| 00:04.33 | *** join/#asterisk salvatore2 (n=asd@host-19-4.mielccc.clients.pavlovmedia.com) |
| 00:04.38 | salvatore2 | hello |
| 00:04.47 | salvatore2 | no reply to our critical packet, and no sound in my calls, why? |
| 00:06.37 | Strom_M | salvatore2: let me guess |
| 00:06.40 | Strom_M | SIP through NAT |
| 00:09.06 | JT | ylon: what technology is the channel type? |
| 00:09.57 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.198) |
| 00:10.06 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 00:12.51 | *** join/#asterisk AGWA (n=andrew@AndrewsEast-148.resnet.brown.edu) |
| 00:13.42 | *** part/#asterisk AGWA (n=andrew@AndrewsEast-148.resnet.brown.edu) |
| 00:14.11 | ylon | kand: that looks excellent |
| 00:14.16 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:14.30 | ylon | JT: you'll have to forgive me, but I really am novice with asterisk. how can I find out? |
| 00:14.54 | ylon | JT: are you referring to zaptel or another? |
| 00:15.01 | ylon | or something totally different? |
| 00:15.40 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 00:17.23 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 00:20.52 | JT | ylon: it must be some form of connection |
| 00:21.58 | fujin_ | technology = sip/zap/iax2.. etc. |
| 00:22.14 | fujin_ | gah, it'd be awesome if people would familiarise themselves with the technology before trying Asterisk |
| 00:22.23 | fujin_ | mind you, I guess I did the same, came in here talking about sip trunks |
| 00:24.02 | GreggB | fujin_: watch out, trunk is a bad word around here |
| 00:24.37 | fujin_ | that's exactly what I just said |
| 00:24.55 | fujin_ | I'm more than aware that a fair percentage of this channel believes that sip trunk is an incorrect or non applicable term |
| 00:25.04 | fujin_ | It's a shame that the marketing people have coined it as a common phrase |
| 00:25.14 | fujin_ | where trunk = a way to deliver calls |
| 00:25.19 | fujin_ | vs. the true sense |
| 00:25.56 | coppice | if there's no elephant at the end, its not a real trunk |
| 00:26.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:27.20 | GreggB | LOL, so I guess ILEC = elephant ;) |
| 00:27.26 | sigmounte | hello ! little problem , my busy lamp field only flash when a phone receive a call , not when it make a call itself , anyone know the problem ? |
| 00:28.08 | GreggB | sigmounte: what type/model of phone? |
| 00:28.18 | sigmounte | grandstream gxp 2000 |
| 00:28.31 | sigmounte | blf work fine , but only when the phone receive call :( |
| 00:28.38 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:29.00 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:29.12 | *** part/#asterisk SM0TVI (i=teodor@c83-253-122-78.bredband.comhem.se) |
| 00:29.18 | *** join/#asterisk chinku (n=chinku@c-69-142-143-206.hsd1.nj.comcast.net) |
| 00:29.28 | GreggB | ours dont blink when making a call. On the blf buttons, the line just goes solid when someone on a blf-monitored line is making an outbound call (hmm, I should double-check that though - brb). |
| 00:30.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:31.02 | grandpapadot | Hi all. I have a location that has 15 Polycom 501's running the latest firmware. I have hundreds of these phones out there, all geographically disperese using the internet to connect to asterisk servers (1.2.24). This one location is using DSL from CenturyTel and a Westell 2100 DSL router. They are behind NAT (sip.conf, nat=yes, qualify=3000). |
| 00:31.23 | grandpapadot | <PROTECTED> |
| 00:31.31 | sigmounte | mine blink when the monitored phone ringing , stay red when receiving call , but when the monitored phone make a call , no blf button change |
| 00:31.32 | JT | fujin_: i actually mean type of connection |
| 00:31.38 | JT | zaptel can be a bunch of things |
| 00:31.47 | grandpapadot | With sip debug on and debugging one of the unavailable peers, it sends the options packets 4 times every 10 seconds. |
| 00:31.48 | GreggB | sigmounte: yep, that's how it works - light just goes solid when the monitored line is making an outbound call, and blinks on inbound calls (when ringing on inbound). |
| 00:32.09 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 00:32.11 | hmmhesays | bah |
| 00:32.13 | grandpapadot | It takes about 3 minutes, but eventually the phone will respond to the options packet request. |
| 00:32.15 | grandpapadot | Any ideas? |
| 00:32.19 | sigmounte | solid red or green when making outgoing call ?? |
| 00:33.00 | grandpapadot | Oh, and the UDP ports being used are all 50100-50200. |
| 00:33.01 | GreggB | sigmounte: solid red when either making an outbound call (including ringing), or on an inbound call. Blinking red when ringing on inbound. |
| 00:33.08 | grandpapadot | (dynamic) |
| 00:33.46 | sigmounte | GreggB do you have the grandstream model with bi color led ? |
| 00:33.52 | GreggB | sigmounte: FW ver: 1.1.4.17 |
| 00:33.56 | grandpapadot | grandstream = crap |
| 00:33.57 | GreggB | sigmounte: yep |
| 00:34.08 | GreggB | grandpapadot: yea, no argument there |
| 00:34.26 | sigmounte | ok GrebbB and when nothing append to a phone led is off ? |
| 00:34.29 | GreggB | sigmounte: green when the monitored line is idle |
| 00:34.59 | kand | grandpapadot: what kind of router does the EU have? |
| 00:35.11 | sigmounte | ok , mine doesn't change on inbound , look like you have a better config than mine , can we chat in pv ? |
| 00:35.20 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:35.27 | GreggB | sigmounte: going home soon, but sure |
| 00:35.37 | sigmounte | thanks a lot |
| 00:39.42 | CBU[^_^]M`` | is it possible to edit sip.conf using cli?? :) |
| 00:41.31 | [hC] | did polycom discontinue the ip601? my distributor suddenly doesnt carry them anymore and replaced it with the 650 |
| 00:42.24 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-320a3582992ce4d4) |
| 00:42.54 | JT | [hC]: it would seem logical |
| 00:43.21 | GreggB | [hC]: voipsupply stil has them listed |
| 00:43.24 | *** join/#asterisk oej (n=olle@74.9.53.154) |
| 00:44.48 | JT | who would buy from voipsupply though |
| 00:45.29 | kand | [hC]: thevoipconnection.com still has cases listed |
| 00:45.50 | generalhan | ouch ... i buy from voipsupply |
| 00:46.00 | GreggB | JT: they treat me great, no problems with returns, and a great discount as a non-profit... |
| 00:46.27 | JT | wow, first positive feedback i've ever heard for them |
| 00:46.50 | generalhan | wow, glad i got lucky then !! |
| 00:46.51 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 00:47.04 | GreggB | JT: I've purchased ~30 pieces of equipment from them over the past year. sofarsogood... |
| 00:47.54 | JT | many people in here have had issues with voipsupply sending the wrong equipment, sending late, and being a pain with returns/changing to correct equipment |
| 00:47.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:48.03 | JT | they're also not that cheap ;) |
| 00:49.33 | GreggB | JT: hmph, I YMMV then, they came in pretty inexpensive for us once our discount kicked in. |
| 00:50.56 | hmmhesays | IF YOU HANDS BEGIN TO SHAKE, STOP REMOVING THE EYELINER BECAUSE YOU MIGHT STAB YOURSELF IN THE EYE |
| 00:53.03 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 00:56.10 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 01:05.00 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 01:05.05 | adeel | does * 1.4 have better dtmf handling than 1.2? |
| 01:07.33 | *** join/#asterisk asdx (n=diego@adsl-156-138.click.com.py) |
| 01:08.59 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 01:09.35 | hmmhesays | some days |
| 01:09.38 | kand | adeel: not really |
| 01:10.31 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 01:12.58 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:13.04 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 01:16.52 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 01:16.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:17.13 | *** join/#asterisk Innatech (n=it@netblock-208-127-172-97.dslextreme.com) |
| 01:19.46 | Innatech | So--I've been asking in #freepbx and #trixbox, but I think the problem I'm having has something to do with the glue between them, so I thought I might ask here too. I have a Rhino CEROS with a R4FXO-EC and a Teliax account. I've eliminated everything but the Teliax SIP trunk from the outbound route, but it still picks up Zap to dial. Meanwhile, I never see an attempt to register or call out on the SIP trunk. Meanwhile, this box i |
| 01:19.46 | Innatech | s meant to be up tomorrow and needs to get VOIP up to supplement the POTS lines. |
| 01:20.05 | Innatech | *the glue between them and * , that is. |
| 01:20.40 | Innatech | any pointers regarding what I should be looking for would be much appreciated. |
| 01:21.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:23.14 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:23.17 | fujin_ | you're doing it wrong |
| 01:23.21 | fujin_ | use less freepbx or trixbox |
| 01:28.37 | Innatech | So far as I can tell everything is fine. I realize it's likely a FreePBX issue, as I said. The dial script as shown in the CLI never tries anything but ZAP trunks. I suppose I'm looking for advice on a raw * workaround, such as placing a register line manually in sip.conf . Any constructive advice would be appreciated, convincing this client to go with plain * isn't going to happen. Flames to /dev/null : I'm too stressed to care |
| 01:28.37 | Innatech | . |
| 01:32.07 | [TK]D-Fender | Innatech, if its picking up zap to dial its FreePBX's fault (or your configuration of it). So basically nothing to be asked here. |
| 01:35.37 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 01:36.59 | *** join/#asterisk orkid (n=mike@unaffiliated/orkid) |
| 01:46.06 | Innatech | TK D-Fender yes, that was my immediate reaction too. It's just that I can't find anything in FreePBX that looks out of place. I have another box on site working fine with the same config. Oh well, I appreciate the definitive answer at least. |
| 01:48.34 | [TK]D-Fender | Innatech, Dial(SIP/something) willl NOT call Zap...... |
| 01:48.50 | [TK]D-Fender | Innatech, So ummm... DUH, yeah its FreePBX or your setup of it. |
| 01:50.03 | Innatech | yeah. I know. After staring at it for a few hours, you can't help but wonder. |
| 01:51.25 | [TK]D-Fender | Innatech, only if you're on crack :) |
| 01:51.41 | Innatech | At this point, I think that might help. |
| 01:51.52 | [TK]D-Fender | Innatech, * doesn't change your dialplan as its executing it. |
| 01:51.55 | Innatech | Seriously, something here is broken that just doesn't make sense. |
| 01:58.16 | Innatech | Part of me wants to start hand-hacking at the conf, but FreePBX is such a mess under the hood I'd be useless. I suppose I'll just finish up the Zap routing, the IVRs and the extensions and try to get VOIP up in the morning when people are answering their phones. Thanks for the sanity check, TK. |
| 01:58.50 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 01:59.47 | [T]ank | wondering if there is a shortcut or easier way to do multiple extensions. here is how i am doing it, but I have 300 extensions i want to try to do it with. (Trying paging) exten => 1901,1,Page(SIP/1000&SIP/1001&SIP/1003) |
| 02:00.10 | [TK]D-Fender | [T]ank, you want to page 300 extensions? |
| 02:00.24 | [T]ank | sounds crazy i know, but yes |
| 02:00.26 | [T]ank | :-D |
| 02:00.32 | Innatech | The man has a lot to say! |
| 02:00.38 | [TK]D-Fender | [T]ank, Ok, several extreme problem with that |
| 02:00.49 | [T]ank | shoot.. i am listening |
| 02:01.21 | [TK]D-Fender | [T]ank, 1st : Meetme (which runs this behind the scenes) will have a FLAMING HEART ATTACK trying to MUX that. |
| 02:01.31 | [TK]D-Fender | [T]ank, 2nd : raw channel load. |
| 02:01.40 | [TK]D-Fender | [T]ank, 3nd : call SETUP delay. |
| 02:02.00 | [T]ank | well, i can break it down in groups. could I do 50 at a time? |
| 02:02.05 | [TK]D-Fender | [T]ank, You'd have to wait for ALL sets to report back in and in so doing the massive flood would burst * like a ripe zit. |
| 02:02.17 | [T]ank | mmm zits |
| 02:02.30 | [TK]D-Fender | [T]ank, (imagery is a free of charge bonus_ |
| 02:02.38 | [T]ank | lol |
| 02:02.48 | [TK]D-Fender | [T]ank, Solution : store & forward paging. |
| 02:02.55 | [T]ank | what I could do is groups of phones. like every 3 in the same area |
| 02:03.01 | [T]ank | tell me more about store and forward. |
| 02:03.33 | [TK]D-Fender | [T]ank, do a Record, then ORIGINATE a call and playback to each independantly. True they won't play back all time synched to the sec, but it'll SURVIVE. |
| 02:03.59 | [TK]D-Fender | [T]ank, just the MUX load kills the idea of using Page |
| 02:04.27 | [T]ank | good idea... i think for what my client is looking for I can just do a few phones within earshot of each other and do a few phones. maybe? could that work? |
| 02:05.49 | [TK]D-Fender | [T]ank, yeah, if geography allows. Oh yeah.. the OTHER challenge... |
| 02:06.02 | [T]ank | i have also seen the voip intercom speakers. then i can just call their extension directly. have you seen or heard of anyone using those with success? |
| 02:06.05 | [TK]D-Fender | [T]ank, #4 : dialplan STRING LENGTH limit. |
| 02:06.35 | sigmounte | hello , really strange problem : 3 phones (raph,sigmounte,seb) when using any of the three phone the asterisk console show : Executing [400@salya-rezo:1] Playback("SIP/sigmounte-grandstream-081f46f0", "were-sorry") <-- the channel is wrong i'm using another one (seb or raph) |
| 02:06.47 | [TK]D-Fender | [T]ank, VoIP intercom speakers are just 1-way SIP phones with AA |
| 02:07.33 | [T]ank | yeah, but I can mount one in the ceiling and do an entire room. I have only seen the one listed on the wiki for just shy of $300. gotta be something cheaper |
| 02:08.00 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:08.00 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:08.24 | [TK]D-Fender | [T]ank, best way : get a real paging amp & Viking FXO paging unit and wire up loudspeakers throughout. |
| 02:08.40 | [TK]D-Fender | [T]ank, AA Phones for paging is good in small env's, not large |
| 02:09.08 | [T]ank | looking now for viking fxo. thanks for the chat. |
| 02:11.19 | [TK]D-Fender | [T]ank, np |
| 02:12.16 | [T]ank | question... |
| 02:12.27 | sigmounte | no idea why asterisk always use the same channel and not the real one ? |
| 02:12.43 | [T]ank | [TK]D-Fender: so i would need an FXS port on the asterisk server to do this, right? |
| 02:13.08 | [TK]D-Fender | [T]ank, For that you'd splurge on an ATA :) |
| 02:13.36 | [T]ank | i would have to dial out to an analog extension and into the FXO on the viking |
| 02:14.11 | [TK]D-Fender | [T]ank, Just buy an ATA, and dial it Dial(SIP/MyPagingElCheapoLinksys) |
| 02:14.32 | [TK]D-Fender | [T]ank, the Viking unit would be designed to always automatically answer. |
| 02:14.37 | [T]ank | ok |
| 02:14.38 | [TK]D-Fender | [T]ank, and presto |
| 02:14.59 | [T]ank | makes sense. thats how i will do it. |
| 02:15.01 | [T]ank | thanks for the info |
| 02:17.37 | *** join/#asterisk BeeBuu (n=chatzill@218.13.83.156) |
| 02:17.45 | BeeBuu | hello,all. |
| 02:19.26 | *** join/#asterisk jtexter3 (n=jamest@ppp-70-128-111-195.dsl.tulsok.swbell.net) |
| 02:20.25 | BeeBuu | exten=100,1,Set(CALLERID(all)=Jane Smith <88888888>) |
| 02:20.27 | BeeBuu | but i still got the original phone id,why i can't change to 8888888? someone help me,please. |
| 02:31.30 | *** join/#asterisk shamusmcirish (n=dcasey@ool-44c70dce.dyn.optonline.net) |
| 02:32.52 | shamusmcirish | hello |
| 02:33.07 | BeeBuu | shamusmcirish: can you help that? |
| 02:33.44 | shamusmcirish | saying hello, or being irish ? |
| 02:34.48 | BeeBuu | exten=>100,1,Set(CALLERID(all)=Jane Smith <88888888>) |
| 02:34.49 | BeeBuu | exten=>100,n,Dail(Zap/1-1/xxxxx) |
| 02:34.51 | BeeBuu | but i still got the original phone id,why i can't change to 8888888? |
| 02:35.56 | shamusmcirish | Is this over a t1? Perhaps the telco is overriding it.. Or ni2 signalling needs to be used. |
| 02:36.55 | BeeBuu | what's ni2? |
| 02:38.16 | shamusmcirish | NI-1 signaling allows you to set caller id number, NI-2 signaling allows you to set caller id name and number. |
| 02:38.30 | shamusmcirish | National ISDN-2 |
| 02:38.40 | BeeBuu | Oh... |
| 02:39.04 | BeeBuu | i got you. |
| 02:39.14 | shamusmcirish | It's funny you ask that. I ran into that problem today. |
| 02:39.28 | BeeBuu | you 2? |
| 02:39.40 | shamusmcirish | thats what the telco told me (XO Communications).. I told them to just set it for me on their switch |
| 02:40.27 | BeeBuu | so do i ? |
| 02:40.34 | shamusmcirish | Is your caller id being changed to something else, or just to Unknown. |
| 02:41.40 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.14) |
| 02:43.17 | shamusmcirish | Any one have experience with routing based on DNIS numbers? |
| 02:47.34 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 02:47.43 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:47.53 | iPod-nano | Alright, I need to make one Asterisk server log into another, what's the best way to do so? |
| 02:49.47 | iPod-nano | One server is in my friend's dorm room, which for obvious reasons cannot allow port forwarding, and my server is behind a router where ports are forwarded. How can we make his computer connect to mine and allow for two-way communication? |
| 02:50.34 | shamusmcirish | can you use vpn? |
| 02:50.45 | iPod-nano | No |
| 02:50.55 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 02:51.01 | shamusmcirish | ssh? |
| 02:51.20 | iPod-nano | Well, one way we can. |
| 02:51.30 | iPod-nano | He can connect to my server in that way. |
| 02:51.46 | shamusmcirish | and this is assuming that you cannot use iax because of the firewall |
| 02:51.57 | iPod-nano | I would assume so. |
| 02:52.35 | shamusmcirish | i would see if you could get iax working. |
| 02:52.56 | iPod-nano | What we need to do most is extend my server. |
| 02:52.58 | shamusmcirish | You can use ssh to make a two-way vpn. but that would be quite a hack job |
| 02:53.29 | iPod-nano | So when he logs into his server, he can log into his extension, even though it's hosted by mine. |
| 02:53.43 | lvl`home | have you tried hamachi? |
| 02:53.49 | iPod-nano | Never heard of it. |
| 02:54.10 | shamusmcirish | iax would be ideal if you had some more flexability |
| 02:54.32 | lvl`home | it's a p2p vpn thing |
| 02:54.45 | lvl`home | does a pretty good job at punching through routers/firewalls |
| 02:55.45 | iPod-nano | I just really need to make the two become one. |
| 02:56.13 | iPod-nano | Basically so he can log into mine throuh his. |
| 02:56.18 | shamusmcirish | interesting |
| 02:56.18 | iPod-nano | *through |
| 02:56.36 | iPod-nano | Want to hear the reason why we're trying to come up with such nonsense? |
| 02:56.51 | lvl`home | well, if hamachi manages to establish a connection, you can just connect to the other computer as if it has a public ip |
| 02:57.03 | shamusmcirish | right. you would need to use a vpn or somesort, perhaps hamachi.. than iax to connect the two servers |
| 02:57.31 | lvl`home | and it 'll spare you any sip/nat headaches |
| 02:57.51 | shamusmcirish | yes.. my head still hurts. |
| 02:57.55 | iPod-nano | It's all because of his stupid D-Link ATA. |
| 02:58.24 | iPod-nano | Locally, we know it can connect to an Asterisk server and communicate in either direction just fine. |
| 02:58.43 | iPod-nano | I call it, it rings. It calls, I ring. |
| 02:59.12 | iPod-nano | Remotely, however, it calls, the other end rings. Other end calls, nothing happens. |
| 02:59.26 | shamusmcirish | right. thats the headache with sip |
| 03:00.03 | shamusmcirish | SIP is a nasty protocol to put behind a firewall. You need nat traversal set up properly or a stun server |
| 03:01.00 | iPod-nano | He's got that stupid thing in his dorm and tries to connect to me. |
| 03:01.26 | iPod-nano | But since we know Asterisk can communicate with it locally, we want to make the two Asterisk boxes connect remotely. |
| 03:01.26 | shamusmcirish | if your connecting form an ATA then i assume that your only protocol choice is SIP. If this were asterisk to asterisk you could try iax which is a little easier to nat. |
| 03:01.55 | iPod-nano | So, he'd be running Asterisk at his end and his ATA would be communicating with that. |
| 03:02.03 | shamusmcirish | sorry , got to run |
| 03:02.09 | iPod-nano | And his Asterisk would be connecting to mine. |
| 03:02.19 | shamusmcirish | try iax2 trunks between the two. if it doesn't work, resort to vpn |
| 03:10.59 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:14.59 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 03:21.14 | *** join/#asterisk orkid (n=mike@unaffiliated/orkid) |
| 03:32.39 | alpha232 | Evenin yall |
| 03:34.35 | orkid | hello |
| 03:38.07 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 03:39.49 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:46.21 | alpha232 | 23:40 -!- TokyoJimu [n=TokyoJim@ikebukuro.nccom.com] has quit ["TokyoJimu has left the building"] |
| 03:46.33 | alpha232 | lol i saw that hostname and I almost thought it said IlikeBukake |
| 03:47.09 | *** join/#asterisk ugob (n=chatzill@modemcable180.221-59-74.mc.videotron.ca) |
| 03:47.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:47.31 | ugob | Hi everyone. I have a weird problem regarding NAT |
| 03:48.04 | ugob | I used externip = my.ip.ip.ip |
| 03:48.39 | ugob | and localnet for 10.0.0.0/8, 172.16.0.0/12 and 192.168.100.0/24 |
| 03:48.50 | ugob | (the server has 192.168.100.18) |
| 03:50.04 | ugob | curiously, the asterisk server is receiving the sip request, but it is not sending it out the the phone's public IP address (phones are also behind nat) |
| 03:50.18 | ugob | I'm not too familiar with irc... is anyone here? |
| 03:50.33 | Nugget | Nope. Nobody. |
| 03:51.07 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 03:51.40 | ugob | Thanks for your help... |
| 03:52.23 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:52.49 | alpha232 | hrrm |
| 03:52.59 | alpha232 | hardware hardware hardware why does it always cost money |
| 03:54.38 | ugob | I can't really answer taht question, alpha232... |
| 03:54.51 | ugob | Some people also have to gain some money to eat ;) |
| 03:57.36 | fujin_ | ~cheap |
| 03:57.37 | jbot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:59.10 | *** join/#asterisk FunkyGMT (i=gmt@office.kiwi-interactif.com) |
| 04:03.20 | *** join/#asterisk Kyreeth (n=ashley@dragonfruit.ddns.nominum.com) |
| 04:04.25 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 04:07.04 | *** join/#asterisk cnbrk (n=cnbrk@host-19-57.mielccc.clients.pavlovmedia.com) |
| 04:07.37 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:07.38 | cnbrk | hi |
| 04:08.08 | Kyreeth | Hihi.. So as with most people dropping in here, I've got a problem: I've got a set of PRIs that are configured as primary and secondary, grouped together with a trunkgroup statement. That's working nicely. When I disconnect one of the PRIs, incoming calls fall over to the other, this is good too. Unfortunately, while zap show status shows that that span is in red alarm, none of the b-channels associated with it show as being 'inalarm' fro |
| 04:08.08 | Kyreeth | m zap show channel 71, say, where 71 is a channel on the PRI that's in red alarm state. |
| 04:08.41 | Kyreeth | I'm on libpri 1.4.2, zaptel 1.4.6, asterisk 1.4.13. |
| 04:09.27 | variable_office | what does sip error 489 mean? |
| 04:09.29 | Kyreeth | If I had to guess, I'd say I'm not setting a flag in either zaptel.conf or zapata.conf to propagate that alarm status, but I've nothing to support that theory. :> |
| 04:10.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:20.48 | variable_office | i get the 489 errors sent by asterisk when the ata sends notify even: keep-alive |
| 04:26.03 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 04:26.56 | variable_office | are keep-alives not supported by asterisk? |
| 04:26.58 | Kyreeth | Aha, yes. The problem was indeed a flag that hadn't been set in zaptel.conf - it needs the yellow flag set on the span declaration in order to flag all the channels as in alarm when the PRI is down. |
| 04:27.28 | Kyreeth | Well, now that I've solved my problem, I'm going to head home. :> |
| 04:27.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:30.15 | JT | Kyreeth: how did you set the flag? |
| 04:31.51 | Kyreeth | JT: Edited /etc/zaptel.conf, tacked ',yellow' on the end of the span definitions. |
| 04:33.33 | alpha232 | yello? |
| 04:33.36 | alpha232 | melloyello |
| 04:34.57 | JT | Kyreeth: never seen that parameter before |
| 04:35.02 | JT | where is it documented? |
| 04:35.27 | Kyreeth | JT: Well, it mentions it in the sample zaptel.conf file that comes with the source. |
| 04:35.52 | alpha232 | lucky brat, you have a PRI :( |
| 04:35.57 | Kyreeth | Two. :> |
| 04:39.12 | JT | Kyreeth: ah ok |
| 04:39.14 | Kyreeth | (I'm currently running asterisk in between the telco and the legacy PBX) |
| 04:39.37 | JT | Kyreeth: heh i've got a setup doing that between BRI and PRI |
| 04:55.03 | alpha232 | JT: stop mentioning BRI *cry* |
| 05:00.32 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:00.39 | JT | alpha232: haha |
| 05:01.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:08.07 | alpha232 | hey JT you have fwd? |
| 05:08.19 | alpha232 | I need a test rodent |
| 05:11.23 | JT | i have, but haven't used it in ages, it's not currently running |
| 05:11.34 | alpha232 | oh |
| 05:11.42 | alpha232 | ok... thought you might have had it in your dial plan |
| 05:14.57 | JT | nah, it's in a server i don't use for asterisk at the moment |
| 05:15.33 | alpha232 | lol |
| 05:20.31 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 05:24.21 | *** join/#asterisk techie (n=techie@adsl-76-214-27-184.dsl.lsan03.sbcglobal.net) |
| 05:33.37 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9c7bd8bcb17d3b99) |
| 05:37.40 | JT | alpha232: it's in a server at home with a T1 card and channel bank |
| 05:37.46 | JT | but i don't use that right now |
| 05:38.00 | JT | i just have ip phones hooked up to my co-lo server in a datacentre |
| 05:50.35 | hmmhesays | 2nd date with the math phd, good |
| 05:51.10 | dlynes | What might be causing a SIP 489 "Bad event" to be generated when I have two asterisk 1.4 boxes talking SIP to each other? |
| 05:51.17 | alpha232 | grrrr |
| 05:51.48 | alpha232 | JT: skip the T1 and channel bank |
| 05:52.02 | hmmhesays | now its beer thirty |
| 05:52.04 | alpha232 | just get me a bri card for NI1 |
| 05:52.12 | alpha232 | afk |
| 05:55.30 | *** part/#asterisk techie (n=techie@adsl-76-214-27-184.dsl.lsan03.sbcglobal.net) |
| 05:59.24 | hmmhesays | its weird we're from two completely different worlds |
| 05:59.28 | hmmhesays | but seem to get along really well |
| 06:01.29 | *** join/#asterisk techie (n=techie@adsl-76-214-27-184.dsl.lsan03.sbcglobal.net) |
| 06:02.32 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 06:05.41 | *** join/#asterisk IPetrov (i=IPetrov@ppp85-140-235-189.pppoe.mtu-net.ru) |
| 06:16.45 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 06:18.28 | alpha232 | moooo |
| 06:18.45 | alpha232 | hmmhesays: hows the sex |
| 06:32.33 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:44.27 | hmmhesays | alpha, non yet, soon I'm guessing |
| 06:44.36 | hmmhesays | she really digs that I'm big into music |
| 06:59.07 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9c7bd8bcb17d3b99) |
| 07:00.27 | *** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 07:04.26 | blackhole | Hi, I am looking to develop a software which connects to my asterisk server and allows users to place calls as someone would do using a2b from web. Now i believe we need to develop a software which connect to my Asterisk SIP addr and hence place calls using my Server. |
| 07:04.38 | blackhole | Please let me know if i am correct... |
| 07:04.56 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-61369fcec219864b) |
| 07:12.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 07:17.34 | *** join/#asterisk peanut- (n=tokarev@cpe-70-113-100-193.austin.res.rr.com) |
| 07:18.58 | hmmhesays | heh |
| 07:19.39 | BeeBuu | hi,all |
| 07:20.25 | BeeBuu | can i use astdb store big datas? |
| 07:30.44 | blackhole | Hi, I am looking to develop a software which connects to my asterisk server and allows users to place calls as someone would do using a2b from web. Now i believe we need to develop a software which connect to my Asterisk SIP addr and hence place calls using my Server. |
| 07:30.48 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 07:31.50 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 07:33.57 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:38.58 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.52) |
| 07:39.08 | CBU[^_^]M`` | what the use of DID?? |
| 07:41.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:48.55 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 07:49.22 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666655.dsl.bell.ca) |
| 08:06.22 | alpha232 | CBU[^_^]M``: how do you mean? |
| 08:06.31 | alpha232 | Direct Inward Dial is used for many things |
| 08:06.41 | alpha232 | Department Direct IVR/AA |
| 08:06.52 | alpha232 | Direct to User |
| 08:07.44 | alpha232 | like say your corporate # is xxx-xxx-5000, that goes to the main menu |
| 08:08.05 | alpha232 | -5100 could goto the Sales Menu directly so all your sales poeple have the -5100 number |
| 08:08.45 | alpha232 | -5101-51xx could be for sales managers where calls go directly to their phones without the auto attendant or menus |
| 08:08.55 | alpha232 | same applies for other departments |
| 08:09.22 | alpha232 | it could also be used to track sources of business, say you run 10 ads, each has a different number so you know what prompted the call |
| 08:10.10 | blackhole | Hi, I am looking to develop a software which connects to my asterisk server and allows users to place calls as someone would do using a2b from web. Now i believe we need to develop a software which connect to my Asterisk SIP addr and hence place calls using my Server. |
| 08:10.55 | Strom_M | blackhole: so....you want to develop a softphone? |
| 08:11.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:11.05 | blackhole | Strom_M, yes.. |
| 08:11.23 | blackhole | Strom_M, A Mobile client application which acts as a softphone for users using that.. |
| 08:11.34 | blackhole | Strom_M, They just have to fill username pwd and sip addr.. |
| 08:11.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:12.01 | blackhole | Strom_M, So what you say.. I should use j2me SIP ? |
| 08:12.41 | Strom_M | hell if I know; I don't write a lot of code |
| 08:13.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:13.44 | blackhole | Strom_M, Hmm, Okay! |
| 08:16.23 | Strom_M | but I can advise you on protocols, telephony theory, etc :) |
| 08:30.45 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 08:35.16 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 08:37.00 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 08:39.16 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 08:39.24 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:42.47 | Aurs | anyone here using astmanproxy? |
| 08:44.19 | *** join/#asterisk shtoom (n=godson@59.93.119.235) |
| 08:47.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:48.06 | CBU[^_^]M`` | hello.. whats the use of pickupgroup? |
| 08:51.14 | JT | CBU[^_^]M``: hell, what's the use of RESEARCH and GOOGLE? |
| 08:51.18 | JT | s/hell/hello/ |
| 08:55.01 | *** join/#asterisk lbt (n=david@i-83-67-36-194.freedom2surf.net) |
| 08:56.05 | *** join/#asterisk awk (n=awk@kia.inet-corp.com) |
| 08:56.31 | awk | hrm, tzafrir I see you have posted on this issue before, what wasthe solution ? |
| 08:56.51 | blackhole | Strom_M, Do so.. please..? |
| 08:56.53 | awk | Nov 2 10:55:02 localhost kernel: qozap: dropped audio card 1 cardid 0 bytes 3 z1 124 z2 105 fifo 6 |
| 08:56.58 | awk | millions of these.. |
| 08:58.00 | *** join/#asterisk blq (n=Bl@dslb-088-064-135-076.pools.arcor-ip.net) |
| 08:58.38 | Strom_M | blackhole: well...you need to ask me specific questions :) |
| 09:00.00 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
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| 09:16.27 | *** join/#asterisk ToTo (n=ToTo@host189-87-dynamic.56-82-r.retail.telecomitalia.it) |
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| 09:17.42 | *** part/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 09:17.45 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 09:21.26 | blackhole | Strom_M, Say my client connect to sip addr i have now What headers i should sent to place a call..? |
| 09:22.01 | Strom_M | blackhole: that question can be easily answered by reading the SIP RFC |
| 09:23.39 | blackhole | Strom_M, can you point me to correct link. |
| 09:25.05 | Strom_M | search.yahoo.com |
| 09:37.19 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
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| 09:51.55 | *** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net) |
| 09:54.14 | badcfe | hello. is it possible to get the SIP Call-ID somehow from within the dialplan execution? |
| 09:59.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:06.15 | *** join/#asterisk _MarkH_ (n=chatzill@fentech.gotadsl.co.uk) |
| 10:12.37 | _MarkH_ | Hi, I'm working on a messaging system that has to automatically dial a number and then play a message to the callee (currently using the Cepstral TTS voices...). I'll be doing this using the HTTP/AMI interface. It's not clear to me how to do this as the Originate call seems to require a 'calling' extension too - which I don't have. I know I can pass a context for the call but I still can't... |
| 10:12.38 | _MarkH_ | ...work out exactly what to do. Simplistically what I want to do is 1. Call a number. 2. If answered play a message and enter a menu context or if not answered reschedule the call. 3. Hangup |
| 10:15.03 | Strom_M | _MarkH_: well, the call does consist of two legs |
| 10:15.38 | Strom_M | the leg that goes from the asterisk core to the called party (don't ever use the term "callee"), and the leg that goes from the core to the message playback function |
| 10:18.32 | _MarkH_ | Strom_M: So how do I identify the leg from the core to the message playback system? |
| 10:18.49 | Strom_M | you could use a Local channel |
| 10:20.09 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:20.43 | _MarkH_ | OK. I almost guessed there'd be somthing like that. So I'll refer to it as LOCAL/nnn (say) but how do I link that to the Swift/Cepstral system.? |
| 10:21.07 | Strom_M | you point it at an extension in your dialplan |
| 10:22.38 | _MarkH_ | Hmmmm, I must be being a bit blonde... How do I do that? |
| 10:23.05 | Strom_M | http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 10:23.25 | Strom_M | LOCAL/extension@context |
| 10:24.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:25.21 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 10:26.23 | _MarkH_ | I read that page before but didn't work out how to do it. Re-reading it I can see how it works now - thanks for the steer. |
| 10:30.24 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-42-248.red.bezeqint.net) |
| 10:30.38 | Dovid | does anyone know if the SPA-942 offers QoS ? |
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| 10:35.50 | BeeBuu | hello,all |
| 10:39.36 | *** join/#asterisk stmaher (n=stmaher@mail.voxpilot.com) |
| 10:46.22 | BeeBuu | is there anyone teach me how to return SQL results in dialplan? |
| 10:47.39 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 10:51.49 | *** join/#asterisk Alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 10:51.54 | Alpha232 | well that sucked hairy balls |
| 11:06.56 | IPetrov | BeeBuu: for example u can use perl or php with AGI |
| 11:07.54 | IPetrov | http://www.voip-info.org/wiki-Asterisk+AGI |
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| 11:19.16 | BeeBuu | i'm reading,thnaks,IPetrov |
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| 11:32.57 | _MarkH_ | I'm now successfully using AMI:Originate to call a number and connect a local extension (which is a speech system). The trouble is the LOCAL extension starts speaking immediately (without waiting for the called party to answer). Is there something in the local extension I can block on to detect whether the called party has picked up? |
| 11:35.21 | Strom_M | _MarkH_: rather than using originate, provide the outbound channel and then a context/extension/priority instead of your local channel |
| 11:35.30 | Strom_M | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 11:36.47 | Alpha232 | yummy |
| 11:37.10 | Alpha232 | hmm FPU's are important |
| 11:37.42 | Alpha232 | has anyone thought of using a gpu? |
| 11:37.51 | Alpha232 | i mean from a linux/asterisk pov |
| 11:38.17 | Strom_M | Alpha232: ? |
| 11:38.25 | Alpha232 | put a good gfx card in and only use it for text console or simple gui, and leave GPU mostly available for doing codec |
| 11:38.52 | Alpha232 | Strom_M: it's an idea thats been worked on recently, using the GPU on a video card for doing math |
| 11:39.06 | Alpha232 | be it encryption or codec work |
| 11:39.18 | Strom_M | perhaps |
| 11:39.29 | Alpha232 | think of it as an off cpu fpu |
| 11:39.34 | Strom_M | yes, I know |
| 11:39.40 | ai-a | sounds slower and more expensive. |
| 11:39.43 | Strom_M | your original statements lacked context |
| 11:40.30 | Alpha232 | ai-a: slower? do you know how many flops a good gpu can do? |
| 11:41.12 | Strom_M | Alpha232: may I suggest you bring this up in #asterisk-dev where it will probably be a more useful conversation? :) |
| 11:41.50 | _MarkH_ | Strom_M: I thought I _had_ to use a channel. How do I test the called party status (i.e. block till they have answered) in the context |
| 11:41.52 | Alpha232 | Strom_M: you could... but it likely wouldn't do any good anyways |
| 11:42.05 | Alpha232 | Strom_M: just thinkin ;) |
| 11:42.38 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 11:42.59 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 11:43.18 | Strom_M | _MarkH_: read the page I linked you to |
| 11:43.39 | Strom_M | the context/exten/priority you specify don't begin execution until after the channel supervises |
| 11:45.36 | Rawplayer | anyone in here compiled app_ldap on debian with asterisk from the debian reps? |
| 11:51.26 | *** join/#asterisk oej (n=olle@74.9.53.154) |
| 11:54.29 | *** join/#asterisk gmustafa (n=gmustafa@202.5.145.13) |
| 11:54.41 | gmustafa | i m trying to configure ldap realtime |
| 11:55.20 | gmustafa | can someone tell me what value AsteriskCategory and AsteriskCategoryMetric accepts in ldap |
| 11:55.50 | _MarkH_ | Strom_M: Thanks for that - works great now. I really do read the wiki pages before asking but was struggling to get my head around the AMI:Originate stuff. I was not picturing the notion of connecting the speech system to the external number when the solution lies in connecting the external number to the speech system... |
| 12:06.50 | Alpha232 | _MarkH_: cart, horse, horse, cart |
| 12:07.33 | _MarkH_ | :) |
| 12:13.14 | ai-a | Im making a empty asterisk box. which files are required. ive created extension.conf with echo test, and a sip.conf with ext 101. phone says ' register timed out' and nothing on the cli. |
| 12:14.07 | ai-a | dont think sip module is loaded. |
| 12:14.21 | ai-a | says no modules.conf |
| 12:15.28 | ai-a | ok, copied the modules.conf and stripped it, now seems something is working. |
| 12:22.18 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 12:29.08 | hmmhesays | some people say that theres a woman to blame, but I know |
| 12:29.15 | hmmhesays | it was all asterisk fault |
| 12:29.50 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:30.35 | hmmhesays | oh [TK]D-Fender: you just missed my jimmy buffet serenade |
| 12:32.01 | *** join/#asterisk Woifi1988 (n=anon@M1303P013.adsl.highway.telekom.at) |
| 12:32.59 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 12:34.48 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 12:35.04 | Alpha232 | hmmhesays: uhhhuh |
| 12:35.15 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 12:35.16 | *** join/#asterisk sdac (n=chatzill@217-211-57-252-no38.tbcn.telia.com) |
| 12:36.25 | sdac | Hi, is this case possible: asterisk receives a call, it then itself calls 5 numbers, the one who answers first gets the call. And if it's possible, any guidelines available? |
| 12:36.55 | Alpha232 | ya know people selling VoIP hardware are the scum of the earth |
| 12:37.52 | Strom_M | Alpha232: depends on the retailer |
| 12:38.20 | [TK]D-Fender | sdac: Yes, dial out through a VoIP provider with call progress or get a PRI. And yes, this could be 1 line of dialplan. |
| 12:39.30 | Alpha232 | Strom_M: ebay |
| 12:39.42 | Alpha232 | http://cgi.ebay.com/Audiocodes-MP-118-FXO-Analog-Gateway-VoIP_W0QQitemZ180173190780QQihZ008QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem take this item |
| 12:39.49 | Alpha232 | Audiocodes MP-118 FXO Analog Gateway VoIP |
| 12:39.51 | [TK]D-Fender | Alpha232: Feel free to continue kisking yourself |
| 12:40.27 | sdac | [TK]D-Fender: PRI? you got any helpful link or should I just google it? |
| 12:40.30 | [TK]D-Fender | kicking* |
| 12:40.41 | [TK]D-Fender | sdac: PRI is a T1 signalling method. |
| 12:40.56 | sdac | so, what's the requirements? |
| 12:40.59 | [TK]D-Fender | sdac: So basically a Digital link to your telco |
| 12:41.30 | [TK]D-Fender | sdac: for PRI would be PRI service & an of a large number of supported cards from Digium, Sangoma, Rhino, etc |
| 12:42.03 | [TK]D-Fender | Alpha232: They got that unit backwards. they Said FXO and then " Connect up to 8 analog telephone sets or fax machines." |
| 12:42.48 | sdac | [TK]D-Fender: thank you for your information, I'll google and further investigate this |
| 12:42.56 | Alpha232 | [TK]D-Fender: yup... thats my point |
| 12:42.57 | [TK]D-Fender | sdac: here |
| 12:42.59 | [TK]D-Fender | ~101 |
| 12:43.00 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 12:43.04 | Alpha232 | [TK]D-Fender: the back of the unit states FXS ports |
| 12:43.05 | [TK]D-Fender | ~book |
| 12:43.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 12:43.28 | sdac | oh, thanks! |
| 12:43.29 | sdac | great |
| 12:43.35 | Alpha232 | [TK]D-Fender: this one makes me *growl* http://cgi.ebay.com/GSM-Cellular-Terminal-Gateway-FXO-FXS-Asterisk-IP-PBX_W0QQitemZ120178447208QQihZ002QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 12:43.51 | Alpha232 | now if only it had SIP on board |
| 12:44.40 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 12:45.09 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
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| 12:51.58 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 12:52.36 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:52.36 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:01.15 | Uatec | sorry that it's off topic, but it's not like the channel is busy... |
| 13:02.18 | Uatec | If i transfer a .biz domain to a different registrar, will the DNS servers of the domain be retained when the transfer occurs? Basically, will there be any interruption to the services of the domain name during the change? |
| 13:04.44 | *** part/#asterisk mocker (n=user@198.247.173.227) |
| 13:05.09 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 13:06.19 | [TK]D-Fender | Uatec: have the DNS ready on the new side and both pointing to where it should be before starting the actual transfer |
| 13:06.43 | wwalker | is there a way to get asterisk to start a full log without restarting it? |
| 13:07.01 | Alpha232 | hrrm why don't the damn ubuntu packages for asterisk have ztdummy |
| 13:07.16 | Strom_M | Alpha232: because that's part of zaptel |
| 13:07.19 | [TK]D-Fender | Ubuntu for a SERVER..... *shudder* |
| 13:07.19 | Corydon76-dig | wwalker: alter logger.conf and 'logger reload' |
| 13:07.41 | Corydon76-dig | [TK]D-Fender: ubuntu works just fine as a server |
| 13:07.48 | Alpha232 | Strom_M: but according to somthing somewhere if i'm using 2.6 i don't need the dummy module as the kernel does 1khz timing now |
| 13:08.00 | [TK]D-Fender | Corydon76-dig: Yeah yeah.... still... meh |
| 13:08.02 | Alpha232 | [TK]D-Fender: don't knock what you havn't tried |
| 13:08.12 | Alpha232 | [TK]D-Fender: if you want to complain about an OS... try IRIX |
| 13:08.20 | Corydon76-dig | If you want an awful distro for a server, try Gentoo |
| 13:08.48 | [TK]D-Fender | Alpha232: You seem to have cornered the market on complaining.... wouldn't want to horn in on your territory ;) |
| 13:08.53 | *** join/#asterisk Op3r (n=Op3r@125.212.50.122) |
| 13:09.09 | Alpha232 | uhh huh |
| 13:09.19 | [TK]D-Fender | Corydon76-dig: When you absolutely MUST get that extra 0.5% out of your system, no matter the cost! |
| 13:09.20 | Strom_M | Alpha232: you misread |
| 13:09.33 | Uatec | [TK]D-Fender, the domain currently is pointing to our DNS server, but we can't set the DNS of the domain itself till it has been transfered... |
| 13:09.46 | Strom_M | Alpha232: with 2.6 kernels, ztdummy gets the timing from the kernel rather than the USB UHCI cotroller |
| 13:09.46 | Alpha232 | Strom_M: yeah i did |
| 13:09.49 | Corydon76-dig | If we're going for truly awful OS's, go with HP/UX |
| 13:10.09 | Alpha232 | Strom_M: yep..i just read tfa instead of just the googleclip |
| 13:10.52 | Strom_M | cocks |
| 13:10.54 | Corydon76-dig | There's a reason why the pronunciation of "HP/UX" sounds like you're vomiting |
| 13:10.56 | Alpha232 | Strom_M: exactly |
| 13:11.56 | Alpha232 | Corydon76-dig: ? it's always been called Hockey Pucks |
| 13:11.56 | hmmhesays | LOL |
| 13:11.56 | Alpha232 | Corydon76-dig: it sounds like you're vomiting or you're yiddish |
| 13:12.51 | wwalker | Corydon76-dig: thx I had tried just reload |
| 13:13.03 | Katty | mew. |
| 13:13.04 | [TK]D-Fender | Alpha232: Both should be slapped |
| 13:14.51 | mocker | Friday morning Khan, http://www.khaaan.com/ |
| 13:16.32 | Alpha232 | hmm is it me or is the default hold music for asterisk not attributed anywhere |
| 13:16.34 | [TK]D-Fender | Katty: Mew. |
| 13:19.26 | Qwell | Alpha232: what do you mean attributed? |
| 13:20.20 | mocker | Alpha232: Check out CREDITS in the source distribution. |
| 13:20.29 | Qwell | and http://svn.digium.com/view/asterisk/branches/1.4/doc/musiconhold-fpm.txt?revision=58955&view=markup |
| 13:21.04 | Katty | friday morning cute overload instead! |
| 13:21.14 | mocker | jbot: moh is originally from www.freeplaymusic.com |
| 13:21.15 | jbot | ...but moh is already something else... |
| 13:21.18 | mocker | ~moh |
| 13:21.19 | jbot | [moh] Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 13:21.40 | Katty | are there any asterisk based rack mountable appliances yet? |
| 13:21.48 | mocker | jbot: moh is also originally from www.freeplaymusic.com |
| 13:21.49 | jbot | mocker: okay |
| 13:21.50 | Katty | besides the digium one, that is still hella spensive |
| 13:21.56 | mocker | ~moh |
| 13:21.59 | jbot | i guess moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf, or originally from http://www.freeplaymusic.com |
| 13:22.28 | mocker | But I've seen that question several times recently. |
| 13:24.15 | [TK]D-Fender | Katty: More cuteness awaits you in Canuckland! http://www.youtube.com/watch?v=4i6CSoct4lE |
| 13:25.11 | Strom_M | speaking of canuckland: http://flickr.com/photos/stromcarlson/1826760072/ |
| 13:26.05 | [TK]D-Fender | Strom_M: degelasse! |
| 13:26.20 | blitzrage | w00t Canucks! |
| 13:26.32 | blitzrage | Oh I love poutine! |
| 13:26.48 | *** join/#asterisk ZeNN (i=ZeNN@ip5657bd18.direct-adsl.nl) |
| 13:27.24 | Strom_M | that was my first real poutine |
| 13:27.30 | blitzrage | tasty stuff |
| 13:27.35 | Strom_M | so I figured it would be a good idea to photograph it |
| 13:27.38 | blitzrage | totally not good for you :) |
| 13:27.39 | Strom_M | blitzrage: indeed :) |
| 13:27.53 | Strom_M | i need to figure out how to make it here in Los Angeles |
| 13:28.20 | blitzrage | just make it... its' just fries, gravy, and cheese curds |
| 13:28.46 | [TK]D-Fender | blitzrage: very specific cheese & gravy though... |
| 13:28.54 | Strom_M | yes, but from what I understand, there is a specific variety of gravy...and lord knows where i'm going to get cheese curds here |
| 13:29.03 | Strom_M | i don't recall seeing cheese curds at my supermarket |
| 13:29.34 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 13:30.03 | [TK]D-Fender | Strom_M: You might have to import... |
| 13:30.18 | Strom_M | [TK]D-Fender: i'm sure i just have to find a different supermarket |
| 13:30.34 | [TK]D-Fender | Strom_M: That sounds like a good thing regardless. |
| 13:31.52 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:33.32 | hmmhesays | sweet jeebus, firefox just updated itself like 10 times |
| 13:34.06 | cpm | heh |
| 13:38.41 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:39.13 | hmmhesays | Awww, good morning Katty |
| 13:39.22 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:39.32 | Katty | morning. |
| 13:39.34 | Katty | fskrotzki: herro (= |
| 13:39.54 | sigmounte | morning |
| 13:40.14 | hmmhesays | strumming my six string, on my front porch swing |
| 13:40.20 | fskrotzki | mornin, you holding up ok? |
| 13:42.07 | hmmhesays | wasting away again in margaritaville |
| 13:42.31 | [TK]D-Fender | hmmhesays: blarg |
| 13:42.58 | [TK]D-Fender | hmmhesays: Jimmy Buffet is giving me indigestion... |
| 13:43.06 | cpm | that happens |
| 13:43.29 | [TK]D-Fender | hmmhesays: If you want to go accoustic, you need some new influences! |
| 13:44.23 | hmmhesays | lol |
| 13:44.27 | hmmhesays | I like jimmy buffet |
| 13:44.36 | hmmhesays | more importantly chicks like jimmy buffet |
| 13:45.40 | [TK]D-Fender | hmmhesays: http://www.youtube.com/results?search_query=antoine+dufour&search=Search |
| 13:45.54 | [TK]D-Fender | hmmhesays: http://www.youtube.com/results?search_query=andy+mckee&search=Search |
| 13:47.09 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 13:48.01 | hmmhesays | insane |
| 13:48.03 | sigmounte | Hi ! Anyone use Busy Lamp Field ? |
| 13:48.27 | [TK]D-Fender | sigmounte: Yes. |
| 13:48.38 | mocker | sigmounte: Deja vui |
| 13:48.41 | mocker | +spelling |
| 13:49.13 | sigmounte | hi mocker and [TK]D-Fender , my blf work finally , but only on incoming call , not on outgoing , any idea why ? |
| 13:49.35 | hmmhesays | antoine dufour I'm guessing is a bassist at heart |
| 13:49.47 | [TK]D-Fender | hmmhesays: he rocks... |
| 13:50.00 | [TK]D-Fender | hmmhesays: checkout "Drifting" from Andy Mckee... |
| 13:50.16 | hmmhesays | that makes me want to give up guitar |
| 13:50.24 | [TK]D-Fender | lol |
| 13:51.46 | hmmhesays | i think I like that antoine dufour link I clicked |
| 13:51.47 | hmmhesays | better |
| 13:52.16 | [TK]D-Fender | hmmhesays: Yeah I find him a little more interesting as well.. |
| 13:52.17 | hmmhesays | had move of a groove |
| 13:52.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:53.05 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:53.05 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:53.12 | hmmhesays | *more of even |
| 13:53.13 | mocker | sigmounte: Check that when you play an outbound call it changes your status in asterisk. |
| 13:54.11 | sigmounte | mocker, i've made an ext with just a muscionhold for test , their is not difference with and outgoing call ? |
| 13:54.55 | [TK]D-Fender | sigmounte: What are you calling using? |
| 13:55.05 | sigmounte | mocker, with a musiconhold or a real outgoing (outside pbx) call , it look like asterisk don't change the hint status |
| 13:55.24 | mocker | sigmounte: sip show subscriptions will show if you are no longer in an Idle state on the * server. |
| 13:55.24 | Katty | ohhhh |
| 13:55.27 | Katty | it's friday :> |
| 13:55.29 | mocker | Ahh, gotcha. |
| 13:55.31 | Katty | i keep forgetting :> |
| 13:55.31 | *** join/#asterisk enalert (i=trelane@2001:4830:150c:0:20d:61ff:fe31:a58) |
| 13:55.35 | hmmhesays | god I suck at guitar |
| 13:55.56 | Katty | hmmhesays: berr chickaberrberr |
| 13:55.58 | mocker | sigmounte: Are your hints configured in the same context that your phone registers to? |
| 13:56.06 | hmmhesays | Katty: wooohooo! |
| 13:56.22 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 13:57.06 | sigmounte | mocker, the howto i've found tell me to create a context "blf" at the end of my extentions.conf , and use it to subscribe in my sip.conf |
| 13:57.26 | mocker | sigmounte: I've never done it that way. |
| 13:57.52 | mocker | are all your hints in the blf context? |
| 13:57.54 | Katty | hmmhesays: i'm oddly listening to josh turner this morning >.< |
| 13:58.03 | Katty | hmmhesays: my co-workers are going to say i've insaned. |
| 13:58.18 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 13:58.39 | hmmhesays | fun, i've been grooving to avenged sevenfolds new album |
| 13:58.58 | sigmounte | mocker, yes , all the hints in the blf context , i'll try the old way |
| 13:59.16 | Katty | hmmhesays: oh, did i tell you my boy sold all his PA gear? |
| 13:59.31 | mocker | sigmounte: Just try includeing blf where your sip phones register. |
| 13:59.46 | mocker | include => blf |
| 13:59.50 | hmmhesays | You did not we haven't talked much, why did he do such a thing? |
| 13:59.54 | Katty | hmmhesays: cept for like an amp and sub |
| 14:00.32 | Katty | and the stage snake. |
| 14:00.42 | hmmhesays | for what reason? |
| 14:00.51 | Katty | no more band. |
| 14:01.02 | [TK]D-Fender | Katty: Yeah.. that snake cound have fetched some real dough! |
| 14:01.06 | Katty | he got sick of them trying to do their Write Your Own Music and get Rich and Famous |
| 14:01.16 | *** join/#asterisk ManxPower (n=manxpowe@189.sub-70-220-142.myvzw.com) |
| 14:01.18 | Katty | [TK]D-Fender: he sure did pay a pretty penny for it. |
| 14:01.23 | Katty | all of it sold for about 10 grand. |
| 14:01.39 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:01.39 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:01.56 | Katty | blitzrage: ! |
| 14:01.58 | Katty | blitzrage: FOUND YOU! |
| 14:02.05 | [TK]D-Fender | Katty: That blasted truck will get paid off! (if he isn't a twit!) |
| 14:02.10 | blitzrage | Katty: *gasp*!!! |
| 14:02.16 | Katty | [TK]D-Fender: oh yes. |
| 14:02.26 | Katty | [TK]D-Fender: yes it is getting paid off. much to my utter shock. |
| 14:02.37 | Katty | blitzrage: prepare yourself!! |
| 14:02.46 | blitzrage | oooo... this sounds like it's gonna get good |
| 14:03.02 | sigmounte | mocker, last version: all my hints are in the context the phones register and i've add subscribecontext= & context= to each account of my sip.conf |
| 14:03.10 | blitzrage | oh snap |
| 14:03.34 | Katty | [TK]D-Fender: ever since i sat him down and said NOW LOOK HERE |
| 14:03.40 | Katty | [TK]D-Fender: he's straightened up... a good deal. |
| 14:03.47 | hmmhesays | no more band, thats like loosing a girl/guy |
| 14:03.56 | Katty | hmmhesays: eh, it was slowly dying anyway.... |
| 14:04.05 | Katty | hmmhesays: they went through 3 drummers and 5 lead guitar players. |
| 14:04.12 | hmmhesays | that sucks |
| 14:04.16 | Katty | yep |
| 14:04.21 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:05.07 | Katty | hmmhesays: ryan just wanted to play at bars, make some money, drink, and have fun |
| 14:05.13 | Katty | hmmhesays: as did the base guitar player. |
| 14:05.27 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:05.28 | Katty | hmmhesays: the rest of them wanted to write their own music and go nowhere. |
| 14:06.09 | Katty | it got so bad, that for awhile ryan said he'd play lead and sing both :< |
| 14:06.26 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 14:07.00 | *** join/#asterisk Runlvl (n=gustavo@190.2.40.57) |
| 14:07.01 | Runlvl | Comunidad de Asterisk en español, http://www.asterisk-la.org |
| 14:08.39 | Katty | Runlvl: hola (= |
| 14:09.08 | Runlvl | Katty, buen dia:) |
| 14:09.26 | hmmhesays | yeah if you want to make any money you almost have to play covers |
| 14:09.34 | hmmhesays | to build up an audience |
| 14:09.38 | hmmhesays | then you start throwing originals in |
| 14:09.54 | Katty | Runlvl: buenas tardes! |
| 14:10.00 | Katty | que tal? |
| 14:10.07 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.9) |
| 14:10.09 | Runlvl | Muy bien, y vos? |
| 14:10.15 | Katty | bien (= |
| 14:10.29 | blitzrage | ca va? |
| 14:11.10 | Katty | Runlvl: todavia estoy estudiando :> |
| 14:11.28 | Runlvl | Katty, asterisk? |
| 14:11.32 | Katty | Runlvl: si |
| 14:11.40 | Strom_M | claro, estan muchos zapatos en mis postres hoy |
| 14:11.53 | Katty | Runlvl: soy ingles (= |
| 14:12.08 | Runlvl | Katty, hablas muy bien español |
| 14:12.21 | Katty | Runlvl: no :< |
| 14:12.30 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net) |
| 14:13.15 | Strom_M | Katty: ?Asterisk esta como afeitando con un gato en su grabadora? |
| 14:13.23 | Runlvl | jajajaja :P |
| 14:13.44 | Katty | Strom_M: dude. |
| 14:13.51 | Katty | Strom_M: come on. i don't know THAT much spanish :P |
| 14:14.02 | Runlvl | eso no tiene sentido |
| 14:14.06 | Strom_M | I asked you if Asterisk was like shaving with a cat in your tape recorder. |
| 14:14.08 | Runlvl | don't make sense |
| 14:14.17 | Katty | hehehehe |
| 14:14.26 | Katty | Strom_M: puede ayudarme? :P |
| 14:14.26 | JayTee52 | Hi, I'm looking to talk with anyone who has deployed Asterisk with over 200 users. Has anyone in here done that successfully? |
| 14:14.38 | Runlvl | JayTee52, me :) |
| 14:14.55 | Strom_M | Katty: see, now that i dont understand anymore |
| 14:15.10 | Katty | Strom_M: "can you help me" |
| 14:15.14 | ManxPower | JayTee52: the number of users is far, far less important than the codecs used and the number of simul calls needed. |
| 14:15.30 | Strom_M | Katty: ah |
| 14:15.36 | Runlvl | JayTee52, what kind of problem you have? |
| 14:16.14 | ManxPower | for example, 200 users on analog phones is totally different than 200 users on SIP phones. |
| 14:16.33 | JayTee52 | I don't have a problem at present, my boss is leaning toward using Asterisk for all our voice needs but we need to know of other companies doing it on the same scale successfully. |
| 14:16.47 | Katty | Strom_M: i don't speak much spanish anyway. |
| 14:16.48 | ZeNN | i'm trying to patch and compile Asterisk-1.2.19 but keep getting an error message stating that: asterisk.c:94: error: expected declaration specifiers or '...' before 'capget'. complete log at: http://pastebin.com/d49446520 |
| 14:17.01 | Katty | Strom_M: just enough to understand what my mom goes babbling on about when she gets mad. |
| 14:17.13 | JayTee52 | ManxPower, it would be about 120 SIP phones and probably 30 or so analog phones with SIP adapters. |
| 14:17.18 | ZeNN | patching asterisk with vISDN diffs |
| 14:17.27 | ManxPower | JayTee52: A better place to ask would be the asterisk-users mailing list. |
| 14:18.01 | JayTee52 | ManxPower, um do I need to be added to that list to send mail there? |
| 14:18.06 | ManxPower | JayTee52: in that case, then you are mostly concerned with the location of the phones (all local to the server?) and how many calls you need to handle all at once. |
| 14:18.10 | ManxPower | JayTee52: yes. |
| 14:18.49 | JayTee52 | ManxPower, where do I sign up for the mailing list? |
| 14:18.50 | Strom_M | Katty: i took four years of it in high school, and have spoken it conversationally perhaps three times in the last six years |
| 14:19.07 | Katty | Strom_M: ahhh. |
| 14:19.16 | Katty | Strom_M: it's a little different when you grow up around it (= |
| 14:19.17 | ManxPower | JayTee52: lists.digium.com |
| 14:19.25 | Strom_M | Katty: i'm sure :) |
| 14:19.36 | Runlvl | JayTee52, you can have some help here too |
| 14:19.55 | Runlvl | and you can use the asterisk forumns |
| 14:22.02 | JayTee52 | ManxPower, thanks so much for your help! I've got a small Asterisk system running 8 SIP phones and one fax machine using Digium's 4 port FXO cards using analog lines from a Nortel Meridian PBX. It's a kludge but it works and we are looking at moving Asterisk to take all inbound outbound from our 2 PRI/ISDN circuits and route them through Asterisk to Nortel and back out using Digium's 4 port T1 card. |
| 14:22.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:22.45 | ManxPower | JayTee52: using asterisk as a front end to another PBX is complicated and time consuming (usually because of the non-Asterisk PBX) |
| 14:22.56 | JayTee52 | we currently have Microsoft's Unified Messaging working with Asterisk using sipX as a connector for SIP TCP to UPD conversion. |
| 14:23.31 | ManxPower | Don't use FXO cards, they don't work well and a are a pain in the ass. |
| 14:23.45 | JayTee52 | ManxPower, we'd only have the Nortel for a short span while we migrate users in phases instead of 100% cutover. |
| 14:24.30 | ManxPower | JayTee52: I've done that. We spent more time trying to make the Nortel work the way we wanted than it would have taken to migrate everyone over to Asterisk. |
| 14:25.56 | JayTee52 | ManxPower, I hear ya! Wish I had the option to do it all at once but it doesn't look like that would be the case. |
| 14:26.55 | ManxPower | Once we moved from analog to T-1 for the connection to the Nortel, things at least worked reliably. |
| 14:27.35 | ManxPower | JayTee52: I suggest buying two, 2-port cards, actually. You won't need 4 ports when you have moved everyone over to Asterisk and having a 2nd 2-port card means you have a spare card on hand. |
| 14:30.58 | Katty | [TK]D-Fender: what's the bootrom for? |
| 14:31.17 | Katty | [TK]D-Fender: i don't understand the difference between 'bootrom' and 'firmware'? |
| 14:31.56 | [TK]D-Fender | Katty: Bootrom is what determins what kind of provisioning methods you can have. Thats the part the picks UP new SIP images, etc. Its in 2 parts for that reason |
| 14:32.50 | [TK]D-Fender | Katty: So a newer BR could allow for provisioning via new methods, be able to negociate new network aprameters (like CDP / PoE / VLAN, etc) |
| 14:32.57 | Katty | ooooh |
| 14:32.58 | Katty | i see. |
| 14:32.59 | JayTee52 | ManxPower, Excellent suggestion! I'll also hit the mailing lists to get some case histories for scaling large size. |
| 14:32.59 | [TK]D-Fender | Katty: All the LOW LEVEL stuff that isn't just SIP. |
| 14:34.09 | JayTee52 | ManxPower, Runlvl, thank you both for your help. I've got to run now. Have a great weekend! |
| 14:34.24 | *** part/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net) |
| 14:34.35 | Runlvl | Cya, take care |
| 14:34.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:34.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:34.45 | Katty | oh noes. |
| 14:34.46 | Katty | RUN |
| 14:35.12 | [TK]D-Fender | *crash* |
| 14:35.28 | Katty | so i just take this firmware zip file.. |
| 14:35.34 | Katty | and unzip it into the ftp directory? |
| 14:36.31 | [TK]D-Fender | Katty: Depends which. Thats what the release nots are for. |
| 14:36.48 | [TK]D-Fender | Katty: BR 4.0 is a big departure from the old style |
| 14:37.40 | *** join/#asterisk shamusmcirish (n=no@gw-wp.debtresolve.com) |
| 14:39.21 | FreezeS | I have a problem with phpagi (launched with DeadAgi). It doesn't hangup the current channel: http://pastebin.ca/758733 |
| 14:40.27 | shamusmcirish | anyone know how to route an incoming call based on DNIS? |
| 14:41.14 | FreezeS | how can I hangup from AGI before answering ? |
| 14:44.27 | [TK]D-Fender | shamusmcirish: "show function CALLERID" , "show application gotoif" |
| 14:46.42 | ManxPower | FreezeS: issue the AGI HANGUP command |
| 14:47.09 | ManxPower | FreezeS: also, you can't hangup an already hungup channel. |
| 14:47.30 | ManxPower | just exit the agi and if the dialplan has run out of priorites, it should exit just fine. |
| 14:47.44 | *** join/#asterisk huey23 (n=yea@64.192.209.132) |
| 14:49.18 | huey23 | could someone take a look @ http://pastebin.com/m15bdfcba and see if something in here would be causing my time to display new years eve on every reboot? |
| 14:49.58 | [TK]D-Fender | huey23: tcpIpApp.sntp.resyncPeriod="" <-ick |
| 14:50.10 | [TK]D-Fender | huey23: tcpIpApp.sntp.gmtOffset.overrideDHCP="0" <- ICK! |
| 14:50.28 | [TK]D-Fender | huey23: tcpIpApp.sntp.gmtOffset="-5" <- LOL! |
| 14:50.33 | huey23 | :P |
| 14:50.55 | mocker | gmtOffset has to be in seconds I think |
| 14:50.59 | [TK]D-Fender | huey23: Last one shows you are indeed "a little slow" :p |
| 14:51.13 | huey23 | lol |
| 14:51.27 | huey23 | geographic region can do that sometimes |
| 14:51.28 | [TK]D-Fender | huey23: And I already corrected you on this now less :) |
| 14:51.51 | FreezeS | does anyone know if I can hangup in agi before answering a call ? |
| 14:52.07 | [TK]D-Fender | huey23: Yes, extreme altitude can cause the kind of aneurisms that would be responsible for that config :) |
| 14:52.26 | mocker | [TK]D-Fender: Oh, that reminds me. |
| 14:52.29 | huey23 | good thing i am in the heartland :) |
| 14:52.36 | mocker | Do you know a way to increase the default volume on Polycoms? |
| 14:52.43 | [TK]D-Fender | FreezeS: ManxPower>FreezeS: issue the AGI HANGUP command <--------- and? |
| 14:53.12 | FreezeS | [TK]D-Fender: http://pastebin.ca/758733 |
| 14:53.15 | [TK]D-Fender | mocker: default? I would suggest setting the "persist" values in sip.cfg |
| 14:53.19 | huey23 | fender: this is new SIP software...just trying to get a good config before i go completely active...thanks man |
| 14:53.46 | mocker | [TK]D-Fender: Actually just found that. |
| 14:53.49 | mocker | Thanks. |
| 14:53.50 | FreezeS | the channel is not hung up |
| 14:53.52 | FreezeS | it's ringing |
| 14:54.31 | [TK]D-Fender | FreezeS: neato |
| 14:56.09 | FreezeS | so, what's the problem with it ? |
| 14:56.18 | [TK]D-Fender | FreezeS: not a clue. |
| 14:56.53 | FreezeS | if I don't issue the hangup command in agi but in the context, the channel is answered very shortly and then hungup |
| 14:57.30 | FreezeS | and the mobile operator bills me one minute (of international call) because of that |
| 15:00.22 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:00.44 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 15:01.45 | waverly360 | Aloha :P |
| 15:01.57 | Spida | which (kernel-) driver should I use for a HFC Bri ISDN card? hisax, zaptel, misdn, ...? |
| 15:03.07 | ManxPower | FreezeS: sounds to me like you have an exten => _.1,Whatever which is a very, very bad thing. |
| 15:03.14 | ManxPower | you would want at least _X. |
| 15:03.21 | waverly360 | Hmm..I don't come in here nearly enough anymore. How sad. |
| 15:05.45 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 15:06.33 | *** join/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net) |
| 15:06.57 | waverly360 | Do any of you guys have recommendations for a good SIP cordless phone? Not WiFi, but one with a VoIP base station that uses normal cordless phone technology. |
| 15:08.06 | [TK]D-Fender | waverly360: I've heard Seimens makes a fairly decent DECT one |
| 15:08.14 | waverly360 | The most promising one I've seen is the Siemens C460IP Dect SIP Phone |
| 15:08.21 | waverly360 | [TK]D-Fender: oh cool :) |
| 15:08.58 | [TK]D-Fender | waverly360: Just what I heard... never tried. |
| 15:09.03 | huey23 | waverly360: i recently set up an Aastra and it was fairly simple and I was able to get an engineer on the horn within 5 minutes when I had a question |
| 15:09.05 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 15:09.34 | waverly360 | huey23: you mean the 480i? |
| 15:09.44 | huey23 | ie i believe |
| 15:09.58 | huey23 | 480 ie (i think) |
| 15:10.04 | waverly360 | huey23: I've been playing around with the 57i, and got kinda burned on it...I'm kinda shying away from aastra for the time being. |
| 15:10.59 | huey23 | waverly360: ok...just a suggestion, it was pretty easy once i got the correct firmware loaded |
| 15:11.40 | waverly360 | huey23: Cool. I guess that one would use an entirely different firmware than their 5i series. |
| 15:12.03 | FreezeS | ManxPower: I actually have something else: exten => _000999.,1,DeadAgi(quickdial2.php) |
| 15:12.03 | FreezeS | exten => _000999.,2,HangUp |
| 15:12.35 | FreezeS | seems that it has the following problem: when I run the DeadAgi command, it automatically answers the channel |
| 15:12.42 | ManxPower | FreezeS: well there is your problem! |
| 15:12.54 | huey23 | waverly360: i am nowhere near an expert (ask fender) but this was Aastra was pretty easy |
| 15:12.55 | ManxPower | DeadAGI is supposed to be run from exten => h,1,Whatever |
| 15:13.26 | Spida | waverly360: siemens has some gigaset phones that use DECT on the wireless side, and sip on the cable |
| 15:13.34 | FreezeS | ManxPower: I want to hangup a call and then process the cli and destination |
| 15:13.41 | FreezeS | how should I rewrite the dialplan ? |
| 15:13.52 | waverly360 | huey23: Hmm...any ideas whether it's configurable via tftp? I've gotten the 57i configured without much problem. |
| 15:14.32 | waverly360 | Spida: Yeah, that's what I'm looking for. I'm leaning heavily towards that C460 |
| 15:14.59 | [TK]D-Fender | waverly360: %i = meh |
| 15:15.02 | [TK]D-Fender | 5i |
| 15:15.09 | huey23 | waverly360: tftp is how i loaded the firmware |
| 15:15.19 | waverly360 | [TK]D-Fender: You don't like them either? |
| 15:15.26 | [TK]D-Fender | waverly360: unless you WANT the 57i base or 480i for that matter, I'd keep looking at the seimens |
| 15:15.43 | [TK]D-Fender | waverly360: I have numerous gripes about the 5i series. |
| 15:15.44 | waverly360 | [TK]D-Fender: my first reaction when someone brought the spec sheet to me was no. |
| 15:15.52 | waverly360 | [TK]D-Fender: Yay...that makes at least two of us. |
| 15:15.56 | huey23 | waverly360: i received the phone in web recovery mode...i had to load the firmware through tfpt |
| 15:16.13 | [TK]D-Fender | waverly360: Actaully the 1st impression of the 5i was "Looks GREAT!". Then you start USING it and.... bleh. |
| 15:16.37 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 15:16.51 | waverly360 | [TK]D-Fender: I couldn't agree more. We originally bough the phone because it supported a larger number of users on the sidecars. I mean...180 total users was our operator console answer. |
| 15:17.17 | waverly360 | [TK]D-Fender: Worked fine with the first customer..they had about 50 users...but at the next..with 150..well..the phone locked up over and over and couldn't make calls |
| 15:17.29 | *** join/#asterisk thewiizle (n=nick@87.127.85.42) |
| 15:17.33 | thewiizle | hi hi |
| 15:17.59 | thewiizle | what would cause asterisk to hit the rtp timeout if a call is connecting successfully in the sense that i get audio both ways |
| 15:18.04 | waverly360 | [TK]D-Fender: turns out the phone couldn't handle all of the presence indications..and their newest firmware hard limits blf to monitoring 50 people..the rest of the spots are worthless unless you use them for speeddial |
| 15:18.45 | [TK]D-Fender | waverly360: BLARG! |
| 15:18.57 | [TK]D-Fender | wave thats the POINT of the 57i! Receptionist! |
| 15:18.59 | waverly360 | [TK]D-Fender: Then there's the matter of calls being dropped from it randomly for some users. |
| 15:19.08 | waverly360 | [TK]D-Fender: Tell them that. |
| 15:19.23 | [TK]D-Fender | huey23: pass the buck! |
| 15:19.28 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.39 | waverly360 | [TK]D-Fender: We were astonished...and we'd promised that phone to a company who was MAJOR pissed when we told them about the limitations |
| 15:20.26 | [TK]D-Fender | waverly360: can't even use a SINGLE LCD attendant console to capacity let alone the *3* you're supposed to be able to chain |
| 15:20.36 | waverly360 | [TK]D-Fender: exactly. |
| 15:20.40 | [TK]D-Fender | waverly360: falls under the "#$^% that) category |
| 15:20.49 | waverly360 | [TK]D-Fender: I'll be damned if I willingly purchase another one of their phones. |
| 15:21.25 | thewiizle | hey TK |
| 15:25.48 | [TK]D-Fender | thewiizle: y0 |
| 15:26.43 | thewiizle | :) |
| 15:26.58 | thewiizle | can i pick ur brain for 2 mins? |
| 15:27.06 | thewiizle | ill be gentle |
| 15:27.10 | waverly360 | [TK]D-Fender: Was that a joke? or are you seriously speccing out a katana? :) |
| 15:29.10 | coppice | [TK]D-Fender: do you have some spare peasants to test it on? |
| 15:29.55 | [TK]D-Fender | coppice: No, I'm awaiting the NAFTA certificates to get some Mexicans in ;) |
| 15:30.06 | thewiizle | what would cause asterisk to hit the rtp timeout if a call is connecting successfully in the sense that i get audio both ways? |
| 15:30.11 | [TK]D-Fender | waverly360: Very seriously |
| 15:30.44 | [TK]D-Fender | waverly360: and if I really like what I get I'll have them redress my Oni Forge Bushi. |
| 15:30.56 | waverly360 | [TK]D-Fender: Well that sounds cool and all..but I don't know enough about samurai culture to know what that is :) |
| 15:31.35 | [TK]D-Fender | waverly360: http://gallery.aocomputing.net/index.php?album=2007-03-02+Oni+Forge+Bushi |
| 15:32.21 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:32.22 | coppice | a man with a ken do attitude |
| 15:32.51 | cpm | ouch |
| 15:33.01 | [TK]D-Fender | coppice: katori shinto actually :) |
| 15:33.02 | waverly360 | [TK]D-Fender: That's a very pretty katana. |
| 15:33.29 | waverly360 | [TK]D-Fender: So is it for display, or do you practice the art? |
| 15:33.43 | *** join/#asterisk ix33 (n=ix@4a.9e.5546.static.theplanet.com) |
| 15:33.47 | [TK]D-Fender | waverly360: It is... I'm likely to have it polished and refitted. |
| 15:33.55 | [TK]D-Fender | waverly360: It sees regular use |
| 15:34.02 | [TK]D-Fender | waverly360: iai-jutsu |
| 15:34.27 | [TK]D-Fender | waverly360: These are very inexpensive sword in my view. |
| 15:35.18 | waverly360 | I love japanese swords |
| 15:35.20 | [TK]D-Fender | waverly360: Bushi was $525, my new custom one is coming out at $550 so far (USD which comes out CHEAPER :p) and I'm still tweaking the build which will likely add another 50-150$ |
| 15:35.23 | ix33 | why can't i leave a voicemail by sending it from the voicemail menu but i can otherwise? |
| 15:35.29 | ix33 | i get the following: leave_voicemail: No entry in voicemail config file for '1003' |
| 15:35.30 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 15:35.49 | waverly360 | [TK]D-Fender: Actually..that does sound quite cheap to me. I would've expected something closer to the thousands |
| 15:36.14 | thewiizle | SwitchVox :) |
| 15:36.20 | waverly360 | [TK]D-Fender: I always assumed a very nice blade would be closer to 2-3 grand. |
| 15:36.22 | thewiizle | been using that for a few now |
| 15:36.50 | coppice | the traditionally made katanas are more like $100k |
| 15:37.08 | waverly360 | [TK]D-Fender: now, I wasn't familiar with the term iai-jutsu, so I looked it up. That's the part of fighting that involves drawing the blade...correct? |
| 15:37.38 | nettie | Hi guys, I'm runnig a bristuffed 1.2 version of asterisk I'm having issue with DTMF tones recognition using SIP channels, zap channels are fine. I tried every possible combination without luck. Anyone know what else I should try please? |
| 15:37.40 | [TK]D-Fender | waverly360: Yes, |
| 15:37.52 | ManxPower | nettie: what phones? |
| 15:37.52 | coppice | you mean sketching? |
| 15:38.41 | [TK]D-Fender | coppice: For the finest originals yes, but FAR less for a newly crafted shin sakuto direct from NBHK (sp?) accredited smith. |
| 15:39.13 | [TK]D-Fender | coppice: All still safely out of my budget :) |
| 15:39.22 | nettie | ManxPower nah, the problem is with the voip line and asterisk IVR. The voip carrier assigned me an overseas number which connects our IVR. |
| 15:39.36 | waverly360 | [TK]D-Fender: I'd love to own one personally. |
| 15:39.50 | nettie | LINE---voipcarrie---sipasterisk-->phones |
| 15:40.07 | waverly360 | [TK]D-Fender: You say you've used it quite often. have you ever struck another blade with yours? I'm curious as to how easily they chip... |
| 15:40.13 | [TK]D-Fender | waverly360: http://www.roninswords.com/ <- under the shin sakuto, I'm speccing a bushi (not the same as mine). |
| 15:40.31 | [TK]D-Fender | waverly360: these are NEVER used for blade on blade. |
| 15:40.54 | coppice | they are kept for blade on peasant |
| 15:41.10 | [TK]D-Fender | waverly360: Here, for relatively full understanding of my art : http://video.google.ca/url?docid=-3592341485993959661&esrc=sr2&ev=v&len=1197&q=katori%2Bshinto&srcurl=http%3A%2F%2Fvideo.google.ca%2Fvideoplay%3Fdocid%3D-3592341485993959661&vidurl=%2Fvideoplay%3Fdocid%3D-3592341485993959661%26q%3Dkatori%2Bshinto%26total%3D104%26start%3D0%26num%3D10%26so%3D0%26type%3Dsearch%26plindex%3D1&usg=AL29H20-6 |
| 15:41.12 | [TK]D-Fender | 4gSiSQoWVwDTZeHr94xFE3F6g |
| 15:41.17 | [TK]D-Fender | coppice: lol |
| 15:41.21 | waverly360 | lol |
| 15:41.26 | waverly360 | lemme resplice that |
| 15:41.27 | [TK]D-Fender | coppice: more like blade on convicted criminal. |
| 15:41.33 | ManxPower | nettie: OK. What DTMF Mode is the provider using? |
| 15:41.40 | nettie | when we call the overseas number, our asterisk IVR answers, we hit a digit to get trough it and it just dont recognise the tones. |
| 15:42.09 | ManxPower | nettie: that is a common symptom of Asterisk and the provier or phone not being set to the same dtmf mode. |
| 15:42.36 | nettie | ManxPower: I asked them didnt get an official answer yet but I tried rfc2833, inband with a g711 codec, relaxdtmf |
| 15:42.43 | thewiizle | sip info |
| 15:42.44 | thewiizle | ? |
| 15:42.49 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 15:42.51 | ManxPower | nettie: randomly trying is not the solution. |
| 15:43.09 | nettie | ManxPower: I know, it's not professional nor costructive.. |
| 15:43.13 | ManxPower | do not use relaxdtmf |
| 15:43.21 | ManxPower | nettie: come back when you know. |
| 15:43.21 | nettie | ManxPower: and very time consuming too |
| 15:43.33 | coppice | randomly trying is research 2.0 |
| 15:43.49 | nettie | ManxPower: K, I'll ping them again |
| 15:43.56 | ManxPower | I'm starting to dislike gmail |
| 15:44.34 | nettie | ManxPower: just wait, the new interface will be out soon :) and they just gave us IMAP, isnt that bad .. |
| 15:44.41 | ManxPower | Their karma from enabling IMAP service, as well as using SSL for IMAP and SMTP is rapidly being lost. I don't WANT to see all my e-mail ever received in my inbox. |
| 15:44.44 | nettie | ManxPower: webmail.us is great though |
| 15:45.35 | ManxPower | nettie: the reason I have not used gmail before is their lack of IMAP w/SSLorTLS |
| 15:46.01 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 15:46.06 | waverly360 | Crap |
| 15:46.13 | waverly360 | I accidently closed my irc client.. |
| 15:46.16 | nettie | gmail for domain is pretty cool .. |
| 15:46.19 | waverly360 | [TK]D-Fender: can you send me that link again please? |
| 15:46.41 | ManxPower | nettie: Oh, I have my own domain. |
| 15:47.00 | [TK]D-Fender | waverly360: Shorter link : http://video.google.ca/videoplay?docid=-3592341485993959661&q=katori+shinto&total=104&start=0&num=10&so=0&type=search&plindex=1 |
| 15:47.16 | nettie | we stick with webmail.us, they're very professional and is just great being able to resell their product with skinned for our customers |
| 15:47.40 | [TK]D-Fender | waverly360: And for the place I'm "shopping" : www.roninswords.com |
| 15:47.45 | nettie | search via webif is pretty cool they use lucene to index emails and stuff.. |
| 15:47.51 | nettie | pretty impressive.. |
| 15:48.11 | nettie | great infrastructure they're at rackspace (rackspace actually just aquired them) |
| 15:48.31 | ManxPower | nettie: I'm just used to only seeing e-mails in that folder when I click on that folder. |
| 15:48.39 | nettie | give them a look, of course they'e not free but their service is very worth every penny |
| 15:48.49 | nettie | ehehe |
| 15:48.54 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:48.55 | ManxPower | if I wanted the damn message in multiple folders, I would have put it in multiple folders. |
| 15:49.07 | nettie | ManxPower: yeah it's gmail labeling thing, take a while to get used |
| 15:49.24 | ManxPower | nettie: I will never get used to it. |
| 15:49.32 | nettie | if people can get used of vista, I bealive you can get used to labels hahaah |
| 15:49.34 | ManxPower | gmail seems to want to show you all messages. |
| 15:50.08 | nettie | yeah it's "their way".. |
| 15:50.51 | nettie | they should include something like: "enable human being look" option eheh |
| 15:51.15 | ManxPower | I am now beginning to understand why people can't find anything on gmail. |
| 15:51.16 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 15:51.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:51.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:51.35 | nettie | ManxPower do you know if I can figure out any dtmf configuration from their (voip carrier) sip header? |
| 15:52.06 | waverly360 | [TK]D-Fender: That's a really neat fighting style |
| 15:52.19 | ManxPower | nettie: not really, it depends on what Asterisk is sending and how the carrier handles it. |
| 15:52.38 | ManxPower | you can see what asterisk is currently receiving, but that could change when you change your DTMF mode. |
| 15:52.43 | *** join/#asterisk uwe (i=d46a490f@gateway/web/cgi-irc/ircatwork.com/x-f3b427ef8fb446ea) |
| 15:53.15 | ManxPower | nettie: I am unable to even find where I would filter on SENDER, not FROM |
| 15:53.20 | waverly360 | [TK]D-Fender: or maybe a combination of them...still watching :) |
| 15:53.41 | nettie | what worries me is the fact that they're an italian carrier selling an overseas number so the hops might be actually 2 not just one. What MY carrier is using and that HIS carrier is using.. |
| 15:54.25 | *** part/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net) |
| 15:56.38 | Poincare | Is there a way to get the result of a 'debug' output into a logfile or so, without the other messages? |
| 15:56.48 | nettie | in their howto's I can see dtmfmode is "inband" |
| 15:56.54 | ManxPower | Poincare: /etc/asterisk/logger.conf |
| 15:57.13 | ManxPower | nettie: inband DTMF will only work with ulaw or alaw codec. |
| 15:57.16 | Poincare | ManxPower: I should have known :-) thanks |
| 15:57.49 | ManxPower | nettie: I think the best way for me is just to delete all my archived mail. |
| 15:57.55 | nettie | ManxPower yes I know.. I alredy tried that without luck.. anyway in another howto they suggest to use rfc2833 |
| 15:58.18 | nettie | uhmm |
| 15:58.27 | ManxPower | nettie: Why are we still talking about this. You know what you have to do. |
| 15:58.45 | nettie | see if you could exclude some labels from the search |
| 15:58.50 | nettie | never tried though |
| 15:58.56 | nettie | not sure could be possible |
| 15:59.02 | nettie | or search with -archive |
| 15:59.03 | nettie | eheh |
| 15:59.04 | nettie | try |
| 15:59.07 | nettie | :) |
| 15:59.11 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 15:59.32 | nettie | create an archive label and tag "archived emails" with it |
| 15:59.38 | nettie | then search with -archive |
| 15:59.53 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 15:59.54 | nettie | maybe find an unusual name for the archived email |
| 16:00.06 | nettie | otherwise you wont find any email with the workd archive in it |
| 16:01.03 | *** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net) |
| 16:01.22 | *** join/#asterisk SexyKen (n=sexy@c-24-4-238-80.hsd1.ca.comcast.net) |
| 16:01.35 | ManxPower | nettie: 99% of the stuff I'll never need again. I have mail going back to 1999 in my current imap account. |
| 16:01.36 | *** join/#asterisk mtaht4 (n=m@87-104-62-200.enitel.net.ni) |
| 16:01.41 | SexyKen | Hey guys -- when setting up SRV record -- how do you tell it what you want clients to connect to in order to have the SRV record used? |
| 16:01.48 | SexyKen | For isntance, service.domain.com |
| 16:02.06 | SexyKen | But the SRV records dont reference what should be srv'd |
| 16:02.14 | *** part/#asterisk mtaht4 (n=m@87-104-62-200.enitel.net.ni) |
| 16:03.09 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 16:04.35 | Poincare | ManxPower: I rather want the results of a 'sip debug' into a file, no software debugging... If you have a clue |
| 16:04.44 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 16:04.44 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 16:05.17 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:05.22 | ManxPower | Poincare: that is not what you asked. |
| 16:05.26 | ManxPower | you can't do that. |
| 16:05.38 | Zeeek | Ô my |
| 16:05.50 | Poincare | ManxPower: ok, thanks :-) |
| 16:06.14 | ManxPower | use sip debug peer X or sip debug ip X |
| 16:07.14 | Poincare | yeah, I know, but I only know how to have that output on the console. And I cant keep my eyes all the time on the console |
| 16:07.44 | Poincare | would be easier if I could get just the sip debug output alone in a file |
| 16:09.02 | *** join/#asterisk Woifi1988 (n=anon@M1303P013.adsl.highway.telekom.at) |
| 16:09.16 | Woifi1988 | do i still need a make26 for kernel 2.6? |
| 16:09.43 | ManxPower | Woifi1988: do the install instructions tell you to do so? |
| 16:10.34 | Woifi1988 | yes but i read that this isn't required any longer |
| 16:10.42 | Woifi1988 | and the instructions are rather old |
| 16:11.12 | ManxPower | Woifi1988: Then report it as a bug. The install instructions that came with the source should not be out of date. |
| 16:11.13 | Woifi1988 | but i ask for zaptel, not for asterisk |
| 16:11.28 | Zeeek | Boyz and ..., in about 20 minutes from now, the voIP Users Conference, about asterisk, will begin live: http://x2z.eu |
| 16:11.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:11.42 | Zeeek | IRC #voip-users-conference |
| 16:11.51 | Zeeek | right here on FreeNode.net |
| 16:11.54 | Strom_M | boys and ellipsis? |
| 16:12.06 | ManxPower | Woifi1988: and Zaptel has a readme. |
| 16:12.09 | Zeeek | the rest of you know who you are! :) |
| 16:12.22 | Woifi1988 | okay i'll look |
| 16:12.27 | Zeeek | we welcome any one of the three dots |
| 16:12.32 | ManxPower | in fact in 1.2 it has SIX different readmes. |
| 16:12.57 | ManxPower | why do they even bother with them if nobody reads them. |
| 16:13.21 | Zeeek | ManxPower the tree in the forest |
| 16:13.32 | coppice | because the autotools won't work unless you have a README |
| 16:13.50 | Zeeek | russellb are you around? |
| 16:13.58 | Woifi1988 | where should the readme be locatet? |
| 16:14.27 | Woifi1988 | oh i see |
| 16:14.39 | Woifi1988 | big letters ;-> |
| 16:14.58 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 16:15.08 | Strom_M | ManxPower: i think they should employ reverse psychology and call it IGNOREME |
| 16:15.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:15.20 | Zeeek | that works for me |
| 16:15.46 | coppice | STUFFYOUWILLLATERWISHYOUHADREADATTHESTART |
| 16:15.57 | ManxPower | Strom_M: YES! |
| 16:16.54 | Woifi1988 | sorry that i asked but i am a 6 year old boy and i am too stupid to read |
| 16:18.09 | orkid | lol |
| 16:18.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:18.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:18.39 | Zeeek | not lol, LOL |
| 16:18.54 | ManxPower | We get a lot of those here. |
| 16:18.56 | Zeeek | not readme, README |
| 16:19.01 | ManxPower | some of them even throw tantrums |
| 16:19.07 | Zeeek | Whaaaaaahhhhhhhhhhh |
| 16:19.16 | Zeeek | you wrecked my dialplan |
| 16:19.48 | e` | How does asterisk pull the voicemail time? The "voicemail sent" time is off by 1 hour. |
| 16:19.50 | Woifi1988 | but can i install zaptel after asterisk? |
| 16:20.05 | Woifi1988 | or do i have to recompile asterisk afterwards? |
| 16:20.40 | ManxPower | Woifi1988: you can install zaptel after asterisk, but it won't do you any good until you rebuild asterisk. |
| 16:20.48 | Zeeek | OMG an intelligent answer from EBAY cust serv. What next? |
| 16:20.53 | Zeeek | EBay is the devil |
| 16:21.00 | ManxPower | if asterisk does not see zaptel already installed then the asterisk build process will disable any zaptel related code. |
| 16:21.05 | Woifi1988 | ManxPower: thx |
| 16:21.11 | Qwell | e`: sounds like a timezone issue |
| 16:21.19 | Qwell | ...one which will probably go away in 2 days, heh |
| 16:21.26 | Zeeek | ok I'll leave you to it, come to the conference if you'd care to http://x2z.eu |
| 16:21.29 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 16:21.58 | e` | Qwell, ah you are correct, the timezone date was changed this year |
| 16:22.15 | Qwell | e`: timezone can be specified in voicemail.conf though |
| 16:22.31 | uwe | hello, im trying to configure a catch-all thingie in the dialplan, it seems that i need to use "i" for that, but it is not working, did i get something wrong and its not "i" or is it that i did something wrong and should invistegate further in my current conf ? |
| 16:22.32 | *** join/#asterisk rpm (n=russell@dsl-crshts-66-18-230-165-cgy.nucleus.com) |
| 16:28.25 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
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| 16:34.11 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 16:38.11 | [TK]D-Fender | uwu : "i" is only usable for IVR's. You'd likely need a pattern match that checks against ANOTHER context to see if its valid and jump there or continue on fail. |
| 16:38.16 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 16:38.45 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:38.58 | mrtelephone | what does it mean that odbc voicemail storage is not realtime? |
| 16:39.55 | Alan_Hicks | Is there an easy application to play the gsm files shipped with asterisk? |
| 16:40.20 | thewiizle | PlaySound? |
| 16:40.37 | Alan_Hicks | I should have specified something other than "application". :^) |
| 16:40.42 | thewiizle | Playback even |
| 16:40.45 | Alan_Hicks | I mean if I just want to go through each one-by-one. |
| 16:40.54 | thewiizle | urm |
| 16:40.54 | Alan_Hicks | I could edit the dialplan, but that would be tedious. |
| 16:40.55 | thewiizle | FTP |
| 16:40.55 | thewiizle | :) |
| 16:40.57 | lirakis | hmmm .. i am seeing " WARNING[19077]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7004c88', 9 retries!" on my cli a lot ... any one know whats up? |
| 16:41.39 | Alan_Hicks | Something like mpg123 maybe? gsm123 /var/lib/asterisk/sounds/* |
| 16:41.49 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:41.55 | Alan_Hicks | Just so I know what each gsm file actualy plays. |
| 16:42.24 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:42.29 | [TK]D-Fender | Alan_Hicks: Winamp |
| 16:43.16 | Alan_Hicks | Winamp knows gsm? |
| 16:43.24 | Alan_Hicks | Who'd've thunk it. |
| 16:47.11 | ManxPower | lirakis: that is common when using ChanSpy, Monitor, MixMonitor, etc. |
| 16:47.51 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 16:48.14 | mrtelephone | hmm |
| 16:48.23 | mrtelephone | manx, have you tried odbc voiemail storage? |
| 16:48.53 | lirakis | ManxPower: okay .. i mean.. can i ignore it.. or is it a problem? |
| 16:49.29 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-235-73.lv.lv.cox.net) |
| 16:50.08 | ManxPower | lirakis: it is only a problem if it causes a problem. |
| 16:50.34 | mrtelephone | im trying to make redundant sip boxes but voicemail is an issue |
| 16:50.35 | lirakis | ManxPower: okay thanks |
| 16:50.37 | ManxPower | mrtelephone: I have never in my entire life interfaced Asterisk with any database whatsoever. Does that answer your question> |
| 16:50.52 | mrtelephone | manxpower, i was trying to avoid that myself |
| 16:53.36 | ai-a | anyone here use snom300 on a working environment ? - how do you hang up a call (when on headset) if another call is coming in? if you pick the receiver up and put it down, you have a race condition between you hangup up correctly, or the other person hanging up and you then hanging up on the next caller instead Heh.. If you press (X) it dismisses the next call,, sure you can press (X) again and it hangs up, but the next caller ha |
| 16:54.33 | ManxPower | ai-a: I'll let you in on a secret. Don't tell anyone. The answer you are seeking is located in the phone's user guide. |
| 16:55.16 | ai-a | ManxPower: read it, no help. |
| 16:55.43 | ManxPower | ai-a: then you would need to contact SNOM. |
| 16:57.06 | [TK]D-Fender | ai-a: And sacrifice a goat |
| 16:57.22 | ai-a | Heh |
| 16:57.54 | ai-a | thing is we have it live and the telesales people are going nuts at IT.. im like,, WTF.. saw it.. we are not saying press X twice, let someone else answer the call thats ringing. |
| 16:58.04 | ai-a | however, if its a direct call to them,, they will send them to voice mail. |
| 16:58.05 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 16:58.55 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:00.02 | ManxPower | ai-a: too bad you didn't test this stuff before deployment. |
| 17:00.09 | ManxPower | too bad you won't call snom tech support. |
| 17:00.16 | ManxPower | too bad you have not searched google. |
| 17:00.33 | ManxPower | I assume you are trying to get fired? |
| 17:01.00 | ai-a | ManxPower: calm down mate. |
| 17:01.31 | ccaron | I try to replace my old PBX with asterisk but I dont know which type of ISDN signal bus go out of the 2 NT |
| 17:01.51 | ccaron | anybody with ISDN knowledge on this forum ? |
| 17:01.55 | ManxPower | ccaron: what card do you have? |
| 17:02.10 | ccaron | I try with a fritz isdn card |
| 17:02.52 | ccaron | it work fine on the S bus of the office but not if I conect it directly in in the NT ehere the PBX is curently conected |
| 17:03.20 | ccaron | I am in switzerland and PBX is from ascotel |
| 17:04.35 | ccaron | and misdn is configured to use the card in te_ptmp mode |
| 17:06.05 | *** join/#asterisk MoutaPT (n=mmouta@dhcp11.corp.fccn.pt) |
| 17:06.26 | *** join/#asterisk lyoungz (n=larry@64.52.119.81) |
| 17:10.30 | lyoungz | Can someone tell me how to send a '*' dtmf from a SIP phone through an Asterisk server (trixbox in this case) so that the other party to the call (in this case a computer) gets the DTMF? Right now I think Trixbox's voicemail system is intercepting the '*' |
| 17:11.46 | MoutaPT | probably you have '*' as firs character in any feature code in features.conf of your asterisk server |
| 17:11.47 | MoutaPT | no? |
| 17:12.52 | [TK]D-Fender | lyoungz: First if you're on a call with another person, what's VOICEMAIL have to do with anything? |
| 17:12.56 | [TK]D-Fender | lyoungz: Second.. |
| 17:12.59 | [TK]D-Fender | ~trixbox |
| 17:13.00 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 17:13.29 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 17:14.27 | rene- | hey is there a way to get the number of channel a call has come tru? i know it can be done using AMI but i would like to know if there is a simple way of doing it using dialplan functions |
| 17:15.16 | *** join/#asterisk adeel (n=adeeln@c-24-7-132-155.hsd1.ca.comcast.net) |
| 17:15.22 | ManxPower | rene-: you never read README.variables have you. |
| 17:15.38 | rene- | hehe |
| 17:15.44 | rene- | reading the wiki now |
| 17:15.46 | rene- | sorry |
| 17:15.47 | rene- | thx |
| 17:15.50 | ManxPower | DONT! |
| 17:15.55 | ManxPower | the wiki is not the readme. |
| 17:15.57 | rene- | ok |
| 17:16.12 | ManxPower | the readme is in the asterisk source code. That is the only place you should be reading asterisk or zaptel readmes |
| 17:16.31 | adeel | i can only get dtmf to work when I use dtmfmode=info, but i'm using polycom phones that use rfc2833, does that matter? will * use the info method or rfc2833? |
| 17:17.01 | ManxPower | in fact, I kow specifically that as of a week ago the Wiki's variable information had only about 10% of the variables listed that are supproted in 1.2 or 1.4 |
| 17:17.30 | ManxPower | adeel: asterisk will use whatever mode you TELL IT TO USE. |
| 17:18.03 | ManxPower | "make install" should ask the user a series of questions that can only be answered by reading the readmes. |
| 17:18.06 | adeel | ManxPower, so i shouldn't have problems with the phones using rfc2833, and * using info? * will automatically do the conversion? |
| 17:18.49 | ManxPower | adeel: if the phones are using rfc2833 and asterisk is using INFO for those phones it's not going to work very well. |
| 17:19.35 | adeel | ManxPower, hmm...this might present a problem then; as i can only seem to get dtmf working reliably when i use info |
| 17:19.41 | adeel | i'll keep fiddling with it |
| 17:19.49 | ManxPower | adeel: issue info for what? |
| 17:19.57 | adeel | dtmfmode=info |
| 17:20.15 | ManxPower | phone(rfc2833) <-> (rfc2833)Asterisk(INFO) <-> Provider |
| 17:20.17 | ManxPower | that will work. |
| 17:20.42 | ManxPower | phone(rfc2833) <-> (INFO)Asterisk(INFO)<->Provider will NOT WORK |
| 17:20.54 | ManxPower | you say "asterisk" like Asterisk can only do one thing. |
| 17:21.11 | ManxPower | or like these settings are global. they are not global. |
| 17:21.25 | adeel | okay, so in the first scenario you describe, i can't have dtmfmode=info in the general sip.conf heading correct? |
| 17:21.49 | ManxPower | no, you put it in the [sectionofsipconfforyourprovider] |
| 17:22.53 | adeel | ManxPower, okay, thanks |
| 17:23.46 | *** part/#asterisk Woifi1988 (n=anon@M1303P013.adsl.highway.telekom.at) |
| 17:24.32 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:25.07 | *** join/#asterisk PC_Clone (n=pc_clone@209-252-117-210.ip.mcleodusa.net) |
| 17:26.20 | PC_Clone | hi....is there a setup wizard for extensions voicemail so that when they call for the first time it will walk them through recording their greetings? |
| 17:26.35 | adeel | PC_Clone, yes, check the voicemail.conf file |
| 17:27.19 | adeel | PC_Clone: specifically, set the option 'forcegreetings=yes' |
| 17:27.29 | PC_Clone | adeel: |
| 17:27.34 | PC_Clone | adeel: thanks |
| 17:28.11 | adeel | PC_Clone: one thing to note, it might require that the voicemail password be the same as the extension, before that prompt is played, and once the user changes the password, that prompt will stop |
| 17:28.47 | PC_Clone | adeel: I have already done that |
| 17:28.55 | PC_Clone | and was wondering why it wasn't doing the setup |
| 17:28.58 | PC_Clone | thanks though |
| 17:29.41 | adeel | PC_Clone: no prob |
| 17:30.56 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
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| 17:47.10 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
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| 17:57.21 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 17:57.52 | *** join/#asterisk uwe (i=d46a490f@gateway/web/cgi-irc/ircatwork.com/x-7ae8e9937008b34c) |
| 17:57.54 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:58.14 | uwe | um, how can i test sip trunk ? sip show register ? |
| 17:58.29 | uwe | registry* |
| 17:58.48 | [TK]D-Fender | uwe: Have you considered trying to PLACE or RECEIVE calls? :) |
| 18:01.03 | uwe | well ... im unable to do that , thats why im trying to dubug it [TK]D-Fender |
| 18:02.02 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 18:02.10 | uwe | as usual im trying something from an already existing config, and thus , again , as always , running into problem related to other things that seem to be irelevent :) |
| 18:02.11 | [TK]D-Fender | uwe: PASTEBIN is your friend. |
| 18:02.36 | uwe | :) ok |
| 18:03.14 | *** join/#asterisk agx (n=badpengu@81-174-8-141.dynamic.ngi.it) |
| 18:04.00 | uwe | well, a little question first, is it actually alright if i make a sip trunk on the same asterisk machine that im trying to dial out from ... just for the sake of testing :) |
| 18:04.31 | uwe | ie, dial in the same machine im dialing out using a sip trunk |
| 18:05.25 | *** part/#asterisk MoutaPT (n=mmouta@dhcp11.corp.fccn.pt) |
| 18:05.49 | [TK]D-Fender | uwe: Don't |
| 18:05.54 | [TK]D-Fender | Test each independantly |
| 18:07.13 | ManxPower | ~trunk |
| 18:07.13 | jbot | trunk is probably a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment." There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 18:08.06 | uwe | oh ... hmmm |
| 18:08.25 | ManxPower | the usual term for what you mean is "sip peer" |
| 18:08.49 | ManxPower | which people use to mean type=peer or user or friend |
| 18:09.30 | uwe | i see ... well, is there something like a /dev/null trunk that i can use :) just to send all calls to it ! :D |
| 18:10.28 | uwe | its because i need a trunk , but have no other machine to test with or hardware or anything similar i can think of ! |
| 18:11.46 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 18:13.19 | ManxPower | um, send the call to an ivr |
| 18:13.34 | ManxPower | I see you are totally ignoring the community standards here. |
| 18:14.54 | orkid | what's a good cheap fxs/fxo? (for now, i just want an fxs, but maybe in the future an fxo will be a good idea). |
| 18:15.20 | uwe | ManxPower, why do you think im ignoring the community standards ? |
| 18:15.22 | ManxPower | orkid: there are none. The closest you are going to get is a cheap sipura SIP/analog ATA adapter. |
| 18:15.34 | ManxPower | uwe: you are still using the word trunk to mean "sip" |
| 18:17.14 | orkid | hmm ok, how about iaxy? |
| 18:17.26 | uwe | well, sorry, im just used to using it to refer to something that asterisk can send calls to it and is not an extension ... i appologize |
| 18:17.30 | ManxPower | it's like saying "bicycle" to mean "car" |
| 18:17.46 | ManxPower | uwe: that would be anything in asterisk does does anything except Dial. |
| 18:17.56 | ManxPower | orkid: I will never buy another iaxy ever ever ever |
| 18:18.06 | ManxPower | orkid: the iaxy is more expensive than the sipura |
| 18:18.27 | lirakis | I have agents who are logged into a queue. When the agents make a call outbound and are still logged in, Asterisk thinks they are still available and tries to send them calls... is there a way to make * not send calls to agents who are on the phone for non queue originated calls? |
| 18:19.08 | ManxPower | btw, extensions are things in extensions.conf. A phone is not actually an extension, a phone or PSTN line is an "endpoint" or "phone" or "pstn". extensions.conf maps extensions to the physical device. |
| 18:19.35 | ManxPower | Asterisk requires all incoming calls to match an exten => line. |
| 18:19.46 | orkid | is the 3102 better than the 2102? (SPA-) |
| 18:20.03 | ManxPower | exten => 5551512,1,Playback(tt-monkeys) assuming your DID with your SIP provider is 5551212 |
| 18:20.16 | ManxPower | orkid: they are all the same except for the interfaces |
| 18:20.59 | ManxPower | cell phone calling 5551212 -> SIP provider -> Asterisk -> exten 5551212 -> Playback |
| 18:21.18 | mocker | -> screaming monkeys |
| 18:21.19 | ManxPower | you might want to add a Hangup after the Playback |
| 18:22.42 | orkid | how about a cheap dlink dph-50u? (usb) |
| 18:22.53 | orkid | or is that skype only? |
| 18:23.17 | ManxPower | if it is usb don't expect it to work with asterisk |
| 18:23.56 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:24.44 | ManxPower | all the linksys/sipura atas out there you can get for less than $100 |
| 18:25.56 | ManxPower | I would get the SPA3000 if you don't need a built in router, and a SPA3120 if you do need a built in router. |
| 18:26.14 | ManxPower | sorry, SPA3102 |
| 18:30.38 | orkid | how about those pci cards derived from motorola modems? |
| 18:30.51 | orkid | is the voice quality call reliability etc good with them? |
| 18:31.45 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:31.50 | ManxPower | best of luck finding one. |
| 18:32.09 | orkid | really? i thought there were clones around? |
| 18:32.13 | ManxPower | they have not been built in several years. The intel ship they used was discontinued. |
| 18:32.21 | [TK]D-Fender | ManxPower: the 3000 is completely discontinued. Only the 3102 should exist at this point except for old stock |
| 18:32.21 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.68) |
| 18:32.31 | *** join/#asterisk ReDNeQ- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:32.34 | ManxPower | rumor has it many of the "x100P" compat cards are really old cards that failed QA testing |
| 18:32.40 | [TK]D-Fender | orkid: What are you looking for? FXO, or FXS? |
| 18:32.47 | ManxPower | [TK]D-Fender: linksys has been doing alot of that recently. |
| 18:33.12 | ManxPower | you can no longer get a single AP with support for a, b, and g standards. |
| 18:33.17 | ManxPower | (from linksys) |
| 18:33.42 | orkid | fxs at first, maybe fxo later on. so possibly both at once would be best. it'd be for home use, i'd rather use something low power, possibly to intergrate with a small (miniITX or embedded type) computer, but maybe an old (like a K6-233) full size PC. |
| 18:35.40 | *** join/#asterisk kkjoe (n=kkjoe@p578b42d3.dip0.t-ipconnect.de) |
| 18:35.56 | [TK]D-Fender | orkid: Then get an SPA-3102 and if you don't need more FXO's and only need FXS grab SPA-2102's here |
| 18:36.29 | kkjoe | is there any misdn irc channel ? |
| 18:36.33 | orkid | so this is a no good (depending on my luck)? http://www.x100p.com/products/FXO.php |
| 18:37.41 | [TK]D-Fender | orkid: I'd avoid that cheap junk if I were you. |
| 18:38.14 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 18:38.47 | ManxPower | orkid: telecom is expensive |
| 18:38.54 | orkid | ok. sipura it will probably be then (or well, linksys now). so i can use it with asterisk 'directly', as i would an x100p? |
| 18:39.43 | ManxPower | orkid: if you wanted 4-ports on most PBXs expect to pay $1000 |
| 18:39.43 | orkid | yeah. it looks like it, kinda surprising seeing how cheap internet stuff is now (at least wifi/network cards, although stuff that goes over 2-wire copper, like adsl modems, are still a bit pricier). |
| 18:39.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:40.18 | ManxPower | orkid: an x100p is a PCI card, SIPURA are SIP adapters, no pci slit required |
| 18:41.04 | [TK]D-Fender | orkid: its a SIP device. You place calls to it and get calls fromit no differently than any other SIP communication. |
| 18:41.42 | orkid | oh ic, so it's not like an x100p? i'm still trying to wrap my head around voip (even after reading the online book on voip from asterisk) |
| 18:41.58 | [TK]D-Fender | orkid: Its a little box you plung onto your LAN. Thats all |
| 18:42.02 | orkid | so the spa3102 acts like it's own 'server'? |
| 18:42.33 | ManxPower | orkid: think of it as a device that converts SIP/VoIP into analog for analog phones. |
| 18:42.41 | [TK]D-Fender | orkid: http://www.telephonydepot.com/product_p/105-054-312.htm |
| 18:42.49 | [TK]D-Fender | orkid: There... $35 / Port |
| 18:43.10 | [TK]D-Fender | orkid: it is a SIP device, nothing more. What makes something a "server"? |
| 18:43.29 | orkid | ok, same thing for an x100p though, right? except that it's integrated into the computer (that in this case would have asterisk). so if i weret to put it in another computer (the x100p) it would require some software to get running and talking to asterisk. |
| 18:43.44 | orkid | basically, how much am i tied in with the spa3102 firmware, is my question. |
| 18:43.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:44.04 | [TK]D-Fender | orkid: That question makes no sense |
| 18:44.08 | orkid | it doesn't? |
| 18:44.11 | [TK]D-Fender | orkid: What do you mean "tied it"? |
| 18:44.15 | [TK]D-Fender | tied in* |
| 18:44.24 | orkid | hmm. i'm confused. |
| 18:44.46 | orkid | i mean, how much procesing does the spa3102 do? what software does it use, etc? |
| 18:45.23 | [TK]D-Fender | orkid: indeed you are. It is just a dumb device like a SIP phone. You tell it its account credentials, where to send calls and what to auth outgoing calls and thats it. |
| 18:45.29 | orkid | it has to convert the analog into SIP/VoIP, so I'm assuming it has software on it for configuration purposes. But this software is probably not 'free' |
| 18:45.42 | [TK]D-Fender | orkid: Its just like a SIP phone execpt it leads to a LINE. this isn't Raw Caw science. |
| 18:45.46 | orkid | how about codecs? |
| 18:45.58 | [TK]D-Fender | orkid: It has its own firmware like every other device onthe planet... |
| 18:46.00 | sigmounte | hello ! is it normal that in the console whatever phone i use , the console always tell me it's the same channel used (also not the good one ) ? |
| 18:46.02 | orkid | raw caw? lol wtf is that! |
| 18:46.15 | orkid | sure, but some stuff (like wrt54gl) can be flashed with free firmware (openwrt, ddwrt, etc) |
| 18:46.18 | [TK]D-Fender | orkid: hukt on fonix werkt 4 u |
| 18:46.37 | [TK]D-Fender | orkid: No, there is no free firmware that I'm aware of for these, and there is little point. |
| 18:46.58 | *** join/#asterisk ginvent (n=ginvent@76.203.119.54) |
| 18:47.07 | [TK]D-Fender | orkid: Yes, DD-WRT is fun, but what gives a shit about trying to convince a dog it's an ELEPHANT? |
| 18:47.42 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 18:48.06 | [TK]D-Fender | orkid: The firmware works ok? Do you rip apart your cars computer trying to hack in OSS into it for the gentoo ricing warm fuzzy feeling? It ain't broke.... |
| 18:48.34 | [TK]D-Fender | NEXT!@!@@! (c) BKW |
| 18:48.37 | orkid | lol. i might break into it to add some juice if i want. or reprogram ignition timing, etc. |
| 18:48.57 | ginvent | I keep Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:48.59 | orkid | but anyway, i heard there are some codecs that are 'non-free'. |
| 18:49.05 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 18:49.05 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 18:49.12 | orkid | would the spa3102 support these? |
| 18:49.14 | ginvent | I installed on ubuntu |
| 18:49.21 | orkid | where would you suggest i go to learn more about this? |
| 18:49.24 | ginvent | when I try and run it... I keep getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:50.18 | [TK]D-Fender | orkid: Have you ever considered LOOKING rather than just bombarding all of these questions upon us? You'd have all this info if you bothered to look at all. |
| 18:50.41 | [TK]D-Fender | orkid: And it supports G.711, G.723, G.726, and G.729 |
| 18:50.47 | ginvent | weird... got it. |
| 18:50.53 | [TK]D-Fender | ginvent: What user are you logged as? |
| 18:51.00 | ginvent | root |
| 18:51.08 | [TK]D-Fender | ginvent: Hrm, that would be wierd |
| 18:51.24 | JackEStorm | [TK]D-Fender: not if it's not running |
| 18:51.33 | [TK]D-Fender | JackEStorm: indeed |
| 18:54.19 | orkid | no need to get angry. just thought someone might offer some links to get acquainted with the codecs involved, differences between them, and other such information. |
| 18:54.35 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-218-77.cmdnnj.east.verizon.net) |
| 18:55.20 | *** part/#asterisk horsesgofaster (n=dcantera@pool-72-82-218-77.cmdnnj.east.verizon.net) |
| 18:56.03 | [TK]D-Fender | orkid: Good tip : try showing SOME motivations. Thats all. You start off as "cheap" and then quickly jump to looking "lazy". Not a great impression in these parts. |
| 18:56.14 | [TK]D-Fender | orkid: But you're not yet baked in my books :) |
| 18:58.24 | orkid | lol. i've been around for a while, and been reading voip-info and other sites to learn. but since i don't have much experience in telephone (beyond using a phone :P it all seems kind of confusing to me. and there are tons of providers out there. it's hard to know which one is good, what makes up 'good' and 'reliable' etc. maybe it's best now to just try stuff out and learn by experience. |
| 18:59.13 | mocker | orkid: Remember that all ITSPs suck, just by varying degrees. |
| 18:59.25 | ManxPower | orkid: to be good at Asterisk, you need to be good at Linux, Networking (routing, NAT, TCP, UDP), Telecom, and Asterisk |
| 19:00.11 | rene- | and an affinity to be seated staring at a computer for unhealthy amounts of time |
| 19:00.36 | mocker | rene-: That's the easy part, crawling under desks and hooking up phones is the hard part. :) |
| 19:02.30 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:02.30 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:03.11 | ManxPower | mocker: that is what cable guys and helpdesk people are for. |
| 19:04.04 | mocker | ManxPower: In a perfect world. |
| 19:04.10 | mocker | I need an underling. |
| 19:04.22 | ManxPower | PFYs should be pretty cheap. |
| 19:05.37 | mocker | True. |
| 19:06.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:07.46 | *** join/#asterisk techie (n=techie@adsl-76-214-27-184.dsl.lsan03.sbcglobal.net) |
| 19:07.52 | e` | Do you think there will be an issue with having Cisco and polycom phones on the same asterisk machine? |
| 19:09.15 | ManxPower | e`: only an issue with your sanity. |
| 19:09.30 | ManxPower | are the cisco phones running SIP, H323, MGCP, or Skinny/SCCP? |
| 19:10.46 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 19:11.51 | orkid | hmm, 3102 is not on linksys.com ? weird. |
| 19:12.17 | [TK]D-Fender | orkid: neither is most of D-Link's on theirs |
| 19:12.30 | [TK]D-Fender | orkid: www.voxilla.com <- go check out the forums to learn more. |
| 19:15.59 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 19:16.17 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 19:16.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:17.41 | orkid | ty |
| 19:18.46 | pepse | hi sorry for stupid offtopic question, but maybe someone knows off the top of their head... 1 / -2 = 0 r -2 or 0 r +2? (i just need confirmation for peace of mind) |
| 19:19.51 | Qwell | pepse: eh? |
| 19:20.20 | pepse | one thing i see says remainder is always same sign as dividend, another thing i see says divisor |
| 19:21.09 | Qwell | divisor |
| 19:21.24 | pepse | thanks :) |
| 19:21.31 | Qwell | -.5 |
| 19:21.46 | pepse | yeh i know the quotient is neg if opposite signs |
| 19:22.13 | pepse | hooray for doing multiply/divide manually in assembly |
| 19:22.50 | pepse | now to figure out how to arrange the jumps. |
| 19:25.28 | russellb | it's actually 0 r 1 .... |
| 19:25.32 | russellb | :-p |
| 19:25.59 | Shaun2222 | are there any good tools for generating statistics from asterisk.. basically trying to do like performance reports. |
| 19:26.13 | mocker | asterisk-stat is pretty popular |
| 19:26.54 | Shaun2222 | ya i tried installing that but it looks to be broken. |
| 19:27.23 | [TK]D-Fender | Shaun2222: MS Excel |
| 19:27.35 | Shaun2222 | get mysql errors showing access denied, but the user/pass it's using works... i even tested it by editing code in the script to see if the vars are being passed. |
| 19:31.03 | Shaun2222 | anybody else seen this with asterisk-stat on centos 4 with default apache/php rpms installed? |
| 19:31.41 | hmmhesays | http://www.pastebin.ca/759015 <-- i'm having problems with my feature codes not working |
| 19:32.09 | hmmhesays | I thought that I had the featuredigittimeout set to low, so I set it to 1.2 seconds and it doesn't seem to have affected anything |
| 19:32.25 | ManxPower | hmmhesays: you are using w or W on the Dial line? |
| 19:32.36 | hmmhesays | ManxPower negative |
| 19:32.46 | ManxPower | hmmhesays: then don't expect features.conf stuff to work. |
| 19:33.07 | hmmhesays | I thought wW was just for one touch recording? |
| 19:33.09 | ManxPower | (other than the stuff handled by t/T. Also, you want to set DYNAMIC_FEATURES variable, I think. |
| 19:33.22 | hmmhesays | ManxPower: i'm just looking at TT |
| 19:33.25 | hmmhesays | *Tt |
| 19:33.25 | ManxPower | hmmhesays: well that's what the docs SAY, but I have found it to not be true. |
| 19:33.37 | hmmhesays | really? I'll add it and try |
| 19:33.54 | ManxPower | if not, look up the DYNAMIC_FEATURES in README.variables |
| 19:33.57 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 19:34.14 | ManxPower | hmmhesays: it has been a while since I tried working with that. |
| 19:34.19 | hmmhesays | you know what i'm freaking retarded, I left Tt off my zap dial |
| 19:35.54 | hmmhesays | yeah whoops |
| 19:36.46 | ginvent | Ugh. my phone keeps trying to register and asterisk keeps saying :No matching peer found |
| 19:37.00 | ManxPower | ginvent: it should be saying a little more than that. |
| 19:37.07 | ManxPower | pastebin the errors |
| 19:37.08 | ManxPower | ~pb |
| 19:37.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:37.20 | ginvent | [Nov 2 12:35:02] NOTICE[27407] chan_sip.c: Registration from '<sip:200@192.168.1.102>' failed for '192.168.1.202' - No matching peer found |
| 19:37.35 | ManxPower | you do not have a [200] section of sup.conf |
| 19:37.40 | *** join/#asterisk yoanis (n=fred@200.55.139.218) |
| 19:37.53 | ManxPower | as that is the username the phone is sending. |
| 19:37.53 | ginvent | I do. |
| 19:38.03 | ManxPower | type=peer or friend or user? |
| 19:38.10 | orkid | i've looked at the pap2t and spa3102 info pages on store.voxilla.com. i have a question about encryption: is encryption widely used in voip/sip communications nowadays? what would have to support the encryption for it to be end to end? |
| 19:38.46 | ManxPower | orkid: no it is not common. |
| 19:38.46 | orkid | (ie. the device, the sip network you connect, to etc). |
| 19:38.46 | ManxPower | Asterisk does not support it, for example. |
| 19:38.56 | orkid | how about ZRTP? |
| 19:39.02 | ManxPower | We generally feel if you want an encrypted connection you should use a VPN, not some encryption feature tacked on as an afterthought. |
| 19:39.54 | file | VoIP over OpenVPN works quite well |
| 19:39.58 | hmmhesays | cheers to that |
| 19:40.10 | GreggB | Anyone using a Polycom IP 4000? I wanted to find out if the power-injector must be a Polycom provided one, or if I can just use any standard PoE injector? |
| 19:41.19 | [TK]D-Fender | GreggB: Any standard 802.3af will do |
| 19:41.34 | keith4 | does anyone still use non-standard PoE anymore? |
| 19:41.41 | Qwell | cisco |
| 19:41.42 | [TK]D-Fender | keith4: Only Cisco |
| 19:41.43 | Qwell | nuff said |
| 19:41.47 | keith4 | i had some cisco crap that didn't.... okay nevermind |
| 19:41.55 | keith4 | i guess some things never change |
| 19:42.07 | ginvent | The type is friend, what should it be? |
| 19:42.14 | ginvent | peer? |
| 19:42.26 | ManxPower | ginvent: that should work. pastebin the [200] section of sip.conf |
| 19:42.43 | pepse | friggin cisco |
| 19:43.20 | GreggB | [TK]D-Fender: thanks keith4: I have a Polycom SoundStation2 which looks to use non-standard PoE...but then that's an analog phone, so it's not really "ethernet" that it's using anyway... |
| 19:43.22 | ginvent | http://www.pastebin.ca/759032 |
| 19:43.28 | irule | has anyone had experience with the ip04? |
| 19:43.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:43.48 | [TK]D-Fender | GreggB: Can't be PoE if it isn't ETHERNET :p |
| 19:43.59 | Qwell | PoCAT5 |
| 19:44.13 | [TK]D-Fender | GreggB: We call those "power bricks" or "wall warts" :p |
| 19:44.35 | [TK]D-Fender | ginvent: FreePBX, yay! |
| 19:44.38 | [TK]D-Fender | ~freepbx |
| 19:44.39 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:45.02 | ManxPower | ginvent: first of all we do not support GUIs here. try this pastebin: http://www.pastebin.ca/759033 |
| 19:45.10 | orkid | sure, but voip over vpn. where does the vpn end? |
| 19:45.26 | ManxPower | orkid: hopefully somewhere you can trust. |
| 19:45.35 | ginvent | Ok Ok... just thought I would try. |
| 19:46.47 | [TK]D-Fender | ginvent: You did, thats why God invented failure :) |
| 19:47.48 | GreggB | [TK]D-Fender: Well, I guess it's PoCAT5 then :-P (thanks Qwell), my PoCAT5 "wall warts" are hidden in server room; users just think it's magic that the "...old analog phone..." powers up without the rats nest following it around nowadays. |
| 19:48.31 | [TK]D-Fender | GreggB: Ah, so is uses the same style jack as the IP 501 then. |
| 19:48.45 | keith4 | I guess you could call it PoRJ48, too then |
| 19:49.36 | GreggB | lol - yea now the analog phone uses...PoCAT5RJ45 |
| 19:49.53 | Shaun2222 | with voicemail imap, if all voicemail is read in the mailbox will the notification light stop blinking on the phone? |
| 19:50.01 | Shaun2222 | or do they still have to log in and clear them |
| 19:51.36 | putnopvut | Shaun2222: It *should* clear itself without having to log in. |
| 19:52.25 | GreggB | ...or Po((CAT5|CAT5e|CAT6)&(RJ45|RJ48))....whew... I love standards, there's so many of them. I better stop drifting OT before I get myself into trouble... |
| 19:53.46 | ManxPower | I thought Cat5e and Cat6 were "marketing standards" not technical standards" |
| 19:54.08 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 19:54.13 | ZaVoid | morning all |
| 19:54.19 | Strom_M | ManxPower: you are wrong :) |
| 19:54.40 | ZaVoid | so what would cause random "stuttery" sound file playing |
| 19:54.43 | keith4 | Cat6 specifies more twists per inch than 5e, doesn't it? |
| 19:54.55 | ZaVoid | it does keith4 |
| 19:55.11 | Strom_M | keith4: no, it specifies frequency and noise tolerances; it's up to the manufacturer to figure out how to meet those tolerances |
| 19:56.02 | ManxPower | 25-pair Cat5 cables have MANY twists per inch. |
| 19:56.43 | ManxPower | (we used them for telco DEMARC <-> DSLAM) |
| 19:57.03 | Shaun2222 | ok, asterisk-stat still borked with latest version of apache and php.. |
| 19:57.09 | Shaun2222 | anybody have asterisk-stat working? |
| 19:59.50 | yoanis | hello |
| 19:59.53 | ManxPower | Shaun2222: perhaps the asterisk-stat mailing list would be more helpful to you. |
| 20:00.14 | yoanis | i have a voice modem and telephone line, i would like to know what asterisk can do for me? |
| 20:01.03 | Shaun2222 | ManxPower: where's this list? |
| 20:01.06 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 20:02.22 | ManxPower | Shaun2222: no idea, but obviously nobody runs that software here. |
| 20:02.35 | ManxPower | perhaps you could check the docs for the non-asterisk software you are trying to use. |
| 20:02.44 | ManxPower | non-asterisk, non-digium software. |
| 20:03.31 | [TK]D-Fender | waverly360: Just finished negotiating all the fittings and some custom blade characteristics.... Katana order is going to happen :) |
| 20:03.44 | [TK]D-Fender | waverly360: Massive custom job :) |
| 20:03.51 | ManxPower | I think I'll go to a Windows channel and ask about linux. They both run on computers, right? |
| 20:03.55 | waverly360 | [TK]D-Fender: Sweet! Congratulations :) |
| 20:04.11 | waverly360 | [TK]D-Fender: post pictures when you finally get it |
| 20:04.22 | [TK]D-Fender | waverly360: You can bet on it.... |
| 20:04.26 | GreggB | yoanis: If you eBay your voice modem, and buy a LinkSys SPA3102 instead, Asterisk can do lots for you 9.9 ...never tried to use a voice modem with Asterisk..doesn't Asterisk just work with one particular type of voice modem anyway? |
| 20:04.27 | ManxPower | Even better yet, I'll go to a windows channel and ask for support on Adbobe. |
| 20:05.45 | ManxPower | yoanis: nothing. There are two specific modem chipsets that might work with Asterisk. One of those chipsets was discontinued something like 5 years ago and the other one 2 or 3 years ago. |
| 20:05.51 | [TK]D-Fender | waverly360: In fact... I changed virtually EVERYTHING from the based models shown. Custom painted saya, replaced all the fittings with stuff not listed incl custom gold work, and blade redimensioning. All for about $700 USD pre-ship |
| 20:06.16 | ManxPower | so unless you bought the "modem" from a vendor that specifically claims "asterisk compat" then you are out of luck. |
| 20:06.16 | yoanis | GreggB: i don't know that's why i'm asking |
| 20:06.23 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:06.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:06.30 | waverly360 | [TK]D-Fender: That's really not bad at all. I'm amazed you could get a custom katana made for less than $1000 |
| 20:07.24 | ManxPower | yoanis: and no, Asterisk does not support voice modems. |
| 20:08.08 | ManxPower | Even on those two supported chipsets Asterisk does NOT use the voice modem commands. |
| 20:08.20 | Qwell | ManxPower: 2? |
| 20:08.32 | [TK]D-Fender | waverly360: a little challenging, but doable. http://www.lastlegend.com has some good stuff, but what I'm doing is more like a consigned piece of art. LL's are more "colour by numbers" cutting machines. |
| 20:09.15 | ManxPower | Qwell: the motorola and ambient |
| 20:09.34 | Qwell | [TK]D-Fender: usable/sharp? |
| 20:09.35 | ManxPower | the X100P was motorola and the X101P was Intel ambient |
| 20:09.38 | Qwell | oh |
| 20:10.09 | [TK]D-Fender | Qwell: Full-on folded, differentially tempered high-carbon steel. |
| 20:10.24 | Qwell | so...yes? |
| 20:10.25 | [TK]D-Fender | Qwell: Next best thing to nihonto :) |
| 20:10.29 | [TK]D-Fender | Qwell: YES |
| 20:10.35 | Qwell | pre-sharpened? |
| 20:10.38 | [TK]D-Fender | Qwell: Supremely usable. |
| 20:10.39 | waverly360 | [TK]D-Fender: I'd love to own a nice one...but before I'd shell out the cash I'd really like to learn to use one effectively. |
| 20:10.53 | [TK]D-Fender | Qwell: Very |
| 20:11.04 | Qwell | neat |
| 20:11.18 | Qwell | but it isn't actually FOR use? |
| 20:11.26 | [TK]D-Fender | Qwell: I can shred loose-held paper with my current 2 cats. |
| 20:11.37 | [TK]D-Fender | Qwell: Indeed it is. |
| 20:11.43 | Qwell | not just for show? |
| 20:11.58 | waverly360 | [TK]D-Fender, Qwell: Home protection :) |
| 20:11.58 | ManxPower | what would you actually USE it for. Deal with stupid users? |
| 20:12.02 | [TK]D-Fender | Qwell: For use in iai-jutsu. |
| 20:12.24 | [TK]D-Fender | Qwell: not wieded openly in public. For use at home, dojo, etc. |
| 20:12.33 | Qwell | home depot? |
| 20:12.35 | Qwell | :p |
| 20:13.04 | Shaun2222 | ManxPower: ya guess not, just find it hard to believe that nobody uses any type of statistics software to generate call usage/performance reports on there employee's. |
| 20:13.06 | [TK]D-Fender | Qwell: now... gotta find me a water tower and..... |
| 20:18.00 | ManxPower | Shaun2222: I suspect any such software would be developed inhouse, much like CDR billing software. i.e. both are so specific to the company it is silly to create a generic software to do it. |
| 20:20.11 | [TK]D-Fender | Shaun2222: there are a number of existing packages already listed on the WIKI. Get searching.... |
| 20:20.43 | *** part/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 20:21.29 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 20:22.22 | *** join/#asterisk Yourname`` (i=Miranda@unaffiliated/yourname/x-837320) |
| 20:22.34 | Yourname`` | So, they say.. if you want something done, do it yourself. |
| 20:22.36 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:22.37 | Yourname`` | < - firm believer. |
| 20:23.43 | [TK]D-Fender | .... |
| 20:23.47 | [TK]D-Fender | too easy :D |
| 20:24.00 | GreggB | Yourname``: I notice you left out the "...right..." LOL |
| 20:24.19 | Yourname`` | lol |
| 20:24.42 | Yourname`` | Nope. Didn't need other people to believe that along with me. ;-) |
| 20:24.58 | Yourname`` | Besides, I think most people believe it secretly anyways. |
| 20:26.38 | Yourname`` | [TK]D-Fender: Very easy! :P |
| 20:36.53 | [TK]D-Fender | heading home, BBIAB |
| 20:40.57 | dlynes | !seen flauto |
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| 20:52.59 | *** part/#asterisk Tclp (n=tcalp@S01060016b680c0e0.ed.shawcable.net) |
| 20:53.18 | *** join/#asterisk Tclp (n=tcalp@S01060016b680c0e0.ed.shawcable.net) |
| 20:55.02 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:56.47 | Tclp | can anyone recommend a voip provider for Asterisk that uses SIP for Canada .. |
| 20:57.59 | *** join/#asterisk asdx (n=diego@adsl-128-192.click.com.py) |
| 20:58.11 | *** join/#asterisk blq (n=Bl@dslb-088-064-157-095.pools.arcor-ip.net) |
| 20:59.27 | ManxPower | ~itsp |
| 20:59.27 | jbot | somebody said itsp was an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 20:59.33 | ManxPower | All ITSPs suck. |
| 20:59.56 | J4k3 | my ITSP sucks less than my ILEC |
| 21:00.04 | J4k3 | and is entirely more reliable |
| 21:00.13 | J4k3 | of course, so is two tin cans and a wet string |
| 21:01.26 | ManxPower | My ILEC doesn't go down very often, but when it does they do, they are down for a while. |
| 21:01.40 | J4k3 | also, if you one of those dumbasses who thinks his cable modem/dsl is the end-all-be-all of an internet connection, you're not only confused, you're ignorant. |
| 21:01.52 | J4k3 | my ILEC is serving dialtone out of 35 year old gear |
| 21:02.02 | ManxPower | J4k3: exactly. |
| 21:02.05 | J4k3 | barely-digital remotes from the 70s, etc. |
| 21:02.26 | ManxPower | J4k3: no CLECs around? |
| 21:02.36 | J4k3 | facilities based? hell no. |
| 21:02.44 | J4k3 | hell we don't even have LNP. |
| 21:03.00 | ManxPower | J4k3: where are you? |
| 21:03.05 | J4k3 | hell, some parts of this county didn't have tone dialing until the late 90s. |
| 21:03.11 | J4k3 | Grapeland, TX, USA. |
| 21:03.40 | J4k3 | about 700 people in the city limits, about 1000 households in the exchange |
| 21:05.28 | ManxPower | J4k3: sounds about the same for the area I'm in. The CO covers a lot of area, so the number of users might be higher. |
| 21:14.23 | J4k3 | yeah, this county (5000 households) has a DMS-100/200 and a few big remotes, and our little DMS-10 and about 8 small remotes hanging off it |
| 21:14.41 | J4k3 | the biggest reliability issue is a single fiber run for all the ILEC traffic... |
| 21:18.35 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-78db5d584ebf2189) |
| 21:18.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:20.54 | *** part/#asterisk shamusmcirish (n=no@gw-wp.debtresolve.com) |
| 21:26.19 | *** join/#asterisk kkjoe (n=kkjoe@p578b42d3.dip0.t-ipconnect.de) |
| 21:26.25 | kkjoe | is there any channel for misdn support ? |
| 21:34.27 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 21:35.53 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 21:36.10 | ginvent | Ok, anyone good with sip debugging? http://www.pastebin.ca/759134 |
| 21:38.14 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 21:39.19 | J4k3 | fuckity fuck, how the hell do you get wifi to work on a wm5 device?!?! :) |
| 21:39.27 | J4k3 | damn youse, microsofts! |
| 21:41.37 | Qwell | J4k3: buy me one, and I'll tell you. |
| 21:41.51 | Qwell | oh, vm == windows mobile... nevermind |
| 21:41.59 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:42.09 | J4k3 | yeah, this thing should run linux too |
| 21:42.33 | Qwell | buy me an iPhone, and I'll help you get it going. |
| 21:42.34 | J4k3 | xscale, keyboard, ti wlan, standard cdma/evdo radio attached... |
| 21:42.36 | J4k3 | ick |
| 21:42.42 | J4k3 | iphones are SEXY, yet total shit |
| 21:42.51 | Qwell | and wm isn't? |
| 21:42.53 | J4k3 | I have two friends with iphones, and they're up to 3 returns in total. |
| 21:43.10 | J4k3 | I love the apple software, its the hardware thats the problem in their case |
| 21:43.22 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 21:43.48 | Voicemeup | can i remap the # so it doesnt temrinate dtmf |
| 21:45.12 | blitzrage | Voicemeup: can you be more clear? |
| 21:45.21 | Voicemeup | i want # not to temrinate dtmf input |
| 21:45.24 | blitzrage | that might be a function of your SIP phone |
| 21:45.30 | blitzrage | using what dialplan application? |
| 21:46.21 | Voicemeup | hmm gi |
| 21:46.32 | Voicemeup | <PROTECTED> |
| 21:49.37 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:50.41 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-126-184.dsl.tulsok.swbell.net) |
| 21:50.47 | Voicemeup | hey brian pm mme |
| 21:52.48 | Katty | blitzrage: blitzrage blitzrage blitzrage |
| 21:56.44 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 21:57.49 | devel | greetings all, just starting to mess with t.38 in asterisk 1.4, is there any way to tell during a call that it is indeed doing passthrough? |
| 21:57.58 | blitzrage | Katty: Katty Katty Katty |
| 21:58.14 | blitzrage | devel: if it works... then it has to be passing through :) |
| 21:58.17 | Katty | blitzrage: herroherro. i accidnetally called toronottototototo today |
| 21:58.23 | devel | ah blitzrage, would that it were so. |
| 21:58.37 | blitzrage | Katty: oh noes! |
| 21:58.39 | blitzrage | why for? |
| 21:58.43 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:58.43 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 21:58.54 | Katty | well i was attempting to call twisted's company |
| 21:58.59 | Katty | but i transposed the last two digits. |
| 22:00.05 | blitzrage | hehehe :) |
| 22:00.17 | [TK]D-Fender | wouldn't that be the first TWO digits? |
| 22:00.17 | Katty | oh. it's 5 |
| 22:00.18 | Qwell | 265? |
| 22:00.29 | Qwell | 625? |
| 22:00.49 | blitzrage | toronto is 416 and 647 |
| 22:01.06 | Qwell | transposed, and subtracted one |
| 22:13.11 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 22:16.18 | J4k3 | any suggestions for a wm5 voip client? |
| 22:18.18 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 22:23.25 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 22:35.13 | russellb_ | JunK-Y: should be fixed now |
| 22:35.19 | russellb_ | wrong channel! |
| 22:37.05 | alrs | ~book |
| 22:37.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 22:40.09 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
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| 22:42.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:42.52 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 22:43.05 | MrTelephone | any reason why sip inuse doesn't decrease even with rtptimeout set? |
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| 23:14.48 | Qwell | spare? wasn't it forget? |
| 23:14.58 | Qwell | I'm pretty sure it was |
| 23:15.41 | MrTelephone | qwell |
| 23:15.52 | Qwell | I'm not here. |
| 23:15.56 | MrTelephone | what is supposed to happen when no call-limit is set |
| 23:15.58 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 23:15.59 | MrTelephone | default of 2? |
| 23:16.06 | Qwell | no, I don't think so |
| 23:16.08 | Qwell | but bbl |
| 23:16.16 | MrTelephone | what decreases call-limit |
| 23:17.24 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 23:18.09 | *** join/#asterisk IPetrov (i=IPetrov@ppp85-140-235-104.pppoe.mtu-net.ru) |
| 23:24.15 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 23:26.10 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 23:26.39 | dlynes_laptop | I haven't changed my callprogress from no to yes for a while...I'm just wondering if it's still flaky on yes? |
| 23:26.52 | dlynes_laptop | I'm talking specifically for North American analog lines |
| 23:28.41 | jameswf | << finds it amusing fonality doesn't monitor there own room.... |
| 23:29.03 | [TK]D-Fender | jameswf, which? |
| 23:29.11 | jameswf | trixbox.. |
| 23:29.14 | jameswf | # |
| 23:29.25 | MrTelephone | fender bender |
| 23:29.50 | [TK]D-Fender | jameswf, do THEY own it, or is it simply named after one of their newly acquired products and run by their FANBOYS? :) |
| 23:30.31 | jameswf | well they are the ops... kerry has not been on since TB pro, think he is scared |
| 23:32.01 | jameswf | kerry made a big stink on ward's site about trixbox not being closed, yet they remove the kernel development package from the repo... |
| 23:32.07 | [TK]D-Fender | jameswf, well the question is that he was an "independant" promoter of all things "cheap shit chump VoIP" and going all "suit"-like maybe has him watching out not to try to look too smart :) |
| 23:33.31 | jameswf | i would never accuse him of that.... those who can do those who cant teach those who cant teach writs small poorly orginized books.... |
| 23:33.58 | jameswf | s/writs/write/ |
| 23:34.36 | dlynes_laptop | I guess nobody knows about the callprogress zaptel option, then? |
| 23:34.57 | [TK]D-Fender | dlynes_laptop, Oh you mean "hanguponcallsrandomly=yes"? |
| 23:35.04 | [TK]D-Fender | :p |
| 23:35.09 | dlynes_laptop | [TK]D-Fender: ah yeah...so it's still buggy as hell? |
| 23:35.20 | jameswf | that should be an option in 1.6 |
| 23:35.33 | dlynes_laptop | jameswf: hanguponcallsrandomly=yes? |
| 23:35.39 | jameswf | :) |
| 23:35.51 | [TK]D-Fender | jameswf, failure is NOT and option.... it comes BUNDLED with the software :p |
| 23:36.23 | jameswf | their was an old worm called "cookie monster" I am going to adapt it to asterisk... |
| 23:36.26 | GreggB | totally...I wonder if that option can be tied to a db lookup so it only applies to certain callers |
| 23:36.44 | dlynes_laptop | GreggB: it's called app_lookupblacklist |
| 23:36.56 | jameswf | GreggB: so you think it should be a dialplan app |
| 23:37.17 | [TK]D-Fender | GreggB, easy enough to implement |
| 23:37.34 | jameswf | no blacklist, make it play static, then tunnel noises get choppy and finaly hang up |
| 23:37.35 | GreggB | dlynes_laptop: can you do that after the call is answered...like 5 minutes into a discussion? |
| 23:37.47 | dlynes_laptop | GreggB: no...that's when the call is answered |
| 23:38.02 | dlynes_laptop | GreggB: or for that matter any time into the call, where you have dialplan control |
| 23:38.29 | jameswf | press 11 to simulate cingular call :) |
| 23:38.29 | GreggB | jameswf: that sounds like $evilcompany whenever I call to complain about service...lately that's been our ILEC |
| 23:38.40 | dlynes_laptop | GreggB: so if you want to use the ,T or ,t option to dial, I suppose you could activate it from your handset |
| 23:40.09 | jameswf | s/$evilcompany/fonality/ |
| 23:40.10 | jameswf | ? |
| 23:40.16 | jameswf | :)) |
| 23:40.24 | GreggB | s/$evilcompany/Integra/ |
| 23:40.57 | dlynes_laptop | [TK]D-Fender: how about the immediate=yes option? |
| 23:41.02 | *** join/#asterisk l2trace99 (n=asd@fl-67-76-209-28.sta.embarqhsd.net) |
| 23:41.05 | dlynes_laptop | [TK]D-Fender: that's hangup on caller randomly, too? |
| 23:41.17 | jameswf | i told someone today they needed to adjust their asterisk configs, not use the gui... they almost cried |
| 23:41.19 | GreggB | ugh... the relationship was great for the first six months, and then...well...she ^D^D^D they started doing stupid things. |
| 23:43.20 | GreggB | at least the fingers clearly pointed at the ILEC, and our VoIP migration was moved higher on the priority list :D ...asterisk to the rescue in this case |
| 23:45.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:45.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:53.10 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 23:55.49 | l2trace99 | anything wrong with having an extension called QUEUEDIALER ? |
| 23:56.02 | [TK]D-Fender | l2trace99, nope |
| 23:56.14 | l2trace99 | cook |
| 23:56.16 | l2trace99 | cool |
| 23:59.42 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.252) |