n=ambervoi@107.250.broadband5.iol.cz) |
10:32.16 | tardisx | but other handsets don't do that, so I'd be stripping off the important first digit |
10:32.50 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
10:33.15 | tardisx | I tried this: exten => +61.,1,Goto(default,0{EXTEN:3},1) |
10:33.23 | tardisx | but it doesn't seem to ever match |
10:33.38 | lilalinux | you have to use _ if you want an expression |
10:34.08 | lilalinux | :) |
10:35.21 | tardisx | thanks muchly, I think that's working now |
10:35.23 | Aurs | hello |
10:36.55 | *** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
10:37.00 | sshadow | kay2: i think that would work too. haven't tried it, but it should |
10:37.32 | Aurs | any expertise on polycom phones here? :) |
10:40.00 | blitz[laptop] | Aurs: you should probably just ask an actual question |
10:40.53 | QuAtRo[NL] | What might cause that my Asterisk says Phone 3 is ringing but in fact it isn't |
10:41.14 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
10:42.17 | Aurs | I wonder what this means: |
10:42.19 | Aurs | <PROTECTED> |
10:42.41 | Aurs | "persist"... does that mean that the setting will not be reset, or something? |
10:42.55 | lilalinux | how do i turn off "capi debug"? |
10:43.00 | Aurs | (the phones fetch config from ftp) |
10:44.50 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
10:45.46 | hads|home | Aurs: I've never congigured a polycom but I believe you are correct, that line means the volumes will keep the settings the user chooses across calls, and possibly reboots. |
10:47.10 | op3r | whats the command to the CLI to show you the realtime status? |
10:48.09 | *** join/#asterisk oej (n=olle@82.148.165.13) |
10:48.27 | Aurs | guess the only way to find out is to try it |
10:48.55 | P-NuT | hooray for the SPA3000, I have it working! zero echo and great quality. Now I'm going to burn my x100p. |
10:49.13 | bugz | is there a way to have a user push a key combo on the phone to initiate a call recording?? |
10:49.29 | bugz | i could swear i read this somewhere not too long ago |
10:50.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
10:50.51 | *** mode/#asterisk [+o denon] by ChanServ |
10:51.12 | hads|home | bugz: Check out automon in features.conf |
10:51.15 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
10:52.30 | bugz | thanks |
10:52.33 | FuriousGeorge | are there any good softphones that work with windows ce? |
10:52.38 | hads|home | np |
10:52.56 | FuriousGeorge | i believe eyebeam has a ce version, doesnt it? |
10:56.04 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
10:58.55 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
10:59.32 | P-NuT | Hey guys can someone help me with inbound SIP calls? |
11:01.06 | fourcheeze | P-NuT: don't ask to ask just ask |
11:07.52 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
11:08.14 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
11:09.23 | *** join/#asterisk florin2703 (n=aaa@florin-jurma.tm.ew.ro) |
11:09.49 | QuAtRo[NL] | Currently i have some problems with my queues |
11:09.54 | QuAtRo[NL] | My strategy is ringall |
11:10.10 | QuAtRo[NL] | But it seem that Asterisk only rings the least recent |
11:11.38 | bugz | queues are hard to manage after they get big |
11:11.47 | bugz | security is a big issue |
11:11.54 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
11:12.09 | bugz | lock down your phones, make sure assholes arent able to put their phone on DND or otherwise modify the behavior of the phone |
11:12.14 | bugz | like being able to forward the phone |
11:12.48 | bugz | we had to rebuild an entire dial plan for a gigantic call center |
11:13.02 | bugz | we basically did away with the queue all together and came up with something different |
11:14.03 | *** join/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
11:14.15 | Zeeek | ba da boom |
11:15.17 | QuAtRo[NL] | bugz: But the phones are _not_ on dnd |
11:15.24 | QuAtRo[NL] | So it seem to be a bug in Asterisk |
11:20.36 | QuAtRo[NL] | bugz: Or am I wrong? |
11:21.01 | _problem_ | anybody can tell me what is the significance of local context in extensions.conf..and if i omit that then will there be any problem?? |
11:21.51 | bugz | QuAtRo[NL]: no idea, ive abandoned all hope for queues at this time |
11:23.16 | Zeeek | _problem_: look at the docs on dialplans at http://asteriskdocs.org for the answer to this and many other questions |
11:26.31 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
11:26.37 | _problem_ | Zeeek: ok thanks |
11:27.10 | Zeeek | np |
11:27.14 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
11:27.35 | Zeeek | heres an even better overview: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
11:28.09 | Zeeek | not anywhere near here I hope :) |
11:28.34 | *** join/#asterisk viking78 (n=aherbert@66-168-98-144.static.jcsn.tn.charter.com) |
11:28.48 | RoyK | Zeeek: yeah |
11:29.01 | _problem_ | <PROTECTED> |
11:29.01 | _problem_ | <PROTECTED> |
11:29.01 | _problem_ | <PROTECTED> |
11:29.39 | _problem_ | Zeeek: i m getting these errors in cli dont know why it is happenning..the docs reference u gave me i already saw them before |
11:30.01 | Zeeek | are you on @home or AMP or something? |
11:30.20 | _problem_ | asterisk-1.2.4 |
11:30.40 | Zeeek | more info is needed. Pastebin your extension |
11:30.45 | Zeeek | b |
11:30.48 | RoyK | ~pb |
11:30.51 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
11:30.55 | QuAtRo[NL] | Is there a value where is determined after how many time a agent is able to handle the next call ? |
11:30.59 | RoyK | ~lart Zeeek |
11:34.20 | QuAtRo[NL] | FOund, wrapuptime |
11:36.57 | lilalinux | is festival 1.4.3 (shipped with debian) still ok, or should I install 1.9.5 manually? |
11:37.52 | _problem_ | Zeeek: http://pastebin.ca/73042 |
11:38.36 | Zeeek | didn't I say paste your extension? |
11:38.51 | Zeeek | the one that begins exten => .... |
11:38.56 | op3r | QuAtRo[NL]: are you using agentcallbacklogin? |
11:39.15 | Zeeek | _problem_: paste the extension and the sip.conf peer entry that it uses |
11:40.17 | _problem_ | Zeeek:ok |
11:44.52 | _problem_ | Zeeek: http://pastebin.ca/73050 |
11:47.32 | Zeeek | _problem_: is the phone registered with asterisk? |
11:48.02 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
11:48.04 | Bert- | hello there |
11:48.13 | Bert- | I've a little problem with music on hold |
11:48.24 | _problem_ | Zeeek: its a softphone when it goes offline it happens like that..but when it is online then no problem happens |
11:49.11 | Bert- | All seems to be fine, excepted a notice complaining about that :"nmp3thread: Request to schedule in the past?!?!". But people placed on hold hear nothing at all |
11:49.32 | _problem_ | Zeeek: the asterisk cli logs which i posted there are from offline scenario |
11:49.33 | Zeeek | _problem_: when it's offline there's no one there |
11:49.46 | Zeeek | you want it to go to voicemail? |
11:50.11 | _problem_ | Zeeek: yes i want it to go to voicemail. |
11:50.30 | Zeeek | try looking up and using DIALSTATUS and see if that works better |
11:50.37 | _problem_ | Zeeek: without those bad logs |
11:51.16 | _problem_ | Zeeek: ok, but what dialstatus do? I never heard of that |
11:51.23 | Zeeek | look it up |
11:51.43 | Zeeek | it's important to know what it does and too long to go into on an IRC channel |
11:51.44 | _problem_ | Zeeek: ok thnx |
11:51.56 | blitz[laptop] | ${DIALSTATUS} is teh coolest |
11:52.00 | Zeeek | it is a variable - find it on wiki or in docs |
11:52.00 | _problem_ | Zeeek: ok fine |
11:52.12 | Zeeek | blitz how's london? |
11:52.35 | Zeeek | or how was it? |
11:52.59 | blitz[laptop] | Zeeek: London is going very well! I'm currently in the exhibition hall programming :) |
11:53.08 | Zeeek | don't do that! |
11:54.00 | Zeeek | I'm copying and pasting a lot of Japanese text atm |
11:55.40 | Zeeek | brb |
11:55.42 | *** part/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
11:55.58 | *** join/#asterisk littlejohn (n=little@host20-61.pool8711.interbusiness.it) |
11:56.01 | *** join/#asterisk Zeeek_ (n=icechat5@62-240-244-9.adsl.claranet.fr) |
11:56.05 | *** part/#asterisk littlejohn (n=little@host20-61.pool8711.interbusiness.it) |
11:56.30 | SplasPood | Morning all, question.. can Voicemail() be setup so rather than hanging up on the caller after they leave the message and hit #, it instead continues on with the dialplan? |
11:56.50 | Zeeek_ | ah that's better |
11:58.07 | Zeeek_ | SplasPood doesn't show application voicemail tell you that? |
11:59.07 | SplasPood | Zeeek_: Not sure if it does, but I just did a quick test and answered my question |
11:59.22 | SplasPood | for some reason I was convinced that Voicemail() was doing its own hangup |
11:59.33 | Zeeek_ | heh best way, sometimes! I think there's a key that allows that yeah, either star or pound |
11:59.55 | SplasPood | well if you hit # it says "Goodbye" .. but if there's more dialplan, it continues.. that was news to me |
12:00.15 | Zeeek_ | isn't there a flag that allows it as well? Can't remember |
12:02.06 | QuAtRo[NL] | Damn, ringall seem to just don't work in Asterisk |
12:02.19 | QuAtRo[NL] | It always uses leastrecent |
12:02.52 | QuAtRo[NL] | Which makes Asterisk unusable for a lot of helpdesks |
12:04.53 | SplasPood | QuAtRo[NL]: hrm, ringall works for me |
12:05.35 | QuAtRo[NL] | SplasPood: When i call to the helpdesk (which is a queue) it rings two phones... |
12:05.52 | QuAtRo[NL] | I pick up phone 1 |
12:06.14 | QuAtRo[NL] | Then I hang up and call (5 seconds later) again |
12:06.23 | QuAtRo[NL] | And only phone 2 rings |
12:06.46 | SplasPood | maybe its wrapup time |
12:06.50 | SplasPood | in the queue config |
12:06.55 | QuAtRo[NL] | My wrapuptime is 10 |
12:07.11 | QuAtRo[NL] | And wrapuptime is miliseconds |
12:07.18 | SplasPood | so if you called 5 seconds later phone 1 would still be in wrapup... |
12:07.23 | SplasPood | is it? I'm pretty sure it's seconds |
12:07.45 | lilalinux | the voicemailbox has to apps: VoiceMail and VoiceMailMain, Is it possible to get into VoiceMailMain from within VoiceMail? |
12:08.10 | SplasPood | lilalinux: yes, during the greeting one can hit... * or #, (i forget) and it'll prompt them to login to that mailbox |
12:08.24 | lilalinux | SplasPood: thx |
12:08.30 | SplasPood | lilalinux: you'll need a exten 'a' and stuff.. check out Voicemail() on voip-info.org |
12:09.02 | QuAtRo[NL] | SplasPood: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+agents.conf |
12:09.19 | SplasPood | QuAtRo[NL]: thats agents.conf.. |
12:09.21 | *** join/#asterisk jpeeler (n=jpeeler4@host81-149-2-72.in-addr.btopenworld.com) |
12:09.43 | QuAtRo[NL] | Ah... |
12:09.47 | QuAtRo[NL] | Let me have a look |
12:09.48 | SplasPood | QuAtRo[NL]: there's a wrapup setting in queues.conf as well |
12:10.38 | QuAtRo[NL] | I saw ;) |
12:11.56 | *** join/#asterisk coppice (n=chatzill@223.193.17.210.dyn.pacific.net.hk) |
12:12.08 | *** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com) |
12:12.49 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
12:13.04 | Tili | hey where do i set zap to show cli as international to switch |
12:15.05 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
12:15.46 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
12:16.20 | lilalinux | How can I get the originally called number (meanwhile beeing in a different exten) |
12:17.29 | SplasPood | lilalinux: I *believe* one of the variables has it... do a search for variables on voip-info.org |
12:17.33 | RoyK | lilalinux: i don't think you can. just Set(ORIGINALNUMBER=${EXTEN}) |
12:17.43 | SplasPood | or, what RoyK said |
12:17.54 | bugz | Jun 27 07:16:39 NOTICE[5597]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
12:17.56 | Zeeek_ | or pass it in a macro |
12:18.01 | *** join/#asterisk olor1n (n=void@LAubervilliers-151-11-80-139.w193-251.abo.wanadoo.fr) |
12:18.03 | olor1n | hello |
12:18.25 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
12:18.55 | lilalinux | thx |
12:19.14 | QuAtRo[NL] | SplasPood: Thank you! It works fine :) |
12:20.14 | SplasPood | QuAtRo[NL]: No problem :) Queues pissed me the hell off when i first tried to set them up.. Another thing i found is that if you use roundrobin, but want it to continue if the agent doesn't answer (but is available) you need to have timeoutrestart=yes in queues.conf |
12:20.50 | QuAtRo[NL] | SplasPood: Do you have any idea what makes my phones log off after a certain time= |
12:21.20 | RoyK | don't listen to Zeeek_ |
12:21.27 | Zeeek_ | they get tired of hearing the same shit? |
12:21.42 | SplasPood | QuAtRo[NL]: there's an option in agents.conf that logs agents out if they don't answer... |
12:22.04 | topping | anyone good with RealTime? I've got the tables, login and ODBC set up, but i don't see res_odbc loading in the debug output |
12:22.21 | SplasPood | load res_odbc.so ? |
12:22.38 | op3r | does anyone know this error? Jun 27 20:22:09 WARNING[23303]: app.c:644 ast_play_and_record_full: No audio available on IAX2/u26450 |
12:22.43 | jpeeler | anybody know why i would get "Primary D-Channel on span 1 down" when the lines are up? |
12:23.10 | topping | SplasPood: maybe I don't have it compiled in? |
12:23.14 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
12:23.15 | SplasPood | its a module |
12:23.16 | RoyK[no] | jpeeler: could be akk sirts if reasons |
12:23.23 | SplasPood | if you don't autoload in modules.conf |
12:23.31 | SplasPood | you'd need to explicitly load it |
12:23.38 | SplasPood | what happens if you do load res_odbc.so |
12:23.39 | topping | i just don't see it in usr/lib/asterisk |
12:23.44 | SplasPood | oh |
12:23.59 | RoyK[no] | topping: then perhaps you don't have the odbc libs installed. then it'll never be built |
12:24.00 | jpeeler | RoyK[no], yeah so where to start? |
12:24.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
12:24.07 | SplasPood | well like RoyK, I don't use it.. but maybe you l... what RoyK said |
12:24.19 | topping | ahhh ok lemme try to rebuild now that i have odbc installed thx |
12:25.21 | op3r | does anyone know this error? Jun 27 20:22:09 WARNING[23303]: app.c:644 ast_play_and_record_full: No audio available on IAX2/u26450? |
12:25.41 | SplasPood | anyone know if the privacy option to dial can be configure to not offer the 'Torture' option? |
12:26.12 | SplasPood | op3r: I'd assume if one of the 5 people here right now knew the answer someone would have responded by now. |
12:26.51 | SplasPood | op3r: might want more info.. what was going on when you got that error, etc.. |
12:26.56 | op3r | ok hehehheeh |
12:27.12 | op3r | SplasPood: thats when Im trying to call an inbound DID number |
12:27.21 | op3r | and redirected it to the voicemail |
12:27.22 | op3r | :( |
12:27.46 | SplasPood | and you get that error when it.....? |
12:28.31 | NotJohnDavid | the asterisk-user list is busier than i thought it'd be |
12:28.59 | RoyK[no] | ops. customs... getting closer to the .se border |
12:29.09 | RoyK[no] | they're going through my bag... |
12:29.14 | QuAtRo[NL] | SplasPood: They do not only log off from the queue... When you call one of those phones Asterisk says the phone is ringing |
12:29.19 | QuAtRo[NL] | But it isn't |
12:29.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
12:29.47 | SplasPood | hrm... network issue maybe? some type of NAT/firewall interference? |
12:30.07 | jpeeler | can anybody tell me what to start looking at with all the primary d-channel spans down? |
12:30.15 | QuAtRo[NL] | The strange thing is, it happends after a certain time... |
12:30.19 | jpeeler | it's not the card or the lines |
12:30.43 | bugz | christ |
12:30.53 | bugz | is there a way to remotely troubleshoot a te110p |
12:31.29 | olor1n | I try to make something with asterisk but i am not quite sure it can be done so please comment: the idea is when a call is received to put the caller in an interactive menu (choice 1,2 etc.) and in the same time to keep anouncing on the called party telephone that there is an incoming event. Is it possible |
12:31.56 | QuAtRo[NL] | SplasPood: And when i use the 'restart' function of my phone.. It all works fine again.. Untill a certain time |
12:32.53 | SplasPood | QuAtRo[NL]: always the same time? |
12:33.18 | QuAtRo[NL] | Didn't measure it.. |
12:34.55 | SplasPood | roughly? |
12:34.59 | SplasPood | I'd do some packet sniffing |
12:35.02 | SplasPood | and see what was up |
12:35.10 | SplasPood | is there a firewall/NAT involved here anywhere? |
12:35.16 | QuAtRo[NL] | Yes, there is |
12:35.23 | SplasPood | that'd be my first guess |
12:35.31 | QuAtRo[NL] | The PBX is in the datacenter and we are on a ADSL line |
12:35.56 | QuAtRo[NL] | Behind a Eminent router |
12:36.12 | SplasPood | heh a Pre-Eminent router |
12:36.33 | *** join/#asterisk Cresl1n (n=matt@217.41.237.104) |
12:38.15 | *** join/#asterisk Ifaistos (n=stelios@dslcustomer169.vivodi.gr) |
12:39.17 | topping | SplasPood: OK, I can do DB queries from CLI now, but I don't seem to be seeing the extensions in the dialplan yet, any ideas? |
12:39.49 | SplasPood | topping: I dunno res_odbc, when you say 'from CLI' what do you mean? |
12:40.07 | topping | 'realtime load extensions exten 411' gives back the correct data |
12:40.11 | SplasPood | ahh |
12:40.11 | topping | from the database |
12:40.30 | QuAtRo[NL] | SplasPood: I guess it is my NAT keepalive function of my phone |
12:40.31 | topping | but when i dial that extension, i get a reorder |
12:40.36 | SplasPood | you have the switch statements and what not in extensions.conf |
12:40.51 | Cresl1n | blitzrage: !!!! |
12:40.54 | Cresl1n | blitz[laptop]: !!!!! |
12:40.54 | topping | don't think so, i'll rtfm, thanks! |
12:41.04 | blitz[laptop] | Cresl1n: !!!!!!!! |
12:41.15 | *** join/#asterisk TeePOG (n=Ender@dsl-145-132-214.telkomadsl.co.za) |
12:41.16 | blitz[laptop] | Cresl1n: back to work :) |
12:41.22 | SplasPood | topping: Warning, the way asterisk realtime extensions work is highly lame IMHO |
12:41.26 | TeePOG | afternoon all |
12:41.37 | Cresl1n | afternoon |
12:41.48 | topping | SplasPood: yah, i sensed that, but isn't it still the best way to get to a database? |
12:41.57 | blitz[laptop] | Cresl1n: did it die again? |
12:41.58 | SplasPood | I suppose |
12:42.00 | SplasPood | without a reload.. |
12:42.11 | blitz[laptop] | Cresl1n: who's laptop are you using now? |
12:42.11 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) |
12:42.14 | acrg | hi |
12:42.19 | acrg | having an odd problem |
12:42.23 | Cresl1n | blitz[laptop]: when I jump up and down it does |
12:42.30 | acrg | if a caller puts someone on hold |
12:42.34 | acrg | and then takes the call back |
12:42.44 | acrg | the other party can not hear them |
12:42.52 | acrg | but he can hear the other party |
12:42.58 | blitz[laptop] | Cresl1n: for real? |
12:43.04 | Cresl1n | yeah |
12:43.06 | Cresl1n | sux0rs |
12:43.11 | acrg | anyone know what's up ? |
12:43.17 | blitz[laptop] | Cresl1n: wow... could the CPU or something be a bit loose? |
12:43.23 | blitz[laptop] | or a cable? |
12:43.32 | TeePOG | hi guys, do you need any special hardware to run asterisk on a *nix box? I want to use asterisk as a voip-to-phone gateway |
12:43.55 | SplasPood | TeePOG: you wanna use a normal phone line? |
12:44.27 | SplasPood | a POTS line.. |
12:44.36 | TeePOG | yes, but i suppose it's going to be a bitch getting an internal pci 56k modem to work under linux? |
12:44.40 | Bert- | hmm what is the 'cleaner solution', resolving the 'mpg123issue' plz ??? |
12:44.47 | Bert- | I'm unable to find it on voip-info |
12:44.53 | Bert- | ~mpg123 |
12:45.06 | jbot | methinks mpg123 is Real time MPEG Audio Player for Layer 1,2 and Layer3. URL: http://www.mpg123.de/. ONLY MPG123-R will work with asterisk. PERIOD. use 'make mpg123' in the asterisk source dir |
12:45.08 | SplasPood | TeePOG: well... technically the low end digium cards were (are?) modems.. but a specific kind |
12:45.08 | Cresl1n | why can't I build trunk chan_zap |
12:45.08 | Cresl1n | somebody hit me |
12:45.24 | SplasPood | TeePOG: So you'll either need an external FXO gateway, or you'll need one of the analog line digium cards |
12:45.25 | TeePOG | ok SplasPood -- so I need a digium card? I thought those were only for ISDN lines |
12:45.31 | TeePOG | 'ok |
12:45.39 | Bert- | hmm what is the 'cleaner solution', resolving the 'mpg123issue' plz ??? |
12:45.49 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
12:46.13 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
12:46.34 | NotJohnDavid | teepog: the x100p works just fine under linux. that'll connect a POTS line to the asterisk server then you need a way to connect handsets to asterisk (if you so choose) |
12:46.55 | Bert- | grr disconnected again and again :( |
12:47.35 | acrg | bert do you mean the leftover mpg123 processes running at 100% cpu? |
12:48.25 | Bert- | no |
12:48.30 | Bert- | I mean |
12:48.39 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
12:48.43 | Bert- | as I can read on website, there is no need to use mpg123 anymore |
12:48.45 | TeePOG | well, we're like 3 shops under the same management, next to each other. would it be better to let their *doze SIP clients connect to my asterisk box as extensions? |
12:48.54 | Bert- | asterisk 1.2 and higher provide a new solution |
12:49.03 | Bert- | but there is nothing about that on the website :( |
12:49.34 | blitz[laptop] | ouch! |
12:49.54 | Bert- | "Asterisk 1.2 has solved the "mpg123 issue" and comes with a cleaner solution" |
12:50.08 | Bert- | then I'm looking on this cleaner solution |
12:50.13 | acrg | Bert yes, it can read mp3s natively now with the format_mp3 module |
12:50.28 | Bert- | does it works with MusicOnHold ? |
12:50.31 | acrg | it is in the asterisk-addons package |
12:50.32 | acrg | yes |
12:50.43 | blitz[laptop] | although converting your mp3's to ulaw/alaw/gsm/g729/etc... and just using the native format can take a lot of load off the CPU |
12:51.02 | Bert- | yep for sure |
12:51.14 | Bert- | for now I only wnat to have musiconhold working |
12:51.24 | Bert- | whatever the soulution used:) |
12:51.37 | NotJohnDavid | what can you use to play gsm? and convert mpg->gsm ? |
12:51.48 | Bert- | It is just a test, as I have to make a demo to my boss |
12:51.56 | jpeeler | Bert-, I followed blitz's advice about converting and got my music on hold working |
12:51.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
12:51.57 | *** mode/#asterisk [+o anthm] by ChanServ |
12:52.13 | Bert- | you convert it with lame ?? sox ?? |
12:52.14 | blitz[laptop] | jpeeler: w00t!!! |
12:52.17 | blitz[laptop] | sox |
12:52.21 | jpeeler | correct sox |
12:52.25 | Bert- | ok let me try |
12:52.28 | Bert- | thx guys :) |
12:52.47 | jpeeler | blitz[laptop], yeah thanks a lot |
12:52.49 | NotJohnDavid | sox is like a swiss army knife |
12:53.09 | blitz[laptop] | jpeeler: glad it worked for you |
12:53.24 | mut | yea, it's useful for a minute or two until it falls apart |
12:53.27 | jpeeler | blitz[laptop], did you see any of my earlier problems today? |
12:53.43 | blitz[laptop] | jpeeler: sorry, I did not -- been running a conference :) |
12:54.00 | topping | SplasPood: I think this is the last question... everything is almost working, but I get "macro_exec: No such context 'macro-tenant,1415xxxxxxx' for macro 'tenant,1415xxxxxxx'" |
12:54.14 | jpeeler | blitz[laptop], I'm kind of at a loss here. I keep getting these errors: Primary D-Channel on span 1 down |
12:54.21 | topping | my database is populated with calls to 'Macro' |
12:54.23 | jpeeler | blitz[laptop], the lines are up and the card is good |
12:54.29 | topping | can i do that? |
12:54.48 | QuAtRo[NL] | SplasPood: It was the so called: 'NAT keepalive' setting in my phone ;) |
12:55.02 | blitz[laptop] | topping: how are you calling the macro app? It thinks the macro name is 'tenant,1415xxxxxxxx' |
12:55.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:55.19 | topping | lemme pastebin it |
12:55.25 | blitz[laptop] | and obviously you are trying to send Macro an argument |
12:55.57 | topping | yes |
12:56.19 | blitz[laptop] | jpeeler: hrmmmm... I'm not really a T1/E1 pro... sorry |
12:56.25 | *** part/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
12:56.25 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
12:57.46 | Druken | morning peoples |
12:58.06 | topping | http://rafb.net/paste/results/cQgsUK62.html |
12:58.29 | topping | the first half is what i have in extensions.conf, the second half is a record from the db |
12:58.41 | jpeeler | blitz[laptop], well thanks anyway. it's hard to get help with problems i don't even know where to start |
12:59.35 | topping | i have a macro in the extensions.conf that i should have listed... http://rafb.net/paste/results/dtH4Jq29.html |
12:59.51 | blitz[laptop] | jpeeler: well... the D-channel is used to provide signaling between you and the other end (CO, Channel Bank, etc...) and it sounds like something is happening that is causing the D-channel to drop |
13:00.00 | blitz[laptop] | jpeeler: how often does it happen? |
13:00.35 | [TK]D-Fender | blitz[laptop] : ! ! ! |
13:00.57 | jpeeler | blitz[laptop], when something with the lines change |
13:01.13 | jpeeler | blitz[laptop], like when i reboot or unplug a line |
13:01.41 | blitz[laptop] | [TK]D-Fender: I don't want to meet your mom! |
13:02.08 | [TK]D-Fender | Up in Mississauga now. we should schedule |
13:02.13 | lilalinux | how do I configure asterisk for _incoming_ calls from sipgate? |
13:02.28 | blitz[laptop] | [TK]D-Fender: yes! I will be returning tomorrow from London actually |
13:02.38 | topping | blitz[laptop]: do you think this could be because the macro is not in the database with the calls to the macro? |
13:02.41 | [TK]D-Fender | blitz[laptop] : ON or UK? |
13:02.59 | blitz[laptop] | [TK]D-Fender: From London, UK to Mississauga, ON |
13:03.37 | blitz[laptop] | topping: not too sure -- I don't use realtime, but somehow Asterisk isn't parsing the comma |
13:03.39 | [TK]D-Fender | blitz[laptop] : And you got more than just a lousy T-shirt, right? ;) |
13:03.51 | topping | ahh, ic |
13:03.59 | blitz[laptop] | [TK]D-Fender: yah... I bought a bunch of chocolate I hope to give to a hot asian girl I went on a date with once, lol |
13:04.02 | Druken | there's a mississauga UK ? |
13:04.12 | blitz[laptop] | Druken: don't think so :) |
13:04.17 | blitz[laptop] | ok .. I'm going back to programming -- peas |
13:04.40 | [TK]D-Fender | Druken : Not to my knowledge |
13:04.42 | topping | thanks for the help blitz[laptop] |
13:04.48 | Druken | :) |
13:04.53 | Druken | i was gonna say :) |
13:06.44 | Druken | do i want to know why blitz[laptop] would be meating [TK]D-Fender's mom ? |
13:06.59 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
13:09.22 | acrg | Having an odd problem: if a caller puts someone on hold and then takes the call back, the other party can not hear them. But he can hear the other party. |
13:09.39 | kay2 | [TK]D-Fender: do you know why I get that: NOTICE[20945]: chan_iax2.c:5068 register_verify: Peer 'key2' is not dynamic (from 62.39.9.251) |
13:09.50 | topping | SplasPood blitz[laptop]: turns out if I make the database appdata field have a '|' instead of a comma that it all works. Looks like there's some problem with parsing the comma, as blitz[laptop] pointed out. Everything works now, THANKS A BUNCH! :-D |
13:11.20 | blitz[laptop] | topping: oh yah!! I have to remember that because Asterisk will definately parse on the pipe, but for some reason it can't always parse on the comma because in the dialplan it actually converts the comma to a pipe |
13:11.55 | topping | cool stuff! |
13:11.56 | topping | hehe |
13:12.15 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.21.Dial1.SanJose1.Level3.net) |
13:12.18 | topping | it was easy for me to change because I am reading the data through a view |
13:12.25 | topping | so i just changed the output of the view |
13:12.38 | topping | postgresql rocks |
13:13.04 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.21.Dial1.SanJose1.Level3.net) |
13:13.06 | kay2 | anyone knows why I get NOTICE[20945]: "chan_iax2.c:5068 register_verify: Peer 'key2' is not dynamic (from 62.39.9.251)" if in iax.conf I put host=62.39.9.251 |
13:13.21 | topping | i have these sentex call boxes outside my apartment building, and people are constantly asking me to reprogram them |
13:13.35 | topping | now i can put their numbers in a database and let them update the door boxes via the web |
13:13.57 | *** join/#asterisk ToyMan (n=stuq@74-32-78-58.dsl1.mdl.ny.frontiernet.net) |
13:15.38 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
13:18.24 | Bert- | hmm |
13:18.26 | Bert- | doesn't work |
13:18.39 | Bert- | I use the default musiconhold context |
13:18.56 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
13:19.33 | benjamin7062 | Any of you guys using SCCP and noticing Asterisk 1.2.9.1 crashes if the phone terminates in a queue? |
13:19.39 | Bert- | I set it like that : |
13:19.40 | Bert- | [default] |
13:19.40 | Bert- | mode => quietmp3 |
13:19.40 | Bert- | directory => /var/lib/asterisk/mohmp3/planetvoip |
13:19.40 | Bert- | ;application=/usr/bin/sox |
13:19.44 | benjamin7062 | SCCP phone I mean? |
13:19.52 | *** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
13:19.54 | BertZ | grr |
13:20.15 | Druken | ~pastebin |
13:20.18 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
13:21.06 | benjamin7062 | ~fart |
13:21.08 | jbot | ACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!" |
13:23.02 | *** join/#asterisk creadurx (n=creadure@196.82-134-19.bkkb.no) |
13:23.58 | Zeeek_ | jbot stop flooding the channel with inane remarks |
13:23.59 | jbot | ACTION leaps to his feet and stops flooding the channel with inane remarks |
13:24.05 | Zeeek_ | whew! |
13:24.28 | creadurx | inane is better than insane |
13:25.08 | Zeeek_ | not for the person who is insane |
13:25.09 | TeePOG | inane is a boring type of insane |
13:25.11 | Druken | uhg.... god damn light |
13:25.23 | Zeeek_ | insane is a dangerous form of inanity |
13:25.30 | Druken | if not for light, i'd say fuck today and go back asleep |
13:25.30 | TeePOG | snap |
13:25.46 | Zeeek_ | sleeping masks with digium logo rock |
13:25.47 | topping | Druken: are you on the west coast too? |
13:25.57 | Druken | ontario |
13:26.11 | topping | oh so it's extra-bright there lol |
13:26.21 | topping | people still have their headlights on here |
13:26.22 | creadurx | how would i solve the problem of knowing which DID is responsible for a SIP phone ringing when using queue()? one SIP phone is a member in 2 different queues, each set up to two different DIDs |
13:26.34 | Druken | no.. it's a god damn overcast miserable rainy day |
13:26.43 | topping | heh |
13:26.48 | Zeeek_ | insult to injury! |
13:27.18 | *** join/#asterisk op3r (n=op3r@124.107.26.34) |
13:27.31 | *** join/#asterisk wese103 (n=wschaffe@c069.centercall.com) |
13:27.57 | Druken | creadurx: you want to know what did it came from or what queue? |
13:28.06 | op3r | does any one know how to remove the message that announce the the caller's queue's position? |
13:28.06 | topping | ok i'm going to try sleeping lol nite |
13:28.23 | Druken | bastard! |
13:28.44 | creadurx | Druken: both :) |
13:29.09 | creadurx | im hacking the shit outta the manager interface and astmanproxy |
13:29.16 | creadurx | but this problem stumbled me |
13:29.24 | benjamin7062 | So * keeps crashing if I terminate an SCCP phone inside a queue. With Verbose and Debug all the way up it doesn't spit anything unusual up since it's crashing. Any way for me to get it to spit more info? |
13:31.17 | *** join/#asterisk [pyro] (i=_pyro_@tor/regular/bracketed-pyro) |
13:31.21 | op3r | benjamin7062: do you know how to remove the announcement of the caller's queue position? |
13:31.22 | Druken | creadurx: well, asterisk can announce what queue the call came from, as for what did... no idea... i guess my question would be, what does it matter? |
13:31.44 | Druken | op3r: did you actually read the queue.conf file? |
13:31.51 | op3r | Druken |
13:31.55 | benjamin7062 | op3r, You mean for my problem? Or do you have a seperate problem. The config for that is queue.conf |
13:31.59 | op3r | but it kept on telling the queue position? |
13:32.16 | benjamin7062 | op3r, asterisk -rx'reload' |
13:32.51 | *** join/#asterisk BertoX (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
13:32.53 | BertoX | :( |
13:32.55 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
13:33.00 | BertoX | big internet issue from my side :( |
13:33.18 | BertoX | How to see where musiconhold fails plz ? |
13:33.21 | BertoX | no log, no error |
13:33.25 | BertoX | but no music ... :( |
13:33.55 | benjamin7062 | Bertox -- Turn up verbose and debug... it will show you everything... from the console... set verbose 100 and set debug 100 |
13:34.27 | blitz[laptop] | LOL |
13:34.30 | creadurx | Druken: well, I cant even phrase my own problem! i gotta do some more thinking. |
13:34.33 | blitz[laptop] | 100? nothing above 4 is useful |
13:34.47 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
13:35.05 | BertoX | I lol |
13:35.07 | blitz[laptop] | debugging information is also controlled in logger.conf |
13:35.11 | benjamin7062 | Yes, I know... but if you say 100 then when they bump it some day you don't have to re-train.. =) |
13:35.12 | BertoX | set level 100 for both |
13:35.15 | BertoX | nothing more ... |
13:35.38 | blitz[laptop] | benjamin7062: if you need to retrain someone regarding how to use 'set' on the CLI they shouldn't be on the CLI in the first place ;) |
13:35.54 | BertoX | what exactly mean that please ? |
13:35.55 | BertoX | monmp3thread: Request to schedule in the past?!?! |
13:36.02 | BertoX | maybe my problem comes from here |
13:36.04 | benjamin7062 | blitz[laptop], I can't argue with that... =) |
13:36.12 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
13:36.21 | benjamin7062 | blitz[laptop], but 13 = 100 as far as logging.. =) |
13:36.27 | Hmmhesays | yo ho |
13:36.33 | blitz[laptop] | and 4 == 100 |
13:36.50 | Zeeek_ | 40 + 2 = ? |
13:36.52 | blitz[laptop] | or just use 5 which is an easy number to remember |
13:36.55 | benjamin7062 | 45? |
13:36.57 | blitz[laptop] | Zeeek_: you nerd :) |
13:37.08 | benjamin7062 | No, I got it... 46 |
13:37.17 | Druken | 69? |
13:37.22 | benjamin7062 | Damn |
13:37.24 | benjamin7062 | I was way off |
13:37.34 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
13:37.39 | kay2 | why do I get that msg ? NOTICE[22371]: chan_iax2.c:6802 socket_read: Rejected connect attempt from 62.39.9.251, who was trying to reach '400@ddefault' |
13:37.45 | benjamin7062 | BertoX, Does it show it 'trying' to play music? |
13:37.46 | kay2 | why does it get Rejected |
13:37.47 | kay2 | ? |
13:37.54 | Katty | morning |
13:37.59 | BertoX | no |
13:38.02 | BertoX | benjamin7062 : no |
13:38.16 | benjamin7062 | kay2, I have a little experience with rejection.. but it involves women.. not * |
13:38.24 | Katty | haha |
13:38.28 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
13:38.38 | kay2 | benjamin7062: how weird, I have the oposite |
13:38.48 | Zeeek_ | key katty, long time no C++ |
13:39.11 | Katty | hey Zeeek (= |
13:39.14 | benjamin7062 | kay2, Hopefully, you can fix your problem cause there is no code fix that will help me. =) |
13:39.45 | iDunno | the world is trying to take me over :) |
13:40.10 | benjamin7062 | BertoX, you have a line in your dialplan to play music but never see it hit that line in the log at all? |
13:40.20 | BertoX | benjamin7062 : just -- Started music on hold, class 'default', on channel 'SIP/Nextone_OUT-06c9' |
13:40.20 | BertoX | <PROTECTED> |
13:40.22 | jbalcomb | why is #perl is haven for unhelpful, underage folks? |
13:40.31 | *** join/#asterisk Winkie (n=urmom@gateway.duclicsic.com) |
13:40.42 | Druken | BertoX: do you have a zap interface? |
13:40.48 | Zeeek_ | did you type #perljam by accident? |
13:41.04 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
13:41.13 | benjamin7062 | BertoX, Hmm.. Do you have Zap...? |
13:41.14 | Katty | Zeeek_: you framilier with mounting windows 2003 shares? |
13:41.25 | Zeeek_ | Ewwwwwww |
13:41.33 | benjamin7062 | Katty, if you do... don't write to them (if they are NTFS) |
13:41.38 | kay2 | Katty; i am |
13:41.43 | jbalcomb | Zeeek_: haha.. it's seems almost likely. |
13:41.47 | Katty | benjamin7062: i'll do whatever i please ;) |
13:41.58 | Katty | Zeeek_: i'll take that as a no, heh. |
13:41.59 | Zeeek_ | katty you've been warned! |
13:42.04 | Druken | typical women.... |
13:42.14 | Katty | twisted[asteria]: you around? |
13:42.16 | Zeeek_ | yes, ask for advice and then do what they please |
13:42.18 | Katty | twisted[asteria]: i know you're helpful. |
13:42.24 | Katty | Zeeek_: i wasn't asking for advice. |
13:42.29 | Katty | Zeeek_: i'm going to do what i'm going to do. |
13:42.29 | benjamin7062 | Katty, K... Good luck... I predict mass corruption... I need some popcorn to watch |
13:42.32 | Zeeek_ | no, true, you wouldn't |
13:42.37 | Katty | Zeeek_: i /simply/ asked if if you'd done it. |
13:42.51 | Katty | how annoying, geeks being bitter. |
13:42.57 | Katty | do lighten up guys. |
13:43.04 | benjamin7062 | LOL |
13:43.08 | Zeeek_ | "for sale one used encylopedia britanica. Reason: no longer needed, married, wife knows everything" |
13:43.17 | Druken | hahaha |
13:43.18 | Katty | exactly ;) |
13:43.29 | Katty | ariel_: you around? |
13:43.29 | jbalcomb | Katty: We have no problemds with our NTFS mounts. |
13:43.31 | Zeeek_ | I'm confirming it Katty, not denying :) |
13:43.46 | Katty | Zeeek_: eddie izzard moment? |
13:43.53 | Katty | Zeeek_: just keep confirming and denying |
13:44.22 | Zeeek_ | Error 42 -unknown culture specific reference or allusion. Please contact your culture manager |
13:44.46 | benjamin7062 | Heavy writing to NTFS works fine. It's reading it from W2K3 again that becomes the problem over time |
13:44.49 | Katty | jbalcomb: is it a 2003 share? |
13:44.50 | Zeeek_ | wtf is eddie_izzard? |
13:44.57 | Katty | Zeeek_: consult google, dear. |
13:45.05 | Zeeek_ | naw |
13:45.17 | Zeeek_ | well, ok, fer u |
13:45.17 | Druken | Katty: i'm tired.... mind if i sleep with you? :) hehe |
13:45.32 | Zeeek_ | fools rush in! |
13:45.44 | benjamin7062 | haha |
13:45.45 | jbalcomb | Katty: Well, actually I beleive they are 2000. I'm not familiar with there being a difference there though. |
13:45.46 | Katty | Druken: i think the significant other would mind. |
13:45.50 | Zeeek_ | where asterisk-angels fear to tread |
13:45.55 | Katty | jbalcomb: and yes, well there is.. |
13:45.58 | Druken | it's a possibility |
13:46.01 | Katty | jbalcomb: thanks anyway (= |
13:46.17 | Zeeek_ | but asterisk-angels never fall in love. They just go to bed |
13:46.17 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
13:46.19 | Katty | jbalcomb: i do appreciate you at least /attempting/ to offer help. |
13:46.29 | Katty | jbalcomb: rather than just saying zomgdon'tdoit |
13:46.37 | Katty | jbalcomb: so go you (= |
13:46.37 | jbalcomb | Katty: ah, well, yes, thank you. Sorry we can't be of more help on this particular matter. Please consider our assistance with future quandries. |
13:47.07 | Zeeek_ | We know that you have a choice in IRC channels. Thank you for coming to #asterisk |
13:47.08 | benjamin7062 | Katty, you didn't say remotely mounting NTFS -- You said mounting. There's a difference. |
13:47.16 | Katty | oh for goodness sake. |
13:47.20 | Katty | i said mount and windows 2003 share. |
13:47.30 | creadurx | Druken: this is why i need the DID.. each of our customers have their own DID that gets routed to either queue1 or queue2. the SIP phone that is a member of queue1 needs to know which DID was called so our inhouse app automatically opens the correct customer database |
13:47.37 | Katty | if you're going to nit pick me to death, just go stand in a corner :P |
13:47.52 | Zeeek_ | and if he's in a round building? |
13:48.00 | benjamin7062 | Katty, that works fine... |
13:48.10 | Katty | yes i KNOW it works fine |
13:48.14 | Katty | that's not the question here |
13:48.26 | Katty | it never was in the first place |
13:48.32 | benjamin7062 | Katty, Have you actually asked a question yet or are you here to bitch? |
13:48.36 | Zeeek_ | if anyone knows how to talk to the file windows server éà03 system in php I'd be interested |
13:48.38 | benjamin7062 | women |
13:48.39 | benjamin7062 | ... |
13:48.40 | Katty | hahaha |
13:48.44 | Druken | creadurx: so your doing an answering service then ? |
13:48.48 | Katty | benjamin7062: you're cute. |
13:48.54 | Katty | benjamin7062: you ever read what people ask? |
13:49.08 | Katty | benjamin7062: sounds like you're the bitchy one ;) |
13:49.23 | *** join/#asterisk Ifaistos (n=stelios@dslcustomer169.vivodi.gr) |
13:49.25 | Katty | benjamin7062: don't worry about, k? i'll get twisted[asteria] to help me. i know he knows what he's doing. |
13:49.56 | creadurx | Druken: no.. im doing db lookups based on the CID, and also trying to do the same with the DID. this is all client side, via the manager interface |
13:50.11 | benjamin7062 | Katty, mkay, you do that. I pitty him |
13:50.22 | Katty | benjamin7062: you shouldn't pitty him. |
13:50.32 | Katty | benjamin7062: he probably has more fun in a day with his job than you do all year ;) |
13:51.20 | sercz | hmm i'm looking for a very cheap way to connect an analogue phone to an asterisk box- any ideas? |
13:51.21 | benjamin7062 | Katty, But that's simply not true, there are people like you in the world that make life complete |
13:51.34 | benjamin7062 | sercz, $60.00 cheap? |
13:51.36 | Katty | benjamin7062: glad to be of service. |
13:51.51 | sercz | less than 60$ benjamin7062 :) |
13:52.04 | Katty | benjamin7062: despite what you think, i'm not a bitch ;) |
13:52.05 | Zeeek_ | sercz cheap IAX chinese ATA |
13:52.05 | Druken | my job isn't fun.... granted, i only work about 3-4 hours a day... |
13:52.07 | sercz | my phone was 40$ |
13:52.12 | Katty | benjamin7062: i just refuse to deal with idiots that don't listen. |
13:52.29 | Hmmhesays | heh |
13:52.32 | Hmmhesays | that's all I deal with |
13:52.33 | Katty | benjamin7062: or try to tell me something can't be done, when i know full well that it can be... |
13:52.33 | Zeeek_ | Katty I resemble that remark! |
13:52.51 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
13:52.55 | lilalinux | does kphone support md5 secrets? |
13:53.04 | Zeeek_ | Druken you should dstop being a streetwalker and get a normal job |
13:53.09 | kay2 | When does a iax2 call get rejected for the reason "No Authority Found" ? |
13:53.55 | benjamin7062 | Katty, sure.. k.. |
13:54.05 | Druken | Zeeek_, i wish i had the looks to be a street walker :) |
13:54.46 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
13:55.48 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
13:56.24 | littleball | hello,who can recommend a good sip soft phone (free one:-)) |
13:56.27 | Druken | ok... well i guess i need to get ready to go... that 15 mins of work needs to be done sooner or later :) |
13:56.34 | op3r | littleball: xlite? |
13:56.57 | littleball | i have tried two. xlite and firefly |
13:57.00 | benjamin7062 | littleball, I like twinkle if you are running unix |
13:57.01 | littleball | both work |
13:57.25 | op3r | then choose q |
13:57.25 | littleball | benjamin7062, i already run sip service now. :-) |
13:57.29 | op3r | 1 rather |
13:57.33 | littleball | q? |
13:57.35 | littleball | what is it? |
13:57.52 | kay2 | mog: so why shouldnt it work |
13:58.36 | benjamin7062 | littleball, sweet. |
13:58.57 | littleball | firefly is goood becasue i got g729 dll file |
13:59.19 | Hmmhesays | care to share? |
14:00.00 | littleball | from where i can download sip phone q? i try to collect all free soft sip phones and recommend to my customers |
14:00.10 | littleball | why not? Hmmhesays |
14:00.15 | mog | lol |
14:00.31 | ManxPower | Oddly, I find tariffs interesting |
14:00.49 | littleball | http://www.mobmeee.com/portal/index.html?ctrl:cmd=render&ctrl:window=default.Forums.ForumsPortletWindow&op=showForum&f=10659 |
14:01.06 | littleball | hi, from where to download sip phone Q? |
14:01.36 | Hmmhesays | so I got my 32 channel mixer yesterday, I have no idea what to do with it |
14:02.07 | Hmmhesays | lots and lots of settings |
14:02.07 | fourcheeze | what kind of mixer? |
14:02.07 | mut | just download all kind of animal sounds and mix em |
14:02.09 | Hmmhesays | behringer 32 channel |
14:02.15 | mut | call it the rain forest in new york |
14:02.23 | mut | horns honking |
14:02.25 | mut | hun shots |
14:02.27 | mut | screams |
14:02.40 | mut | birds chirping and a nice waterfall |
14:02.59 | mut | gun shots too |
14:03.01 | fourcheeze | Hmmhesays: you got 32 sound sources? |
14:03.01 | mut | o_O |
14:03.10 | Hmmhesays | no, only about 20 |
14:03.14 | fourcheeze | cool |
14:03.16 | fourcheeze | what sort? |
14:03.42 | *** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
14:04.07 | fourcheeze | occasionally playing keyboard from the back of the room as well |
14:04.19 | Hmmhesays | 2 guitar, 1 bass, 3 voice mic's and about 10 on the drum set |
14:04.24 | fourcheeze | nice |
14:04.25 | Hmmhesays | so I guess thats about 16 |
14:04.31 | Hmmhesays | but.. the price was right |
14:04.39 | fourcheeze | you always need more |
14:04.49 | mut | MORE COWBELL! |
14:04.55 | Hmmhesays | yeah, room for expansion there, 32 channels and 4 sub groups for monitors |
14:05.05 | fourcheeze | well you might want things for effects to go in etc |
14:05.06 | mut | I NEED MORE COWBELL! |
14:05.21 | Hmmhesays | http://www.musiciansfriend.com/product/Behringer-Eurodesk-SL3242FXPRO-Mixer?sku=631246 |
14:05.27 | fourcheeze | yeah |
14:05.29 | Hmmhesays | its got 20 or so build in effects |
14:05.33 | fourcheeze | cool |
14:05.36 | fourcheeze | how much are those |
14:05.59 | fourcheeze | wow $600 |
14:06.00 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
14:06.18 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
14:06.44 | fourcheeze | Hmmhesays: well the 4 on the right are just groups anyway |
14:06.46 | benjamin7062 | I need to buy 50+ Polycom 601's... USA... Vendor suggestions before I just 'pick' on |
14:06.48 | benjamin7062 | one |
14:07.01 | Hmmhesays | yeah |
14:07.12 | Hmmhesays | what I can't figure out is where we put monitors on this thing |
14:07.23 | Hmmhesays | the old board had monitor outs |
14:07.29 | Hmmhesays | labeled "MONITOR OUT" |
14:07.39 | fourcheeze | generally there's a prefade / postfade out |
14:07.57 | Hmmhesays | I'm pretty new to the whole PA thing |
14:08.12 | fourcheeze | or you could use a group out |
14:08.17 | Hmmhesays | can you explain to me whats going to sound like a very n00b question? |
14:08.24 | fourcheeze | hehe |
14:08.55 | Hmmhesays | ok, the 1/4 in's on this say they can use balanced or unbalanced... I googled that a lot and got a whole lot of nothing |
14:09.21 | fourcheeze | hmm |
14:09.22 | fourcheeze | ok |
14:09.26 | [TK]D-Fender | benjamin7062 : 601's? All heavy volume users, or just a high budget? |
14:09.26 | Zeeek_ | like Canon connectors |
14:09.39 | Hmmhesays | I know all my 1/4 instrument cables are un balanced |
14:09.40 | fourcheeze | generally you would use the Canon XLRs for balanced |
14:09.44 | fourcheeze | those big 3 pin things |
14:09.49 | Hmmhesays | yeah... |
14:09.57 | fourcheeze | I'm guessing that there may be an option of using what would be a stereo 1/4 jack |
14:10.40 | Hmmhesays | look at the top of that board.... http://img3.musiciansfriend.com/dbase/pics/products/4/9/7/282497.jpg |
14:10.51 | fourcheeze | ok |
14:10.54 | ManxPower | Does anyone here know where I can find a list of CLECs that have service in a specific CO (CLLI)? |
14:10.54 | Hmmhesays | yeah there is.. that is what says "balanced or un balanced" |
14:11.05 | SpaceBass | Hey Hmmhesays long time no see |
14:11.09 | Hmmhesays | hey SpaceBass |
14:11.26 | SpaceBass | playing a lot these days? |
14:11.42 | Hmmhesays | yeah.. just about done getting the rest of the pa stuff |
14:11.55 | SpaceBass | fun |
14:11.56 | fourcheeze | Hmmhesays: balanced shold be those 3 pin things |
14:11.58 | Hmmhesays | is there an advantage to using either? |
14:12.04 | fourcheeze | yeah, use balanced |
14:12.07 | Hmmhesays | so there is no balanced 1/4 cables? |
14:12.10 | kay2 | Does anyone here know why asterisk doesnt use my file extension.conf when it loads ? is there any way from the CLI to see what config file it loaded ? |
14:12.11 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
14:12.31 | fourcheeze | Hmmhesays: if your mic or whatever terminates in an ordinary 1/4" jack it's unbalanced |
14:12.36 | *** join/#asterisk SHad|Work (n=kvirc@84.255.228.2) |
14:12.39 | fourcheeze | balanced needs 3 connections |
14:12.43 | SHad|Work | hi |
14:12.56 | Hmmhesays | kay2 because extension.conf is not right? |
14:13.00 | fourcheeze | and it basically eliminates interference down the line |
14:13.11 | Hmmhesays | ok |
14:13.20 | Hmmhesays | cause there is a ground |
14:13.21 | SHad|Work | Does anyone have any experience with configuring an octoBRI and quad GSM cards from Junghanns in the same machine? |
14:13.28 | Hmmhesays | so if I can use balanced then I should |
14:13.30 | SHad|Work | I can't figure out the zaptel/zapata configs |
14:13.36 | fourcheeze | 1/4" jack is signal + ground |
14:13.47 | Dr-Linux | hi all |
14:13.49 | fourcheeze | balanced XLR is -signal, ground and +signal |
14:13.53 | anthm | someone msg florin2703 with an example of a basic dialplan ivr |
14:14.05 | Dr-Linux | hi all |
14:14.07 | kay2 | Hmmhesays: what u mean it's not rght ? |
14:14.14 | Dr-Linux | anyone? i need some info about , that how can we handle dialplan using mysql database? |
14:14.32 | Vorondil | heh, i did that too once. global variables aren't set when they're in the "global" context and not the "globals" one. |
14:14.33 | fourcheeze | Hmmhesays: so basically what you need is a connector that goes into the end of your mic that terminates in a canon XLR |
14:14.37 | Hmmhesays | according to the behringer manual here, there are balanced 1/4's |
14:14.44 | SHad|Work | Dr-Linux: compile asterisk-addons and then use the MYSQL command |
14:14.46 | Hmmhesays | kay2: extensions.conf |
14:14.50 | fourcheeze | Hmmhesays: if they are then they must have 3 connections |
14:15.01 | Hmmhesays | and they do |
14:15.09 | fourcheeze | I don't see why anyone would use that rather than an XLR |
14:15.19 | fourcheeze | canon-style |
14:15.26 | fourcheeze | but maybe I'm too trad or something |
14:15.29 | Dr-Linux | SHad|Work: asterisk-addones are already compiled and i'm using Mysql DB for CDR reports .. |
14:15.45 | SHad|Work | Dr-Linux: what exactly would you like to do? |
14:15.57 | Dr-Linux | SHad|Work: but i'm not sure how can i handle dialplan using mysql db.. |
14:16.03 | Katty | Hmmhesays: hey hun (= |
14:16.08 | Hmmhesays | hey Katty |
14:16.10 | fourcheeze | Hmmhesays: as far as monitors go |
14:16.11 | Dr-Linux | SHad|Work: what things i need to do? |
14:16.20 | fourcheeze | I would use group 3/4 |
14:16.22 | SHad|Work | Dr-Linux: what exactly do you mean by "handle"? |
14:16.32 | Dr-Linux | SHad|Work: only i need mysql() in dialplan .. |
14:16.34 | Hmmhesays | fourcheeze why is that? |
14:16.46 | Dr-Linux | SHad|Work: wait dude, lemme confirm it from my manager. |
14:16.52 | fourcheeze | Hmmhesays: because it's there and you can fade them down together with main if required |
14:16.57 | fourcheeze | using 1 hand :-) |
14:17.06 | Hmmhesays | hmm ok |
14:17.09 | fourcheeze | Hmmhesays: what do you want monitor to do? |
14:17.24 | Hmmhesays | send me sound so I can hear what i'm playing |
14:17.46 | Hmmhesays | i want my guitar up above everyone else |
14:17.49 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:17.49 | Hmmhesays | in the monitor |
14:18.03 | fourcheeze | don't you have an amp for that? |
14:18.12 | fourcheeze | does it have anything called aux out? |
14:18.38 | Hmmhesays | yes however I need the rest of the band in the mix too |
14:18.57 | fourcheeze | ok, so do you want one set of monitors for the whole band? |
14:18.59 | Hmmhesays | this was easy on the hold mixer cause everything was labeled for idiots like me |
14:19.23 | Hmmhesays | no we had two channels set up, 1 for the bass and drummer 1 for the guitarists |
14:19.27 | fourcheeze | ok |
14:19.36 | fourcheeze | do you have things like aux1 and aux2 out or send ? |
14:19.47 | fourcheeze | probably 1/4" jacks |
14:19.58 | Hmmhesays | on my amp I have 1/4' line out |
14:20.18 | *** join/#asterisk dangerarea (n=kevin@podcast-audio.chrysalis.com) |
14:20.21 | fourcheeze | no, on the mixer |
14:20.35 | dangerarea | hey all |
14:20.45 | fourcheeze | I'm guessing that the first 2 pink knobs on each line are aux1/2 |
14:20.56 | dangerarea | anyone an expert on getting ztdummy to work on debian? |
14:20.58 | Hmmhesays | i'm not sure i haven't looked at it carefully enough I guess |
14:21.04 | Hmmhesays | dangerarea its a no brainer |
14:21.04 | fourcheeze | should send stuff out to a line out in the mixer output section |
14:21.08 | Joe__11 | Hi all! I trying to understand how chan_local.c (Local channel) works. Anybody knows why it uses two channels (owner and chan) not one? |
14:21.25 | fourcheeze | Hmmhesays: ahh yes look: http://media.zzounds.com/media/brand,zzounds/sl3242fx-b29baf30e8501cd84ec154272bc6c6df.jpg |
14:21.26 | Hmmhesays | I'm going to have to scourer the manual |
14:21.28 | dangerarea | Hmmhesays in a good or bad way :) |
14:21.50 | op3r | dangerarea: how about just use centos? |
14:21.51 | fourcheeze | Hmmhesays: look top right |
14:21.52 | Hmmhesays | dangerarea good |
14:22.05 | fourcheeze | from the right |
14:22.05 | Hmmhesays | holy hi rez batman |
14:22.14 | dangerarea | i'm having issues loading it in |
14:22.21 | fourcheeze | see where it says "aux send" |
14:22.32 | dangerarea | can't get my usb-uhci module to load |
14:22.45 | fourcheeze | Hmmhesays: pick one for you and one for the drummer |
14:23.04 | fourcheeze | Hmmhesays: then on each line in you set how much of that line to send to each aux out |
14:23.07 | fourcheeze | dead easy |
14:23.13 | Hmmhesays | with the pink knobs |
14:23.22 | fourcheeze | yeah |
14:23.24 | Katty | Hmmhesays: you framilier with mounting w2k3 shares? |
14:23.34 | Hmmhesays | Katty not really |
14:23.39 | Katty | Hmmhesays: m'kay |
14:24.06 | benjamin7062 | Katty, I assume mounting from unix? What is your problem? |
14:24.17 | Hmmhesays | fourcheeze i'm going to have to play with this tonight |
14:25.05 | SHad|Work | anyone here has any exerience with octoBRI or GSM cards? |
14:25.15 | SHad|Work | or any zaptel interfaces |
14:25.30 | fourcheeze | Hmmhesays: yeah, I wish I was :-) |
14:25.53 | Hmmhesays | I'm thinking of going with some kustom floor wedges... they are cheap |
14:25.56 | Hmmhesays | and the reviews are decent |
14:25.59 | [TK]D-Fender | Katty : "man smb"? The manpage gives a good quick 1-liner for it IIRC |
14:25.59 | benjamin7062 | Katty, are you trying to configure samba? Do you have active directory? etc? |
14:26.53 | benjamin7062 | Katty, if you want it to be a local mount point make sure you have the smbfs support for the kernel |
14:28.36 | SplasPood | Hrm.. is there any way to handle Background() within a macro? Ie.. I want to Record() something, then give them a menu asking what they wanna do... |
14:28.50 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
14:28.53 | benjamin7062 | Katty, if you are running debian... it's as easy as apt-get install smbfs |
14:28.56 | Katty | [TK]D-Fender: i can mount 2k shares fine. it's a 2k3 share that's giving me issues (= |
14:28.58 | Persilon | Hi |
14:29.14 | Persilon | I'm getting: pbx.c:1700 pbx_extension_helper: No application 'MeetMe' |
14:29.22 | Katty | [TK]D-Fender: i have details, if you'd like them and are framilier with mounting 2k3 shares. |
14:29.56 | nortex | Katty, Are you in a Windows domain or are the share on a domain controller? |
14:30.31 | [TK]D-Fender | Katty : No, don't know that much... did jsut do it with Samba and WinXP Pro shares. |
14:30.38 | *** part/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
14:30.53 | Katty | [TK]D-Fender: m'kay. |
14:30.56 | nortex | Katty, 2K3 introduced a specific security feature that cripples connections from DOS and Linux boxes |
14:31.18 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
14:31.18 | Katty | nortex: i'm sure i'll find someone to help me with it. |
14:31.18 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
14:32.09 | Persilon | *CLI> show application MeetMe --> Your application(s) is (are) not registered |
14:32.12 | nortex | Katty, Do you have access to the 2K3 box? I have changed the setting on ours and can give you the details as to what we found. |
14:32.22 | Persilon | is meetme bundled in stantard asterisk distribution ? |
14:32.37 | Katty | nortex: i've got access to every box in the house, bring it on (= |
14:32.37 | NotJohnDavid | espresso, oh how I love thee |
14:32.48 | *** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
14:35.00 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
14:35.01 | ManxPower | Persilon, Zaptel MUST be installed when you build Asterisk or MeetMe will not be built. |
14:35.01 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
14:35.42 | smackus | i have been trying for about a week now to get music on hold to work. I have reinstalled my asterisk and messed with about every little thing I can think of. When i put a call on hold, I get the following message: |
14:35.43 | Persilon | ManxPower: can I use meetme over a sip channel ? |
14:35.43 | smackus | <PROTECTED> |
14:35.43 | smackus | <PROTECTED> |
14:35.53 | smackus | it starts and stops instantly. |
14:35.56 | ManxPower | Persilon, Yes, but it requires Zaptel for timing |
14:36.00 | smackus | what could this be from? |
14:36.02 | dangerarea | anyone know how to get round this... |
14:36.02 | dangerarea | pbx:~# modprobe ztdummy |
14:36.02 | dangerarea | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
14:36.03 | dangerarea | <PROTECTED> |
14:36.06 | Persilon | ManxPower: ok, thank you |
14:36.24 | dangerarea | oops |
14:36.56 | [TK]D-Fender | smackus : Pastebin your musiconhold.conf |
14:36.58 | [TK]D-Fender | ~pb |
14:37.00 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
14:37.03 | *** join/#asterisk jj- (n=jj-@82.128.184.75) |
14:37.11 | jbalcomb | why does my syntax highlighting work one server and not the other? I've copied .bashrc and .vimrc. |
14:37.32 | nortex | Katty, Okay the setting is changed in the Domain Controller Security Settings > Local Policies > Secutiry Options. Look for Microsoft network server: Digitally sign communications (always) and disable it. |
14:40.08 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
14:40.36 | ManxPower | jbalcomb, does ldd /usr/sbin/asterisk show that libcurses is linked in? |
14:41.02 | ManxPower | or libncurses |
14:41.21 | dangerarea | http://pastebin.com/733145 |
14:41.25 | dangerarea | any suggestions? |
14:42.12 | jbalcomb | ManxPower: well, sometimes I don't think right until I ask. The working system has vim, the non-working system has nvi. |
14:43.09 | *** join/#asterisk MattH (n=matt@noc-wireless.chilitech.net) |
14:43.09 | smackus | sorry... dropped my network connection. |
14:43.13 | fourcheeze | Hmmhesays: what's the little button between the 2 sets of pink knobs ? |
14:43.24 | MattH | Hi.. does anyone know what the sip registration string should be to get an F3000 phone to connect to asterisk? I can't seem to get mine online |
14:43.30 | jbalcomb | So `apt-get install vim` fixed that problem. Hard to beleive a text editor should be 15.2 MB thoughh. |
14:43.30 | MattH | er.. rather it's online but I can't get it to register |
14:43.49 | Hmmhesays | "PRE" |
14:43.52 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
14:43.55 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
14:44.10 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
14:44.35 | smackus | here is my musiconhold.conf http://pastebin.ca/73156 |
14:45.03 | fourcheeze | Hmmhesays: ok that's probably a pre-fade listen thing to monitor the line |
14:45.11 | Hmmhesays | yeah I believe so |
14:45.21 | fourcheeze | in which case the 2 pink knobs below that will be post-fade auxes for effects |
14:45.36 | fourcheeze | it's amazing how little mixer design changes |
14:46.17 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
14:47.07 | jbalcomb | Grr.. what kinda bonehead forces an anti-virus definitions update during work hours for 100+ PCs? |
14:47.36 | Hmmhesays | your average computer science major graduate? |
14:47.38 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
14:48.44 | jbalcomb | Hmmhesays: perhaps, or gumptionless old admin who doesn't really know how to do anything? |
14:49.52 | jbalcomb | Hmmhesays: As I understand it, this guys main qualification is that he has no life and only lives five minutes from the building. |
14:49.53 | [TK]D-Fender | smackus : Got MP3's in that folder? Without ID3 tags I hope? Did you install asterisk-addons to provide MP3 support? |
14:50.12 | smackus | i did install asterisk-addons. |
14:50.14 | Hmmhesays | cool jbalcomb pefect admin |
14:50.36 | jbalcomb | [TK]D-Fender: What, no ID3 tages? You need ID3 tags to feed to festival and announce what your playing! |
14:50.48 | jbalcomb | Hmmhesays: Indeed. =) |
14:51.35 | *** join/#asterisk fugitivo (n=ajf@190.48.166.75) |
14:51.37 | fugitivo | hello |
14:51.42 | jbalcomb | [TK]D-Fender: oh wait, it says 'no ID3 tags' on the wiki.. Asterisk is no fun. |
14:51.50 | smackus | here is the content of my directory http://pastebin.ca/73160 |
14:52.26 | [TK]D-Fender | smackus : get rid of that "backup" file... |
14:52.45 | fugitivo | I need agents in a queue to have a limit of 2 outgoing calls and a limit of 1 incoming call, but I see that in asterisk 1.2.x there's only a call-limit parameter in sip.conf for both incoming and outgoings |
14:52.46 | [TK]D-Fender | smackus :err ..folder.. hmm.. shouldn't matter |
14:52.50 | jbalcomb | [TK]D-Fender Maybe you could set up a conference server so people can stream the satelite radio over the phone? |
14:53.10 | [TK]D-Fender | ... |
14:54.19 | fugitivo | any way to do that? |
14:54.31 | Katty | nortex: thanks for heads up on the local policy setting, but ours is already disabled. |
14:55.05 | Katty | nortex: would you like a pastebin? |
14:55.51 | nortex | Katty, Sure |
14:56.32 | Persilon | is there anyway of setting permissions on monitor recorded files as it records them ? |
14:59.01 | fugitivo | or a way to not hear the call waiting tone on queues |
14:59.36 | Katty | nortex: http://pastebin.ca/73165 |
15:00.03 | Katty | nortex: i have a feeling i know where i need to go with this... |
15:00.08 | Katty | nortex: but a second opinion is always handy. |
15:01.39 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
15:07.02 | fugitivo | so, isn't that possible? |
15:07.12 | *** join/#asterisk NLinington (n=nfl@82-69-27-212.dsl.in-addr.zen.co.uk) |
15:07.23 | nortex | Katty, I did mount -t cifs -o username=siusername,password='si(password' //192.168.0.3/asterisk /mnt/asterisk with one of my 2K3 boxes and it worked. |
15:07.56 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
15:08.03 | Tili | how can i force a callerid from sip.conf |
15:08.10 | Tili | i set callerid=whatiwant |
15:08.14 | Tili | but still i dont get that one |
15:08.54 | nortex | Katty, I am not sure, but the error about cifs not supported would be my guess as to the problem. But I'm more a windows person then linux. I'm running Centos which may make the difference. |
15:10.21 | EinsteinTaylor | morning all |
15:10.28 | EinsteinTaylor | someone pass the caffeine please |
15:11.00 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
15:13.25 | smackus | i have chmod'd and chowned the files and the directory in my mohmp3 directory and I still get: |
15:13.25 | smackus | <PROTECTED> |
15:13.26 | smackus | <PROTECTED> |
15:15.06 | lilalinux | does cdr_pgsql work with unix sockets, too? |
15:15.16 | lilalinux | or do I need to enable tcp/ip in postgresql? |
15:15.22 | Katty | nortex: i will look into cifs, like i expected then |
15:16.01 | Persilon | I need some help with ChanIsAvail... it doesn't jump n+101 |
15:16.53 | MACscr | is their any type of app or website that can be used to test the number of simultaneus phone calls your internet connection can handle with SIP? |
15:17.00 | smackus | any other ideas for the music on hold issue? |
15:17.15 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
15:17.20 | pdtmobile | smackus: i suggest native, just convert the stuff to gsm |
15:17.34 | pdtmobile | simple sox command and your done |
15:17.40 | smackus | does native not do mp3? |
15:17.41 | *** join/#asterisk skraelings001 (n=skraelin@201.230.140.95) |
15:17.59 | anthm | if you want it to it can |
15:18.02 | Katty | nortex: any thoughts on how to get cifs? |
15:18.04 | smackus | ok |
15:18.22 | smackus | so I am using native, and I have the directory pointed to the mohmp3 directory with 3 mp3 files in it |
15:18.28 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
15:18.29 | *** join/#asterisk __undef (i=uxjf@rzstud2.stud.uni-karlsruhe.de) |
15:18.30 | skraelings001 | Hi everyone |
15:18.31 | anthm | I made format_mp3 in asterisk-addons you can install |
15:18.34 | __undef | hi |
15:18.36 | fugitivo | MACscr: google for voip bandwidth calculator |
15:18.47 | anthm | but you are needlessly decoding it over and over again fyi |
15:18.56 | MACscr | thanks Fugitivo |
15:19.03 | pdtmobile | thats why I suggest a one time convert to gsm and call it a day |
15:19.10 | pdtmobile | lower processor overhead blah blah |
15:19.19 | anthm | or convert to slin |
15:19.28 | pdtmobile | ya |
15:19.33 | anthm | or if you are really stingy convert it to every codec |
15:19.38 | pdtmobile | hehe |
15:19.39 | Katty | nortex: actually, this is a local box. |
15:19.40 | anthm | then it will pick the one that matches your channel |
15:19.42 | coppice | just don't keep converting. |
15:19.48 | Katty | nortex: cifs shouldn't even be needed. |
15:20.30 | Persilon | I'm getting app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
15:20.35 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
15:20.57 | MACscr | crap, i dont know what algorithm to pick. Is their any particular standard for SIP that is most widely used? |
15:20.57 | smackus | so if i am doing native, and i have all the files, and it is still not working, where else can I look? |
15:21.34 | mut | g711 ulaw |
15:21.48 | anthm | not working how, I made the original cut of native music but I can't say they didn't change it since then |
15:22.01 | smackus | <PROTECTED> |
15:22.01 | smackus | <PROTECTED> |
15:22.04 | smackus | never plays anything |
15:22.12 | smackus | starts and stops in an instant |
15:22.22 | anthm | is there still the cli command to list the files ? |
15:22.37 | anthm | moh files show |
15:22.45 | smackus | ok, hang on |
15:23.00 | smackus | shows nothing. |
15:23.15 | smackus | that would explain why it does not play |
15:23.26 | smackus | but how do i explain why nothing shows in moh files show? |
15:23.52 | anthm | I think they changed the way you config lately |
15:23.56 | anthm | did you have it a while ? |
15:24.17 | smackus | I am installing from scratch. |
15:24.34 | anthm | you said they are mp3? |
15:24.36 | smackus | yes |
15:24.41 | anthm | if you dont have format_mp3 loaded |
15:24.50 | smackus | in modules? |
15:24.51 | anthm | they will be skipped since they are invalid |
15:24.57 | __undef | is there any way to generate a call on a pri_net span? |
15:25.09 | blitz[laptop] | __undef: from where? |
15:25.15 | anthm | there is an initial pass to verify the extensions against the various formats you have loaded |
15:25.18 | __undef | blitz[laptop]: from the host itself ;) |
15:25.24 | blitz[laptop] | __undef: callfiles |
15:25.36 | __undef | blitz[laptop]: okay, thanks... |
15:25.40 | blitz[laptop] | np |
15:26.17 | anthm | so either install format_mp3 or sox them to raw or cp some of those stock audio files as a test |
15:26.20 | smackus | show modules lists: format_mp3.so |
15:26.47 | anthm | if that's the case you must have something configed wrong |
15:27.05 | smackus | i would assume so, where can we start looking it over. |
15:27.13 | anthm | look at the console when it loads it |
15:27.25 | anthm | if you wnat to make it easier |
15:27.37 | anthm | add noload => res_musiconhold.so to the modules.conf |
15:27.47 | smackus | here is my musiconhold.conf http://pastebin.ca/73156 |
15:27.48 | anthm | then start with debug log on and lots of v |
15:27.59 | anthm | then load it from the cli so you can see the msgs |
15:29.21 | anthm | my guess is moh loads before mp3 does |
15:29.30 | Hmmhesays | i'm kind of like this FC5 for a Workstation |
15:29.32 | anthm | so it doesnt know it's a valid format till it's too late |
15:29.35 | Hmmhesays | it is pretty |
15:29.44 | anthm | so you can stick in manual load => lines at the top |
15:29.54 | anthm | so you load all your file formats first |
15:29.58 | anthm | then the music |
15:30.03 | Hmmhesays | do i need to load res_crypto if i'm not using any of the md5 stuff? |
15:30.24 | smackus | anthm: ok, hang on |
15:30.43 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
15:30.44 | anthm | isnt res_crypto one of those lazy linked ones that other code calls stuff from? |
15:30.46 | smackus | i have added the noload => res_musiconhold.so |
15:30.52 | Hmmhesays | i'm really not sure |
15:31.03 | anthm | that will prove it |
15:31.04 | Hmmhesays | its sure been a pain in the ass for me trying to compile for mipsel though |
15:31.07 | anthm | so if you start up now |
15:31.15 | anthm | and it works by hand loading music |
15:31.20 | Tili | i can't seem to override callerid by setting it in iax.conf or sip.conf |
15:31.35 | smackus | this is where you lost me... and it works by hand loading music |
15:31.53 | anthm | enter >load res_musiconhold.so |
15:31.56 | anthm | at cli |
15:32.01 | anthm | aka hand loading it |
15:32.09 | smackus | ok |
15:32.35 | anthm | because you are loading it after everything else |
15:32.45 | anthm | it will probably work now |
15:33.08 | smackus | slk-apbx-01*CLI> load res_musiconhold.so |
15:33.08 | smackus | Unable to load module res_musiconhold.so |
15:33.08 | smackus | Jun 27 09:32:45 WARNING[7801]: loader.c:305 __load_resource: Module 'res_musiconhold.so' already exists |
15:33.33 | anthm | then you didnt noload it right |
15:33.41 | smackus | ok... hang on, |
15:34.00 | smackus | noload => res_musiconhold.so |
15:34.03 | smackus | right? |
15:34.04 | Qwell | did you restart? |
15:34.14 | smackus | I reloaded... i must restart? |
15:34.17 | anthm | yes |
15:34.19 | smackus | ok... |
15:34.20 | Qwell | well..yeah |
15:34.28 | smackus | that will take some time. I will get back with you |
15:34.37 | Qwell | not the whole machine |
15:34.48 | smackus | right... just have alot of callers on it right now |
15:34.54 | anthm | if you cant do that then do moh reload |
15:34.57 | anthm | see if that woeks |
15:35.01 | anthm | it may rescan the files |
15:35.03 | smackus | did a restart when convenient |
15:35.30 | CunningPike | Heh - if I did a 'restart when convenient' now, I could wait for 12 hours........ |
15:35.44 | anthm | try "moh reload" |
15:35.48 | smackus | ok |
15:36.05 | anthm | you are just trying to prove it's the load catch-22 |
15:36.32 | anthm | most likely you will need to add a bunch of .... load => format_* |
15:36.37 | anthm | for all the formats |
15:36.39 | smackus | must not have reloaded, cuz i got the same error |
15:36.48 | anthm | then load => res_musiconhold |
15:36.54 | NLinington | Hi, has anybody got faxing working using the app_rxfax module? |
15:36.57 | anthm | so you are sure you have all the file formats loaded before music |
15:37.09 | smackus | no |
15:37.26 | smackus | how do i make sure? |
15:37.49 | anthm | did moh reload say it reload |
15:37.55 | anthm | then try moh show files again |
15:38.07 | anthm | moh files show i mean |
15:38.39 | smackus | now there are files. |
15:39.07 | anthm | so then it should work now |
15:39.11 | smackus | yes it does. |
15:39.23 | smackus | so just adding the noload was the fix? |
15:39.26 | anthm | no |
15:39.33 | anthm | the fix will be |
15:39.39 | anthm | adding actual load lines |
15:39.46 | smackus | ok |
15:39.57 | smackus | walk me through this, because I am a little confused. |
15:40.12 | smackus | so do i need to take out the noload line? |
15:40.35 | anthm | ya |
15:40.37 | anthm | take it out |
15:40.41 | smackus | ok |
15:40.53 | smackus | done |
15:41.37 | anthm | in a shell |
15:41.48 | anthm | cd /usr/lib/asterisk/modules |
15:42.00 | stephane_ | soir |
15:42.03 | MACscr | is their managed asterisk hosting out there? |
15:42.07 | anthm | execute /bin/ls -1 format_* | awk '{print "load => "$1}' |
15:42.16 | MACscr | using that particular text, i didnt fine much with google |
15:42.18 | smackus | done |
15:42.19 | anthm | and cut and paste the output to the top of modules.conf |
15:42.44 | anthm | then add load => res_musiconhold.so under that |
15:42.57 | anthm | that should do it |
15:43.59 | smackus | ok, thanks |
15:44.04 | smackus | testing now. |
15:44.14 | anthm | this way you load all the formats first |
15:44.17 | anthm | then music |
15:44.21 | mut | LMFAO! |
15:44.22 | mut | http://www.youtube.com/watch?v=2S89Y4shxtE&search=Maury%20Show%20phobia |
15:44.27 | anthm | so they do not skip any |
15:44.39 | anthm | cos say format_mp3 is loaded after music |
15:44.51 | smackus | thank you very much |
15:44.55 | anthm | then music will skip all mp3 as invalid cos it's not loaded yet |
15:44.57 | anthm | np |
15:44.59 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
15:46.17 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
15:48.06 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
15:48.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.230.137) |
15:52.50 | *** join/#asterisk monkeyshine (n=countjas@216.64.160.84) |
15:52.54 | monkeyshine | i need some help |
15:53.43 | monkeyshine | my sipura boxes are acting as a dhcp server and my internet line has a different gateway |
15:53.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.219.203) |
15:53.58 | monkeyshine | can i bridge 2 gatgeways |
15:54.14 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
15:59.33 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
15:59.50 | generalhan | whats up all ! |
16:01.03 | NLinington | my * box is doing a seg fault every time I call app_rxfax, has anybody some idea how to fix this? |
16:01.31 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
16:01.38 | h3x | nling: are you doing t.30 with it? |
16:02.07 | NLinington | h3x nope, I have a single pstn inbound line I have fax detection on |
16:02.24 | h3x | I would say thats T.30 :) |
16:02.49 | NLinington | oops |
16:03.05 | NLinington | thought you meant fax over voip |
16:03.11 | h3x | can you tell if its really app_rxfax that causes the fault or is it fax detection |
16:03.24 | h3x | that is t.37 or t.38 |
16:03.39 | h3x | or t.30 over g711 :) |
16:04.06 | {zombie} | NLinington: make sure the version of spandsp (libtiff) used to compile app_rxfax is the same as is installed on your system |
16:04.10 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
16:04.16 | h3x | that too |
16:04.17 | {zombie} | oh, and that app_rxfax was compiled for your asterisk |
16:04.22 | NLinington | the fax detection works ok as I can see the script running ok and it plays a wav file saying 'fax detected' |
16:04.27 | monkeyshine | is there a way to bridge gateway |
16:04.33 | *** join/#asterisk skirmisha (n=vlado@87.126.55.7) |
16:04.41 | skirmisha | hi guys |
16:04.59 | NLinington | I have completly re-compiled asterisk, spansp, all the libe and reinsrtalled from scratch |
16:05.11 | *** part/#asterisk fourcheeze (n=rich@82.153.215.21) |
16:05.16 | skirmisha | can someone tell me where the asterisk PBX comes from in the email header under Sender field when i send voicemail msg |
16:05.26 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
16:05.35 | sonic69 | i need a script that will delete voicemail messages folder when i delete an account in php interface... can someone help me with that?? |
16:05.39 | {zombie} | skirmisha: voicemail.conf |
16:06.04 | skirmisha | it's not from there |
16:06.13 | skirmisha | i've been playing 2 days |
16:06.14 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
16:06.31 | skirmisha | and it comes from somewhere and i don;t know how to rewrite it with mail server |
16:06.35 | {zombie} | ; Change the From: string |
16:06.35 | {zombie} | fromstring=Asterisk PBX |
16:06.46 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
16:06.49 | {zombie} | that's where I changed mine |
16:07.06 | {zombie} | did you restart asterisk after making the change? |
16:07.14 | skirmisha | yes i did |
16:07.27 | h3x | sonic: I'd hang on to the voicemails for extortion purposes :) |
16:07.29 | skirmisha | the problem is that from string is ok , but sender string is not |
16:07.52 | skirmisha | Sender: asterisk PBX <voicemail@ |
16:07.56 | jbalcomb | is pastebin broken? |
16:08.05 | skirmisha | this is what i get in mail header |
16:08.10 | h3x | sonic: actually you should set up a cronjob to delete voicemails on accounts that were deleted 30 days ago or something |
16:08.49 | jbalcomb | h3x: can you send us yours? |
16:09.10 | h3x | my what? |
16:09.17 | jbalcomb | your cronjob |
16:09.22 | h3x | i dont have one |
16:09.28 | jbalcomb | why not? |
16:09.41 | h3x | coz i have better customer retention than you do ? hahaha |
16:10.09 | skirmisha | anyone who can help me? |
16:10.09 | jbalcomb | haha.. yeah, that's funny. |
16:10.19 | h3x | actually i dont have many customers that have voicemail |
16:10.43 | _problem_ | skirmisha: whats ur problem u were been told by {zombie} |
16:10.58 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
16:11.11 | _problem_ | skirmisha: thats what i did what he says and it works for me also |
16:11.21 | skirmisha | {zombie} told me to check in the config file |
16:11.27 | skirmisha | but the problem comes from source |
16:11.30 | tRSS | do we have anyone interested in developing a fully functional inbound/outbound predictive dialer on top of asterisk? |
16:11.32 | CunningPike | skirmisha: Did you try sendmail.cf like I suggested? |
16:11.41 | generalhan | hey can someone explain to me how to get information from the CDR and keep it in that format? |
16:11.44 | _problem_ | skirmisha: its all in the voicemail.conf |
16:11.46 | generalhan | yea htat sounded real clear ... lol |
16:11.48 | {zombie} | skirmisha: I don't get a "Sender:" field, so your MTA must be adding that |
16:11.54 | skirmisha | CunningPike yes i did manage to chamge the address |
16:12.05 | _problem_ | skirmisha: pastebin ur voicemail.conf |
16:12.19 | skirmisha | but it does not change related info |
16:12.30 | generalhan | Like if i use an "fgrep something" in that file it comes out in just text so its hard to read ... i need to be able to do that but keep it in a .csv format |
16:12.30 | CunningPike | skirmisha: So your Sender: header is now correct? |
16:12.38 | skirmisha | it comes as Sender: asterisk PBX <voicemail@your domain> |
16:12.53 | skirmisha | i want to change this asterisk pbx |
16:13.03 | skirmisha | i did managed to change the email |
16:13.04 | {zombie} | is your MTA grabbing that from /etc/passwd maybe? |
16:13.14 | skirmisha | i am using exim |
16:13.26 | *** join/#asterisk oej (n=olle@212.17.152.81) |
16:13.37 | {zombie} | try "chfn asterisk" or "chfn voicemail" |
16:14.52 | skirmisha | {zombie} where should i type this |
16:15.07 | {zombie} | from the linux commandline (as root) |
16:17.30 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
16:18.00 | skirmisha | ahhh here it is |
16:18.03 | skirmisha | thanks guys |
16:18.17 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
16:18.22 | TripleFFFF | : is there a way to sip debug a peer to file |
16:18.29 | skirmisha | damn mail server |
16:18.31 | TripleFFFF | thanks.. |
16:19.47 | jbalcomb | TripleFFFF: maybe asterisk -rc sip debug peer <exten> > <file> ? |
16:19.59 | TripleFFFF | oh |
16:20.22 | wunderkin | you are thinking of x not c, and no |
16:20.40 | TripleFFFF | ? |
16:20.56 | TripleFFFF | so that would not work |
16:20.56 | TripleFFFF | hmm |
16:21.09 | TripleFFFF | would be frigign usefull to imlpement.. loger.conf. |
16:21.18 | TripleFFFF | debugpeer1=username |
16:21.20 | TripleFFFF | ro somethign |
16:21.27 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
16:21.47 | NotJohnDavid | Wish a decent Socket A motherboard was cheaper than they are currently |
16:22.23 | jbalcomb | NotJohnDavid: what CPU is Socket A for? |
16:22.26 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
16:22.49 | *** join/#asterisk burnproof (n=burnproo@210.213.199.85) |
16:22.55 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:23.29 | NotJohnDavid | AMD Athlon |
16:24.20 | jbalcomb | ah, I have several Socket A boards then.. |
16:24.31 | TripleFFFF | is there a way to put a chanel on hold without a hold button ? |
16:24.33 | TripleFFFF | like *99 |
16:24.37 | TripleFFFF | but how to get back ? |
16:24.40 | TripleFFFF | *99 again ? |
16:25.31 | [TK]D-Fender | TripleFFFF : Depends on the phone |
16:25.35 | TripleFFFF | normal phone |
16:25.41 | TripleFFFF | no hold button |
16:25.42 | {zombie} | hookflash |
16:25.45 | TripleFFFF | soyeah |
16:25.45 | [TK]D-Fender | TripleFFFF : Connected how? |
16:25.49 | TripleFFFF | pap2 |
16:26.07 | [TK]D-Fender | TripleFFFF : PAP2 should have a * code feature built in for that like the SPA series does |
16:26.17 | skirmisha | {zombie} thank all is working fine now |
16:26.21 | {zombie} | cool |
16:26.23 | [TK]D-Fender | TripleFFFF : Download the admin/user guide |
16:26.31 | TripleFFFF | lol |
16:26.38 | TripleFFFF | linksys pap2 na too ? |
16:27.06 | {zombie} | TripleFFFF: hookflash to put them on hold, hookflash again to unhold. if you hookflash then dial a number you can transfer the call. |
16:28.34 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
16:29.04 | Greek-Boy | I'm trying to record all calls/channels on my asterisk box. How do I go about it? I had a look at http://www.voip-info.org/wiki/view/Monitor+stereo-example but it didn't work for me. I need to record all outgoing calls from every extension |
16:29.59 | skirmisha | there is config file Greek-Boy |
16:30.25 | Greek-Boy | which conf file? |
16:30.45 | skirmisha | let me check it |
16:31.17 | skirmisha | futures.conf |
16:32.15 | *** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140) |
16:32.25 | skirmisha | automon=*1 |
16:32.42 | TripleFFFF | hey |
16:32.47 | TripleFFFF | whats the no bridge option ? |
16:33.08 | TripleFFFF | level 1: start=2006-06-27 12:23:10 |
16:33.08 | TripleFFFF | level 1: answer=2006-06-27 12:23:21 |
16:33.08 | TripleFFFF | level 1: end=2006-06-27 12:23:21 |
16:33.13 | TripleFFFF | and channel still up |
16:35.38 | TripleFFFF | so i can |
16:35.40 | TripleFFFF | know |
16:36.02 | lilalinux | does anyone use twinkle as a sip client/ |
16:36.30 | TripleFFFF | <PROTECTED> |
16:36.35 | TripleFFFF | so how can i know bridge |
16:36.39 | TripleFFFF | un bridge |
16:38.31 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
16:41.47 | *** join/#asterisk dacleric (n=dacleric@p54822D4E.dip0.t-ipconnect.de) |
16:44.17 | dlynes_office | twinkle, twinkle little star... |
16:44.32 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
16:45.46 | *** join/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
16:48.19 | lilalinux | dlynes_office: do you know if twinkle supports md5 passwords? |
16:48.30 | dlynes_office | lilalinux: no idea |
16:48.38 | dlynes_office | lilalinux: i don't even know what it is |
16:48.48 | lilalinux | a sip client |
16:48.55 | dlynes_office | ah |
16:49.11 | dlynes_office | sounds like a sip client for fairies |
16:49.45 | Greek-Boy | what is wrong with this? |
16:49.45 | Greek-Boy | exten => 301,1,Dial(SIP/kc-cashdesk01,,r) |
16:49.45 | Greek-Boy | exten => 301,2,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP}) |
16:49.45 | Greek-Boy | exten => 301,3,Monitor(wav,${CALLFILENAME},m |
16:49.53 | Greek-Boy | ? |
16:50.05 | Greek-Boy | i'm not getting any output files |
16:50.27 | Greek-Boy | I also have MONITOR_EXEC=/usr/local/bin/2wav2mp3 under my globals as a custom script |
16:50.27 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
16:50.39 | nortex | Greek-Boy, No closing ) on the monitor command |
16:50.47 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
16:51.05 | *** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-10.perspektivbredband.net) |
16:52.04 | Greek-Boy | what should permissions be on /usr/local/bin/2wav2mp3? |
16:52.23 | nortex | What are some typical causes of one sided audio between PRI channels and SIP clients? |
16:52.33 | h3x | NAT |
16:52.40 | h3x | double NAT |
16:52.43 | h3x | heh |
16:53.36 | darkskiez | firewall |
16:54.06 | darkskiez | broken microphones/earpieces |
16:54.16 | darkskiez | volume settings |
16:54.33 | Greek-Boy | I now have: |
16:54.34 | Greek-Boy | exten => 301,1,Dial(SIP/kc-cashdesk01,,r) |
16:54.34 | Greek-Boy | exten => 301,2,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP}) |
16:54.34 | Greek-Boy | exten => 301,3,Monitor(wav,${CALLFILENAME},m) |
16:54.44 | Greek-Boy | but still no output files |
16:55.00 | Greek-Boy | what could be wrong? |
16:55.12 | vader-- | anyone have any issues where their sip phones won't do anything in the features.conf? |
16:55.14 | vader-- | like *1 for recording and stuff |
16:55.20 | vader-- | i have cisco 7940G phones |
16:55.26 | nortex | h3x, No nat or firewall present between Server and client. |
16:55.32 | darkskiez | vader--: make sure the right params enable that in the dial options |
16:56.19 | *** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
16:56.50 | *** part/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
16:57.12 | *** join/#asterisk heath__ (n=heath__@71-87-34-39.dhcp.stcd.mn.charter.com) |
16:57.31 | heath__ | how new does one's version have to be for atxfer to work in features.conf? |
16:57.47 | MACscr | why are so many voip providers calling Auto Attendants, IVRs |
16:57.55 | *** part/#asterisk skraelings001 (n=skraelin@201.230.140.95) |
16:57.55 | h3x | heath lives |
16:58.02 | MACscr | it doesnt make sense |
16:58.36 | vader-- | darkskiez you mean in the extensions.conf? |
17:01.06 | rnovotny22 | Can incoming PSTN and SIP calls be routed to the same context? |
17:01.08 | TripleFFFF | so .. |
17:01.19 | TripleFFFF | if you have a normal 1 sip phone pap2 |
17:01.30 | TripleFFFF | and got 4 lines on sip account.. how can you have callwaiting ? |
17:05.01 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-145-155.gdrpmi.dsl-w.verizon.net) |
17:07.10 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
17:07.34 | variable_office | musiconhold immediatly stops for me, but reports no errors, what could be causing this? |
17:08.45 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
17:08.51 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
17:08.52 | dlynes_office | variable_office: your organ grinder monkey died |
17:09.19 | variable_office | ic |
17:09.56 | variable_office | really though, i have asterisk-addons and sounds installed, and i am just trying to use the default class that comes with the sample conf file |
17:11.05 | dlynes_office | variable_office: you don't believe me? |
17:11.09 | dlynes_office | variable_office: i'm hurt :(( |
17:11.28 | coppice | I thought it was the RIAA who stopped it |
17:11.36 | *** part/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
17:11.45 | dlynes_office | coppice: yeah...probably...found out he had metallica mp3s on his moh |
17:12.11 | coppice | the system itself should stop that to reduce unneccessary suffering |
17:12.45 | nortex | coppice, Some would say that about the default mp3s |
17:13.03 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
17:13.11 | Stephnie | hi |
17:13.24 | variable_office | no ideas then i take it eh? |
17:13.31 | nortex | Stephnie, hello |
17:14.07 | TripleFFFF | wonders how a girl naick gets attention ;) |
17:14.46 | nortex | TripleFFFF, I could answer her question :) |
17:15.28 | *** join/#asterisk pengyong (n=lala@218.93.154.125) |
17:15.31 | TripleFFFF | ;) |
17:15.44 | TripleFFFF | i could answer her needs';) |
17:15.48 | TripleFFFF | j/k |
17:16.03 | TripleFFFF | anyone have pap2 t na ? |
17:16.16 | TripleFFFF | im getting probs with the 2nd line ref |
17:16.17 | TripleFFFF | reg |
17:17.01 | Stephnie | hi |
17:17.09 | nortex | TripleFFFF, don't have one, but are both lines try to reg using the 5060 port. |
17:17.40 | TripleFFFF | no |
17:17.42 | TripleFFFF | 5061 |
17:17.44 | TripleFFFF | for 2 |
17:17.46 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
17:18.04 | TripleFFFF | and is Bridged Call(SIP/blah) normal ? |
17:18.07 | mdiehl | Hi all. |
17:18.08 | TripleFFFF | bridged i mean |
17:18.11 | TripleFFFF | hi |
17:18.25 | nortex | hi |
17:18.39 | mdiehl | Got a question. |
17:19.09 | nortex | TripleFFFF, You should see bridged call when asterisk connects the two parts of the call |
17:19.16 | Qwell[] | mdiehl: we've got dumbfounded looks |
17:19.17 | nortex | mdiehl, Fire away |
17:19.20 | mdiehl | I just removed a tdm card from my * server, now call transfer doesn't work, voicemail doesn't work. |
17:19.31 | mdiehl | And conferenceing STILL doesn't work. |
17:19.44 | mdiehl | I've got ztdummy loaded. |
17:20.03 | mdiehl | ...says that voicemailbox can't accept new messages. |
17:20.04 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
17:20.09 | FlyboySR22 | g us for it. |
17:20.24 | TripleFFFF | k |
17:20.44 | TripleFFFF | so ..line 1 connect.. line 2 cant connect to server... |
17:20.44 | TripleFFFF | if i make line 2 login onto line 1 ..it sworks |
17:20.46 | Dr-Linux | question about spa 2100, what should be "NAT Keep Alive Intvl:?" by default it's 15 |
17:20.54 | TripleFFFF | so the pap just doens want to use line 2 reg |
17:20.57 | mdiehl | Things wer working better with the card installed.... eventhough it was sharing an irq with 3 nic's. |
17:21.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.196) |
17:21.13 | Dr-Linux | sometime line goes dead, anybody can tell me if this is a reason? |
17:21.26 | rob0 | Qwell[]: you work for Radio Shark? :) |
17:22.15 | mdiehl | Any ideas as to where to begin to fix this? |
17:22.49 | dlynes_office | variable_office: nah...it's not that...i was just busy in the office :p |
17:22.59 | rob0 | Problem with IRC: the dumbfounded looks are hard to communicate. Is there an emoticon for that? |
17:23.07 | Qwell[] | rob0: ":)" |
17:23.11 | nortex | TripleFFFF, can you pastebin your sip.conf file? |
17:23.30 | Stephnie | http://pastebin.ca/73259 . . . .anyone? |
17:24.09 | rob0 | No, that's *happy* -- when you catch a supposed tech person who's pretending, they're seldom happy about that. |
17:24.28 | dlynes_office | variable_office: can you give me a bit more information as to what your moh environment is like? i.e. what's your configuration, ...? |
17:25.32 | TripleFFFF | nortex not really |
17:25.40 | TripleFFFF | i dfoudn hte prob |
17:25.40 | TripleFFFF | thanks |
17:25.44 | Stephnie | anyone using ASTCC? |
17:25.48 | _problem_ | Stephnie: there are many prepaid proprietary solutions available to use with asterisk ..u can check them on voip-info |
17:26.19 | vader-- | anyone have any issues where their sip phones won't do anything in the features.conf? |
17:26.21 | vader-- | i have cisco 7940G phones |
17:26.21 | nortex | TripleFFFF, Good deal! |
17:26.22 | Stephnie | _problem_ : what about astcc? |
17:26.31 | rnovotny22 | Can incoming PSTN and SIP calls be routed to the same context in extensions.conf? |
17:26.37 | TripleFFFF | yes |
17:26.46 | nortex | rnovotny22, yep |
17:26.48 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
17:27.02 | _problem_ | Stephnie: no i dont know abt that..never used that |
17:27.02 | TripleFFFF | just make the ztconf etc use same exten |
17:27.02 | TripleFFFF | lol |
17:27.10 | dlynes_office | _problem_: i don't think she's really wanting to know what's available...just what's reliable for a heavy load, specifically astcc, and if not astcc, then an equivalent |
17:27.13 | TripleFFFF | so... |
17:27.21 | TripleFFFF | zap i mean |
17:27.37 | _problem_ | dlynes_office: ok |
17:27.38 | Stephnie | dlynes_office : yeah .. |
17:28.08 | dlynes_office | Stephnie: astcc's just a perl script, right? |
17:28.42 | Stephnie | dlynes_office : right....and in my production server I'll use AGI with Perl...is it gud? |
17:29.08 | Stephnie | AGI + PERL + MYSQL ..... |
17:29.08 | dlynes_office | Stephnie: yeah, should be fine |
17:29.12 | dlynes_office | Stephnie: but lose the mysql |
17:29.23 | dlynes_office | Stephnie: use postgresql instead...it's much more robust |
17:29.36 | Stephnie | I thought to go for MS-SQL |
17:29.37 | dlynes_office | Stephnie: as soon as you start doing any kind of joins, you'll see what i mean |
17:29.47 | dlynes_office | Stephnie: mssql's fine, too |
17:29.49 | Stephnie | aaa aou....okey |
17:29.56 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
17:30.07 | dlynes_office | Stephnie: but if you want to handle a lot of traffic, and you want to be able to handle business logic |
17:30.18 | TripleFFFF | anyone familiar with cdrool froma g crap |
17:30.20 | TripleFFFF | ? |
17:30.21 | dlynes_office | Stephnie: imho, mysql is a horrible database; it's not terribly scalable |
17:30.30 | TripleFFFF | dlynes_office what ? |
17:30.33 | *** join/#asterisk skraelings001 (n=skraelin@201.230.140.95) |
17:30.37 | dlynes_office | TripleFFFF: cdrool? |
17:30.38 | TripleFFFF | dlynes_office use clustering engine |
17:30.38 | salaud | Anyone here have experience with H.323 and asterisk? I could use some help interfacing IAX channels and H.323 |
17:30.45 | TripleFFFF | dlynes_home yes |
17:30.46 | dlynes_office | TripleFFFF: you didn't listen to me |
17:30.51 | Stephnie | I see!! then ? u suggest me to use POSTGRESQL |
17:30.53 | TripleFFFF | it's not terribly scalable |
17:30.55 | dlynes_office | TripleFFFF: mysql's join support is absolutely horrible |
17:30.55 | h3x | sala: try Yate for that |
17:31.01 | TripleFFFF | oh |
17:31.02 | TripleFFFF | yeah |
17:31.06 | dlynes_office | TripleFFFF: you only listened to half of what i said |
17:31.10 | TripleFFFF | any db sucks.. imho |
17:31.13 | rnovotny22 | nortex: Thanks, having problems with incoming sip calls getting busy all the time. |
17:31.15 | h3x | Supposedly Yate does a better job with H.323 |
17:31.28 | salaud | h3x: Yate? |
17:31.30 | TripleFFFF | rnovotny22 .. what provider |
17:31.31 | dlynes_office | Stephnie: I only suggested postgresql because i assumed you wanted to go opensource |
17:31.33 | rpm | i can't find the kb article that talks about having multiple digium cards in a single machine, can someone point me to the url? |
17:31.35 | TripleFFFF | what hardware |
17:31.36 | salaud | h3x: A different PBX? |
17:31.36 | h3x | but personally I would use H.323 to SIP than H.323 to IAX |
17:31.37 | dlynes_office | Stephnie: mssql's fine, too |
17:31.41 | h3x | Sort of yes |
17:31.47 | TripleFFFF | rpm dont do it |
17:31.57 | dlynes_office | Stephnie: postgresql is about the closest you can get to oracle in the opensource world |
17:32.07 | rnovotny22 | TripleFFFF: BroadVoice |
17:32.10 | h3x | whoracle |
17:32.20 | Stephnie | dlynes_office : thanks .... |
17:32.23 | TripleFFFF | and a sip ? or asterisk |
17:32.24 | salaud | h3x: My thing is that I have an existing IAX channel structure to VOIP providers and I just need to do H.323 with another phone system |
17:32.25 | nortex | rnovotny22, are you registering to your SIP provider? |
17:32.36 | TripleFFFF | sip debug peer (PERNAMEOFBROADVOICE) |
17:32.45 | TripleFFFF | then make call |
17:32.49 | dlynes_office | Stephnie: basically what you want is something scalable, and that can handle a heavy load |
17:32.49 | TripleFFFF | check to see hwat happens |
17:33.02 | h3x | well keep in mind that SIP and H.323 both use RTP |
17:33.04 | h3x | and IAX2 dosent |
17:33.06 | dlynes_office | Stephnie: oracle is probably total overkill for you, though |
17:33.08 | TripleFFFF | me thinks stephanie looking for scratch card calling card solution |
17:33.13 | rnovotny22 | nortex: yes, current config is incoming on pstn, outgoing on sip. Would like to use sip for incoming also. |
17:33.25 | Stephnie | dlynes_office : so do u think that using AGI + PERL + POSTGRESQL or MYSQL is a professional way to use in production server for heavy traffic as they wont mess up the billing...... |
17:33.28 | salaud | h3x: I'm trying to put asterisk in the middle to do the translation layer... |
17:33.42 | salaud | h3x: The solution only has to work for a few weeks until some PRI's come in |
17:33.46 | dlynes_office | Stephnie: like i said...take mysql out of the solution...it's horrible for business logic |
17:33.51 | TripleFFFF | http://www.xtenn.com/ |
17:33.58 | TripleFFFF | WOW.. nice copryight issue |
17:33.58 | h3x | so stick yate in the middle of it |
17:34.09 | h3x | or find some other software that does h.323 -> sip like ummm |
17:34.10 | rnovotny22 | dlynes_office: Whats wrong with mysql. It rivals oracle and is open source? |
17:34.12 | Stephnie | TripleFFFF: yeah..... Pinless service...(authentication on Caller ID) |
17:34.19 | h3x | In any case, asterisk and H.323 is awful |
17:34.34 | TripleFFFF | k |
17:34.40 | TripleFFFF | good idea |
17:34.46 | salaud | h3x: So H.323 PBX -> Yate (SIP) -> Asterisk -> VOIP Providers (IAX) ? |
17:34.49 | TripleFFFF | Stepni |
17:34.52 | Stephnie | dlynes_office : okey....forget about DATABASE .... what about AGI + PERL |
17:35.08 | dlynes_office | movotny22: mysql rivals oracle? are you joking? |
17:35.14 | h3x | or scrap the h.323 pbx and use asterisk :) |
17:35.28 | salaud | h3x: Client is currently using asterisk... |
17:35.33 | dlynes_office | Stephnie: yeah, agi+perl is fine |
17:35.39 | rnovotny22 | dlynes_office: nope, use it quite heavily here with no problems. |
17:35.41 | salaud | h3x: Unfortunately they are idiots and bought some Altigen solution |
17:35.49 | h3x | ew |
17:35.55 | dlynes_office | movotny22: you don't use joins, do you? |
17:35.59 | salaud | h3x: But... they still want to pay us to bridge the two systems |
17:36.14 | Stephnie | dlynes_office : great!!! so I think I should change ASTCC Scripts to use them in my production server.... |
17:36.18 | salaud | h3x: Stupid people with money... never ceases to amaze me |
17:36.26 | h3x | Thats like somebody wanting to use uucp for email in 2006 |
17:36.27 | h3x | heh |
17:36.27 | Stephnie | dlynes_office : what about oracle? |
17:36.40 | salaud | h3x: I'm thinking smoke signals instead of e-mail |
17:36.44 | h3x | haha |
17:36.46 | rnovotny22 | dlynes_office: You just have to set them up correctly, then they are quite fast. Faster in my case than oracle. |
17:36.47 | dlynes_office | Stephnie: oracle will definitely do the job, but as I said, it's total overkill for what you need |
17:37.00 | salaud | h3x: Is there a package (debian) for Yate? |
17:37.13 | dlynes_office | movotny22: Yeah, I know mysql can be faster than Oracle |
17:37.17 | h3x | i dont know, probably |
17:37.32 | dlynes_office | movotny22: but my problem with it, is if I do any kinda complex queries in mysql, it slows to a crawl |
17:37.33 | Stephnie | dlynes_office : then I should go for mysql ... |
17:37.36 | salaud | h3x: Ok... I'll start digging around for Yate... |
17:37.55 | salaud | h3x: But you are pretty sure the H.323 channels for asterisk won't work at all? |
17:38.04 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
17:38.05 | h3x | they work but not very well |
17:38.12 | h3x | capacity is a problem for sure |
17:38.22 | salaud | h3x: We only have 8 channels |
17:38.31 | h3x | the max capacity ive heard is like 10 |
17:38.32 | salaud | h3x: I mean 12 channels |
17:38.34 | h3x | so thats even pushing it |
17:38.44 | salaud | h3x: But.. Yate doesn't have that limitation? |
17:38.45 | rnovotny22 | dlynes_office: I use joins all the time with multiple tables and have not seen any slowdown. Several tables have close to a million records. |
17:40.29 | *** join/#asterisk InfraRed (n=subhi@bb-87-81-46-122.ukonline.co.uk) |
17:40.31 | InfraRed | hi all |
17:40.33 | h3x | they claim it dosent |
17:40.41 | h3x | but it still sucks im sure just because its openh323 |
17:40.41 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
17:40.43 | InfraRed | anyone here using sip registration from behind nat ? |
17:40.51 | Juggie | everyone? |
17:40.53 | [TK]D-Fender | InfraRed : Yup, plenty of us |
17:40.53 | h3x | openh323 is huge |
17:41.03 | dlynes_office | Juggie: ? |
17:41.03 | h3x | it takes about 4 hours to compile on a relatively fast machine |
17:41.07 | h3x | to give you an idea of how bloated it is |
17:41.14 | InfraRed | ok what voodoo must i do to make it work ? |
17:41.28 | Juggie | dlynes_home, my answer was everyone, almost everyone uses sip behind nat. |
17:41.34 | dlynes_office | Juggie: oh |
17:41.34 | [TK]D-Fender | InfraRed : Forward 2 port ranges, add 4 settings to sip.conf and thats it |
17:41.36 | Juggie | InfraRed, nat=yes for your client. |
17:41.40 | dlynes_office | Juggie: Yeah...I did before |
17:41.44 | dlynes_office | Juggie: don't now |
17:41.45 | InfraRed | i added the * server as DMZ |
17:42.08 | Juggie | InfraRed, so you have * behind nat, but DMZ on your router |
17:42.10 | InfraRed | Juggie: excellent idea |
17:42.12 | Stephnie | dlynes_office : thank you for your suggestions.... |
17:42.12 | Juggie | and then also clients behind nat |
17:42.16 | Juggie | correct? |
17:42.19 | InfraRed | Juggie: yes |
17:42.24 | InfraRed | double nat |
17:42.26 | Juggie | set externip= in sip.conf |
17:42.26 | [TK]D-Fender | InfraRed : Then all you need is "localnet", "externip", and "nat=yes" added to your [general] section of sip.conf |
17:42.29 | Stephnie | thanks everybody...:) tc...Bye |
17:42.31 | Juggie | to your real external ip. |
17:42.35 | Juggie | and nat=yes for your clients |
17:42.38 | Juggie | also qualify=yes |
17:42.38 | InfraRed | fantastic |
17:42.40 | Juggie | and you should be good to go |
17:43.05 | InfraRed | what is the "localnet" directive? |
17:43.13 | salaud | h3x: That doesn't sound good |
17:43.18 | dlynes_office | or qualify=300 so that you're compatible with most home office grade routers out there (qualify=150, for some really crappy cisco routers) |
17:43.33 | salaud | h3x: Is there anything that doesn't suck? |
17:43.37 | Juggie | dont worry about that unless you have clients on your local network |
17:43.38 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
17:43.38 | h3x | I think vodiva has a h323 to sip stack |
17:43.42 | Juggie | but they are on a different subnet |
17:43.55 | salaud | h3x: is that commercial? |
17:43.55 | [TK]D-Fender | InfraRed : For SIP to work, * has to forge your external IP address into the headers. Any client calling you OUTSIDE your "local" networks will receive the forged one specified in "externip" |
17:44.24 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
17:44.30 | Juggie | you dont need to touch local unless your internal network is more then one subnet. |
17:44.30 | *** part/#asterisk Damin (n=damin@nucleus.nacs.net) |
17:44.39 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
17:44.53 | Juggie | eg, if your * box is 192.168.1.100 but you also have 192.168.2.x as a local network |
17:44.55 | InfraRed | ok |
17:44.57 | h3x | http://www1.cs.columbia.edu/~kns10/research/gw/ |
17:45.48 | h3x | that looks like a clean implementation |
17:46.29 | salaud | h3x: So it is essentially only necessary to translate signaling and then everything is RTP from that point. |
17:46.38 | h3x | right |
17:46.48 | Juggie | rtp is cake so long as your nat implementation doesnt suck |
17:46.54 | h3x | asterisk does alright at sip rtp to iax2 usually |
17:47.09 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
17:47.32 | lunk | do iax trunks support the dtmfmode option? |
17:47.36 | salaud | Well.... I'll try a few things... Yate looks like an entire infrastructure to itself built on openh323 |
17:48.02 | salaud | h3x: I thought there was an asterisk channel driver based on openh323 also |
17:48.05 | SpaceBass | anyone know of a way to mimic the Cisco VT video stuff on os x? IE call someone with a hard phone and have a video softphone start automaticallly? |
17:48.18 | h3x | there is |
17:48.22 | h3x | thats the one im talking about :) |
17:48.27 | h3x | the builtin one is a joke |
17:49.06 | sonic69 | i am trying to dial with my polycom ip 4000 phone and it says ressource full!!! and when i am doing sip show peer in the CLI i see that the phone is connected |
17:49.07 | salaud | h3x: Ok... So the one in Yate and the openH323 based channel driver for asterisk are equivalent, in your mind? |
17:49.58 | h3x | well the last i knew the openh323 stuff for asterisk was a dangling project |
17:50.07 | salaud | h3x: ah.. |
17:50.17 | Juggie | bug jerjer. |
17:50.26 | h3x | jerjer is going to say |
17:50.29 | h3x | "dont use h323" |
17:50.38 | Juggie | probally |
17:50.44 | Juggie | but didnt he write it? or at least one implementation of it? |
17:50.46 | h3x | and then some explicitives about how you should go F your mother |
17:50.51 | h3x | yes |
17:50.53 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
17:51.16 | salaud | h3x: This is where software "available" link goes on the sip323 implementation you put the link to.. http://www.sipquest.com/ |
17:51.18 | h3x | he wrote it so people could do h.323 with nufone |
17:51.26 | Spy000007 | I thought jerjer was busy driving to Mexico to avoid paying his Nufone bills... |
17:51.32 | salaud | h3x: looks like it is commercial now... not 100% sure |
17:51.51 | h3x | available? |
17:51.52 | h3x | i clicked on software |
17:51.53 | Damin | Juggie: Not that I can see.. |
17:52.01 | h3x | ohyeah |
17:52.04 | h3x | THAT software link |
17:52.15 | Juggie | Damin, answer me in the correct channel! :) |
17:53.02 | salaud | h3x: right... software link sends you to another page with an "available" link which goes to a commercial site... where it is unavailable |
17:54.01 | anthm | we use woomera for h323 into asterisk sometimes |
17:55.12 | h3x | well thats a good idea |
17:55.18 | h3x | i didnt know woomera did h323 |
17:55.43 | salaud | h3x: is woomera another PBX? |
17:55.54 | salaud | anthm: is it easier to setup than Yate? |
17:56.02 | h3x | The Woomera protocol, designed by Craig Southeren of OpenH323 fame, makes it possible to put your |
17:56.06 | anthm | woomera is openh323 running as a daemon |
17:56.07 | h3x | haha well that makes sense |
17:56.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
17:56.39 | salaud | anthm: cool... does it translate to SIP? |
17:56.46 | anthm | in it's own process then it abstracts it down to a mega simple voice proto that asterisk can handle |
17:57.14 | salaud | anthm: how does it interface to asterisk? via a channel module? |
17:57.21 | anthm | ya chan_woomera |
17:57.50 | salaud | that might be the most stripped down solution... can it do 12 channels? |
17:57.50 | h3x | you could just use that asterisk fork project |
17:58.13 | h3x | freepbx.org |
17:58.16 | salaud | h3x: I can't replace the existing asterisk infrastructure... well... I guess I COULD...but... |
17:58.21 | Qwell[] | h3x: openpbx.. |
17:58.26 | Qwell[] | ~freepbx |
17:58.27 | jbot | rumour has it, freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
17:58.32 | h3x | oh |
17:58.37 | Qwell[] | ~openpbx |
17:58.38 | jbot | i guess openpbx is an asterisk fork without asterisk's limitations of using other GPLed code. see http://openpbx.org/ for more info, or join #openpbx |
17:58.39 | dlynes_office | Juggie: yeah...he only wrote one implementation of it...the paid implementation (chan_ooh323c, the one in asterisk-addons) |
17:58.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net) |
17:59.06 | anthm | yah probably it can do 12 chans easily it's going to switch to opal lib sometime soon for now it does h323 |
17:59.13 | salaud | where do you get chan_woomera? |
17:59.28 | MACscr | lol, how can something be a fork of a GPLed project and not use its license |
17:59.32 | MACscr | that doesnt make sense |
17:59.46 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
17:59.48 | salaud | ha! found it |
17:59.51 | Qwell[] | salaud: pbxfreeware? |
18:00.00 | salaud | Qwell[]: no chan_woomera |
18:00.01 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:00.13 | Qwell[] | ... |
18:00.24 | Qwell[] | pbxfreeware |
18:00.41 | coppice | MACscr: what doesn't make sense? |
18:02.32 | anthm | i think what they mean is they are not opposed to using gpl libraries where asterisk actually will not use any so they can still own it all but you would have to ask them to be sure |
18:02.37 | salaud | so... basically... H.323 sucks for asterisk |
18:02.50 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
18:03.09 | MACscr | coppice: i was thinking asterisk was gpl, thus anything fork of it, etc, would have to be gpl'ed as well |
18:03.33 | Qwell[] | MACscr: it is gpl |
18:03.40 | coppice | MACser you completely miss the point. asterisk is not GPL, and is unable to use anything GPL |
18:03.54 | anthm | asterisk is gpl+cake+eat |
18:04.06 | MACscr | what license is asterisk then |
18:04.11 | Qwell[] | MACscr: gpl |
18:04.16 | Qwell[] | and commercial |
18:04.16 | coppice | gpl+something to make it totally incompatible with anything gpl |
18:04.21 | rob0 | H.323 is an Equal Opportunity Suck. It sucks for all. :) |
18:04.28 | anthm | the cake-and-eat-it-too license |
18:04.30 | MACscr | lol, coppice is confusing me |
18:05.35 | MACscr | lol, i guess so |
18:05.35 | coppice | asterisk is GPL+something that means asterisk cannot use any GPL code |
18:05.35 | salaud | Maybe * has the "go F yourself telcos" license |
18:05.35 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
18:05.35 | salaud | ie... we are gonna get this job done |
18:05.40 | coppice | so asterisk has a lousy SIP, a lousy RTP, a lousy almost everything as it all has to be built from scratch |
18:06.10 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
18:06.17 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
18:06.19 | salaud | coppice: I imagine those are fighting words... but I'm a lover not a fighter ;) |
18:06.33 | mroth_imm | could anyone tell me what output i should expect from lspci for a Sangoma A200? |
18:06.33 | anthm | h.323 is actually the closest to perfect voip proto of them all it's just that ppl dont realize they need some of what it has to offer till later when they learn more |
18:06.48 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
18:07.09 | coppice | even the developers will admit that things like chan_sip are horrible junk. its not exactly a point of contention :-) |
18:07.36 | salaud | anthm: It seems more sophisticated... but like other sophisticated stuff it is more in the commercial domain... which makes it useless |
18:07.43 | MACscr | http://www.voip-info.org/wiki/view/Asterisk+GPL+Compliance |
18:07.53 | MACscr | check out the editline library part |
18:08.16 | anthm | it's a vicious circle really |
18:08.20 | coppice | salaud: not really. SIP started simple but has now surpassed everything else for complexity, and without a clean design |
18:08.30 | salaud | coppice: As a non-voip-developer, but fairly successful implementer.. I would say... it works |
18:08.56 | coppice | they have finally realised that SIP is a disaster, and demand TCP but don't require UDP |
18:09.17 | mroth_imm | salaud, i've got a big single server installation (355 concurrent SIP calls yesterday) and i can tell you, asterisk is not exactly stable yet |
18:09.24 | anthm | you start with telco circuit and think of a way to make it go over ip and not lose anything then you think, nah lets just get it to work, then you say oops i need that |
18:10.01 | anthm | it's like playing 7 degrees of separation to why h323 was not so crazy when they did it that way |
18:10.11 | salaud | anthm: Betamax |
18:10.20 | salaud | anthm: nuff said |
18:10.22 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
18:10.23 | salaud | ;) |
18:10.27 | coppice | SIP has still got things wrong. they changed from UDP to TCP at a time when SCTP had become mature enough to use. SCTP is the way to communicate signalling |
18:10.53 | anthm | ya and now they are saying "we should compress the signalling to a binary format" =D |
18:10.59 | salaud | mroth_imm: scalability is an issue for sure.. need multiple servers and that sucks |
18:11.15 | rpm | are there any polycom configuration tools out there? |
18:11.24 | rpm | for ip-601s/501s? |
18:11.29 | Qwell[] | rpm: vi |
18:11.35 | coppice | anthm: really? i hadn't heard that. i've heard suggestions of streaming XML |
18:11.36 | rpm | dirty. |
18:11.41 | mroth_imm | it scaled fine...we were 35% to 45% idle...the problem is that that many calls shakes out bugs |
18:11.51 | mroth_imm | it seems that sometimes chan_sip just goes away |
18:11.53 | anthm | yah they are looking for ways to compress it now |
18:12.09 | anthm | they wanted text to begin with cos it's "easy to debug" =p |
18:12.45 | salaud | mroth_imm: maybe 35% idle is still too much.... asterisk definitely breaks down with high volume to one server... no doubt |
18:12.50 | coppice | the text is easy to debug argument is really brain dead. text causes most of the damn bugs in the frist place :-) |
18:13.36 | anthm | yah especially with tethereal |
18:13.48 | mroth_imm | noone out there with a Sangoma A200...i just need to know how it identifies itself so i can compare it to my lspci output |
18:13.52 | anthm | umm it turs it into text for you |
18:14.11 | dlynes_office | mroth_imm: one sec |
18:14.26 | coppice | how come ethereal has better protocol analysis than any of the packages people try to debug with it? :-) |
18:14.43 | anthm | yah no kidding |
18:15.02 | Juggie | ethereal isnt called ethereal anymore |
18:15.14 | dlynes_office | it's called wraith now? |
18:15.23 | coppice | its called loan shark or something |
18:15.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:16.01 | dlynes_office | mroth_imm: if it's got a hwec on it, it looks like thus: |
18:16.03 | dlynes_office | 00:08.0 Network controller: Unknown device 1923:0040 |
18:16.03 | dlynes_office | <PROTECTED> |
18:16.20 | mroth_imm | dlynes_office: many thanks dlynes |
18:16.25 | anthm | they should make it into a lib you can link against to read protocols into your app |
18:16.39 | mroth_imm | it does not show up on my 6850...and it has the added bonus of making my NIC invisible too :( |
18:17.00 | coppice | so gradually all network communication will be promiscuous :-) |
18:17.06 | dlynes_office | mroth_imm: i would check to make sure your pci card is fully seated (after shutting your machine off, of course) |
18:17.45 | Juggie | ethereal => wireshark |
18:17.51 | mroth_imm | dlynes_office: i'll do it again, but i'm starting to think the Dell PEs just don't play nice with certain PCI devices |
18:18.06 | mroth_imm | I couldn't get a Digium TDM400P to show up in it either... |
18:18.15 | coppice | i think loan shark would have been a better name :-) |
18:18.18 | anthm | anyway the moral is that h.323 is actually not as bad as asterisk makes it appear. |
18:18.33 | dlynes_office | mroth_imm: ah...I would tend to think it's possibly a bios configuration issue, then |
18:18.42 | dlynes_office | mroth_imm: are you using plug and pray support in your bios? |
18:18.50 | coppice | H.323 sucks, but SIP sucks even more |
18:18.51 | mroth_imm | i will start sticking paper clips into it and see if that helps :) |
18:18.54 | De_Mon | anyone familiar with xtunnels.org? |
18:19.02 | anthm | they all have to suck that is not negotiable |
18:19.22 | Juggie | from a developement standpoint, sip sucks alot less then h323 |
18:19.28 | mroth_imm | dlynes_office: i'll have to check that, any other ideas while i'm looking at the BIOS (should PaP be on or off)? |
18:19.31 | anthm | we would run out of things to bitch about then and that would be no fun |
18:19.43 | coppice | why are they interested in compressing SIP signalling, when the media uses RTP :-) |
18:19.50 | dlynes_office | mroth_imm: i would turn it off, to see if that helps the situation |
18:20.07 | mroth_imm | i'm running Fedora, and with most things kudzu just picks it up and adds it to the hwdb, but the telephony cards are causing premature balding for me |
18:20.08 | dlynes_office | mroth_imm: if you bios allows you to see what devices are configured in each slot, see if your bios sees the card |
18:20.08 | *** join/#asterisk saftsack (n=saftsack@p54A7FFFD.dip.t-dialin.net) |
18:20.11 | anthm | imagine it: "good thing evey protocol is perfect all I have to do is sit here looking out the window ho hum..." |
18:20.25 | *** join/#asterisk Vni (n=chatzill@adsl-69-235-247-151.dsl.irvnca.pacbell.net) |
18:20.30 | mroth_imm | it does indeed and it shows that slot as empty... |
18:20.56 | anthm | coppice, cos ppl like to get bothered by moot points |
18:21.16 | coppice | i really hate having to learn stuff to support 2 kids in the room next door |
18:21.30 | xachen | You know... unloading MOH shouldn't have to crash asterisk |
18:21.34 | anthm | like how much db performance you lose using odbc to connect asterisk to a db dispite the fact that if you top out odbc asterisk would have been dead back at the 10% mark |
18:21.44 | *** join/#asterisk dsully (i=daniel@electricrain.com) |
18:21.58 | coppice | anthm: you mean because its easier to play around with SIP, than to attack something containing DSP, right? :-) |
18:22.01 | dsully | g'morning. |
18:22.17 | Vni | I messed up my extensions.conf and I didnt back it up does somebody have one so I can fix mine. Im running Fedora core 4 with astguiclient 1.1.11 and vicidialer??? Please my bosses is gonna fire me |
18:22.25 | salaud | H.323 stuff in yate (routing especially) looks a bit complicated :( |
18:22.29 | dsully | any one have thoughts on why, after asterisk has been up for a while (1-2 days), I start seeing this on the console: |
18:22.32 | dsully | pri_dchannel: Ring requested on channel 0/1 already in use on span 1. |
18:22.35 | Juggie | anthm, are both openpbx & freeswitch being actively developed? or is all work being done on freeswitch now? |
18:22.51 | dsully | and all inbound & outbound calls on the PRI stop? |
18:23.11 | coppice | Juggie: well openpbx can now do T.38 :-) |
18:23.19 | anthm | coppice, or then play with iax cos sip is too hard like the old lady who swallowed a fly |
18:23.29 | jbalcomb | How about this one? Our Polycom SoundStation conference phone attached to a GrandStream HT-386 decides to put on the hold music and then disconnect the call. |
18:23.35 | Juggie | coppice, so can asterisk, at least passthrough. |
18:23.41 | mroth_imm | dlynes_office: thank you for your help... :) |
18:23.47 | Vni | >I messed up my extensions.conf and I didnt back it up does somebody have one so I can fix mine. Im running Fedora core 4 with astguiclient 1.1.11 and vicidialer??? Please my bosses is gonna fire me . No zaptel cards |
18:23.59 | coppice | Juggie: yeah, but that's just the mickey mouse stuff :-) |
18:24.23 | dlynes_office | mroth_imm: it sounds like maybe that slot's defective |
18:24.31 | anthm | ahh dont say that, i just spent 12 days with mickey mouse paying him money at every turn |
18:24.32 | jbalcomb | Vni: you can get one off the wiki or in the asterisk source directory. |
18:24.33 | dlynes_office | mroth_imm: have you tried throwing hte card into a different slot? |
18:25.01 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
18:25.04 | mroth_imm | dlynes_office: i will give it a shot |
18:25.07 | anthm | juggie so ya, it appears openpbx is, I can attest freeswitch is based on my terminal full of code i am looking at |
18:25.12 | dlynes_office | Katty: Meowwwrrr! |
18:25.21 | salaud | I appreciate everyone's help on the H.323 ... I'm going to take my best pass at Yate or Woomera and then it's off to a bunch of Sipura 2000's doing analog/digital and back to analog again into this f*'n altigen system as a final option |
18:25.29 | Katty | dlynes_office: heya |
18:25.31 | coppice | anthm: I can see disneyland from here. that's as close as i want to get |
18:25.36 | smackus | ok, so here is an ftp/polycom question for you. I have the ftp server set up to provision the phones, but they will not boot. here is my boot log for the phone http://pastebin.ca/73313 |
18:25.47 | smackus | can anyone make a suggestion? |
18:25.59 | smackus | i have played with permissions and such, no changes |
18:26.16 | smackus | which file exactly is the bootrom.ld |
18:26.24 | smackus | SIP.cfg? |
18:26.33 | smackus | no, that woulnt be it |
18:26.48 | smackus | sip.ld |
18:26.50 | smackus | right? |
18:27.02 | CunningPike | smackus: Do you have a bootrom.ld file? |
18:27.10 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
18:27.11 | *** join/#asterisk albertito (n=net@host199.201-252-24.telecom.net.ar) |
18:27.11 | smackus | not named as such |
18:27.13 | anthm | turning h323 into sip is probably yate's best trick in thier list of bullet points |
18:27.32 | CunningPike | smackus: You need two files: bootrom.ld is the bootrom and sip.ld is the SIP application |
18:27.40 | anonymouz666 | 3x0 :) |
18:28.06 | smackus | ok, and it has to be named bootrom.ld? |
18:28.09 | *** part/#asterisk skraelings001 (n=skraelin@201.230.140.95) |
18:28.20 | Juggie | anthm, this signup process to get into the tracker is 'special' |
18:28.25 | Juggie | to say the leasdt. |
18:28.27 | Juggie | *least. |
18:28.43 | anthm | for what? |
18:28.55 | Juggie | trac.freeswitch.org |
18:29.04 | jbalcomb | Our Polycom SoundStation conference phone attached to a GrandStream HT-386 decides to put on the hold music and then disconnect the call. Any ideas why or what direction to head? |
18:29.24 | anthm | it's unified all you do is sign up and you get an account for everything |
18:29.35 | Juggie | yeah iknow its jsut a weird signup process. |
18:29.47 | smackus | here are all of the files in the spip zip file, should there be more? http://pastebin.ca/73315 |
18:29.50 | anthm | it's cos it's derived from our isp platform |
18:29.54 | *** part/#asterisk dsully (i=daniel@electricrain.com) |
18:29.54 | coppice | anthm: I thought it was to make people suspicious and go away :-) |
18:29.55 | Juggie | ah. |
18:30.03 | anthm | yah that too |
18:30.26 | Juggie | grr. |
18:30.28 | CunningPike | smackus: Nope - you're missing your bootrom |
18:30.29 | Juggie | i'm impatient |
18:30.33 | Juggie | its slow sending me my email |
18:31.02 | anthm | you dont even have to wait really |
18:31.35 | Juggie | what will my default pass be then? |
18:32.21 | Vni | anyone know how I would put IVR on outbound |
18:33.11 | [TK]D-Fender | smackus : odds are you won't want to mess with your bootrom |
18:33.41 | CunningPike | [TK]D-Fender: Still needs one though :) |
18:33.43 | [TK]D-Fender | jbalcomb : And I doubt your SS decided to go "on-hold" then hangup... |
18:33.54 | [TK]D-Fender | CunningPike : And the phone CAME with one... |
18:34.06 | CunningPike | [TK]D-Fender: You'd think........ |
18:34.09 | smackus | ok... well, at one point I had this working. I have all of my phones running 1.6.5, from the link i have from the wiki, I cannot get that version bootrom. Is there a better link? |
18:34.15 | [TK]D-Fender | CunningPike : I haven't seen a stock phone that can't run through 1.6.6 stock... |
18:34.40 | [TK]D-Fender | 1.6.6 is the SIP version, not the bootrom version. What are you running? |
18:34.42 | CunningPike | smackus: You can use the bootrom from the Polycom web site |
18:35.04 | smackus | so 3.1.0? |
18:35.04 | [TK]D-Fender | smackus : If you are running 2.6.1 or better, don't touch it. |
18:35.08 | Vni | anyone know how I would put IVR on outbound??? |
18:35.12 | CunningPike | smackus: Latest is 3.1.3 |
18:35.14 | [TK]D-Fender | smackus : Did you check on the phone direct? |
18:35.22 | jbalcomb | [TK]D-Fender: How do you mean? Are you suggested my users have done this? |
18:35.33 | Juggie | anthm, somethings broken, i still havnt got my email, unless it has to be approved or something. |
18:35.46 | [TK]D-Fender | jbalcomb : Ask yourself how an analog phone puts a call on hold... |
18:35.55 | [TK]D-Fender | jbalcomb : Esp on taht ATA |
18:35.57 | smackus | Application, main: Label=BOOT, Version=2.6.1.0003 04-Dec-04 14:38 |
18:36.11 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
18:36.12 | [TK]D-Fender | smackus : thats your BR version and its fine. |
18:36.14 | Juggie | [TK]D-Fender, whats stopping analog ports from going on hold? |
18:36.20 | Juggie | if the call is bridged through * |
18:36.32 | [TK]D-Fender | smackus : So whats the actual issue at this point? |
18:36.34 | jbalcomb | [TK]D-Fender: I am. Line noise? Misinterpretted signal? ATA thinks there was a *<hold> or <flash>? |
18:36.51 | [TK]D-Fender | jbalcomb : Dunno... gotta wonder... |
18:37.03 | smackus | the phone wont boot completely, it gets the error Failed to load bootrom.ld. Check filename & FTP parameters and reboots |
18:37.19 | [TK]D-Fender | EW |
18:37.22 | [TK]D-Fender | NOT good. |
18:37.24 | jbalcomb | [TK]D-Fender: it's a conference room phone so it's not likely they are messing with it past dialing into the conference. |
18:37.31 | CunningPike | smackus: Your issue is that if you are using FTP provisioning, the phone needs a bootrom.ld to check against |
18:37.34 | [TK]D-Fender | Might be "bricked" from a bad upgrade |
18:37.39 | jbalcomb | [TK]D-Fender I've heard the complaint before on other phones. |
18:37.53 | smackus | ok, so back to the question... can I use bootrom 3.1.0? |
18:38.00 | [TK]D-Fender | smackus : Go DL any bootrom equal or newer to the one you have and try to force upgrade it. |
18:38.12 | [TK]D-Fender | smackus : What model are you running? |
18:38.16 | jbalcomb | [TK]D-Fender However, I am thinking it's a good opeurtunity to PO a Polycom SoundPoint IP conference phone |
18:38.19 | smackus | 301 and 501 |
18:38.34 | [TK]D-Fender | smackus : Ok, sure... you should get 3.1.3 though as its more current. |
18:38.40 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
18:38.42 | gbodemantv | hey all |
18:38.52 | dlynes_office | mroth_imm: anyways...i'm out for most of the day now...catch ya later |
18:38.52 | gbodemantv | I am in a bind |
18:38.52 | [TK]D-Fender | I should check to see if SIP 2.0 is full release now. |
18:38.56 | rpm | im running 3.1.0 heh |
18:39.02 | gbodemantv | I have been asked to impkement queue reports |
18:39.03 | rpm | i need to get a new bootrom and sip image |
18:39.10 | jbalcomb | [TK]D-Fender: Isn't there some way I can see if they hit the hold button or what the phone system thinks was going on with that extension? |
18:39.12 | smackus | where can i get the newer version? it will only let me dl old bersions |
18:39.31 | gbodemantv | I am wanting to use Queue Statitics from Asterisk Gurus but I am having a hell of a time with the implementation |
18:39.37 | [TK]D-Fender | jbalcomb : "hold" on an analog phone introduces no signalling to the ATA. That can't be it. |
18:39.39 | CunningPike | smackus: What's the most recent they have? |
18:39.40 | gbodemantv | Have never install PHP or PostGres |
18:39.41 | rpm | you gotta be a polycom sales partner or something.. i've been trying to get the images.. you gotta pay for them |
18:39.54 | gbodemantv | any guides or walkthroughs anyone knows of |
18:39.57 | jbalcomb | [TK]D-Fender That's what I was thinking because the hold music came on. |
18:40.00 | [TK]D-Fender | rpm : No you don't.... |
18:40.11 | CunningPike | rpm: You can get them for free from your reseller....... |
18:40.15 | smackus | 3.1.0 |
18:40.18 | [TK]D-Fender | jbalcomb : hold on an analog phone only mutes everything |
18:40.22 | CunningPike | smackus: Good enough |
18:40.24 | jbalcomb | [TK]D-Fender right |
18:40.25 | smackus | ok |
18:40.26 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
18:40.45 | jbalcomb | [TK]D-Fender so for the hold music to come on asterisk has to think something happened and i should be able to see it in the logs right? |
18:40.47 | smackus | just stick it right in the directory with the sip.ld and such, right? |
18:41.02 | [TK]D-Fender | jbalcomb : Perhaps... I don't know the logging system too well. |
18:41.02 | jbalcomb | CunningPike rpm: you can also find them on the internet just like i did |
18:41.23 | [TK]D-Fender | smackus : Yes, but you need to set up your <mac>.cfg file to point to them so it picks up. |
18:41.53 | jbalcomb | smackus: try 'freedomphones' |
18:43.05 | jbalcomb | [TK]D-Fender: ok, i've created /var/log/asterisk/rediculous.asterisk for tailing. It has all log levels going. |
18:43.41 | jbalcomb | [TK]D-Fender: it even shows DTMF events.. |
18:43.56 | [TK]D-Fender | WWF Events? pr0n PPV events? |
18:44.09 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
18:44.57 | jbalcomb | [TK]D-Fender: Yes, you can monitor the World Wildlife Fund and PayPerView pr0ns. Asterisk is rather multifaceted<SP>. |
18:45.56 | [TK]D-Fender | WWE.... |
18:45.59 | [TK]D-Fender | close enough |
18:46.11 | gbodemantv | anybody using Queue Statistics and Asterisk? |
18:46.52 | [TK]D-Fender | gbodemantv : I do on my GUI package... |
18:50.13 | gbodemantv | have been told I have a week to implement queues and reporting |
18:50.19 | gbodemantv | got realtime ques working |
18:50.28 | gbodemantv | but need to get reporting up and running |
18:50.38 | [TK]D-Fender | gbodemantv : Reporting has several packages out there already. |
18:50.40 | gbodemantv | Asterisk Stats from asterisk gurus looks perfect |
18:50.58 | gbodemantv | but their install is not at all specific |
18:51.05 | gbodemantv | I am having a hell of a time |
18:51.20 | gbodemantv | can't even get postgres off the ground |
18:51.44 | gbodemantv | trying to get ahold of them to implement it |
18:51.50 | gbodemantv | but no one has responded |
18:51.54 | jbalcomb | gbodemantv: QueueMetrics is decent |
18:52.15 | gbodemantv | don't need that much info |
18:52.26 | jbalcomb | gbodemantv: Asterisk CDR is good for the reporting |
18:52.47 | gbodemantv | how difficult is it to implement |
18:52.52 | jbalcomb | gbodemantv: Maybe you should contract [TK]D-Fender to get it going for you |
18:53.18 | *** join/#asterisk backblue (n=moo@87-196-47-160.net.novis.pt) |
18:53.19 | gbodemantv | if he is interested I might |
18:53.31 | jbalcomb | gbodemantv: I would say not difficult because I got it working. |
18:53.55 | jbalcomb | gbodemantv: you should /msg him. =) |
18:54.22 | jbalcomb | gbodemantv: I've contracted him and he is effective and friendly. |
18:55.35 | gbodemantv | jbalcomb: does it report realtime queues? |
18:55.58 | jbalcomb | gbodemantv: it reports every call thats been hung up and accounted for if that what you mean. |
18:56.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:56.29 | gbodemantv | http://www.asteriskguru.com/tools/queue_stats.php |
18:56.34 | gbodemantv | this is what i am trying to use |
18:56.38 | jbalcomb | gbodemantv: there maybe a matter of adjusting some line in the source and recompiling so that additional information is logged. |
18:56.52 | gbodemantv | and I know that it will work |
18:57.02 | gbodemantv | just cant get it all to come together |
18:57.02 | jbalcomb | gbodemantv: We added some system calls to echo lines to the log as well. |
18:57.32 | *** join/#asterisk Johnnie (i=odysseus@pdpc/supporter/active/Johnnie) |
18:58.31 | jbalcomb | gbodemantv: Ah, ok. Asterisk CDR is not the comprable product I think. I beleive what we are doing with QueueMetrics is what your looking to do with Asteriskguru Queue Statistics |
18:59.13 | drray | can I axe why you don't just sed/awk/perl a solution? |
18:59.24 | jbalcomb | QueueMetrics has a 30 or 90 dial trial and is fairly easy to set up. The only troubling matter is setting up tomcat for the java machine. |
18:59.36 | [TK]D-Fender | "Too much trouble" |
18:59.43 | jbalcomb | drray: "too much trouble" |
19:00.07 | jbalcomb | man, I gots to take typing course.. too slow. |
19:00.07 | *** join/#asterisk oej (n=oej@212.17.152.81) |
19:00.15 | drray | that sounds a lot like people who use asterisk at home, because asterisk is "too much trouble" but hey, roll on |
19:00.44 | jbalcomb | drray: one can only be so /into/ maintain custom code and systems. |
19:01.19 | jbalcomb | drray: I'm putting together a IP phone management system right now and I very much dread the maintanence of it. |
19:01.21 | *** join/#asterisk oej_ (n=olle@212.17.152.81) |
19:01.21 | drray | that's a fair point, and I'd be on your side if I did not think that the turn-key solutions lacked something |
19:01.28 | drray | :) |
19:01.40 | drray | I hear you |
19:03.03 | jbalcomb | drray: I would think the thing to do is /fix/ the turn-key solution and submit ones code changes for addition |
19:03.46 | drray | I've always found it easier to roll my own than fix what I don't like about someone elses |
19:04.00 | drray | but I wrote my own crappy pvr because I did not like mythtv crashing |
19:04.56 | *** join/#asterisk InfraRed (n=subhi@bb-87-81-46-122.ukonline.co.uk) |
19:05.07 | *** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140) |
19:05.23 | jbalcomb | Anyone have opinions on the Polycom SoundStation IP 3000? |
19:06.14 | *** join/#asterisk NullC (n=greg@wikimedia/Mindspillage) |
19:06.14 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
19:06.17 | InfraRed | hi all |
19:06.25 | InfraRed | thanks for the help earlier with the nat |
19:06.43 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
19:06.55 | NullC | I have a pair of asterisks servers, IAX 2 trunk between them. It works fine, except it doesn't truck (put multiple calls in one packet) in ONE DIRECTION.. one way trunks, the other doesn't. Help! |
19:07.10 | ManxPower | I didn't realize that wasps can nest inside a LNB housing...until today. |
19:07.31 | drray | wasps can nest anywhere |
19:07.36 | drray | even inside of beehives |
19:07.41 | ptinsley | ManxPower, i had the same thing on my old one dirt dobbers I think it was I had tough |
19:07.44 | ptinsley | though |
19:07.52 | InfraRed | now i am having stream issue, during a call i can 'transmit' but not receive. the asterisk box is behind nat in dmz, the phone is behind another nat behind the asterisk box. any ideas? |
19:07.57 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
19:08.00 | ManxPower | NullC, sounds like 1) you don't have zaptel timing on one of the two servers or 2) the incoming connection does not correctly match a user/friend |
19:08.04 | drray | I ran a mile one time when they infested my tool box |
19:08.17 | InfraRed | what i mean by transmit is voice, they can hear me when i can, but i can't hear them |
19:08.26 | InfraRed | when i call |
19:08.29 | ptinsley | drray, lol |
19:08.42 | ManxPower | My aunt and my brother are both allergic. Apparently I'm not yet |
19:09.01 | *** join/#asterisk jgoo (n=jgoo@ppp129-197.adsl.forthnet.gr) |
19:09.15 | ptinsley | my dad almost died when he was a kid by mowing over a hole full of the little bastards |
19:09.30 | NullC | Hmm ... Ztdummy is on both... |
19:09.45 | ManxPower | NullC, but is it LOADED? |
19:10.29 | jgoo | hello all, I am about to partake in the joy that is bristuff install for a card, and i have come up against loveable terms such as NT and TE, which i kind of undertstand... |
19:11.01 | jgoo | the Install docs talks about the script downloading and patching the asterisk build from cvs ... is the cvs-ftp server still running on digium_ |
19:11.03 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
19:11.19 | jgoo | the card is a HFC single port BRI |
19:11.28 | ManxPower | cvs has been turned off, svn has been used for a while. |
19:11.53 | jgoo | that was my specific question - how can I test without breaking things if this install is ok?? voip wiki was a little out of date I fear |
19:12.01 | jgoo | has anyone recently setu pone o fthese babies- |
19:12.13 | jgoo | (new keyboard >.< ) |
19:12.14 | rob0 | Maybe that CVS - SVN thing should be in /topic |
19:12.31 | drray | it's been a year? |
19:12.35 | rob0 | Yes, I have set up new keyboards before. :) |
19:12.42 | rob0 | wow, a year? |
19:12.55 | rob0 | Seems like several questions every day here. |
19:13.16 | jgoo | =] it is a swedish keyboard, and I am not swedish, so it seems most illogical, but then, so do most swedish people ÖP |
19:14.07 | jgoo | so, setting up a BRI card ... the pstn digium was easy enough, but this is an unbranded (rebranded as crypto) hfc card |
19:14.59 | jgoo | I have 3 concerns: 1) it is plugged into a S0 port on the ISDN box. is that normal? |
19:15.23 | [TK]D-Fender | jbalcomb : Strangely I would suggest the IP version.. go for the SS 2 W (wireless) on an ATA... Works great and not locked to any system. |
19:15.32 | jgoo | 2) the junghanns install for this bristuff, is it the only way forward for these hfc cards? and if so is there updated info =] |
19:15.51 | NullC | ManxPower So changing my extensions.conf so that user:pass@hostname rather than user:pass@ipaddr fixed it. |
19:16.38 | florz | jgoo: What is an "ISDN box"? |
19:17.12 | florz | jgoo: And no, you could also use misdn, capi or i4l |
19:17.14 | jgoo | good question, I am an isdn virgin here (except that one time in germany, but that doesn't count I was drunk) |
19:17.31 | jgoo | florz - aha, you are famous, i read something about your patch |
19:17.36 | jgoo | that was my 3rd concern, florz |
19:17.39 | jbalcomb | [TK]D-Fender: well, if i don't go IP there'd be no point. i think i'll atleast switch it to my sipura ATA rather than the ht-386 |
19:17.42 | jgoo | =] |
19:18.27 | jbalcomb | [TK]D-Fender I was just thinking the IP phone would behave better than the ATA |
19:19.48 | florz | jgoo: As far as that bristuff-stuff is concerned: instead of using the download/build script you just as well can apply the patches manually - the script doesn't so all that much anyway ... |
19:20.01 | florz | s/so/do/ |
19:21.47 | [TK]D-Fender | jbalcomb : I would sooner think its a GS device at fault than a Polycom one :) |
19:23.21 | kristalino | is it possible to text to speech in french ? |
19:23.30 | *** part/#asterisk NullC (n=greg@wikimedia/Mindspillage) |
19:24.08 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
19:26.09 | ptinsley | [TK]D-Fender, i second that one, GS == bugggggy |
19:27.11 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
19:29.07 | smackus | just a general question.. I do not want to try to make this work now, but can i dial an agent login like an extension? |
19:29.10 | smackus | is that even possible |
19:30.25 | SplasPood | anyone know teliax's inbound rate for 800 DIDs? |
19:30.53 | *** join/#asterisk Nodren (n=nodren@adsl-75-8-201-246.dsl.frs2ca.sbcglobal.net) |
19:30.57 | Nodren | ~centosbug |
19:30.58 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
19:31.17 | SplasPood | kristalino: Cepstral supports french |
19:31.23 | SplasPood | kristalino: I'm pretty sure |
19:32.12 | jgoo | florz - do you know of any good steps to follow for this patches so I don't miss one? |
19:32.36 | kristalino | SplasPood,what is cepstral ? |
19:32.47 | SplasPood | kristalino: TTS system.. |
19:33.32 | *** join/#asterisk dangerarea (n=kevin@kingfisherwalk.demon.co.uk) |
19:34.07 | florz | jgoo: The script that's in the bristuff package :-) |
19:34.08 | kristalino | ok, thanks |
19:34.33 | dangerarea | evening |
19:34.41 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
19:35.40 | dangerarea | anyone know the rough spec of the standard HP proliant server these days? |
19:35.55 | dangerarea | more to the point how many simultanious calls it could handle with asterisk |
19:36.11 | vader-- | anyone have any issues where their sip phones won't do anything in the features.conf? |
19:36.13 | vader-- | like *1 for recording and stuff |
19:36.14 | vader-- | i have cisco 7940G phones |
19:36.21 | InfraRed | dangerarea: there is a wiki page about 'dimentioning an asterisk server' |
19:36.30 | dangerarea | ooh, ta InfraRed |
19:36.46 | InfraRed | np |
19:38.25 | *** part/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
19:38.55 | InfraRed | i have a problem with asterisk and nat, i have adsl router with nat, asterisk is set as DMZ in the nat, the phones are behind asterisk on another nat (2 nats), the asterisk server also acts as nat server for another subnet seperate from the phones, when i make a phonecall, the call receiver can hear me but i cannot hear anything, any ideas? i have externip in each phone config and nat=1 too |
19:39.06 | [TK]D-Fender | vader-- : Your Dial statements all enable those features? "wWtT", etc? |
19:39.38 | InfraRed | i'm using sip btw |
19:39.42 | [TK]D-Fender | InfraRed : Externip belongs in [general], not in the phone config |
19:39.52 | [TK]D-Fender | InfraRed : And did you set up your localnet clauses? |
19:39.58 | InfraRed | i have it in general |
19:40.08 | [TK]D-Fender | InfraRed : pastebin your sip.conf |
19:40.11 | [TK]D-Fender | ~pb |
19:40.22 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
19:40.22 | InfraRed | ok sec |
19:40.55 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
19:41.39 | InfraRed | is the localnet for the phones subnet ? |
19:41.40 | jbalcomb | [TK]D-Fender I would agree. I've gotten us down to just two HT-386s. If I get around to switch the conf phone to the SPA-2000 that'll drop us to one. |
19:42.03 | jbalcomb | [TK]D-Fender I've been getting a broken PB all day |
19:42.27 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
19:42.55 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
19:43.35 | sparkleytone | anyone ever seen an issue where every time you reboot an asterisk machine the /var/run/asterisk has been deleted? |
19:43.43 | iq | Hi |
19:44.33 | InfraRed | [TK]D-Fender: localnet was what missing |
19:44.50 | InfraRed | thanks for your help :) |
19:45.03 | sparkleytone | i keep having to recreate the directory... |
19:45.37 | InfraRed | sparkleytone: i think /var/run and /tmp get cleared on every reboot |
19:45.45 | sparkleytone | hrrrm |
19:46.02 | InfraRed | try setting the pid file to file inside /var/run rather than /var/run/asterisk/ |
19:47.21 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
19:47.36 | [TK]D-Fender | jbalcomb : PB? |
19:47.43 | [TK]D-Fender | InfraRed : ywc |
19:47.56 | sparkleytone | does this if statement work for directories? `if [ ! -e "/var/run/asterisk" ]` |
19:48.12 | sparkleytone | as much as i can admin a box, i am allergic to scripting |
19:48.14 | sparkleytone | and coding |
19:48.21 | sparkleytone | will that return properly? |
19:48.39 | jbalcomb | [TK]D-Fender PasteBin |
19:49.21 | lunk | has anyone else experienced difficulty with DTMF with VoipJet? |
19:49.50 | lunk | and/or can anyone recommend a good, cheap, bulk termination service |
19:49.59 | lunk | sip preferrably.. |
19:51.07 | Qwell[] | sparkleytone: -d |
19:51.35 | sparkleytone | thx Qwell |
19:51.36 | wese103 | sparkleytone: -e will tell you that the file or directory exists, but will not discern between the two. |
19:51.52 | drray | I don't think good and cheap go together on that |
19:52.22 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
19:54.08 | dangerarea | lunk: i'm looking for one too |
19:55.10 | dangerarea | i'm looking for someone to terminate 1000 simultanious calls if anyone knows of anyone |
19:55.41 | drray | your local telco wont do that for you? |
19:56.11 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
19:56.24 | InfraRed | dangerarea: depends on your location |
19:56.32 | dangerarea | london |
19:56.53 | InfraRed | dangerarea: magrathea(they rock), gradwell |
19:56.54 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
19:57.21 | dangerarea | well they might but I'm looking for a price for someone to terminate them to one non-geo number deliver them over SIP/AIX/IAX2 and then us to do IVR |
19:57.47 | InfraRed | magrathea or gradwell |
19:57.52 | dangerarea | hence trying to find out about maximum call handling |
19:58.12 | dangerarea | not looked at magrathea, i shall do now |
19:58.16 | dangerarea | cheers again InfraRed |
19:58.20 | InfraRed | their website is shite |
19:58.31 | dangerarea | as all good companies should be |
19:58.49 | dangerarea | it means they have techies that know what they're doing |
19:58.49 | InfraRed | but they deliver one of the best voip service sin the uk |
19:59.00 | dangerarea | rather than building websites |
19:59.06 | dangerarea | all techies hate websites |
19:59.08 | dangerarea | :) |
19:59.15 | InfraRed | yep |
19:59.25 | InfraRed | they can deliver geo and non-geo number |
19:59.26 | InfraRed | s |
19:59.57 | dangerarea | cool |
20:00.14 | InfraRed | of any area code in the uk also port number from pstn to voip |
20:00.27 | dangerarea | i've had a look at that wiki page on simultanious calls |
20:00.28 | InfraRed | which is very handy |
20:00.41 | dangerarea | but still not really any the wiser |
20:01.12 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
20:01.47 | InfraRed | well |
20:01.54 | InfraRed | are you doing pure voip ? |
20:02.10 | dangerarea | here's the full brief... |
20:02.12 | dangerarea | 1000 calls in |
20:02.16 | dangerarea | ivr handles them |
20:02.29 | dangerarea | say 30 out to another phone system |
20:02.44 | dangerarea | (avaya ip office) |
20:02.52 | InfraRed | do you have the bandwidth ? |
20:03.24 | dangerarea | not at the mo but we have tonnes of fibre in to our building |
20:03.46 | InfraRed | 1000 x 80Kbps per call |
20:03.46 | dangerarea | can be done at 30 mins notice |
20:03.47 | InfraRed | ~ 80Mbps |
20:03.50 | drray | see if they will coloc your asterisk box |
20:03.59 | dangerarea | really 80k per call? |
20:04.06 | InfraRed | depends on the codec |
20:04.14 | InfraRed | i use 711u |
20:04.20 | InfraRed | since i find 729 is shite |
20:04.50 | dangerarea | well since most of the network uses 711u/711a we'd be using that |
20:04.59 | InfraRed | sorry, 711a |
20:06.22 | dangerarea | yep |
20:08.27 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
20:09.52 | terrapen | so, is there a way to configure password-less agents? |
20:10.10 | terrapen | i'm trying to ring all phones in this store when a call comes in and I don't want anyone to have to log onto the queue, ever |
20:10.21 | Vorondil | okay, just out of curiosity: since 711u is often refered to as u-law, which is pronounced "mu law" (the greek letter), is 711a pronounced "alpha law" or "a law"? |
20:10.27 | CunningPike | terrapen: We use members instead: member => SIP/1234 |
20:10.43 | jbalcomb | Hrmm.. the Polycom SoundStation IP 4000 is $200.00 USD cheaper that the IP 3000. |
20:10.55 | terrapen | cunningpike, thanks |
20:11.05 | jbalcomb | Vorondil no one says mu-law |
20:11.30 | Eonz | guys is possible record the calls with asterisk |
20:11.45 | jbalcomb | terrapen have you tried the meetme hack from the wiki? |
20:11.54 | jbalcomb | Eonz yep, check the wiki for details |
20:12.09 | InfraRed | terrapen: you dont need agents |
20:12.12 | [TK]D-Fender | jbalcomb : Stay analog, go WIRELESS ;) |
20:12.13 | InfraRed | just use extensions |
20:12.24 | Eonz | jbalcomb: k thanks |
20:12.25 | InfraRed | then google the wiki for extensions configuration |
20:12.31 | jbalcomb | [TK]D-Fender strange that its not the ip phone thats wirelss |
20:12.34 | InfraRed | you can have a ring all facility |
20:12.57 | jbalcomb | [TK]D-Fender i do want the linksts wifi sip phone for myself though |
20:13.10 | jbalcomb | [TK]D-Fender i gotta finish revamping our wifi network first though |
20:15.10 | jbalcomb | [TK]D-Fender I finished my perl script that reads the arp.dat from ArpWatch and put the MAC and IP in a MySQL DB!! |
20:15.24 | Vorondil | jbalcomb: nobody? i guess i'm just weird then. i mean, it *is* the greek letter mu, right? |
20:15.56 | wese103 | The ITU-T G.711 spec actually has the greek letter mu in it. |
20:15.57 | jbalcomb | Vorondil: You are likely a bit /unique/ and I have no idea if its the greek letter mu. |
20:16.06 | wese103 | And for alaw, it uses simply an 'a'. |
20:16.12 | wese103 | Well, an 'A'. |
20:16.18 | *** join/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net) |
20:16.37 | jbalcomb | its "yew-law" and "aye-law" that i use and hear used. |
20:17.00 | wese103 | We've always said "mu" and "aye" here. |
20:17.11 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
20:17.31 | japerry | anyone here well inveloped within Asterisk, and lives in the seattle area? |
20:17.37 | drray | I do |
20:17.39 | jbalcomb | I'm not the autority on anything though cause people throw things at me something because i say "ga-nu" for GNU. |
20:18.09 | jbalcomb | Tom Robbins lives in Seattle. |
20:18.27 | japerry | nice nice.. we're looking for someone to consult our asterisk system |
20:18.55 | wese103 | Yea, I actually try to avoid verbalizing GNU. :) |
20:19.09 | jbalcomb | japerry: I'll take the cheap flight if'n need you can't find anyone else. |
20:19.13 | drray | wese103 - How do you say voip? |
20:19.26 | japerry | jbalcomb: thanks |
20:19.51 | japerry | jbalcomb: checking local refs first, I'll keep you in mind though if we can't :-) |
20:19.56 | jbalcomb | drray Are you looking at "voyp" rather than "v-o-i-p"? |
20:19.56 | MACscr | v-oy-p |
20:19.59 | wese103 | I would incline to say it with an "oy" sound. But I work with a lot of Germans who like to say it with an "oh" sound. |
20:20.12 | drray | voyp annoys me |
20:20.24 | MACscr | to bad its the correct way |
20:20.25 | InfraRed | i saw vooopah |
20:20.26 | wese103 | I find myself mostly saying the entire "voice-over-ip" phrase instead. |
20:20.33 | drray | I say voice over ip |
20:20.45 | jbalcomb | is the e starting ethernet long or short? I hear short and I think they sound like a hick. |
20:20.55 | rob0 | I say toe-may-toe. |
20:21.00 | jbalcomb | agreed |
20:21.13 | jbalcomb | though its almost more ta-may-toe |
20:21.15 | *** part/#asterisk acrg (n=aragon@decoder.geek.sh) |
20:21.20 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
20:22.18 | wese103 | I get poked a little bit for my pronounciations. It makes life a little more colorful. |
20:22.27 | rob0 | I'm really uncomfortable with how I say [un]comfortable. I leave out a syllable. I have a large syllabic deficit by now. |
20:22.42 | jbalcomb | Of course, I should note that I'm from Cleveland, OH which is considered to nearly be the heart of what is known as proper pronunciation of US/Microsoft English. |
20:22.59 | rob0 | nonono Walter Cronkite is from Nebraska! |
20:23.13 | rob0 | And that's the way it is. |
20:23.42 | jbalcomb | Walter Cronkite.. feh. That guy talks like somebody who needs a punch in the face. |
20:23.59 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
20:24.00 | wese103 | I work with a guy from India, and his English is impeccibly the Queen's. We have long discussions comparing my American with his proper English. |
20:24.36 | jbalcomb | wese103: My Japanese Sensei is from India as well and we have similar amusements. |
20:24.58 | jbalcomb | wese103: The best so far has been the Japanese word 'mafura |
20:25.13 | wese103 | meaning? |
20:25.31 | jbalcomb | which is the English word Muffler but really it's the British word for scarf as opposed to the US English sound reduction device under most vehicles. |
20:25.54 | wese103 | Heh. :) |
20:26.12 | wese103 | I had a similar miscommunication with a German coworker about the workd "beamer". |
20:26.17 | wese103 | He said the boss agreed to get us a "beamer". |
20:26.23 | wese103 | I was floored. |
20:26.26 | wese103 | "Wow!" |
20:26.28 | jbalcomb | and you were like HELL YEAH! |
20:26.39 | wese103 | Turns out he meant an overhead projector to connect to a VGA port. |
20:26.57 | jbalcomb | haha... oh man, now that is the def. of disappointment. |
20:26.58 | wese103 | It was rather anticlimactic. |
20:28.00 | wese103 | Ok, back on topic... anybody here work with Digium's TE110P card? |
20:28.23 | Nugget | It's weird to see someone say "def." to obscure the fact that they can't spell "definition." Usually people say "def." because they're embarassed by not being able to spell "definitely" |
20:28.52 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
20:28.55 | CunningPike | wes103: We have one - what's up? |
20:29.24 | wes103 | Is there any way I can verify clocking source? |
20:29.28 | CunningPike | japerry: I'm in Vancouver, BC - so is dlynes_office |
20:29.35 | giesen | Is there a way in asterisk queues to have asterisk prompt the agent receiving the call telling them they have to hit # to accept the call |
20:29.35 | wes103 | Any way to poll it and see how many slip errors it might be getting? |
20:30.18 | wes103 | I have an E1 link, and I have set both ends so that neither is clock source. I expect to see some errors, but don't know where to look. |
20:30.22 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
20:30.31 | CunningPike | wes103: Your CLI (-vvv) will usually show HDLC errors if you are getting timing errors - what makes you think you are? |
20:30.40 | wes103 | Oh, nothing. |
20:30.44 | wes103 | I am just testing out the card. |
20:30.54 | wes103 | Haven't gotten asterisk fully configured yet. Just zaptel drivers. |
20:31.06 | wes103 | Trying to connect it to an IMACS channel bank. |
20:31.26 | wes103 | No alarms, so I have framing and carrier. |
20:31.35 | CunningPike | wes103: Ah - you may see HDLC errors then when you get asterisk running, unless there is a timing source on the PRI |
20:31.43 | wes103 | This is CAS. |
20:31.58 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
20:32.45 | wes103 | Well, I will wait until I get the entire system up, and then see if I "need to worry about it". |
20:32.47 | wes103 | :) |
20:33.02 | CunningPike | wes103: Good plan :D |
20:33.47 | wes103 | But, in zttool, no matter how I set the timing source, it always shows sync source as internal. |
20:34.04 | ptinsley | ok guys, my problem with my pri has come back, even after changing the trunk to descending as I assumed it would |
20:34.16 | ptinsley | is there ANYTHING i should run on it before i restart asterisk to get debug info |
20:34.26 | ptinsley | outbound calls work |
20:34.29 | ptinsley | all inbound calls fail |
20:35.17 | justinu|laptop | wes103: i think that's a very irritating known bug |
20:35.23 | wes103 | Ah. |
20:35.26 | ptinsley | Jun 27 15:35:17 WARNING[17647]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/13 already in use on span 1. Hanging up owner. |
20:35.27 | ptinsley | Jun 27 15:35:17 WARNING[17647]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/14 already in use on span 1. Hanging up owner. |
20:35.32 | wes103 | I tried looking in google for it, but didn't find anything. |
20:35.45 | wes103 | I will let it rest for now. |
20:35.51 | wes103 | Thanks. |
20:36.48 | CunningPike | wes103: It may be because there isn't a timing source on the line, so it's failing back to internal |
20:37.19 | wes103 | Shouldn't there be a message somewhere to that affect though? |
20:37.25 | ptinsley | oh well, restarted, everything works again, till next time :/ |
20:37.35 | wes103 | Switching the source would cause alarms in a Telco. |
20:38.01 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
20:38.14 | trelane_ | are there any module options for wctdm24xxp? I need to steer it to a different IRQ |
20:38.48 | justinu|laptop | no, you'll have to move the card |
20:39.05 | CunningPike | wes103: You know - I don't know that there is...... |
20:39.15 | wes103 | ok |
20:39.23 | CunningPike | wes103: Your telco should be the timing source...... |
20:39.54 | wes103 | I have a good source coming into the IMACS from the primary E1. I have the secondary E1 going to this TE110P card in a testing system. |
20:40.09 | justinu|laptop | the timing source is either the inbound framed DS1 signal from your telco, or the internal oscillator |
20:40.10 | wes103 | I guess I need to find out if the IMACS is passing it through. |
20:40.26 | wes103 | Hence, my wondering about slip error counts. :) |
20:40.46 | CunningPike | wes103: Yup - doesn't sound like it is right now..... |
20:42.10 | wes103 | Well, troubleshooting that is out of scope here. It sounds like zttool and /proc/zaptel won't report slip errors. So I have to get asterisk configured to see how it sounds. |
20:42.38 | wes103 | Time to bolt. |
20:42.50 | vader-- | so once a user records a call where does it go? and how can they retrieve it? |
20:46.28 | *** join/#asterisk ncef (n=cef@38.119.128.203) |
20:46.38 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
20:46.44 | *** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
20:46.53 | jbalcomb | vader--: Have your set up the Asterisk ARI? |
20:47.04 | InfraRed | ARI? |
20:47.22 | Bullseye_Network | when doing a show channel Zap/whatever there are some level1 CDR Variables listed can I call them in the dial plan like ${dstchannel} or ${lastdata} I didnt see it in the wiki |
20:47.25 | jbalcomb | Asterisk Recording Interface; lets people retrieve voicemail and call recordings. |
20:48.30 | luke-jr_ | Anyone know a good VoIP provider similar to SellVoIP (except with actual support)? |
20:48.33 | jbalcomb | Bullseye_Network: not sure if you can do it natively but if not you can probably make a system call to a script that returns a value to use |
20:49.24 | vader-- | na i haven't set that up |
20:49.28 | vader-- | i have voicemail setup |
20:49.29 | vader-- | but not that |
20:49.32 | Spy000007 | luke-jr_: try connect.voicepulse.com |
20:49.51 | jbalcomb | I have the MAC addresses and IP addresses of all my IP phones. Now I need to get the extension and username. |
20:50.02 | jbalcomb | What is a good way to look that up? |
20:50.26 | InfraRed | snmp |
20:50.27 | InfraRed | :) |
20:50.55 | jbalcomb | I'm thinking I could pull the asterisk db into an array, search for the IP, and regex the extension out. Once I have the extension I can grep the sip.conf to get the username. |
20:51.43 | InfraRed | 3-1 to france |
20:51.48 | jbalcomb | vs. ? |
20:52.08 | InfraRed | spain |
20:52.22 | luke-jr_ | Spy000007: per-minute outgoing looks nice, but that's about it :p |
20:52.23 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
20:52.26 | InfraRed | they'll play brazil next |
20:52.32 | Bullseye_Network | jbalcomb: think I found it ${CDR(<name>)} does that look familiar? |
20:52.50 | Spy000007 | luke-jr_: You asked for an ITSP with actual support |
20:52.58 | Spy000007 | luke-jr_: Or were you looking to get it for free? |
20:54.21 | luke-jr_ | Spy000007: similar to SellVoIP |
20:54.26 | jbalcomb | Bullseye_Network: I haven't used it but its listed on the wiki. |
20:54.26 | jbalcomb | Turns out you can't use Set(CDR(<name>)=value) for anything but userfield and accountcode. |
20:54.26 | jbalcomb | These fields are read-only. |
20:54.36 | Spy000007 | luke-jr_: ok, good luck |
20:54.43 | luke-jr_ | Spy000007: I was looking for per-minute incoming w/ DIDs themselves being cheap |
20:54.45 | jbalcomb | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr |
20:55.37 | vader-- | ARI seems like it's something for AMP or ASterisk@home |
20:56.08 | Bullseye_Network | jbalcomb: I only need to read them |
20:56.23 | jbalcomb | vader--: noe |
20:56.35 | jbalcomb | Bullseye_Network: I imagine that means you're all set then |
20:58.36 | vader-- | is ARI built into asterisk default package? |
20:59.28 | nortex | vader--, ARI is from littlejohnconsulting.com and is part of the FreePBX package, but can be run on any Asterisk install. |
20:59.44 | vader-- | gotcha |
20:59.52 | vader-- | there is no native way to playback these files through asterisk? |
21:00.04 | vader-- | or have them emailed to the user when the call monitoring is finished |
21:00.47 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
21:02.09 | ariel_ | I am trying to setup a callerID change but it's not working can you guys look at this and help me see what I am doing wrong? http://pastebin.ca/73438 |
21:02.27 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
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21:05.11 | directory | ariel_: ${CALLERID(number)} |
21:05.18 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
21:05.25 | directory | that's if you want to get the callerid number... |
21:06.44 | *** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com) |
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21:08.04 | jbalcomb | vader-- not that i know of. you may try google and/or the wiki. |
21:08.34 | jbalcomb | Is there really no one doing any sort of IP phone system management? |
21:09.32 | jbalcomb | I've only got 150 phones and I'm sick of maintaining the phones and users. Surely someone must have done something... |
21:12.11 | ariel_ | directory, thanks |
21:19.06 | *** join/#asterisk electus (i=electus@113.129.8.217.in-addr.arpa) |
21:19.34 | electus | Anyone here that has some experience with the Originate command in asterisk API? |
21:19.39 | electus | got some problems |
21:20.06 | electus | WARNING[21212]: pbx.c:2353 __ast_pbx_run: Channel 'SIP/62-d5d9' sent into invalid extension 's' in context 'default', but no invalid handler |
21:20.39 | electus | when I try to call a number with my intern number 62 |
21:21.13 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-11-67.PHNX.QWEST.NET) |
21:22.05 | *** part/#asterisk colinm_ (n=colol@VDSL-130-13-11-67.PHNX.QWEST.NET) |
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21:23.56 | electus | I got one context I use wich is 'internt' |
21:24.29 | electus | anyone? |
21:24.56 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
21:26.02 | electus | got a very basic config, so maybe Im missing something |
21:26.03 | Druken | anyone want my headache? |
21:26.15 | FuriousGeorge | anyone using snom 360's notice that when they either assign a park orbit button (may only work with a patch) or transfer to the park extension, the user doesnt hear what "spot" the call was parked on? |
21:28.01 | FuriousGeorge | ~seen shmaltz |
21:28.18 | jbot | shmaltz <n=mybox@mail.dmaven.com> was last seen on IRC in channel #asterisk, 22h 51m 36s ago, saying: 'anybody seen any problems with queus when the members are Sip/sipaccount?'. |
21:28.37 | *** part/#asterisk MACscr (n=MACscr@66.73.154.70) |
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21:30.55 | mountainm2k | anybody heard of Advantatech ? Polycom referred me to them... |
21:33.35 | mountainm2k | quiet today... |
21:38.01 | FuriousGeorge | anyone using snoms notice users dont hear what extensiona call is parked on? |
21:38.22 | Strom_C | FuriousGeorge: that will happen with any phone if you do a blind transfer |
21:39.05 | CunningPike | jbalcomb: Give me ssh access and I'll do it for you :) |
21:39.57 | *** join/#asterisk benno|b1 (n=ben@88-96-30-86.dsl.zen.co.uk) |
21:40.19 | CunningPike | Druken: What's your headache, apart from being named after an NHL player who never realized his full potential? |
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21:41.02 | FuriousGeorge | Strom_C: i set uyp a park orbit button, but lemme try attended. if i try to call the extension first the phone says "address incomplete" and the CLI says nothing. im pretty sure this happened before i applied the metermaid patch to ge the LEDs going |
21:41.41 | FuriousGeorge | in fact i applied the patch because parking was not telling me the number as it would not let me call the park extension directly |
21:41.49 | FuriousGeorge | for atxfer |
21:42.35 | Strom_C | *shrug* I've used call parking with attented transfers just fine without any patches |
21:47.41 | nortex | Strom_C, We do a blind transfer to parking, but it is a PBX transfer with the # key. |
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22:01.03 | nortex | I have seen a little bit on this, but I have a dumb question about Sangoma Echo cancelation cards, I have a A104D and yet there are still times that there is echo on the calls, but only on my side. Should there be any echo with a hardware echo cancelling card? |
22:01.19 | Qwell[] | nortex: there can be, sure |
22:02.02 | nortex | Qwell, Is there anything I can do about it? |
22:03.39 | *** part/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
22:06.38 | mindwarp | Hi, I have a problem with my SPA3000 apparently not passing inbound calls to Asterisk. I have a phone connected to the Sipura's "phone" port, and the "line" port is connected to PSTN. When PSTN rings, the phone connected to the Sipura rings, but Asterisk shows no signs of "seeing" the call (no activity in the CLI with extreme verbosity turned on). I did configure the PSTN-to-VOIP bridge on the Sipura. Does anybody know what may be causing this is |
22:07.28 | [hC] | zaptel suddenly doesnt want to compile... any ideas why i'd be getting this? http://pastebin.ca/73487 |
22:07.39 | [hC] | seems like a libc error? |
22:09.33 | benjamin7062 | Can anyone give me a real life benchmark on recorded calls... does a decent powered machine handle... say.. 40+ monitors at the same time? |
22:09.58 | justinu | ime, yes |
22:11.13 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
22:12.08 | [hC] | well for the future |
22:12.16 | [hC] | current libc in debian unstable breaks zaptel. |
22:12.27 | [hC] | i downgraded and it works now. |
22:13.12 | X-Rob_ | nortex, you should be speaking _to sangoma_ -- their tech support is amazingly good. |
22:14.37 | nortex | X-Rob_, I did a few weeks back on a fax over PRI issue and got no where. I'm really just curious if there is a problem or the norm. |
22:15.02 | brijn | benjamin7062, Google for asterisk benchmarks.. Somebody did a bunch of test runs and put it all on a page |
22:15.04 | X-Rob_ | well, that's kinda reasonable. fax over voip is a silly thing to do. |
22:15.17 | X-Rob_ | getting good echo cancellation is not unreasonable. |
22:15.30 | X-Rob_ | yes, you can still get echo, but you shouldn't notice it enough to bring it up in channel 8) |
22:16.23 | benjamin7062 | brijn, Good Call.. I could just turn it on and break everyone if it fails. =) |
22:16.52 | benjamin7062 | Our callcenter gets a crap ton of traffic... don't want the IO to hurt the machine. I'll look for the site. TY |
22:16.59 | nortex | Thanks, I'll check into it with them. The fax thing was not VOIP, it was PRI to channel bank on the same card and failed terribly, |
22:17.35 | CunningPike | mindwarp: Is it registered? |
22:18.15 | terrapen | anybody ever seen a Polycom minibrowser app that can show a list of parked calls? |
22:18.25 | dahunter3 | Guys, I have a question about merging an inbound phone call with an outbound phone call. The volume is really low and it's extremely hard for both sides to hear each other. I assume I need to mess with the gain settings. Is there a way to only do this for inbound + outbound calls? |
22:19.00 | terrapen | merging? you mean forwarding? |
22:19.02 | nortex | terrapen, Yes, look on voip-info there is a pack of XML files that can be twiked |
22:19.04 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
22:19.09 | terrapen | i've never heard of "merging" |
22:19.13 | SexyKen | Hey guys -- is there anyway to see how many channels have been in use at one time? |
22:19.14 | terrapen | thanks nortex |
22:19.29 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
22:19.40 | dahunter3 | I'm not sure what the right terminology is for it. A call comes in and the dialplan connects it with an outside number. |
22:19.44 | CunningPike | SexyKen: You mean zap channels |
22:19.55 | nortex | SexyKen, show channels will give the current stat. |
22:19.57 | SexyKen | Well, No -- becuase I'm using PURE VOIP |
22:20.02 | SexyKen | I dont want current stat |
22:20.16 | SexyKen | I want to know the highest number of hcannels that were in use at one time |
22:20.40 | mindwarp | CunningPike: i tried both with and without registration (i.e. both registering it and setting it to host=dynamic and doing "call/answer without registration" setting it to a static ip) |
22:20.44 | terrapen | nortex, sweet, found what i needed. shoulda looked there before askin' |
22:20.51 | heath__ | run a crazy assed query against your cdrs? |
22:20.56 | nortex | terrapen, No problem |
22:21.21 | terrapen | i'll have to re-write it though. i hate PHP :) |
22:21.31 | dahunter3 | terrapen: So, I guess that's forwarding. |
22:21.34 | mindwarp | CunningPike: i should also note that if i dial with the console ('dial extension@context') then it will work as it should and the console will tell me a bunch of stuff about the incoming call, not sure if that really means anything |
22:21.35 | terrapen | I'll write a simple CGI in C or Perl or somethin |
22:21.46 | terrapen | dahunter: "bridging" |
22:22.08 | nortex | terrapen, I've modified it, but the operators here struggle to use the phone, let alone anything cool like a microbrowser. |
22:22.22 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
22:23.21 | CunningPike | mindwarp: Not really :D At least we know that your dialplan works. It's likely to be something in your ATA's setup |
22:23.29 | terrapen | nortex, i'm afraid that's what I might be up against |
22:23.55 | terrapen | nortex, I wonder if there is a way to make the microbrowser interface allow an agent to pick up a parked call directly from the UI |
22:24.16 | mindwarp | CunningPike: the PSTN-to-VoIP gateway in the Sipura is using a dialplan that looks like (S0<:192.168.0.x) (address of the asterisk server) ... but the asterisk server doesn't seem to be given anything |
22:24.23 | terrapen | i suppose i could do it with the manager API but it would be nicer to do it straight from within the microbroser |
22:24.59 | CunningPike | mindwarp: Ah - that doesn't look like a valid dialplan to me....... hang on a sec |
22:25.25 | mindwarp | CunningPike: thanks |
22:25.38 | terrapen | dahunter, you might be able to find an app that will allow you to adjust the gain |
22:25.51 | terrapen | so before you call Dial(), you'd call the gain adjustment app |
22:25.54 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
22:25.56 | terrapen | dunno if that is available |
22:26.21 | dahunter3 | terrapen: Interesting. Am I pretty much on the right track that it is a gain issue? |
22:26.52 | terrapen | probably |
22:27.02 | benjamin7062 | terrapen, the Microbrowser is easy to code against... and connecting the parked calls works well with the Manager API... fyi |
22:27.36 | terrapen | i'd explain your problem on the list (make sure you mention that you are trying to adjust the gain on two bridged channels) and see if there is a gain adjustment app |
22:27.52 | terrapen | benjamin, any chance you could let me see your API code that connects them? |
22:28.00 | terrapen | i'm just lazy and it would save me some time :) |
22:28.08 | smackus | I have started getting this error: Jun 27 16:19:34 WARNING[10304]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x77f5b0', 10 retries! I have been reading around and have only learned how to log more verbose output for the error. Is there a fix for this? I just started happening. I must have changed something. but I dont know what. |
22:29.35 | *** join/#asterisk jvictorfc (n=jvictorf@201009011114.user.veloxzone.com.br) |
22:30.02 | jvictorfc | hi all |
22:30.17 | jvictorfc | please i give a help |
22:31.38 | jvictorfc | i need a md3200, I am having problems when I go up the module wcfxo for the md3200 |
22:31.50 | CunningPike | mindwarp: Try this: (xx.) |
22:31.57 | mindwarp | CunningPike: trying |
22:32.12 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
22:33.09 | mindwarp | CunningPike: same behavior -- the line rings, the phone connected to the Sipura rings, but Asterisk remains utterly silent |
22:33.24 | jvictorfc | its a problem: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
22:33.25 | jvictorfc | FATAL: Error running install command for wcfxo |
22:33.45 | jvictorfc | can anybody help myself? |
22:33.51 | terrapen | heh |
22:34.04 | CunningPike | mindwarp: Do you have the admin manual? There are some good examples in it..... |
22:34.08 | dahunter3 | terrapen: Thnk you for your help. |
22:34.13 | terrapen | np |
22:34.55 | mindwarp | CunningPike: I guess I should really read that... I haven't been able to find it on the Sipura site yet, but I've heard it's supposed to be there |
22:35.12 | mindwarp | the dialplan above was taken from a tutorial which supposedly is for what I'm trying to do |
22:35.16 | mindwarp | but the settings seem to do nothing |
22:35.34 | CunningPike | mindwarp: It is - the admin guide was helpful to me |
22:36.43 | mindwarp | CunningPike: do you happen to have a link for it handy or could you tell me where you found it? I can't see it on the support page for the SPA3000? |
22:37.29 | mindwarp | CunningPike: ah, nevermind, I found it :) Thanks for your help, I may ask for more once I RTFM |
22:37.55 | CunningPike | mindwarp: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf - yup, sure, no problem |
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22:43.59 | nortex | I'm looking over the Polycom reboot script on voip-info and get the impression that it will only reboot one phone, or am I missing something? I want to nightly reboot all my Polycom phones. |
22:44.02 | *** join/#asterisk rnovotny22 (n=rnovotny@198.57.19.126) |
22:44.32 | terrapen | nortex, yeah, you'd have to hax0r something that got the info from 'sip show peers' |
22:44.46 | terrapen | that might be able to be accessed through the manager api, I'm not sure... |
22:45.43 | mindwarp | uhm, should 'sip show registry' give me a list of registered peers? because it returns nothing (an empty list) and I thought I had 2 peers registered |
22:46.30 | directory | mindwarp: sip show peers |
22:46.30 | nortex | If I had a list of the peer names I could just pass them to the asterisk cli with sip notify though right. |
22:46.41 | terrapen | sip show peers |
22:46.47 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
22:46.48 | mindwarp | directory: thanks |
22:49.42 | *** part/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
22:50.15 | *** join/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net) |
22:50.46 | phigwork | hi, my pbx is being weird with local outgoing calls (out the fxo) |
22:51.21 | phigwork | if I have this: exten => _9.,1,Macro(dialout,${TRUNK},${EXTEN:1}) |
22:51.36 | phigwork | it doesn't actually dial everything but the 9.. or it's leaving something else out |
22:53.28 | CunningPike | nortex: I wrote a simple shell script that uses 'sip show peers' - want it? |
22:55.08 | phigwork | if I leave out the :1, it seems to work |
22:55.09 | phigwork | but why |
22:55.32 | *** join/#asterisk Nurstonix (i=fwuser@200.27.54.240) |
22:56.41 | *** join/#asterisk RoyK[no] (n=roy@svg-acs.ipzone.no) |
22:56.42 | salaud | quick question... how do you / can you setup hunt groups for sip in asterisk? |
22:56.54 | salaud | does the trunk methodology only work for PRI or Zap? |
22:57.09 | salaud | can do something like SIP/trunkid and have it hunt? |
22:59.34 | nortex | CunningPike, Sure that would be great. |
23:02.53 | benjamin7062 | phigwork, Good question.. =) |
23:03.22 | RoyK[se] | <PROTECTED> |
23:04.58 | *** join/#asterisk quadrata (n=quadrata@ool-44c61ecb.dyn.optonline.net) |
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23:10.07 | phigwork | hehe, no ideas? |
23:10.10 | phigwork | maybe just a bug? |
23:11.52 | sliderool | hello. does anyone here have experience using app_swift for realtime text to speech or would know what is causing these errors: "app_swift.c: Poll timed out/errored out" "app_swift.c: No more data." |
23:13.04 | giesen | Is there a way in asterisk queues to have asterisk prompt the agent receiving the call telling them they have to hit # to accept the call |
23:14.43 | giesen | phigwork: that looks right, so if it doesnt work, it's probably a bug |
23:15.04 | giesen | maybe your macro has a bug in it |
23:15.11 | giesen | does it work with a plain jane Dial cmd? |
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23:36.35 | japerry | in /var/log/asterisk/messages? |
23:38.14 | CunningPike | japerry: Yes - or event_log |
23:43.12 | japerry | CunningPike: yup |
23:45.50 | CunningPike | Sorry - brain fart - there is another setting that deals with call supervision...... |
23:46.43 | *** join/#asterisk goldsmurf (n=rgoldber@64-13-22-231.dul.clearwire-dns.net) |
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23:48.57 | CunningPike | japerry: Nope - sorry, red herring |
23:49.29 | CunningPike | I think you're going to have to get lucky enough to get some PRI debug output from one of these dropped calls |
23:51.17 | CunningPike | japerry: You can try messages => notice,warning,error,debug in your logger.conf and restart asterisk. |
23:54.55 | japerry | CunningPike: okay, I'll try that |
23:55.34 | CunningPike | OK - let me know how you make out. On the CID thing, was it inbound or outbound? |
23:55.47 | *** join/#asterisk iq|mobile (n=iq@unaffiliated/iq) |
23:57.10 | japerry | CunningPike: Both.. CID incoming doesn't wrok, and outside phones don't see our number either |
23:57.22 | mrdigital | CunningPike? |
23:57.55 | japerry | CunningPike: I keep getting those debug messages every second to different phones |
23:58.29 | CunningPike | japerry: From the zap channel? |
23:58.34 | CunningPike | mrdigital: ? |
23:58.58 | mrdigital | pm? |
23:59.35 | *** part/#asterisk benno|b1 (n=ben@88-96-30-86.dsl.zen.co.uk) |
23:59.38 | CunningPike | japerry: That's coming from chan_sip, I notice - not from chan_zap...... |
23:59.44 | japerry | Cunningpike: DEBUG[23371] chan_sip.c: Auto destroying call '302ea438a1df97f5@10.0.6.105' and I get DEBUG[23371] chan_sip.c: Stopping retransmission on '4cc86356543fc6ff2d410ac168fc8b31@10.0.6.14' of Request 102: Match Found |