| 00:00.01 | SpaceBass | and know a mini would smoke my 350mhz PII |
| 00:00.05 | Corydon-w | kink0: there's absolutely no reason to be using g729 for packets which are going across a LAN |
| 00:00.10 | SpaceBass | :) |
| 00:00.11 | kink0 | Corydon-w, because calls comming from the outside world to this asterisk, so for bandwidth reasons. |
| 00:00.30 | Corydon-w | kink0: ah |
| 00:00.55 | kink0 | Corydon-w... no sorry for mistake, Asterisk box is in the same room as GSM gateway, but calls are originated from ouside world and routed to this asterisk box. |
| 00:01.15 | Corydon-w | kink0: for calls coming into Asterisk via g729 and going out g729, there's no issue, because Asterisk doesn't have to do any transcoding |
| 00:01.51 | Corydon-w | Only calls terminating to Asterisk will need to have the g729 codec |
| 00:01.55 | kink0 | Corydon-w but codification/decodification will be done in Asterisk, since GSM gateway has not voIP card, just a PRI interface. |
| 00:02.04 | Corydon-w | Ah... |
| 00:02.14 | kink0 | so I planning to use the PRI from digium to connect to this solution |
| 00:02.46 | Corydon-w | You might be able to use 60 concurrent channels, but you should talk to Digium. I think they've done some testing in this area |
| 00:02.47 | kink0 | in this scenary, all codifing jobs , as well as routing jobs, will be done by the Asterisk box |
| 00:03.22 | Corydon-w | You'll need to talk to Digium anyway, to purchase the licenses |
| 00:04.05 | kink0 | yes, I got two licences to try between two Asterisk boxes, install it and check, and runs fine. |
| 00:04.48 | kink0 | well, those two licenses was just to evaluate the codec, but experimental scenarys some time is not near as production scenary |
| 00:06.23 | kink0 | about GSM gateways ... in your opinions, what is your prefered ? 2N, Teles, Valiant ... others ? |
| 00:06.27 | *** join/#asterisk ST-3 (i=ser@dipsy.tch.org) |
| 00:06.28 | rking | if i want to stress-test a MeetMe conf, is there an easy way to make some null client that sends data from a wave file loop and otherwise doesn't freak out that it isn't attached to actual audio devices? |
| 00:06.34 | rking | oh, IAX or SIP are ok |
| 00:07.35 | marcus2 | so whats a good iax softphone for windows? |
| 00:08.27 | tzafrir_laptop | iaxcomm? not exactly good... |
| 00:09.18 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 00:11.25 | SpaceBass | iaxcomm |
| 00:11.32 | SpaceBass | best around.... in that it works |
| 00:11.58 | marcus2 | so all of the others dont? |
| 00:13.18 | marcus2 | how about idefisk and diax? |
| 00:13.20 | rking | marcus2: i just broke down and added SIP ability because of having too many coworkers that couldn't find iax clients |
| 00:13.54 | marcus2 | are your coworkers making sip connections over the internet? |
| 00:14.01 | robl^ | some of them only work if you sacrifice a red & white striped bucket of chicken (original recipie) to some sort of diety |
| 00:15.46 | kink0 | what SIP softphone for win is the recomended then ? |
| 00:16.27 | SpaceBass | xlite ? |
| 00:16.29 | justinu | eyebeam if you can afford it... |
| 00:16.33 | SpaceBass | i mean... its ok... |
| 00:16.35 | rking | marcus2: were you talking to me? |
| 00:16.38 | marcus2 | rking; yeah |
| 00:16.51 | rking | marcus2: the answer is: yes, they will - now they have to shut up and stop using skype. =) |
| 00:17.02 | SpaceBass | really they all work the same... if they make calls, have a hold button, etc.. they are the same |
| 00:17.15 | marcus2 | rking; do you actually have people using sip to asterisk over the internet, already? |
| 00:17.22 | marcus2 | i find that it sucks ass, since asterisk has no sip jitter buffer |
| 00:17.44 | mog_work | go write one ^_^ |
| 00:17.53 | rking | marcus2: ahh, interesting - no, so far it was just me using my sound card 0 on SIP to connect with my sound card 1 on IAX |
| 00:17.55 | robl^ | the difference between most of the soft phones are how easy they are to set up for a newbie.. and sometimes.. a few featues differ.. but MOST work... some are easier than others |
| 00:17.58 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.26) |
| 00:18.24 | denon | robl^: I dunno, I thought the 7960 was easier to set up than xten's soft phone |
| 00:18.29 | justinu | eyebeam actually supports SIP Refer transfers |
| 00:18.46 | justinu | and some of the SIMPLE stuff |
| 00:19.00 | kink0 | marcus2, that means to use SIP over the internet will be not a good idea due to lack of jittering ? |
| 00:19.18 | robl^ | denon: the 7960s are fairly easy to set up.. and I have 3 here I am trying to get rid of.. :) I am migrating this place to Snow. :) |
| 00:19.23 | marcus2 | kink0; i've had pretty poor luck with it |
| 00:19.32 | marcus2 | especially when your asterisk server is terminating calls to a zap device |
| 00:19.40 | marcus2 | then its basically unusable with remote sip clients |
| 00:19.48 | kink0 | marcus2, yes, that is what I pretend, well to a GSM |
| 00:19.57 | kink0 | marcus2, did you use g711 or g729 ? |
| 00:20.04 | marcus2 | 711 |
| 00:20.08 | marcus2 | i dont think the codec matters much tho |
| 00:20.30 | justinu | 711 needs like perfect conditions to work |
| 00:20.31 | rking | a weird thing: every time, after quitting the asterisk shell, my normal shell (zsh), no longer accepts ^D to log out. |
| 00:20.39 | justinu | so codec does matter |
| 00:20.49 | marcus2 | ah |
| 00:20.55 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 00:20.57 | kink0 | ahhh, well bandwidth is important over the internet, I decide to use g729, even that costs a little. |
| 00:21.18 | justinu | gsm is good over the net, and eyebeam has gsm |
| 00:21.22 | marcus2 | i think you still need a jitter buffer to get any sort of decent quality between a sip channel and a zap channel |
| 00:21.32 | justinu | g729 sounds better than gsm but costs $$$ |
| 00:21.34 | robl^ | g729 has GOOD sound quality for its compression... but it also really taxes the CPU |
| 00:21.42 | kink0 | I think g729 would be fine, and better than 711 ( even if not jittering for SIP) |
| 00:21.44 | justinu | marcus2: try it out, you might be surprised. |
| 00:21.54 | justinu | i'm using gsm over wifi internet and it sounds good |
| 00:21.58 | rking | could someone do me a huge favor and dial my meetme to test out the latency and crappiness? (either SIP or IAX would be helpful) |
| 00:22.49 | marcus2 | justin; what are your calls being terminated to? |
| 00:22.59 | rking | connecting to localhost sounds good so far. |
| 00:23.16 | justinu | marcus: sip right back out to a SIP termination provider |
| 00:23.28 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 00:23.37 | kink0 | robl^, are you ussing g729 ? how much bogomips per channel ? :) |
| 00:23.43 | marcus2 | oh |
| 00:23.52 | marcus2 | so your sip termination provider most likely has a jitter buffer |
| 00:23.57 | marcus2 | and they're delaing with the problem |
| 00:24.01 | justinu | could be |
| 00:24.05 | marcus2 | seriously, try terminating directly to a zap channel and you'll see what i mean |
| 00:24.09 | justinu | i think they run asterisk tho :) |
| 00:24.27 | rking | there isn't any "phone home" service i can make dial me and play a sound file, is there? |
| 00:24.34 | marcus2 | it worked "well enough" to do sip -> asterisk -> iax -> pstn provider... but when i started terminating calls directly to a pri from the asterisk server, it went to shit |
| 00:24.50 | justinu | weird. |
| 00:24.57 | robl^ | kink0: use g729 on a SINGLE remote phone with up to 6 channels (Cisco 7960)... I never tested it closely, but, I notice that the CPU spikes big time with that phone is used. |
| 00:25.28 | robl^ | and I have a 2ghz Athalon CPU |
| 00:26.31 | robl^ | ~seen atacomm |
| 00:26.36 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 183d 21h 21m 39s ago, saying: 'lol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol...'. |
| 00:28.35 | RoyK | ~seen zoa |
| 00:28.37 | jbot | zoa <n=zoa@cable-83.217.145.105.coditel.net> was last seen on IRC in channel #asterisk, 1d 13h 49m 21s ago, saying: 'yoooo'. |
| 00:29.23 | RoyK | ~lart himself |
| 00:30.00 | perlmonkee | hey now |
| 00:30.08 | perlmonkee | I have at least a dozen of those in my garage. |
| 00:30.44 | robl^ | I have a full height 8" WInchester drive.. 5MB :) |
| 00:31.00 | justinu | they're fun to shoot with high powered rifles. |
| 00:31.26 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 00:31.59 | justinu | break stuff |
| 00:32.25 | justinu | chat on irc at work |
| 00:32.30 | RoyK | the magnets in those newer drives are really cool |
| 00:32.33 | robl^ | RoyK: Americans all dress in blue vests and offer people shopping carts/trolleys at the local WalMart :) |
| 00:32.42 | MRH2 | upgrading some polycomms - ne1 know should i specify SDP in dialog body or not? |
| 00:32.59 | justinu | MRH2: wtf does SDP have to do with upgrading apolycom? |
| 00:33.14 | MRH2 | new setting in sip 1.6.3 |
| 00:33.17 | robl^ | DSP maybe? |
| 00:33.26 | justinu | MRH2: hmm, haven't seen 1.6.3 yet |
| 00:33.39 | mog_work | hey lets not hate on america |
| 00:33.42 | mog_work | be original |
| 00:33.51 | mog_work | lets hate on i dont know |
| 00:33.54 | mog_work | nepal |
| 00:34.00 | mog_work | theyd never see it comming |
| 00:34.04 | justinu | the mountain kingdom of lesotho |
| 00:34.06 | robl^ | no! lets hate on... SCO! :) |
| 00:37.36 | *** join/#asterisk steff (n=steff@80.125.254.220) |
| 00:38.44 | *** join/#asterisk jahani (n=k@adsl-157-38-192-81.adsl.iam.net.ma) |
| 00:40.20 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
| 00:41.30 | p1tst0p | would there be a way to do WebChat with Asterisk?... Ie, "Click here to Chat to Support" and it rings a Queue on asterisk... somehow.. lol |
| 00:42.07 | file[laptop] | sure/ |
| 00:42.11 | file[laptop] | anything is possible. |
| 00:42.15 | tzafrir_laptop | p1tst0p, basically a problem of client |
| 00:42.29 | perlmonkee | Just write yourself a java client. |
| 00:42.38 | p1tst0p | yeh a java client. |
| 00:42.40 | p1tst0p | hmm |
| 00:42.50 | perlmonkee | and then make it opensource so we can all use it. Also make it not suck, please. |
| 00:43.06 | p1tst0p | hah |
| 00:43.10 | tzafrir_laptop | e.g: you have moziax for mozilla (if you trust the source) and some activex components for explorer (if you trust the author). But frankly nothing for "a browser" |
| 00:44.02 | tzafrir_laptop | A java applet isn't something that "just works" on browsers, sadly |
| 00:44.13 | p1tst0p | does moziax work ? |
| 00:44.15 | tzafrir_laptop | Sun managed to get that buried |
| 00:44.32 | tzafrir_laptop | it is rummored to work. Never tried it |
| 00:45.47 | litage | if an extension talks to its * box using sip and g729, and the * box talks to its provider using h323 and g729, will the sip usage cause any extra processing, load, conversion, etc on the * box? |
| 00:45.49 | p1tst0p | even then though, your tying your customer to mozilla.. Hmm |
| 00:47.25 | tzafrir_laptop | I believe that asterisk can transfer the sip RTL stream as-is from SIP to h323 without transcoding (of both ends agree on codecs), but I'm not sure |
| 00:49.00 | justinu | yeah, it's just proxying rtp thru the rtp.c module |
| 00:50.06 | MRH2 | p1st0p easier to do a call me button where u enter the phone number on the webpage and asterisk calls you |
| 00:50.58 | *** join/#asterisk ResidenteE (i=Angel200@63.74.26.207) |
| 00:51.38 | *** part/#asterisk ResidenteE (i=Angel200@63.74.26.207) |
| 00:51.41 | *** join/#asterisk ResidenteE (i=Angel200@63.74.26.207) |
| 00:52.47 | ResidenteE | Hi, anyone try a Dell PowerEdge 2800 with TE410P ? |
| 00:53.29 | *** join/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 00:55.29 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 00:56.19 | perlmonkee | MRH2: I believe p1st0p wanted to keep it all VOIP |
| 00:57.49 | MRH2 | ah ok, i wouldn't trust customer endpoints due to quality concerns though. |
| 00:58.41 | MRH2 | "trying to download this movie while talking to support" type issues |
| 01:00.02 | docelm0 | ARGH! |
| 01:00.12 | docelm0 | I am about to pull my fricken HAIR OUT! |
| 01:02.45 | perlmonkee | get in line! |
| 01:04.26 | SpaceBass | ahhh welcome to the world of tech |
| 01:09.03 | marcus2 | hrm |
| 01:09.18 | marcus2 | asterisk is ignoring the callerid= info that i'm specifying for my peers in iax.conf |
| 01:10.04 | p1tst0p | MRH2, yeh i thought about web callback too, but i just thought a webchat button would be a very nice touch. |
| 01:10.20 | p1tst0p | moziax is pants btw.. doesnt save my settings ! |
| 01:10.54 | syle | whats a quick way to get ip address |
| 01:11.11 | tzafrir_laptop | echo 127.0.0.1 |
| 01:11.15 | justinu | ifconfig -a |
| 01:11.19 | syle | ast_gethostbyname? |
| 01:12.13 | *** join/#asterisk techie (i=gus@antibala.com) |
| 01:18.00 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 01:18.55 | *** join/#asterisk callsign (n=callsign@c-24-13-77-145.hsd1.il.comcast.net) |
| 01:19.14 | callsign | how do i have asterisk dial out a real phone number when someone calls my extension and connect the callers? |
| 01:19.23 | callsign | using sip for inbound and outbound |
| 01:19.56 | *** join/#asterisk bweschke_ (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 01:19.56 | SpaceBass | so someone dials 1234 and then it calls 555-555-5555 ? |
| 01:20.13 | callsign | yes |
| 01:20.23 | callsign | and connects the caller to person at 555-555-5555 |
| 01:20.28 | bweschke_ | anyone else a broadvoice customer and suddenly having problems calling in / out ? |
| 01:20.44 | SpaceBass | leme check |
| 01:20.47 | SpaceBass | callsign: one sec |
| 01:21.09 | justinu | i just tried it |
| 01:21.12 | justinu | had one call fail |
| 01:21.39 | bweschke_ | I'm getting a "the device you are using is not registered to place calls on the network. please contact your administrator for assistance." |
| 01:22.02 | bweschke_ | calls to support sends me into an IVR that isn't recognizing DTMF digits from a cell phone to get transferred into a supoprt queue |
| 01:22.03 | justinu | i've had that when your registration times out |
| 01:22.15 | *** join/#asterisk IcePenguin (n=G4Master@207-118-163-29.dyn.centurytel.net) |
| 01:22.16 | bweschke_ | strange |
| 01:22.20 | IcePenguin | hi guys |
| 01:22.26 | bweschke_ | i'll restart * and see what happens |
| 01:22.35 | justinu | sip reload should make it reregister |
| 01:22.47 | IcePenguin | IronHelix? |
| 01:23.11 | justinu | callsign: you need a PSTN termination provider |
| 01:23.16 | callsign | i have justinu |
| 01:23.28 | callsign | i just dont know the asterisk config for it |
| 01:23.29 | bweschke_ | yup. that's exactly what it was |
| 01:23.31 | bweschke_ | that's wierd |
| 01:23.39 | callsign | i can dial out using phones ive connected to my asterisk box over sip |
| 01:23.42 | *** join/#asterisk Junbug (n=Darline@69.182.40.197) |
| 01:23.43 | IcePenguin | Hey guys, I have a quick question |
| 01:23.49 | justinu | callsign: then do something like 1234,1,Dial(SIP/15551212@sipprovider.com); |
| 01:23.52 | bweschke_ | it's been fine for months on end .. this is the first time I've seen that happen |
| 01:24.01 | Junbug | IcePenguin: just ask |
| 01:24.03 | justinu | bweschke_: happened to me once as well. |
| 01:24.13 | callsign | justinu how do i connect the incomming callers to that number its dialling? |
| 01:24.29 | IcePenguin | A client of mine is looking at a * system for her business. Really simple. 3 PBX lines tied to 2 telco lines |
| 01:24.32 | SpaceBass | back |
| 01:24.37 | SpaceBass | Broadvoice seems to be working fine |
| 01:24.55 | justinu | callsign: explain more what you want to do... |
| 01:24.57 | bweschke_ | yep. apparently someone canceled my registration over there |
| 01:25.03 | bweschke_ | re-registering worked fine |
| 01:25.10 | SpaceBass | callsign: did you get your exten working? |
| 01:25.13 | callsign | ok very simple justinu |
| 01:25.26 | IcePenguin | I'll have a * server with 2 FXO ins, and the ethernet out goes to a switch. All phones on the switch will be able to connect to either line, correct? |
| 01:25.39 | bweschke_ | icepenguin: yes, correct. |
| 01:25.40 | justinu | yes |
| 01:25.51 | SpaceBass | IcePenguin: yeah |
| 01:25.52 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 01:26.00 | callsign | i want someone to call into the number handled by asterisk, asterisk picks up the number and calls 555-555-5555 waits for the person at that number to answer and connects the two parties |
| 01:26.09 | IcePenguin | okay. will i need 2 RJ-45 ins on the phone, or can both lines be used from a single connection? |
| 01:26.13 | justinu | callsign: ok. |
| 01:26.24 | justinu | then whatever number they dial into asterisk, put an exten line for it. |
| 01:26.45 | justinu | exten => 18185551212,1,Dial(SIP/12125551212@sipprovider.com); |
| 01:26.46 | callsign | justinu how do i do that? |
| 01:26.50 | SpaceBass | IcePenguin: need an FXo port for each line... something like the x100p clone or tdm400 (or what ever it is called) |
| 01:26.58 | IcePenguin | yep |
| 01:27.01 | IcePenguin | i got that |
| 01:27.04 | bweschke_ | Icep: the phone is sent calls from whatever port via the dial plan |
| 01:27.11 | SpaceBass | exten => 1234,dial(technology/number) |
| 01:27.16 | SpaceBass | oops... what justinu said |
| 01:27.19 | bweschke_ | icep: you rely on * to make the appropriate switching from whatever line |
| 01:27.22 | Junbug | IcePenguin: and one rj-45 to switch |
| 01:27.27 | IcePenguin | okay- now, can the phones switch from line one to line 2 over a single cable |
| 01:27.35 | bweschke_ | correct |
| 01:27.37 | IcePenguin | Okay, thanks Junbug |
| 01:27.44 | IcePenguin | you guys rule |
| 01:27.56 | SpaceBass | wouldn't the fxo have to support 2 lines on the same cable? |
| 01:28.04 | docelm0 | How do I debug DTMF on g711u calls? |
| 01:28.11 | SpaceBass | and don't most break out each pair (line) onto a single port? |
| 01:28.12 | justinu | rtp debug ;) |
| 01:28.19 | bweschke_ | docelm0: what kind of debugging ? |
| 01:28.19 | Junbug | IcePenguin: 200 voip phones -> switch, one rj-45 to asterisk |
| 01:28.22 | callsign | justinu i did what you said and its not working |
| 01:28.26 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 01:28.30 | bweschke_ | docelm0: registering the rfc2833 events? |
| 01:28.34 | IcePenguin | sweet |
| 01:28.45 | docelm0 | basically I cant send dtmf to one of my carriers.. they are saying everything is good to go.. I am sending g711u and inband DTMF |
| 01:28.48 | justinu | docelm0: i prefer ethereal for that stuff. |
| 01:28.48 | IcePenguin | yeah good question |
| 01:29.08 | IcePenguin | thanks guys |
| 01:29.10 | docelm0 | Can I debug this inside of asterisk? |
| 01:29.15 | justinu | yeah, rtp debug |
| 01:29.21 | justinu | it prints the rfc2833 events |
| 01:29.30 | docelm0 | is inband == rfc2833? |
| 01:29.32 | bweschke_ | rtp debug will print the rfc events, but not inband dtmf |
| 01:29.33 | justinu | but if you're using inband, asterisk won't show them |
| 01:29.34 | bweschke_ | no |
| 01:29.37 | tzanger | no inband is not RFC2833 |
| 01:29.42 | docelm0 | I didnt think so |
| 01:29.44 | perlmonkee | I can't find a way to register with them without downloading their client. |
| 01:29.58 | bweschke_ | docelm0: Monitor the call... |
| 01:30.06 | bweschke_ | docelm0: then listen to the call.. |
| 01:30.17 | docelm0 | I am using g729 rfc2833 -> asterisk -> g711u inband |
| 01:30.18 | bweschke_ | you should be able to hear the dtmf I'd think |
| 01:30.29 | bweschke_ | docelm0: it should work |
| 01:30.31 | docelm0 | thats the problem.. I am getting 1 way DTMF |
| 01:30.36 | justinu | docelm0: ok, that should work... |
| 01:30.45 | bweschke_ | doclem0: did you patch that bug in from 1.2? 5780 I think? |
| 01:30.52 | justinu | er wait... do you have the g729 license? |
| 01:30.53 | docelm0 | yes |
| 01:31.00 | docelm0 | yes |
| 01:31.06 | justinu | ok... sounds like you're setup right |
| 01:31.18 | justinu | you sure they want dtmf inband on g711? |
| 01:31.21 | bweschke_ | i'd try and monitor the outbound call |
| 01:31.34 | bweschke_ | if you hear the dtmf in the wav, then you know you're sending it |
| 01:31.35 | docelm0 | thats the problem I am sending from 5 asterisk boxes |
| 01:31.51 | callsign | ok nevermind got it |
| 01:31.59 | callsign | how do i do this now.... |
| 01:32.10 | docelm0 | Well when the PSTN sends DTMF it works.. when the voip side sends dtmf to pstn it doesnt work |
| 01:32.15 | docelm0 | its kinda like 1 way audio |
| 01:32.16 | callsign | how do i have someone call in an dial a number based on wether they pused 1 , 2 or 3 |
| 01:32.35 | justinu | docelm0: odd stuff. |
| 01:33.11 | justinu | docelm0: sounds like they migght be configured wrong |
| 01:33.20 | bweschke_ | docelm0: let's try this... |
| 01:33.25 | bweschke_ | i'll give you my cell #... |
| 01:33.29 | bweschke_ | call that and hit some tones |
| 01:33.35 | bweschke_ | either I hear them or I don't.... |
| 01:33.40 | docelm0 | I know you will not |
| 01:33.46 | docelm0 | thats been established.. |
| 01:33.47 | bweschke_ | k |
| 01:33.49 | bweschke_ | umm... |
| 01:33.51 | justinu | docelm0: can you past the sip debug f the invite/200OK exchange with your PSTN gateway? |
| 01:33.53 | Junbug | man i just read the oreilly asterisk book, real good stuff ... finally get all the issues i have been having with dialplan logic |
| 01:33.56 | docelm0 | you could send the tones to me but not me to you |
| 01:34.36 | docelm0 | ya.. I get the ACK and everything is good to go |
| 01:34.41 | docelm0 | thats whats screwey |
| 01:34.43 | justinu | yeah, but lets see it |
| 01:34.51 | docelm0 | the ACK? |
| 01:35.04 | justinu | with sip, you can have two different codecs in each direction |
| 01:35.18 | justinu | so maybe they're running inband to you, but running rfc2833 from you or something... |
| 01:35.28 | *** part/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 01:35.41 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 01:36.54 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
| 01:37.07 | docelm0 | 1 sec.. let me debug.. brb |
| 01:37.39 | bweschke_ | docelm0: here's the other thing... what's the minimum amount of time they require a tone to sound before they recognize it as a tone? |
| 01:38.35 | swm_ | Need help doing something here... |
| 01:38.43 | Junbug | callsign: what do ya need again? |
| 01:38.54 | tzanger | bweschke_: I think the typical time is 100ms minimum |
| 01:39.06 | tzanger | but 250ms is pretty much a given standard tone length from my experience |
| 01:39.22 | callsign | how do i have someone call in and dial an extension and do some actions based on that extension |
| 01:39.36 | swm_ | Anyone know how to do a dialplan that connects two asterisk boxes together... I want to use my X100P on 1 Box and have the other box do the work and use the X100P on the fist box? |
| 01:39.49 | tzanger | callsign: well you just configure it in extensions.conf |
| 01:39.56 | tzanger | callsign: (ask a vague question, get a vague response) |
| 01:40.03 | guzzler | Hi all! I'm trying to use the "Async" option on the "Originate" action of the Asterisk Manager API but it's not working. I'm using "Async: yes". Is this correct? Does anyone have a clue on this? Thanks! |
| 01:40.13 | bweschke_ | what is the default that the vdsp generates on an inband tone? |
| 01:40.14 | justinu | how about, read the diaplan chapter in the oreilly book? |
| 01:40.15 | justinu | ~dos |
| 01:40.17 | jbot | somebody said dos was (Disk Operating System) This OS is what got it all started for PCs. Denial of Service... |
| 01:40.17 | Junbug | callsign: depending on the application u set the exten => 100,1,APP to do it |
| 01:40.19 | justinu | ~docs |
| 01:40.21 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 01:40.34 | tzanger | bweschke_: you're asking questions that require you to look in the source, this stuff isn't well documented |
| 01:40.43 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 01:40.43 | bweschke_ | checking now... |
| 01:40.50 | bweschke_ | wasn't sure whether or not someone knew |
| 01:41.07 | justinu | dsp.c |
| 01:41.11 | morale | anyone using the "comedian voicemail"? im having trouble because everytime i call it, it asks me to re-register my password and name.. |
| 01:41.17 | tzanger | no it won't be in dsp.c I bet |
| 01:41.20 | justinu | no? |
| 01:41.24 | tzanger | tone generation is handled by the destination channel |
| 01:41.29 | tzanger | zaptel does it in the zaptel driver, for instance |
| 01:41.30 | justinu | i thought that was where the DTMF recognizer is |
| 01:41.40 | justinu | oh, generator |
| 01:41.42 | tzanger | that's the channel-agnostic DTMF recognizer sure |
| 01:41.52 | tzanger | but there's one in zaptel too :-) it can do both |
| 01:41.56 | justinu | ah |
| 01:42.07 | justinu | just sip |
| 01:42.19 | tzanger | chan_sip uses the generic generator code I think |
| 01:42.38 | tzanger | you'd have to dig through the SIP receiver code to see what it does when it receives a DTMF tone in SDP (I think that's the protocol name) |
| 01:42.41 | justinu | yeah, i saw a bunch of warnings when I was telling me I couldn't send inband DTMF over g729: :) |
| 01:43.01 | justinu | actually, it comes in the RTP stream for rfc2833 |
| 01:43.18 | tzanger | you can't |
| 01:43.25 | justinu | actually, you can |
| 01:43.25 | tzanger | you can't send DTMF over any compressed voice codec |
| 01:43.29 | tzanger | not reliably |
| 01:43.31 | tzanger | that's the trick |
| 01:43.35 | justinu | no, that's true. |
| 01:43.38 | justinu | but it does work sorta |
| 01:43.40 | justinu | somtimes |
| 01:43.44 | justinu | when the planets are aligned. |
| 01:43.45 | tzanger | all the compressed voic codecs do poorly with continuous tones |
| 01:45.45 | justinu | there's another mode for DTMF that uses SIP INFO messages too |
| 01:46.19 | javar | hi |
| 01:46.39 | javar | can somebody help me with Realtime? |
| 01:46.58 | litage | if an extension talks to its * box using sip and g729, and the * box talks to its provider using h323 and g729, will the sip usage cause any extra processing, load, conversion, etc on the * box? |
| 01:47.13 | justinu | shouldn't |
| 01:47.25 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 01:47.36 | *** join/#asterisk Katty (n=angela@68.112.15.110) |
| 01:48.07 | IcePenguin | Okay guys... one more question, and you might kill me for it |
| 01:48.21 | callsign | if i configured an sip user in sip.conf and have a softphone connecting through it fine how do i know what that softphones extension is ? |
| 01:48.24 | IcePenguin | Is there a nice, free, GUI for *? |
| 01:48.53 | swm_ | Anyone know how to do a dialplan that connects two asterisk boxes together... I want to use my X100P on 1 Box and have the other box do the work and use the X100P on the fist box? |
| 01:48.55 | IcePenguin | I like to see my work, etc... I'm not too abstract |
| 01:49.10 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 01:49.12 | justinu | try AMP |
| 01:49.13 | litage | justinu: how can one check? |
| 01:49.16 | justinu | i never tried it tho |
| 01:49.38 | justinu | litage: um....... good question... |
| 01:50.31 | IcePenguin | justinu: sorry, I tried a google search and all I get is *@Home. Do you have a link? |
| 01:50.38 | swm_ | HOW? X100P < ASTERISK < SIP > ASTERISK > HARDPHONE |
| 01:50.51 | bweschke_ | http://www.coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 01:51.00 | Junbug | IcePenguin: search on voip-info.org |
| 01:51.11 | justinu | http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 01:51.16 | justinu | whoops |
| 01:51.22 | IcePenguin | sweet |
| 01:51.26 | IcePenguin | thanks a ton guys |
| 01:51.49 | *** join/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
| 01:52.01 | swm_ | Help me |
| 01:52.10 | *** part/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
| 01:52.30 | IcePenguin | is CentOS 4.2 the latest release? |
| 01:52.35 | bweschke_ | swm: you're looking to IAX trunk betrween the two boxes |
| 01:52.45 | swm_ | That or SIP I would prefer sip |
| 01:52.54 | justinu | centos 4.2 is the latest stable |
| 01:52.57 | bweschke_ | fair enough |
| 01:53.15 | *** join/#asterisk klicek (n=tk@nat.bratniak.ds.pw.edu.pl) |
| 01:53.16 | IcePenguin | thanks |
| 01:53.18 | klicek | hello |
| 01:53.21 | marcus2 | idefisk would be a lot cooler if it actually displayed callerid info in the main window |
| 01:53.23 | bweschke_ | do both boxes (or at least one) have public IPs so you don't run into the standard SIP/NAT fun? |
| 01:53.35 | swm_ | Yep |
| 01:53.38 | bweschke_ | k |
| 01:54.19 | bweschke_ | Then you basically just setup a SIP user/peer pair on each box to feed calls back and forth - and once you've done that, you can use your dial plan to pass the calls through the trunk you've configured |
| 01:54.37 | IcePenguin | Thanks a bunch guys! See ya |
| 01:54.43 | klicek | anybody succed with asterisk 1.2 and zaphfc 0.3.0-PRE-1 ?? |
| 01:54.58 | *** part/#asterisk pussfeller (n=todd@12.150.129.170) |
| 01:55.07 | morale | does voicemail + mysql actually work in asterisk :) ? |
| 01:56.10 | bweschke_ | morale: yes |
| 01:56.28 | malverian[work] | Any known problems with Asterisk 1.2.0? |
| 01:56.40 | swm_ | would like an example configuration |
| 01:56.45 | malverian[work] | Any reason not to have it on a production system? |
| 01:56.53 | marcus2 | i have it on a production system |
| 01:57.01 | klicek | malverian[work] : i've heard it won't work with isdn4linux |
| 01:57.04 | justinu | there's an issue with RFC2833 dtmf |
| 01:57.28 | morale | bweschke_: do you have to use res_odbc.conf for it to work? |
| 01:57.55 | docelm0 | this is crazy.. |
| 01:57.56 | bweschke_ | I believe so, yes |
| 01:58.03 | malverian[work] | justinu, In 1.2.0? |
| 01:58.05 | docelm0 | Has anyone had issues with 1 way DTMF? |
| 01:58.15 | justinu | malverian[work]: yeah, in 1.2.0 |
| 01:58.21 | justinu | see topic |
| 01:58.48 | justinu | it's a simple patch, otherwise 1.2.0 seems great. |
| 01:59.25 | marcus2 | the rfc2833 bug only affects people using g729 |
| 01:59.47 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 01:59.48 | justinu | it effects you if g729 is offered |
| 01:59.53 | marcus2 | ah |
| 01:59.54 | justinu | not necessarily negotiated |
| 02:00.13 | malverian[work] | I currently have our production server running CVS as of... 2005-10-27 |
| 02:00.14 | Damin | It's 9:00. Do you know where your children are? |
| 02:00.33 | bweschke_ | yes. in bed |
| 02:01.52 | malverian[work] | Yeah.. that code didn't even exist as of 10/27 and I don't use g729, so it doesn't affect me. |
| 02:02.20 | malverian[work] | Stability-wise though? No crashes? |
| 02:02.46 | bweschke_ | malverian: no. i've not come across any where I'm running 1.2 with clients in production |
| 02:03.19 | malverian[work] | With the current CVS I'm using.. we have crashes every few days.. I think it's a problem that surfaces due to the soft phone we created in house. |
| 02:03.35 | malverian[work] | I'm going to get 1.2.0 and try to reproduce it on our backup machine. |
| 02:05.19 | docelm0 | for DTMF to work with g711u does the ;rport= need to be set for inband to work |
| 02:05.59 | justinu | i don't think so |
| 02:08.02 | *** join/#asterisk IcePenguin (n=G4Master@207-118-163-29.dyn.centurytel.net) |
| 02:08.32 | IcePenguin | Quick question: anyone know of a decent 2-line VOIP phone that's around ~100 bucks? |
| 02:08.47 | tzanger | ha |
| 02:08.48 | tzanger | no |
| 02:08.51 | bweschke_ | spa-941 just out from sipura is pretty good near that price point... |
| 02:08.52 | justinu | yeah, not really |
| 02:08.53 | bweschke_ | near |
| 02:09.00 | justinu | gxp2000 is the closest thing you'll get |
| 02:09.01 | bweschke_ | I think it's closer ro $150 |
| 02:09.09 | *** join/#asterisk Cresl1n (n=matt@146.229.188.212) |
| 02:09.15 | justinu | sipura 841 is ok, but has shitty keypad |
| 02:09.29 | bweschke_ | rt - the spa-941 fixes that, among other things |
| 02:09.45 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
| 02:09.45 | bweschke_ | i was pretty impressed when I put one through its paces here a week or so ago |
| 02:09.47 | justinu | the 941 looks really slick |
| 02:09.57 | justinu | i'll need to get a few |
| 02:10.01 | IcePenguin | dedicated VM button? |
| 02:10.07 | bweschke_ | one sec |
| 02:10.11 | bweschke_ | will go get it and dbl check |
| 02:10.12 | justinu | yeah, it has one |
| 02:10.20 | justinu | even the 841 has it |
| 02:10.28 | IcePenguin | sweet |
| 02:10.43 | Cresl1n | ~lart himself |
| 02:10.44 | IcePenguin | would you say stay away from 841? it seems to fit the budget more |
| 02:10.54 | Cresl1n | hrm... |
| 02:10.57 | }btorch{ | hey guys I'm trying to connect * to a Siemens PBX over a T1 ... the zaptel.conf for the span settings. should I use 0 for the timing sync making asterisk the master I guess |
| 02:10.58 | Cresl1n | no jbot anymore |
| 02:11.03 | tzanger | mmm tuna helper with milk that just started to turn :-) |
| 02:11.11 | justinu | IcePenguin: if your users don't mind shitty ass rubber keypads and a display they can hardly see, it's fine. |
| 02:11.15 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 02:11.45 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 02:11.59 | tzanger | ~lart theworld |
| 02:12.04 | tzanger | bah |
| 02:12.06 | tzanger | ~lart the world |
| 02:12.12 | tzanger | ~lart everyone |
| 02:12.17 | tzanger | ha |
| 02:12.26 | bweschke_ | i wouldn't recommend spa-841 for a biz deployment, no |
| 02:12.54 | IcePenguin | its more like small office |
| 02:13.26 | bweschke_ | even still - if people are sensitive and have certain expectations on how their phones should look/feel, the spa-841 probably won't pass muster |
| 02:13.40 | IcePenguin | all righty |
| 02:13.44 | bweschke_ | the phone works.. don't get me wrong... |
| 02:13.48 | IcePenguin | ill work the 941 in |
| 02:14.00 | tzanger | good evening me dear katty |
| 02:14.53 | Katty | hi tzanger |
| 02:15.16 | justinu | if they want to go super cheap, gxp2000 has nicer buttons |
| 02:15.31 | justinu | 4 line appearances |
| 02:15.39 | tzanger | if you want super cheap get POTS phones and FXS interfaces |
| 02:15.49 | justinu | even a built in ethernenet switch |
| 02:15.53 | *** join/#asterisk shakuhashi (n=libre@200.140.14.30) |
| 02:16.11 | IcePenguin | do you guys use voipsupply.com |
| 02:16.15 | IcePenguin | or another site? |
| 02:16.21 | justinu | i have |
| 02:16.26 | justinu | but I wouldn't recommend them |
| 02:16.32 | IcePenguin | no? |
| 02:16.39 | bweschke_ | i'm a voipsupply.com fan |
| 02:16.43 | justinu | they lie about shipping stuff |
| 02:16.48 | bweschke_ | i haven't been steered wrong by them |
| 02:16.53 | justinu | happened to me too many times now |
| 02:17.03 | ResidenteE | the CB24-FXS-110V is a T1 FXS Channel bank, how connect this to * ? |
| 02:17.05 | chris_topo | I have received 2 orders from them ok |
| 02:17.44 | ResidenteE | i need and aditional T1 card in my PC? |
| 02:17.55 | *** join/#asterisk nw_it (n=alex@203.102.166.250) |
| 02:17.57 | chris_topo | only problem I had was I didn't see the sig release button on the order page the first time and had to go to fedex to pick it up. |
| 02:17.57 | justinu | a friend of mine got sent a bad 7960 phone |
| 02:18.08 | justinu | took them WEEKS to approe the RMA and send out a new phone. |
| 02:18.18 | bweschke_ | i've probably done in excess of $40k worth of business with them in the past six months... so far .. so good |
| 02:18.30 | bweschke_ | I don't have experience with their RMA process though - thankfully |
| 02:18.37 | }btorch{ | what does the TE110P yellow alarm mean ? |
| 02:18.39 | bweschke_ | haven't had a need to do so - thus far |
| 02:18.41 | justinu | i dunno, maybe they treat small time like me poorly |
| 02:18.50 | justinu | their prices on polycom sucks too |
| 02:19.16 | bweschke_ | btorch: means the other side has you in "carrier fail" |
| 02:19.43 | bweschke_ | btorch: your carrier likely needs to bounce their interface facing you |
| 02:19.44 | }btorch{ | ok probably because I don't have the cable connected yet then |
| 02:19.58 | bweschke_ | btorch: if your cable isn't connected you should be red instead of yellow |
| 02:20.14 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-228-2.rcmdva.fios.verizon.net) |
| 02:20.20 | bweschke_ | or I guess red and yellow... |
| 02:20.27 | IcePenguin | does anyone know if the gxp2000 has a VM light? |
| 02:20.29 | bweschke_ | i've seen cards sound both alarms at the same time |
| 02:20.32 | docelm0 | «SpaceBass» How's the FIOS? |
| 02:20.40 | }btorch{ | I have a read light blicking but the dmesg told setting a yellow alarm |
| 02:21.00 | IcePenguin | oh yeah it does |
| 02:21.05 | justinu | IcePenguin: it does |
| 02:21.12 | bweschke_ | setting or received? |
| 02:21.20 | chris_topo | anyone know of a wifi phone that can do both sip and iax2 and have the option to tunnel over ssh or openvpn with pub key authentication? |
| 02:21.37 | bweschke_ | chris_topo: no |
| 02:21.40 | justinu | chris_topo: LOL |
| 02:21.41 | IcePenguin | justinu: 2 ethernet portson the back of the gxp2000... can I daisy chain? |
| 02:21.43 | justinu | that's hilarious |
| 02:21.46 | justinu | IcePenguin: yes |
| 02:21.53 | bweschke_ | chris_topo: ya, a laptop |
| 02:22.07 | IcePenguin | thanks justinu |
| 02:22.21 | chris_topo | actually working on that with a zaurus but not much luck yet |
| 02:22.54 | docelm0 | «IcePenguin» yes.. I have 50 of them |
| 02:23.22 | chris_topo | wish someone (maby linksys) would make a linux based phone that a comunity developed around like with the wrtg and nslu2 |
| 02:23.58 | IcePenguin | how does this setup sound, guys? |
| 02:24.45 | justinu | it sounds like a bargain basement voip deployment |
| 02:24.50 | IcePenguin | 2FXO --> * server --> Linksys SD205 5 port switch --> 3x Grandstream gxp2000 |
| 02:25.00 | justinu | but it'll work |
| 02:25.41 | IcePenguin | it doesnt need to be fancy really, its just a small travel agency |
| 02:25.43 | shmaltz | IcePenguin, I cn't know what it sounds like until I call it |
| 02:26.02 | justinu | IcePenguin: travel agents spend a lot of time on the phone |
| 02:26.09 | IcePenguin | indeed |
| 02:26.11 | justinu | they may opt for a nicer phone if you show them a few |
| 02:26.15 | *** part/#asterisk santiago (n=santiago@208.195.215.160) |
| 02:26.24 | justinu | sell them polycom 501s and they'll think you're a god. |
| 02:26.45 | bweschke_ | a word to the wise: don't short change yourself on the phone and make darn certain you've handled any possible echo conditions before you complete that install |
| 02:26.51 | shmaltz | IcePenguin, I agree with justinu, or maybe even a cisco |
| 02:27.06 | malverian[work] | Anyone else get these a lot? |
| 02:27.10 | bweschke_ | the 501 is a good choice.. agree.. |
| 02:27.12 | malverian[work] | WARNING[20684]: chan_sip.c:11253 do_monitor: chan_sip: ast_sched_runq ran 31 all at once |
| 02:27.16 | justinu | everone I've sold the 501 is just amazed |
| 02:27.21 | IcePenguin | what's the processor requirement for decent quality calls? |
| 02:27.23 | justinu | they've never used a phone that good in their life. |
| 02:27.27 | malverian[work] | I'm getting it every few seconds... |
| 02:27.43 | shmaltz | I justinu, the Cisco is much more stable, but lacks half the features of the polycom |
| 02:27.58 | justinu | you've had stability problems with polycom? |
| 02:28.01 | justinu | mine are solid. |
| 02:28.30 | shmaltz | justinu, yep |
| 02:28.37 | shmaltz | justinu, how big is your biggest setup? |
| 02:28.37 | justinu | any details? |
| 02:28.43 | bweschke_ | I would offer up the SPA-941 as probably the lower end option |
| 02:28.45 | justinu | not big.. 15-20 stations. |
| 02:28.56 | IcePenguin | Okay |
| 02:29.02 | bweschke_ | then come up to the IP501 - and/or possible the 7940 |
| 02:29.14 | shmaltz | justinu, like one way audio even without nat, frozen screens, no sound at all etc |
| 02:29.20 | IcePenguin | whats the deal with echo... is that an option on the * server or the phones themselves |
| 02:29.27 | justinu | shmaltz: haven't experienced any of that luckily. |
| 02:29.31 | bweschke_ | the 7960 and 601 or phones with more appearances don't make much sense because you won't have enough fxo to justify it |
| 02:29.38 | justinu | we could throw in snom in there, but why? |
| 02:29.45 | shmaltz | justinu, thats because you havnt dealt with it enough |
| 02:29.50 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 02:29.56 | justinu | shmaltz: certainly possible. |
| 02:30.22 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 02:30.25 | jeffik | hello all: can anyone tell me what this error from provider is? NOTICE[22757]: frame.c:149 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 02:30.53 | justinu | shmaltz: latest bootrom/sip image won't solve that? |
| 02:31.26 | bweschke_ | there are software based echo cancelers that are built into *.. some work great in some situations... other have misconfigured or badly tuned gain/impedance settings when you're going from 2 wires to 4 wires and in situations like that, if you don't fix/tune it, you're bound to have some pretty miserable experiences with echo |
| 02:31.33 | shmaltz | justinu, The latest sip image I tried was 1.52 and it is possible that 1.63 solves it |
| 02:31.50 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 02:31.52 | justinu | shmaltz: i guess i'll find out eventually |
| 02:31.58 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 02:32.22 | IcePenguin | bweschke: i'll be running the two-line system on 2 seperate pairs, on 2 seperate FXOs |
| 02:32.33 | IcePenguin | am I gonna have really bad echo? |
| 02:33.38 | IcePenguin | or not so much? |
| 02:33.41 | SpaceBass | docelm0: you asked about FiOS... a while ago... its amazing |
| 02:33.54 | justinu | SpaceBass: stfu... :P |
| 02:33.59 | docelm0 | ya.. I know.. whois me.. :) |
| 02:34.03 | bweschke_ | icep: you're not really going to know until you get things up and going and make some calls |
| 02:34.03 | docelm0 | 30/5 here |
| 02:34.03 | SpaceBass | sorry... |
| 02:34.21 | SpaceBass | its not so much the 30/10 as the free 10 statics.... ahhhhh... glorious |
| 02:34.49 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 02:35.25 | IcePenguin | bweschke: but the polycom 501 has decent sound? whats the sound quality like on VOIP |
| 02:35.39 | bweschke_ | the 501 has fantastic sound quality |
| 02:35.43 | SpaceBass | althought I've noticed latency is crap when I dont use the connection for a while |
| 02:35.45 | justinu | SpaceBass: i'd love to test out how long they'll let you push 10m up continuously... |
| 02:35.54 | justinu | before you get disconnected. |
| 02:35.59 | bweschke_ | sound quality over voip is completely dependent on the network |
| 02:36.08 | SpaceBass | justinu: ... business level account... I bet I can push it pretty far |
| 02:36.15 | SpaceBass | :) |
| 02:36.16 | justinu | how much does it cost you? |
| 02:36.37 | SpaceBass | me? zero... my company gets billed like $300/month... and they like it |
| 02:36.37 | IcePenguin | bweschke: if the network is 3 switched VOIP phones (computers are on a different system) |
| 02:36.40 | IcePenguin | ? |
| 02:36.41 | shmaltz | bweschke_, what external echo can would you recommend? |
| 02:36.43 | bweschke_ | if you have a clean network between you and where you're sending / receiving calls, you'll likely be very impressed with the quality |
| 02:36.48 | justinu | 300/mo is not much... i'd pay that for my house. |
| 02:37.03 | SpaceBass | they require me to have a business account... personal is $45/month and they WANT the business level |
| 02:37.19 | docelm0 | I pay 285 I have 30/5 w/ 32 static |
| 02:37.38 | SpaceBass | it seems like the statics are free... at least for business... |
| 02:37.58 | IcePenguin | cool. I think I'll go with the 501s then |
| 02:38.00 | SpaceBass | remains to be seen if my firewall can handle them .... supposed too get the other 9 tomorrow |
| 02:38.05 | justinu | IcePenguin: good choice. |
| 02:38.37 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 02:38.38 | bweschke_ | it's a tough call shmaltz. you typically need to be up at DS1 level to get external echo cans, no? I mean you can do DS0 echo cans externally, but you still need to get a shelf on most that requires you to do alot of DS0s to justify it |
| 02:38.58 | shmaltz | bweschke_, i'm asking for tq |
| 02:39.01 | shmaltz | t1 |
| 02:39.04 | IcePenguin | I'm gonna play around at home, install CentOS + * + AMP and see how it'll work |
| 02:39.34 | justinu | buy yourself at least a 501 and use it for a while |
| 02:39.40 | SpaceBass | IcePenguin: wwhy not just AAH? |
| 02:39.41 | justinu | you'll see what we're talking about |
| 02:39.53 | bweschke_ | well tellabs has a long history of manufacturing DS1 echo cans |
| 02:40.00 | justinu | there's also ditech |
| 02:40.08 | *** part/#asterisk nw_it (n=alex@203.102.166.250) |
| 02:40.10 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 02:40.13 | IcePenguin | spacebass how comparable is it to * + AMP? |
| 02:40.34 | SpaceBass | aah IS * + AMP (and other stuff) |
| 02:40.43 | IcePenguin | oh sweet |
| 02:40.53 | chris_topo | aah is also centos |
| 02:40.55 | IcePenguin | is there a torrent of the new 2.0 beta 6? |
| 02:41.01 | jeffik | SpaceBass: got a moment for a question? |
| 02:41.08 | SpaceBass | always |
| 02:41.25 | bzbw | hi, I thought I have a simple implementation, to by pass greeting when the callerID is identified in zaptel, but it does not seem to work, anyone can give me a hint? |
| 02:41.30 | SpaceBass | ask here too though... I'm no expert |
| 02:41.30 | justinu | are there any apps that run over SCTP instead of TCP? |
| 02:41.31 | chris_topo | anyone know what the aah 2.0 centos version is? |
| 02:41.31 | jeffik | ok any idea what this error is? NOTICE[22757]: frame.c:149 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 02:41.56 | shmaltz | IcePenguin, don't dissapoinnt yourself with cheap phones, buy the real stuff (polycom, cisco, snom) and then try the cheaper ones, then you'll see why we don't like the grandstreams etc |
| 02:42.10 | shmaltz | jeffik, only 2 minutes since you last asked |
| 02:42.13 | SpaceBass | jeffik: asking me about that error? |
| 02:42.28 | SpaceBass | seriously... I'm no where near that advanced :) |
| 02:42.30 | jeffik | Space Bass: yes, |
| 02:42.36 | jeffik | shmalta: sorry |
| 02:42.51 | SpaceBass | chris_topo: I'll check in a sec... think its 4 something |
| 02:42.53 | IcePenguin | all righty, thanks shmaltz |
| 02:43.01 | shmaltz | IcePenguin, np |
| 02:43.13 | justinu | jeffik: i think VAD should be off on your g729 device. |
| 02:43.17 | bzbw | shmaltz: what gs phone do you have? |
| 02:43.18 | bweschke_ | jeffik: what version of asterisk? |
| 02:43.39 | jeffik | bweschke_: it |
| 02:43.41 | shmaltz | bzbw, bs102 |
| 02:43.44 | jeffik | cvs head |
| 02:43.46 | shmaltz | err bt102 |
| 02:43.54 | shmaltz | :) |
| 02:43.57 | bweschke_ | jeffik: what's the date on it? |
| 02:44.04 | bzbw | emm, it's pretty good to me, what went wrong? |
| 02:44.06 | jeffik | let me look |
| 02:44.37 | shmaltz | bzbw, I don't know what you are used to, but I'm used to analog, so that phone (BT102) is no replacement for it |
| 02:44.48 | SpaceBass | i've never used a poly phone, but my cisco phones are great... as is my hitachi IP5000... zyxel sucks by the way |
| 02:45.03 | justinu | zyxel does suck |
| 02:45.07 | justinu | battery life is a joke |
| 02:45.11 | shmaltz | SpaceBass, I have a zyxel here its not so bad |
| 02:45.14 | SpaceBass | better than my vtech analogue |
| 02:45.17 | shmaltz | why don't you guys like it? |
| 02:45.23 | IcePenguin | Question: Pentium I: will * run decently on it? |
| 02:45.28 | SpaceBass | shmaltz: no transfer, no hold, no WPA.... |
| 02:45.33 | SpaceBass | no call waiting |
| 02:45.35 | bweschke_ | icep: sure - depends what you're doing.. |
| 02:45.39 | shmaltz | IcePenguin, yeah it should |
| 02:45.41 | denon | IcePenguin: maybe, with very little voip |
| 02:45.46 | shmaltz | but don't try 24 transcodings |
| 02:45.58 | tzafrir_laptop | now why would you install those strange distros like centos? |
| 02:45.59 | SpaceBass | basically the Zyxel is a wifi mic and speaker.... thats it |
| 02:46.03 | denon | IcePenguin: it should do fine with all TDM |
| 02:46.11 | jeffik | bweschke_: how can i tell the release, somebody set it up for me |
| 02:46.16 | bzbw | shmaltz: you mean the BT102 does not have a good sound quality? Or it does not have some functions compare with your analog phone? |
| 02:46.18 | shmaltz | SpaceBass, mine has all of those |
| 02:46.24 | bweschke_ | it'll probably even be OK with one or two fxo |
| 02:46.26 | IcePenguin | TDM? |
| 02:46.27 | Damin | jeffik: cat /etc/redhat-release |
| 02:46.35 | shmaltz | bzbw, it has nothign search my opionion on this in the lists |
| 02:46.40 | justinu | shmaltz: i'm down on it simply because of the battery life |
| 02:46.58 | SpaceBass | shmaltz: really? |
| 02:46.59 | chris_topo | the kernel in the centOS 4.x makes my interupts scream and nobody cares. |
| 02:47.03 | shmaltz | justinu, you talking about a wifi phone |
| 02:47.04 | SpaceBass | wonder if its firmware....? |
| 02:47.05 | shmaltz | ? |
| 02:47.23 | SpaceBass | my IP5000 is great, but doesnt support WPA either |
| 02:47.24 | bweschke_ | do asterisk -r |
| 02:47.26 | shmaltz | SpaceBass, but not WPA its RJ45, not wifi |
| 02:47.32 | bweschke_ | and then see what it reports |
| 02:47.32 | bweschke_ | eg ast[root@prod01 bweschke]# asterisk -r |
| 02:47.32 | bweschke_ | Asterisk CVS-NHEAD-06/29/05-12:58:12, Copyright (C) 1999 - 2005 Digium. |
| 02:47.33 | bweschke_ | W |
| 02:47.44 | SpaceBass | so currently I'm running an open AP with access to my * box |
| 02:47.47 | SpaceBass | uber safe |
| 02:48.02 | IcePenguin | bweschke: Pentuim I running AAH, 2FXO, 3 Soundpoint 501's... what do you think? Will it handle? |
| 02:48.03 | justinu | shmaltz: so? why does it have to suck? |
| 02:48.12 | docelm0 | Hay SpaceBass got anything I can download from ya? Im curious to see how fios -> fios transfers work.. :) |
| 02:48.14 | shmaltz | SpaceBass, you can still do mac restrictions |
| 02:48.28 | shmaltz | justinu, you taling about the bt102? |
| 02:48.33 | SpaceBass | shmaltz: b/c thats safe :) |
| 02:48.35 | bweschke_ | icep: maybe... if you're going from fxo to ulaw.. i wouldn't try and go from fxo to 729 with it |
| 02:48.37 | justinu | i'm talking about the zyxel |
| 02:48.42 | SpaceBass | mac filtering is easier than WEP to crack ! |
| 02:48.48 | SpaceBass | (but, alas, I use it) |
| 02:49.04 | shmaltz | justinu, I wasn't the one that said it sucks |
| 02:49.12 | justinu | i was |
| 02:49.19 | justinu | why does it have to suck simply because it's wifi |
| 02:49.24 | chris_topo | that's why I want the phone with an open vpn client ;) |
| 02:49.36 | tzafrir_laptop | SpaceBass, and you hope nobody knows which MAC address to ipressionate to? |
| 02:49.37 | bzbw | ok, any one can point me to where I can find the info for setting up my zaptel to recognize callerID then route the call accordingly? |
| 02:49.44 | IcePenguin | bweschke: ulaw and 729 are voice encoders? |
| 02:49.50 | bweschke_ | codecs, yes |
| 02:50.11 | SpaceBass | tzafrir_laptop: finding other (working) MAC addresses on the same AP is trival |
| 02:50.11 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net) |
| 02:50.18 | tzafrir_laptop | bzbw, basically this is set in zapata.conf |
| 02:50.36 | IcePenguin | much quality difference? |
| 02:50.43 | shmaltz | here is the thread about the bt102: |
| 02:50.45 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-November/134593.html |
| 02:51.14 | jeffik | bweschke_: Asterisk CVS HEAD, Copyright (C) 1999 - 2005 Digium. |
| 02:51.15 | tzafrir_laptop | bzbw, callerid=yes; IIRC at this time of day |
| 02:51.17 | jeffik | that's it |
| 02:51.17 | SpaceBass | I just don't trust mac filtering |
| 02:51.24 | SpaceBass | although its what I'm using on that insecure AP now |
| 02:51.32 | SpaceBass | at least its cordoned off from my LAN |
| 02:51.55 | shmaltz | SpaceBass, I don't blame you since with a sniffer it can be: |
| 02:51.57 | shmaltz | spoofed |
| 02:51.59 | shmaltz | open for man in the middle attacks |
| 02:52.08 | SpaceBass | yeah |
| 02:52.21 | SpaceBass | and as someone who stays in hotels WITH hubs all the time... i know! |
| 02:52.33 | IcePenguin | bweschke: much quality difference between ulaw and 729? |
| 02:52.37 | chris_topo | bzbw for routing by caller id you just need exten=> s/incomming#,1,goto(wherever) |
| 02:52.51 | bweschke_ | icep: some.. yes.. 729 uses compression.. ulaw doesn't |
| 02:52.53 | shmaltz | IcePenguin, yeah ulaw is much better in almost all cases |
| 02:53.01 | bweschke_ | so you will notice ulaw produces a better sounding call |
| 02:53.15 | bweschke_ | jeffik: when was this machine setup for you? |
| 02:53.20 | swm_ | No G.729 is better |
| 02:53.21 | IcePenguin | and a PI will handle ulaw? |
| 02:53.24 | shmaltz | unles you have some very fancy expensive equipment doing the transcoding, like a dual 64 bit AMD machine |
| 02:53.38 | shmaltz | IcePenguin, P1 withyout a sweat |
| 02:53.41 | jeffik | bweschke_: about two weeks ago |
| 02:53.47 | IcePenguin | excellent |
| 02:53.49 | bweschke_ | jeffik: do "show version" from the CLI of asterisk and let me know what it shows |
| 02:53.50 | IcePenguin | thanks guys |
| 02:53.55 | jeffik | ok |
| 02:54.36 | IcePenguin | awesome |
| 02:54.40 | IcePenguin | thanks for everything guys! |
| 02:54.44 | shmaltz | np |
| 02:54.47 | jeffik | bweschke_: Asterisk CVS HEAD built by root @ asterisk1.local on a i686 running Linux on 200 5-11-07 22:57:16 UTC |
| 02:54.57 | SpaceBass | i can beat that |
| 02:55.00 | IcePenguin | bye |
| 02:55.11 | shmaltz | swm_, what you running on it? |
| 02:55.13 | SpaceBass | PII 350hmz, 128mb, 20gb drive |
| 02:55.18 | swm_ | Asterisk |
| 02:55.30 | bzbw | tzafrir_laptop: usecallerid=yes, there is no "callerid=yes", but it still does not seem to recognize the call. |
| 02:55.31 | shmaltz | swm_, realy? i thought televantage |
| 02:55.39 | shmaltz | why such an expensive machine? |
| 02:55.39 | bweschke_ | k. well the fix for that error was only put in on 11/15.. so my guess is that you're experiencing a bug that has been fixed in the 1.2 release |
| 02:55.42 | SpaceBass | 2 ZAPs, 3 SIP accounts, 1 IAX2 and 5 hard phones, 2 soft phones |
| 02:55.53 | bweschke_ | here's more info on the situation: http://bugs.digium.com/view.php?id=5539 |
| 02:55.54 | swm_ | It's a Intel SE7500WV2 Motherboard with 2 512 ECC Registered Chips and 2 2.0 Ghz Processors/Hyperthreading... Asterisk of course... |
| 02:56.13 | jeffik | bweschke_: thanks, |
| 02:56.19 | }btorch{ | has anyone here connected * to legacy PBX ? |
| 02:56.28 | bweschke_ | btorch: yes. |
| 02:56.36 | tzanger | }btorch{: yep. Norstar MICS |
| 02:57.32 | }btorch{ | bweschke I got a problem, I made a T1 crossover cable and connected to a T1 CSU ACE in the net port and I have the legacy PBX connected to the CPE port . |
| 02:57.42 | swm_ | Televantage looks like crap and it costs $$$ |
| 02:57.42 | shmaltz | }btorch{, yeah |
| 02:57.53 | jeffik | tzanger: Iwas wondering about MICS/norsar phones |
| 02:57.57 | shmaltz | btorch, what type of pbx? |
| 02:58.12 | shmaltz | swm_, so why such an expensive machine for *? |
| 02:58.17 | SpaceBass | t1 cross orver? isn't that just connecting the two fibers? |
| 02:58.25 | }btorch{ | bweschke for the first time asterisk actually hears some signals from the PBX but it warns me that " We think we are the CPE, but they think they are the CPE too" |
| 02:58.44 | h3x | }btorch{: those t1 cards have CSUs built in so you dont need to use one |
| 02:58.49 | tzanger | }btorch{: well yes, the PBX will very much want to be the CPE because it's designed to be the CPE |
| 02:58.50 | }btorch{ | shmaltz siemens officepro |
| 02:58.56 | tzanger | jeffik: ok |
| 02:58.57 | bweschke_ | what's a CPE port? what card is this in * ? |
| 02:59.14 | tzanger | bweschke_: you're getting confused |
| 02:59.15 | shmaltz | btorch, you got to set one for PRI_NET the asterisk side |
| 02:59.15 | }btorch{ | bweschke it's a TE110P |
| 02:59.16 | h3x | even though that has nothing to do with your problem |
| 02:59.24 | shmaltz | and the one to the telco as pri_cpe |
| 02:59.44 | shmaltz | btorch, set it as pri_net not cpe |
| 02:59.51 | jeffik | bweschke_: thanks for the info i'll read the link |
| 03:00.06 | }btorch{ | pri_net in the zapata.conf i guess |
| 03:00.20 | tzanger | swm_: I don't want to do technical things when I'm on shrooms |
| 03:00.33 | shmaltz | sem_, if you got vi running that it should be easy |
| 03:00.34 | }btorch{ | and then just plug the cable that comes out of the PBx into the TE110P card |
| 03:00.56 | shmaltz | swm_, if you got vi then it's easy |
| 03:00.59 | chris_topo | swm_ probabably easier than configuring an spa3000 |
| 03:01.25 | }btorch{ | cool let me try that |
| 03:01.40 | swm_ | Anyone need a Cisco ATA 186? |
| 03:02.10 | swm_ | ~lobotomy }btorch{ |
| 03:02.43 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:02.43 | *** topic/#asterisk is Asterisk 1.2.0 has been released! -- ftp.digium.com (RFC2833 users see bug 5780) || http://www.asterisk.org |
| 03:02.46 | shmaltz | btorch, I don't think so, unless it's not configure right |
| 03:02.52 | bweschke_ | it should match what your siemens pbx is trying to signal with |
| 03:03.15 | swm_ | ~news |
| 03:03.56 | swm_ | ~lobotomy }btorch{ |
| 03:03.57 | jbot | ACTION pulls out a rusty saw to perform a lobotomy on }btorch{ |
| 03:04.02 | shmaltz | ~sleeping |
| 03:04.03 | jbot | [sleeping] the magical thing geeks have forgotten how to do |
| 03:04.17 | bweschke_ | probably also need to make certain that both agree what ds0 in the ds1 is the D channel |
| 03:04.33 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 03:04.39 | file | hi Cresl1n!!! |
| 03:05.00 | shmaltz | bweschke_, you ever seen a full T1 as a PRI that doesn't use channel 24 as the D channel? |
| 03:05.29 | bweschke_ | I agree - that's certainly the "norm", but yes, to answer your question, I have. |
| 03:05.46 | shmaltz | bweschke_, wow, where? in the lower 48? |
| 03:05.55 | justinu | shmaltz: i've seen dchannels on timeslot 1. |
| 03:05.56 | shmaltz | or in alaska, or maybe japan? |
| 03:06.57 | justinu | i've seen carriers provision them on timeslot 1 |
| 03:07.05 | bweschke_ | yes - there's a CLEC in the south that likes to provision on ds0 1 instead of 24 |
| 03:07.05 | justinu | in response to customer request |
| 03:07.25 | shmaltz | bweschke_, any names? |
| 03:07.52 | bweschke_ | hargray |
| 03:08.06 | shmaltz | hargray is that a name of a clec? |
| 03:08.11 | bweschke_ | yes |
| 03:08.43 | shmaltz | ic |
| 03:08.48 | }btorch{ | awosome |
| 03:08.54 | shmaltz | bweschke_, where you from? |
| 03:09.19 | }btorch{ | thank you you all .. got it working now it was just the pri_net signalling |
| 03:09.36 | bweschke_ | again - in over 99% of the installs I've done, it's been 24... but I've always found some wierdness somewhere with the telco/tdm stuff |
| 03:09.40 | bweschke_ | I'm from NJ |
| 03:10.19 | shmaltz | wow, thats my home town |
| 03:10.23 | shmaltz | where are you now? |
| 03:10.27 | bweschke_ | in NJ |
| 03:10.35 | bweschke_ | Secaucus to be exact |
| 03:10.38 | shmaltz | bweschke_, private please |
| 03:10.42 | shmaltz | Lakewood here |
| 03:10.56 | bweschke_ | sure |
| 03:11.59 | *** join/#asterisk bch1 (n=bch@CPE-24-26-172-197.mn.res.rr.com) |
| 03:12.12 | *** join/#asterisk jahani2 (n=k@adsl-30-46-192-81.adsl.iam.net.ma) |
| 03:12.37 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 03:12.47 | Brijn | Good evening |
| 03:13.10 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 03:13.31 | }btorch{ | good night |
| 03:13.47 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 03:14.14 | jeffik | tzanger: have you got norstar phones working with *? |
| 03:14.53 | robl^ | there is an adaper to let you use norstar phones as SIP phones (to work with Asterisk) |
| 03:15.04 | jahani2 | hi |
| 03:15.16 | jeffik | robl^: really? had not heard of it |
| 03:15.24 | robl^ | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-47071662592.htm |
| 03:15.30 | tzanger | jeffik: no, I have the MICS workig with * to an extent |
| 03:15.31 | jahani2 | how can i dial phone number using my phone line with asterisk ? |
| 03:15.37 | tzanger | robl^: yeah there is one |
| 03:15.39 | tzanger | kind of neat |
| 03:15.43 | Brijn | Weird prob here. Setup used to work fine for a few weeks. But now, voicemail is sounding very weird, sluggish.. |
| 03:15.56 | SpaceBass | jahani: can you elaborate? |
| 03:16.10 | konfuzed | tzanger: what ata have you used that works with nortel? |
| 03:16.20 | Brijn | Normal calls OK, voicemail via email = OK, voicemail on Soundpoint or Xlite = slow. Both CVS and stable |
| 03:16.29 | tzanger | konfuzed: none, I use a PRI between * and nor* |
| 03:16.39 | jahani2 | SpaceBass what u mean by elaborate? ? |
| 03:16.56 | tzanger | robl^: unfortunately that link doesn't say how much control you have over the soft buttons and display |
| 03:17.01 | SpaceBass | well, what kind of number do you want to dial? what kind of "line" do you have with asterisk |
| 03:17.01 | tzanger | IIRC it wasn't all tha tmuch :-) |
| 03:17.17 | konfuzed | tzanger: so youve seen or heard of them but never used them |
| 03:17.33 | jeffik | robl^: thanks, but at that price i can buy all new cisco phones |
| 03:17.38 | jahani2 | i have a analog line |
| 03:17.42 | jahani2 | normal telephone line |
| 03:17.44 | tzanger | konfuzed: not the citel unit, no |
| 03:17.55 | konfuzed | kewl |
| 03:18.00 | jahani2 | i have card modem on the machine where asterisk installed |
| 03:19.30 | bzbw | Emm, I still failed to do so: How do I directly forward a call base on callerid from my x100p? I believe both zpata.conf and zaptel.conf is set up correctly. |
| 03:19.42 | robl^ | jeffik: yeah. they are expensive... but they do support 24 phones.. it would cost about $7200 for 24 Cisco phones :) |
| 03:20.07 | SpaceBass | jahani2: a modem wont work |
| 03:20.14 | SpaceBass | you need an fxo type card |
| 03:20.19 | jahani2 | yes this is |
| 03:20.21 | bzbw | my x100p does not seem to get the callerid quickly. |
| 03:20.38 | jahani2 | i call it modem :) |
| 03:20.42 | SpaceBass | ok |
| 03:20.44 | jeffik | robl^: yeah you are right, it's just here in Canada there are so many norsgart installs and all people can think of is that they paid over $550 each for the phones when they were new |
| 03:20.47 | bzbw | my plan is to bypass the greeting if certain callerid calls in |
| 03:20.57 | jahani2 | how can i configure asterisk to use it? |
| 03:21.02 | robl^ | caller ID on a PSTN analog line , in the US at least, is always between ring one an second ring. |
| 03:21.06 | SpaceBass | bzbw typicall x100ps dont pick up until the 2nd ring |
| 03:21.08 | SpaceBass | for that reason |
| 03:21.34 | bzbw | SpaceBass: so if I set it wait for 5 seconds, it should work, right? |
| 03:21.38 | Flauto | spa, that is always the case |
| 03:22.02 | SpaceBass | duno |
| 03:22.03 | chris_topo | so do the x100ps pickup before an answer() is reached in the dial plan? |
| 03:22.03 | bzbw | SpaceBass: from the console, I don't seem to see the callerid from the caller to the x100p card |
| 03:22.15 | Juggie | sigh |
| 03:22.17 | robl^ | jeffik: its one of those "options" you offer a customer. if they are dead set on keeping the norstar phones, you can use this.. otherwise suggest swapping them out with new phones |
| 03:22.19 | Juggie | if callerid is enabled |
| 03:22.36 | Juggie | x100 doesnt go into the dialplan until 2nd ring |
| 03:22.43 | bzbw | usecallerid=yes is set in zapata.conf |
| 03:23.05 | jeffik | robl^: yes, oh do you know an ip phoen that will do the paging through the phone featuer that norstart phones do? |
| 03:23.11 | SpaceBass | night all |
| 03:23.27 | chris_topo | so with an x100p should I be able to txt my callerid info to my cell without going off hook |
| 03:23.30 | chris_topo | ? |
| 03:24.18 | robl^ | jeffik: You can "hacK" that feature into several models.. just need some extra scripting in extensions.conf. Cisco, Snom, Polycom will all work |
| 03:25.17 | jeffik | robl^: ok, i'm not surprised. it's a featuer a lot of norstar users like |
| 03:25.29 | Flauto | anyone would let me know why i have to modprobe zaptel and wcfxo everytime i reboot my computer |
| 03:26.06 | robl^ | Jeffik: basically you set a "line" on one of the sip phones to autoanswer. THen you set an extension number to dump all the auto anser lines into a conference for the page.. |
| 03:26.14 | jahani2 | plz any one can help me to use the fxo card? |
| 03:26.55 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com) |
| 03:27.27 | jeffik | robl^: thanks, i'll look at that |
| 03:27.56 | jeffik | robl^: do you know anythign about ACT P104SLD? |
| 03:28.01 | robl^ | jeffik: that same auo-answer line can be used for a Norstar-like "Voice Call" feature too |
| 03:28.19 | robl^ | hrmm.. what is ACT P104SLD? |
| 03:28.34 | jeffik | robl^: the 'voice call is really needed also' thanks |
| 03:29.18 | chris_topo | jahani2 see about page 60 of asteriskTFOT it is avaliable as a free pdf |
| 03:29.27 | jeffik | http://www.act-tel.com.tw/_pg/products/productItem.asp |
| 03:29.50 | robl^ | most features of a Norstar can be implimented in Asterisk / SIP.. sometimes the the names for the features are different.. or requires a little scripting loginc |
| 03:30.21 | jeffik | robl^: the auto answer line is a clever idea |
| 03:30.32 | robl^ | jeffik: do you use Norstar CAP modules? |
| 03:30.46 | jeffik | robl^: no |
| 03:31.05 | jeffik | and many of them are still used in square config, all lines on all phones |
| 03:31.26 | robl^ | ahh! so no need for a receptionist console? |
| 03:32.02 | robl^ | last Norstar system I used had 24 incoming lines and 56 extensions.. only the receptionist had direct line appearances.. |
| 03:32.23 | jeffik | well usually there is a centeral answer phone but all they do is put the call on hold and call in the intercom to tell them what line to answer |
| 03:32.43 | jeffik | but the put startalk on it and they sort of have voice mail |
| 03:32.49 | jeffik | true hybrid, and a mess |
| 03:33.35 | jeffik | well this is the home of noretel so there is a ton of it out here from large to tiny configs |
| 03:33.40 | robl^ | yeah. Norstar was pretty awesome in its day... still better than MOST proprietary systems |
| 03:33.49 | jeffik | oh yes, i loved it |
| 03:34.35 | jeffik | and for it's time it was so much better than anything out there |
| 03:34.43 | robl^ | My current employer has Nortel Merridian... but we are slwly migrating each office to VoIP |
| 03:34.56 | jeffik | even put on in as a call center, better than the option 11 call center we had |
| 03:35.12 | jeffik | robl^: Where are you? |
| 03:35.18 | robl^ | Texas :) |
| 03:35.30 | jeffik | ahh, Toronto here |
| 03:36.24 | jeffik | but nortel seems to have a not invented here philosophy, if they didnt' invent it it does not exist |
| 03:37.12 | robl^ | jeffik: *if* you need a CAP-like module for asterisk.. look at Snom 360 phone with the "sidecar" |
| 03:37.19 | Flauto | jeffik, i am using vbuzzer |
| 03:37.28 | jeffik | vuzzer? |
| 03:37.35 | Flauto | vbuzzer |
| 03:38.05 | Flauto | it is legging in and out all the time |
| 03:38.08 | jeffik | Flauto? wow |
| 03:38.14 | robl^ | yeah.. Nortel got a huge market share.. and rarely did they add new features... or expand things.. going from coper to fiber was the only major new feature in years |
| 03:38.41 | robl^ | they were stagnant for a long time |
| 03:38.54 | jeffik | robl^: and i feel bad for the customers who paid over $550 pe phone and now it's all obsolete |
| 03:38.57 | Qwell | justinu: I've never called anybody an idiot for not buying Digium hardware, nor have I even implied it. |
| 03:39.25 | jeffik | startalk falsh was awsome, even had a call center app in the 4 port model, now no support |
| 03:39.30 | Brijn | Weird prob here. Setup used to work fine for a few weeks. But now, voicemail is sounding very weird, sluggish.. |
| 03:39.36 | Brijn | Normal calls OK, voicemail via email = OK, voicemail on Soundpoint or Xlite = slow. Both CVS and stable |
| 03:40.13 | jeffik | Flauto: <<it is legging in and out all the time>> legging? |
| 03:40.31 | robl^ | Yeah! Then they had the NAM... but that was old, slow, and clunky. |
| 03:40.50 | Flauto | Nov 28 21:38:32 NOTICE[6118]: chan_sip.c:9693 handle_response_peerpoke: Peer 'vbuzzer1' is now TOO LAGGED! (1092ms / 200ms) |
| 03:41.18 | jeffik | yeah, i just worked on a system with nam, a 30 pound box with 1 ppound worht of stuff inside |
| 03:41.50 | robl^ | yup!! and they take 2 hours to boot and play that "melody" :) |
| 03:41.58 | jeffik | they did over kill on power supplies |
| 03:42.19 | jeffik | FEATURE 983 |
| 03:43.50 | robl^ | hehe! Passwd: 1020000 ;) |
| 03:44.06 | jeffik | that's if your extensions are 4 digits |
| 03:44.25 | robl^ | 3 digits. |
| 03:44.34 | jeffik | for 3 its 102000 |
| 03:44.35 | robl^ | 10020000 is the 4 digit pass |
| 03:44.56 | jeffik | and 2 digit is 120000 |
| 03:45.14 | jeffik | i think that is what i used last week, |
| 03:45.25 | jeffik | and none of this is on the web |
| 03:45.35 | jeffik | except one site tek-tips |
| 03:46.42 | tzanger | yes 1020000 is the default |
| 03:46.56 | *** join/#asterisk blanchet (n=jk@2001:5c0:8fff:fffe:0:0:0:3f89) |
| 03:47.05 | jeffik | tzanger: you worked on them too? |
| 03:47.26 | robl^ | after I left that company where I used to admin the Norstar system.. They refused to pay me 6 weeks of back pay. The new office manager called me and asked what to do when the dispaly of a phone was in spanish instead of english... I told her she has to reboot and do a "FEATURE **RESTART" |
| 03:47.42 | tzanger | jeffik: yep. feature **266344 |
| 03:47.52 | jeffik | ha ha that will fix her |
| 03:47.59 | jeffik | blank system |
| 03:48.00 | tzanger | robl^: hahaha |
| 03:48.17 | tzanger | it did fix it for her I bet... english is the default :-) |
| 03:48.36 | jeffik | i often walk thogh a retil store with 7310 pphoens on the wall and wonder if i could get into it or if they changed it |
| 03:48.45 | Flauto | jeffik, do you use vbuzzer? |
| 03:48.46 | tzanger | I need some good lowlevel docs on the SL1 protocol and also the intermodule comms. It's a modified E1 from what I can tell |
| 03:48.52 | robl^ | tzanger: yup! I told her it would reset it back to default. I asked if that is what she wanted.. |
| 03:49.06 | jeffik | Flauto: never seen until you told me |
| 03:49.17 | Flauto | it is based in toronto |
| 03:49.31 | Flauto | i even got a 416 area code |
| 03:49.32 | jeffik | tzanger: did you try nortel knowledge base? |
| 03:49.48 | jeffik | Flauto: where are you? |
| 03:49.49 | tzanger | jeffik: where's that? |
| 03:50.06 | jeffik | deep deep in the nortel url |
| 03:50.11 | robl^ | I have a Nortel complete documenation set CD.. but the darn PDFS for the admin guides are password protected.. |
| 03:50.49 | jeffik | robl^; oh |
| 03:51.12 | tzanger | robl^: there are pdf password crackers |
| 03:51.20 | tzanger | jeffik: doubt they'd have it publically available |
| 03:51.30 | Qwell | don't things like gpdf ignore passwords? |
| 03:51.43 | Qwell | or perhaps that's just older ones? |
| 03:51.43 | robl^ | I had the password a long time ago.. on a little card inside the jewl case, but it has been missing for a while |
| 03:51.47 | tzanger | besides, the **266344 I already have tricked out, I need general protocol info |
| 03:51.47 | jeffik | well i had to find programming for MIRAN last month, and it was there, but realloy had ot dig for it |
| 03:52.05 | robl^ | well.. they are encrypted.. you have to have the pass to decrypt and open it |
| 03:53.28 | robl^ | not that I mave much need for it anymore.. about all I do ihelp a friend change set names for new employees of his small business |
| 03:53.35 | Druken | uhg... |
| 03:55.00 | robl^ | jeffik: FEATURE **ADMIN has the default password at a Border's Books down the streat from here. I accidentally dialed the wrong number when I asked to borrow the phone |
| 03:55.54 | Damin | So what changes does the next generation VPM400M implement? |
| 03:56.20 | justinu | i have the tech pubs set for the dms-100/250/500 |
| 03:56.32 | tzanger | you accidentally dialled **ADMIN? heh |
| 03:56.40 | robl^ | ohh! is there an easy way to revert a Cisco 7960 to factory default state prior to selling on ebay? I have a few that I want to wipe clean... |
| 03:56.54 | justinu | put skinny 3.5 on them! |
| 03:57.11 | Qwell | robin_sz: What firmware is on it now? |
| 03:57.12 | Qwell | erm |
| 03:57.13 | Qwell | robl^: |
| 03:57.23 | jeffik | robl^: amazing they put it in public and anyody can reset it |
| 03:57.26 | robl^ | tzanger: its a habbit. you see a M7310 and your fingers have a miond of their own.. :) |
| 03:57.37 | justinu | lol |
| 03:57.44 | robl^ | Qwell: Sip 7.3 |
| 03:57.58 | Qwell | robl^: When you factory reset, you have to have the old firmware version for it to grab. |
| 03:58.02 | Qwell | That was the case on the 7970 anyhow. |
| 03:58.22 | Qwell | but, hold # while you plug in the power...after the top lights start flashing, let go, and do |
| 03:58.24 | jeffik | robl^: i thought i was the only one who thought about doing that |
| 03:58.25 | Qwell | 123456789*0# |
| 03:58.29 | jeffik | every walgreens too |
| 03:58.36 | robl^ | I have firmware for every thing since about 5.x, I think.. |
| 03:58.41 | Qwell | You *will* need dhcp, and the firmware you want to load |
| 04:00.06 | robl^ | not a problem.. I have DHCP and TFTP here. I have been using these phones for about a year or so.. upgraded them through at least 7 versions of firmware |
| 04:00.22 | Qwell | yeah...it's a fairly simple process, if you have dhcp on your LAN. |
| 04:00.36 | Qwell | tip: When doing a factory reset...make sure your freaking routes are setup properly on your dhcp server |
| 04:00.43 | Qwell | I cannot stress that enough. :P |
| 04:01.12 | Qwell | You'll see all the tftp traffic...but the phone will just reboot...it *really* gets on your nerves when you see what appears to be valid traffic |
| 04:01.57 | robl^ | yeah.. I saw that happen.. SCCP firmware is even worse that the SIP.. |
| 04:02.07 | Qwell | this was with no firmware |
| 04:02.12 | Qwell | so I couldn't even check logs or anything |
| 04:02.31 | robl^ | ohh!! |
| 04:03.49 | robl^ | maybe I will just wipe out the proxy setting and a few other key things and just have them to re-read the SIP*.conf files :) |
| 04:04.12 | robl^ | set everything to unprovisioned |
| 04:05.53 | jahani2 | any one has asterisk PBX Manager ? |
| 04:05.57 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:06.18 | shmaltz | jahani2, you mean from 3rdlane? |
| 04:07.03 | *** join/#asterisk junbug (n=junya@64.252.185.155) |
| 04:09.04 | jahani2 | yes |
| 04:09.14 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 04:09.29 | e-milio | hi all |
| 04:09.53 | e-milio | i have a simple question |
| 04:10.09 | Qwell | we'll be the judge of that |
| 04:10.35 | e-milio | with release 1.2 and svn I'am supposed to keep many dirs for the many versions ? |
| 04:10.58 | Qwell | e-milio: No, you just need to checkout the one branch/tag(or trunk) that you need |
| 04:11.16 | jahani2 | there is asterisk PBX Manager other than 3rdlane? |
| 04:11.26 | Qwell | for instance; svn co http://svn.digium.com/svn/asterisk/branches/1.2/ asterisk/branches/1.2 |
| 04:11.58 | Qwell | otherwise, you'll probably end up with hundreds of megs of redundant source |
| 04:12.13 | e-milio | Qwelbut it creates asterisk-1.2.0 |
| 04:12.27 | Qwell | e-milio: What are you trying to get? |
| 04:12.57 | e-milio | I am installing * on a server that does NOT have any other access in out our that ssh |
| 04:12.57 | tzafrir_laptop | 1.2 != 1.2.0 . 1.2 is basically the same as the tag v1-2 in the CVS |
| 04:13.07 | Qwell | tzafrir_laptop: exactly the same |
| 04:13.31 | Qwell | e-milio: you're gonna have to repeat that last part... |
| 04:13.33 | tzafrir_laptop | I figure that we will now have (fixed) versions under tags/ and branches under branches/ |
| 04:13.46 | e-milio | Qwell: so i copying the tars |
| 04:14.27 | e-milio | I want to put them in the right notation for when i can do svn there will be no confusion |
| 04:14.38 | Qwell | e-milio: You want the 1.2.0 release? |
| 04:14.39 | tzafrir_laptop | the content of the tar should be the same as svn co http://svn.digium.com/svn/asterisk/tags/1.2.0/ # I believe |
| 04:15.14 | e-milio | Qwell: yes, i downloaded and tar -xzvf and I get with asterisk-1.2.0 |
| 04:15.30 | tzafrir_laptop | Did they use this occasion as an excuse to remove the FPM mp3 files out of the release? |
| 04:15.34 | Qwell | Did you get the code from svn? I'm confused. |
| 04:15.53 | tzafrir_laptop | Qwell, me? not yet |
| 04:15.56 | Qwell | no, him |
| 04:16.03 | e-milio | Qwell:not not, now ftp'd the tars |
| 04:16.10 | Qwell | So why did you bring up svn? |
| 04:16.18 | Qwell | I still don't actually see a problem... |
| 04:16.19 | e-milio | because I will use it |
| 04:16.31 | e-milio | the prob now is that I have asterisk-1.2.0 |
| 04:16.35 | e-milio | and asterisk |
| 04:16.43 | Qwell | so get rid of it |
| 04:16.47 | tzafrir_laptop | e-milio, so don't bother with CVS/svn . Or is there a specific reason you need HEAD/current 1.2? |
| 04:17.07 | e-milio | should I leave it then as asterisk-1.2.0 ? |
| 04:17.24 | Qwell | If you want svn, no...you can get rid of that |
| 04:17.29 | tzafrir_laptop | The name of the directory doesn't really matter |
| 04:17.54 | e-milio | so I will mv'it to asterisk |
| 04:18.01 | tzafrir_laptop | e-milio, don't listen to both me and Qwell . We seem to give conflicting advices |
| 04:18.09 | Qwell | tzafrir_laptop: :p |
| 04:18.19 | Qwell | I'm certain I don't know what he wants |
| 04:18.28 | e-milio | I was thinking that I was to much caffeine |
| 04:18.33 | e-milio | on me |
| 04:19.11 | e-milio | My ? is that If I leave the dir as asterisk-1.2.0 will i have any probs in the future with svn ? |
| 04:19.18 | Qwell | none |
| 04:19.53 | Qwell | but if you do go to subversion, you might as well grab the tags from there, instead of having a bunch of dirs named "asterisk-1.2.blah" |
| 04:20.02 | e-milio | and It is convenient to leave it like that or will be better to move it to just asterisk ? |
| 04:20.52 | Qwell | the release tarball doesn't have any of the svn stuff in it, so it won't do you any good to move it into your svn dir |
| 04:21.26 | e-milio | i understand |
| 04:21.41 | Qwell | I just don't understand why you want to keep the release tarball around, if you're going to be using svn |
| 04:22.01 | tzafrir_laptop | a tarball is simpler to move around and manipulate. OTOH, if you know you'll build future versions of asterisk from the same directory, you should probably use subversion |
| 04:22.07 | e-milio | because tonight is the only way that i can get anything into that server |
| 04:22.59 | e-milio | and probably it will be a week or so before the "network people" open some ports for me |
| 04:23.19 | Qwell | eww! :p |
| 04:23.28 | e-milio | please dont |
| 04:24.06 | e-milio | now that i got your attention |
| 04:24.16 | e-milio | I may ask a third question |
| 04:24.23 | Qwell | was there a second question? |
| 04:24.47 | e-milio | not really, just checking for attention :) |
| 04:25.24 | e-milio | has anybody noted any strange behaivor with vici-dial and 1.2 ?? |
| 04:26.43 | e-milio | anyways |
| 04:26.56 | *** join/#asterisk JakBeatZ (n=JakBeatZ@216.7.194.254) |
| 04:26.58 | e-milio | Qwell, tzafrir_laptop Thanks a lot |
| 04:27.54 | JakBeatZ | Hey.. anyone know if there is an app to email meetme conferences? I seem to recall something on voip-info, but I can't seem to find it so I'm wondering if I was hellucinating when I thought I saw it |
| 04:28.14 | Qwell | JakBeatZ: You could setup a recording channel, then have a simple sendmail call |
| 04:30.46 | e-milio | e-milio: I dont know if this is the right place but if somebody is interested in working in florida full/ part-time on * support contact me. |
| 04:30.59 | JakBeatZ | Well, I am recording the conference and it is being written to a file. A recording channel would do the same thing, right? |
| 04:33.10 | Qwell | JakBeatZ: nope, we're talking about the same thing |
| 04:33.14 | Qwell | just a channel that records, or something |
| 04:34.20 | JakBeatZ | Ya, the issue is that I run the .wav through uuencode and pipe it to mail, but the problem is, there's no mime data added into the message so it munges up spam filters |
| 04:34.59 | JakBeatZ | I haven't been able to find a tool that I can mail the conf from the command line with proper MIME data added into the message |
| 04:37.02 | Brijn | My * is a bit sick.. It worked fine for a while. But now voicemail/MoH are played at 1/10th of the speed. The same voicemail via email is OK.. Any dieas |
| 04:38.09 | Flauto | would anyone tell me how to setup for voicemail to be able to call back and dialout while someone is checking voicemail |
| 04:40.20 | shmaltz | Flauto, /etc/asteirsk/voicemail.conf should help you |
| 04:40.43 | shmaltz | or try: |
| 04:40.45 | shmaltz | http://www.voip-info.org/wiki-asterisk+voicmail.conf |
| 04:41.03 | shmaltz | http://www.voip-info.org/wiki-asterisk+voicemail.conf |
| 04:41.09 | Flauto | shmaltz, i did setup voicemail.conf |
| 04:41.29 | Flauto | but it seems having problems getting the phone number and dialout |
| 04:41.31 | shmaltz | well, the comments should help you |
| 04:41.42 | Flauto | it says no channel or something like that |
| 04:41.56 | Flauto | so i figured that there might be somethign else i need to config |
| 04:42.14 | shmaltz | Flauto, what is the exact message you getting? |
| 04:42.27 | Flauto | i have to try now |
| 04:42.29 | Flauto | give me a sec |
| 04:44.12 | Flauto | Nov 28 22:43:35 WARNING[15575]: pbx.c:2347 __ast_pbx_run: Channel 'SIP/zhaoliu-0d6b' sent into invalid extension '17329108303' in context 'fromvm', but no invalid handler |
| 04:44.28 | Flauto | this is what i get when i tried to call out |
| 04:44.48 | Flauto | the voice was telling me to enter a number i wish to call |
| 04:45.09 | Flauto | okay |
| 04:45.19 | Qwell | [fromvm] |
| 04:45.28 | Flauto | i kind of think that i need a contxt [fromvm] |
| 04:45.29 | Qwell | exten => 173229108303,1,Dial(something) |
| 04:45.49 | Flauto | let me try |
| 04:46.00 | Qwell | ymmv |
| 04:46.36 | Flauto | give me a sec. |
| 04:46.39 | Flauto | let me figure it out |
| 04:46.47 | Qwell | You probably want the extension to be a pattern |
| 04:46.59 | Flauto | yes |
| 04:47.02 | Flauto | qwell |
| 04:47.06 | Flauto | somethign like |
| 04:47.13 | Flauto | s,1,blablah |
| 04:47.16 | Flauto | somethign liek that |
| 04:47.20 | Qwell | well, no, not s |
| 04:47.26 | Flauto | no s. |
| 04:47.28 | Flauto | hmmm.. |
| 04:47.34 | Qwell | exten => _1NXXNXXXXXX,1, |
| 04:47.38 | shmaltz | Flauto, exactly |
| 04:47.43 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:47.47 | shmaltz | Flauto, where in NJ do you live? |
| 04:47.54 | Flauto | i am not in nj |
| 04:47.57 | Flauto | my sister is |
| 04:48.09 | Flauto | well |
| 04:48.17 | Flauto | she was in eddison |
| 04:48.21 | Flauto | but she just moved |
| 04:48.29 | Flauto | i dont' even know what town she lives in now |
| 04:48.42 | Flauto | she got this stupid family plan from tmobile |
| 04:49.04 | Flauto | she got 4 phones and her family does not have enough people |
| 04:49.07 | Flauto | so, i got one |
| 04:50.15 | shmaltz | Flauto, anyhow you need a context defined where to dial out from |
| 04:50.36 | morale | did asterisk finally make the move over to svn? |
| 04:50.54 | Flauto | _1nxxnxxxxxxx takes care of domestic calls |
| 04:51.08 | Flauto | what about if i want international dialign too |
| 04:51.11 | Qwell | morale: yeah |
| 04:51.16 | shmaltz | morale, whats svn? |
| 04:51.20 | morale | subversion |
| 04:51.20 | Qwell | Flauto: then add a pattern for international |
| 04:51.34 | shmaltz | Flauto, _011X. |
| 04:51.35 | Flauto | okay |
| 04:51.39 | Flauto | yes |
| 04:51.42 | morale | shmaltz: a much less complex way of versioning sourcecode. |
| 04:51.48 | e-milio | I have another very simple question |
| 04:51.54 | morale | a replacement for cvs. |
| 04:52.15 | shmaltz | morale, I see |
| 04:52.47 | e-milio | ztcfg >>> line 0: Unable to open master device '/dev/zap/ctl' means that there is no digium card installed ?? |
| 04:53.05 | shmaltz | e-milio, or that your udev is not configured |
| 04:53.06 | Qwell | e-milio: or you're using udev, and you didn't read README.udev |
| 04:53.21 | Brijn | My * is a bit sick.. It worked fine for a while. But now voicemail/MoH are played at 1/10th of the speed. The same voicemail via email is OK.. Any dieas |
| 04:53.28 | e-milio | will do that |
| 04:53.31 | e-milio | thanks |
| 04:53.50 | shmaltz | Brijn, what else you running on that machine? |
| 04:54.08 | Brijn | Samba, Apache |
| 04:54.17 | shmaltz | Brijn, no x? |
| 04:54.23 | Brijn | shmaltz: no :) |
| 04:54.33 | shmaltz | hmmmmmmm |
| 04:54.49 | Brijn | I don't think it a resource issue, "live" calls are fine.. It's just VM/MoH |
| 04:54.52 | Ikarus | Hrm, is it possible to specify a digit matcher that can have between 0 and 8 X's |
| 04:54.55 | shmaltz | Brijn, how you listening to it? |
| 04:55.19 | Brijn | Polycom Soundpoint and/or Xlite softclient |
| 04:55.47 | Brijn | If I leave a voicemail, teh voicemail via email is OK, checked on the phone it's garbled/slow |
| 04:56.29 | Brijn | Weird thing is that it worked fine for weeks.. Suddenly stopped working, I have no clue what changed |
| 04:56.53 | shmaltz | lkarus, yeah: |
| 04:56.54 | shmaltz | exten => _X.,1,GotoIf(${LEN(${EXTEN})} > 0]?20) |
| 04:56.56 | shmaltz | exten => _X.,20,GotoIf(${LEN(${EXTEN})} <= 8]?25) |
| 04:56.58 | shmaltz | exten => _X.,25,Noop(this extensions is more than 1 but less then 9) |
| 04:57.36 | Ikarus | shmaltz: hmmmm, that is less then pretty, I hoped to do it purely with matching |
| 04:58.03 | Brijn | Nobody ever seen something similar |
| 04:58.50 | shmaltz | lkarus, then you can do this: |
| 04:58.52 | shmaltz | exten => _X,1,Noop() |
| 04:58.53 | shmaltz | exten => _XX,1,Noop() |
| 04:58.55 | shmaltz | exten => _XXX,1,Noop() |
| 04:58.56 | shmaltz | exten => _XXXX,1,Noop() |
| 04:58.58 | shmaltz | exten => _XXXXX,1,Noop() |
| 04:58.59 | shmaltz | exten => _XXXXXX,1,Noop() |
| 04:59.01 | Ikarus | shmaltz: I am considering that |
| 04:59.01 | shmaltz | exten => _XXXXXXX,1,Noop() |
| 04:59.03 | shmaltz | exten => _XXXXXXXX,1,Noop() |
| 04:59.10 | Qwell | implement ? as a pattern match |
| 04:59.18 | shmaltz | lkarus, it makes lots of sense |
| 04:59.21 | Qwell | _X???????,1, |
| 04:59.21 | Qwell | heh |
| 04:59.31 | Brijn | Ikarus: Where in .nl are you from? |
| 04:59.37 | Ikarus | Brijn: Rotterdam |
| 04:59.44 | ptiggerdine_ | anyone here work with voicetronix and asterisk? |
| 04:59.51 | Brijn | Mijn oude woonplaats / My old city :) |
| 04:59.52 | shmaltz | Qwell, whats the use of it? |
| 05:00.01 | Ikarus | Qwell: I have always wanted more extensive matching |
| 05:02.48 | Ikarus | Hrm, reading the official numberplan documents I am discovering numbers I didn't even know existed :) |
| 05:04.36 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 05:04.56 | Brijn | Ikarus: I thought the Dutch nr plan was a nice, consistent one :) |
| 05:05.09 | Ikarus | 1234 is supposed to be voice dialing, 1200 customer services, 131 is disable CID, 132 enable CID for this call, 1488 is a special number to reacht the police on, |
| 05:05.18 | Ikarus | Brijn: it is, mostly |
| 05:06.00 | Ikarus | Brijn: but for example 0842-333 for voicemail is officially not assigned (all 084 numbers are 10 digits long) |
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| 05:06.26 | *** part/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 05:07.09 | *** part/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
| 05:07.48 | Ikarus | btw, the official number for voicemail "electronische postbus" is 1233 |
| 05:07.54 | Ikarus | wonder if the KPN also supports that |
| 05:08.47 | Ikarus | I know that *21 and *61 (also not officially allowed in the number plan, they should have used 13 series numbers for that) are relics from the default configuration of the switches |
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| 05:15.36 | Flauto | qwell, and shm, it is working now |
| 05:15.38 | Flauto | thanks guy |
| 05:15.41 | Flauto | s |
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| 06:00.12 | Telamon | Is there any way to show the full record for a user from within asterisk? IE, I want to see if the nat and canreinvite flags are being propertly parsed from the database (using an ODBC backend). |
| 06:10.21 | IronHelix | sip show peer peername |
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| 06:13.46 | Telamon | Sweet, thanks. :) |
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| 06:18.55 | Flauto | qwell, it works great, thanks |
| 06:19.03 | Flauto | i created the fromvm |
| 06:19.05 | Flauto | and this way |
| 06:19.29 | Flauto | it does not bother anything of my other settings for dialing out frommy regular extensions |
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| 06:30.17 | delox99 | hi all |
| 06:30.45 | delox99 | i recompiled my kernel and now i cannot make install zaptel |
| 06:31.21 | delox99 | it tries to install the modules in /lib/modules/uname -r/misc |
| 06:31.31 | delox99 | that misc is a file and not a directory |
| 06:31.40 | delox99 | something went wrong |
| 06:32.04 | delox99 | however zaptel is compiling without a glitch |
| 06:32.13 | delox99 | it s just the install |
| 06:35.58 | axscode | make clean |
| 06:36.19 | axscode | make install |
| 06:37.15 | delox99 | yeah i did |
| 06:37.50 | harryvv | fist snow of the season. |
| 06:37.52 | harryvv | night all |
| 06:37.53 | tzafrir_laptop | hi |
| 06:38.50 | tzafrir_laptop | uname -r literally? or the kernel version? |
| 06:40.03 | delox99 | kernel version |
| 06:40.03 | tzafrir_laptop | misc is a file? a regular file? ls -ld /lib/modules/*/misc |
| 06:41.01 | delox99 | yeah |
| 06:41.10 | delox99 | i dont know whats wrong |
| 06:41.50 | tzafrir_laptop | it should be a directory. rm it and mkdir it |
| 06:42.15 | tzafrir_laptop | (in fact, mkdir should not be required. make install should take care of it) |
| 06:42.36 | delox99 | yeah i tried to just make it a directory then i make install |
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| 06:43.04 | delox99 | the install script runs but at the end there is still nothing in misc =) |
| 06:43.06 | delox99 | weird |
| 06:43.14 | axscode | delox99 what distro is that? |
| 06:44.27 | delox99 | im trying it again |
| 06:44.33 | delox99 | debian |
| 06:44.38 | delox99 | sarge |
| 06:44.48 | axscode | hmmm ok.. |
| 06:44.50 | delox99 | was running on 2.4 |
| 06:45.00 | axscode | you upgrade it to 2.6 |
| 06:45.05 | delox99 | i compiled a custom kernel 2.6.10 |
| 06:45.06 | delox99 | yeah |
| 06:45.11 | axscode | ic... |
| 06:45.21 | delox99 | and then install zaptel asterisk etc.... |
| 06:45.26 | delox99 | all was working well |
| 06:46.04 | delox99 | now i need to add a ups to the server |
| 06:46.35 | delox99 | i had problems with usb and found out that the required kernel modules for usb were not comoiled |
| 06:46.39 | delox99 | compiled |
| 06:46.47 | delox99 | so i recompiled the 2.6.10 kernel |
| 06:47.08 | axscode | you dont have to compile the module for USB |
| 06:47.14 | axscode | it must be an add-up module |
| 06:47.17 | delox99 | i got the usb working like a charm but at the price of losing zaptel and ztdummy |
| 06:47.28 | axscode | modprobe only |
| 06:47.31 | delox99 | noe meetme is down |
| 06:47.33 | delox99 | bad thing |
| 06:47.47 | axscode | try removeing it again from the kernel. |
| 06:48.01 | axscode | the try modprobe.. kernel module only.. dont install it |
| 06:48.33 | axscode | recompile without USB... then modprobe your USB |
| 06:49.18 | delox99 | well i cannot modprobe the module if they are not installed no? |
| 06:49.24 | delox99 | modprobe zaptel you mean? |
| 06:49.48 | tzafrir_laptop | delox99, BTW: I'm uploading my 1.2 debs now. Slightly modified Sid des, reuilt on Sarge |
| 06:50.24 | tzafrir_laptop | Which USB modules were missing? in the default Sarge kernel? |
| 06:50.58 | delox99 | ohci and uhci support in 2.6.10 custom kernel i built from scratch |
| 06:51.19 | delox99 | only ehci support was included by default |
| 06:51.22 | tzafrir_laptop | BTW: why 2.6.10? |
| 06:51.51 | tzafrir_laptop | because it was the latest when you built it? |
| 06:52.01 | delox99 | because i have an asus a8n-sli mobo |
| 06:52.07 | delox99 | and had probs with sata |
| 06:52.13 | delox99 | using 2.6.8 |
| 06:53.34 | delox99 | i rm misc to misc.bak then mkdir misc and chmod it to 777 |
| 06:53.39 | tzafrir_laptop | anyway, for future reference, my debs are being uploaded to http://rapid.dotsrc.org/experimental/ |
| 06:53.41 | delox99 | recompile zaptel |
| 06:53.52 | delox99 | make clean make linux26 make install |
| 06:54.02 | tzafrir_laptop | chmod 777? do you try to make install as non-root? |
| 06:54.27 | delox99 | no just to make sure its wide open |
| 06:55.04 | tzafrir_laptop | Nothing under /lib/modules should be "wide open". This should be writable to root and root alone. |
| 06:55.20 | delox99 | ls /lib/modules/2.6.10.051126/misc is empty |
| 06:55.28 | delox99 | all the install scripts ran normally |
| 06:55.33 | tzafrir_laptop | In fact, I believe modprobe will fail to load modules not owned by root. Or at least some versions of modprobe |
| 06:56.11 | tzafrir_laptop | what about /lib/modules/2.6.10.051126/extra/ ? |
| 06:56.14 | delox99 | whats the rights on yours? |
| 06:56.33 | delox99 | i dont see extra |
| 06:56.47 | delox99 | isaw that extra in the install script |
| 06:57.01 | delox99 | was that an old path for zaptel on previous versions? |
| 06:57.11 | delox99 | or is it on the newst versions? |
| 06:57.20 | tzafrir_laptop | 755 |
| 06:57.57 | tzafrir_laptop | extra is the default path used by the 2.6 build system. misc was the old convention |
| 06:58.22 | tzafrir_laptop | In fact, it can be anywhere under /lib/modules/`uname -r` |
| 06:59.09 | tzafrir_laptop | copy it manually to /lib/modules/`uname -r`/mycustom/ and run depmod -a |
| 06:59.22 | tzafrir_laptop | Also: |
| 06:59.38 | tzafrir_laptop | find /lib/modules/`uname -r`/ -name zaptel.ko |
| 06:59.51 | delox99 | can i take it in /lib/modules/old2.6.10kernel and put it in the new one? |
| 07:01.29 | tzafrir_laptop | delox99, no. don't mix different kernel versions |
| 07:04.16 | delox99 | from where should depmod -a be run? |
| 07:04.34 | delox99 | usr/src |
| 07:04.39 | delox99 | ? |
| 07:04.41 | tzafrir_laptop | doesn't matter |
| 07:04.43 | delox99 | or doesnt matter? |
| 07:04.46 | delox99 | ok |
| 07:06.31 | delox99 | hehe i copied the whole misc dir from /lib/modules/old kernel to new kernel and now it loads and everything works!!!! |
| 07:07.54 | delox99 | i understand it should not be done but i just dont know what else to try |
| 07:10.15 | moonwick | bloody hell. other than specifying the URL for a firmware .bin in the "Upgrade rule" portion of the SPA web utility and selecting yes for "upgrade enable", what else do I need to do to get this silly SPA-841 to do an upgrade? |
| 07:12.07 | *** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st) |
| 07:12.17 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:14.30 | moonwick | ooh, hey, there we go |
| 07:14.35 | moonwick | a URL to force it to upgrade. |
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| 07:34.49 | sandra78 | help pls I don't know why my extensions always ringing after the pstn hangup i have busydetect=yes |
| 07:34.49 | sandra78 | busycount=3 but nothing the extensions ringing 8 times more always. :( |
| 07:38.25 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
| 07:38.44 | ComputerWarm | anyone here built a php billing system? |
| 07:38.50 | ComputerWarm | if so please message me i have a question |
| 07:39.48 | tessier_ | People generally don't like to help out their competition it seems |
| 07:40.00 | tessier_ | But if the competition is using php they probably have little to fear. ;) |
| 07:40.12 | ComputerWarm | lol very true |
| 07:44.26 | JamesDotCom | haha |
| 07:44.35 | JamesDotCom | what's the question? |
| 07:44.39 | JamesDotCom | i know php and i know billing systems |
| 07:45.18 | ComputerWarm | JamesDotCom i need to get some pointers on the math part of it. |
| 07:45.41 | JamesDotCom | gah |
| 07:45.43 | JamesDotCom | ask the question! |
| 07:45.48 | ComputerWarm | billing 30/30 how do i get it to figure out the math... like i mean i guess round it off at the next highest number |
| 07:47.34 | JamesDotCom | so you want to see how many 30 second blocks there are in the duration of a call? |
| 07:47.39 | JamesDotCom | rounded up to the nearest 30 seconds? |
| 07:49.14 | ComputerWarm | i mean so say its 19.4 it will go for 20 even |
| 07:50.11 | ComputerWarm | JamesDotCom actually the way you put it... is exactly what i need |
| 07:51.53 | JamesDotCom | well if you want to round a float up to the nearest integer, you want ceil |
| 07:52.03 | JamesDotCom | and you'll want to look up the % operator |
| 07:52.22 | JamesDotCom | if you're having trouble with this, good luck with the rest of the billing system :P |
| 07:53.35 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:53.41 | ComputerWarm | na that was the only part i couldn`t get my heard around thank you |
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| 07:57.54 | newmember | I cant even RTFM the answer, I cant find teh default login for the 'panel' on AAH |
| 07:57.58 | newmember | any ideas |
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| 08:06.30 | Math` | any ways of handling user-input timeouts in a disa? |
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| 08:10.08 | Smi|k | has anyone here managed to create a system where users in a hold queue that have accounts on file are able to press dtmf tones to bid their way higher in queue position where it is just a bidding war at the top for next? |
| 08:10.59 | JamesDotCom | wtf |
| 08:11.10 | JamesDotCom | bidding for a phone queue position? |
| 08:11.11 | Smi|k | "you are in position X and the current bid is Y dollars, please enter your new bid and press the # key" |
| 08:11.19 | JamesDotCom | ahh |
| 08:11.23 | JamesDotCom | specialty service |
| 08:11.26 | JamesDotCom | was gonna say |
| 08:11.26 | Smi|k | yes, I am trying to deal with customers who are waiting for tech support and getting pissed off |
| 08:11.26 | JamesDotCom | haha |
| 08:11.32 | JamesDotCom | oh |
| 08:11.34 | Smi|k | so some want to pay their way to the front I guess? |
| 08:11.50 | JamesDotCom | that's a good idea for business? |
| 08:12.02 | Smi|k | it seems to work for google adwords |
| 08:12.10 | JamesDotCom | wouldn't it be a better investment to get another staff member? |
| 08:12.17 | JamesDotCom | uhh, kinda different market here |
| 08:12.23 | Smi|k | bid on position etc..etc.. |
| 08:12.34 | Smi|k | its just a buncha guys who want to talk to phone sex operators |
| 08:13.01 | JamesDotCom | eh? |
| 08:13.06 | JamesDotCom | is it for tech support, or phone sex? |
| 08:13.16 | Smi|k | I think of them as one and the same |
| 08:13.27 | JamesDotCom | haha |
| 08:13.28 | Smi|k | phone operators who help callers fix problems |
| 08:13.45 | JamesDotCom | you dont see a difference in market? |
| 08:14.02 | Smi|k | not really |
| 08:14.17 | Smi|k | I guess its mostly guys in tech support and mostly girls in phone sex |
| 08:14.31 | Smi|k | so you notice a difference when you walk through the call center, but other than that I consider them the same |
| 08:15.34 | JamesDotCom | so is it tech support where you call them up and ask any question you like, or tech support as in a company's support line for a product? |
| 08:15.36 | Rawplayer | MI |
| 08:15.58 | Smi|k | questions and answers |
| 08:16.09 | Smi|k | the person on the phone is helping you accomplish something |
| 08:16.14 | Smi|k | = technical support |
| 08:16.26 | JamesDotCom | that's not what i asked |
| 08:16.27 | JamesDotCom | nevermind |
| 08:16.35 | Smi|k | doesnt matter if you get off on your computer working, or get off on your hand, you can still bid your way up in line, no one likes to wait |
| 08:17.12 | JamesDotCom | you dont get what i'm trying to say, and you wont explain what you're trying to say, so dont worry |
| 08:17.27 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
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| 08:17.33 | JamesDotCom | and it would be relatively easy to accomplish what you're asking |
| 08:17.52 | Smi|k | has anyone done it yet? |
| 08:18.35 | Smi|k | its not efficient to have no wait time, means WAyyyy too many reps, yet some people really could care less spending $3 to wait 5min less |
| 08:20.56 | JamesDotCom | yes, but it also means that someone unwilling to pay may never get through to the operator at the expense of others jumping the queue |
| 08:21.09 | JamesDotCom | which could work in some industries |
| 08:21.13 | JamesDotCom | but would be awful for others |
| 08:21.23 | JamesDotCom | which is why i asked |
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| 08:23.50 | Smi|k | willing being the key word |
| 08:24.05 | Smi|k | they may re-evaluate the cost of their own time after holding a little longer |
| 08:24.14 | Smi|k | its just like China. |
| 08:24.43 | lme | anyone got experience with junghanns's quad bri ? |
| 08:25.07 | ComputerWarm | Smi|k which company do you work for/own if you don`t mind me asking |
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| 08:26.09 | JamesDotCom | for a company which sells a product, a reasonable amount of technical support would be expected... if you called that company up and ended up having to bid to get support, with no guarantee that your call could be answered... well i'd think that would be disasterous for a company's reputation at the expense of hiring another staff member to answer the phone |
| 08:26.11 | ComputerWarm | lol i don`t |
| 08:26.14 | Ikarus | Sex lines are pathetic |
| 08:26.16 | JamesDotCom | which is completely different |
| 08:26.42 | Smi|k | they bid on the people |
| 08:26.43 | JamesDotCom | to someone desperate to rub one out at almost any cost |
| 08:26.52 | Smi|k | there is webcams looking at the girls but with no audio |
| 08:27.01 | JamesDotCom | Smi|k: in which case, that makes perfect sense |
| 08:27.03 | Smi|k | so the guys sit and watch and wait |
| 08:27.07 | Smi|k | then they call and bid to talk |
| 08:27.12 | JamesDotCom | but not for a company's technical support line |
| 08:27.15 | JamesDotCom | well |
| 08:27.18 | JamesDotCom | not for most |
| 08:27.34 | Smi|k | that way girls are never left not talking |
| 08:27.48 | Smi|k | in slower times they are all still getting paid per minute |
| 08:27.48 | Junbug | sex lines are very pathetic... live internet sex chat is somethin else :) |
| 08:27.56 | Qwell | tzafrir_laptop: I made one using festival |
| 08:28.01 | Smi|k | maybe not $3 but $.3 |
| 08:28.07 | Smi|k | but paid for the time none the less |
| 08:28.19 | Ikarus | tzafrir_laptop: an erotica elisa ? |
| 08:28.28 | drray | it's more for lonely people |
| 08:28.28 | tzafrir_laptop | yes |
| 08:28.30 | Smi|k | its the best way to use everyones time and money in the most efficient way |
| 08:28.33 | drray | not horny people |
| 08:28.44 | Smi|k | so it should apply to tech support too looked at from that standpoint |
| 08:29.17 | Smi|k | factor it out of the product price and the customer saves money on the product cost becuase they know the support costs are handled in the most efficient way |
| 08:29.59 | tzafrir_laptop | Ikarus, though the text part seems the simplest part. You still need TTS etc. |
| 08:30.04 | Qwell | "Your call is thi..fo..ninth in line." |
| 08:30.20 | Ikarus | tzafrir_laptop: TTS is the simple bit, voice recognition is the problem |
| 08:30.28 | Smi|k | Qwell, that would piss people off! |
| 08:30.48 | Smi|k | no one ever moves backwards in line, only forwards, at a slower pace |
| 08:30.48 | Qwell | "Press 1 to pay $1 to jump to position 8." 1 "You are now 8t...12th in line." |
| 08:31.10 | Smi|k | that way EVERYONE can pay to be #1 |
| 08:31.16 | Smi|k | then they wait for that one guy before them to finish up |
| 08:31.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:31.38 | Smi|k | its pretty much just there to pull in some extra $ and keep the callers busy |
| 08:32.09 | Smi|k | I think I'm going to use the same system on the casino accounting department queues too now that I think about it |
| 08:32.41 | Smi|k | maybe with a hint of random mixed in, bet your position higher |
| 08:37.32 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 08:42.28 | chapeaurouge | hi all |
| 08:43.05 | chapeaurouge | im calling a voip provider this afternoon... what the top 5 questions i need to ask to make sure the guys are good? |
| 08:43.19 | axscode | lol |
| 08:43.20 | chapeaurouge | this is to provide voip for a small buness of 50 ppl (the one i work in) |
| 08:46.19 | newmember | i'll ask, why do you want a voip provider? |
| 08:46.48 | drray | if you already have decent zap handsets, keep those |
| 08:46.50 | denon | chapeaurouge: www.nufone.net -- we already know they're good |
| 08:47.27 | chapeaurouge | denon, heh :) thx for the pointer. however, i think my director will like it better if i use a "local" provider :X |
| 08:47.44 | denon | oh yeah, cause AT&T is local.. |
| 08:47.55 | denon | unless you mean for local DIDs.. |
| 08:47.58 | drray | why would a local provider be better? |
| 08:48.08 | chapeaurouge | support |
| 08:48.11 | chapeaurouge | ? |
| 08:48.11 | denon | heheheh |
| 08:48.16 | drray | unless you want someone to sue in small claims court |
| 08:48.27 | denon | there's no reason a local provider would be any better |
| 08:48.29 | chapeaurouge | still better to call someone in your timezone than 9 hours behind you |
| 08:48.45 | denon | you'll get the same support |
| 08:48.51 | *** join/#asterisk kernelbee (n=Naveed@80.77.12.2) |
| 08:49.03 | denon | besides, the only support you need from a voip provider is "are you up or are you down?" |
| 08:49.11 | denon | everything else should be seamless |
| 08:49.50 | newmember | when was the last time someone got support from a teleco after business hours? |
| 08:53.44 | Ikarus | newmember: last time they killed my line |
| 08:54.18 | *** join/#asterisk SparFu1 (n=player@tor/session/x-0f6c28d01ac22c2c) |
| 08:55.05 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 08:59.21 | h3x | denon: it aint that simple if they are coloed in a whore house |
| 08:59.37 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:00.33 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:01.30 | Smi|k | chapeaurouge, ask them if they provide the bandwidth or you do, ask them if they have an end-to-end SLA and response time to downed systems |
| 09:02.10 | Smi|k | make sure they do not use some kind of wireless link, and ask if a generator in a centrally located area of your office can provide the power to run the hardware/phones so you can still work if the power is out |
| 09:02.50 | Smi|k | and make sure they are not trying to sell buckets of minutes or flat rate $xx per user |
| 09:08.04 | chapeaurouge | Smi|k, ok thx for that |
| 09:10.13 | Smi|k | no prob |
| 09:14.50 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 09:38.15 | *** join/#asterisk lorinc (n=ang@caracas-2086.adsl.interware.hu) |
| 09:45.23 | *** join/#asterisk _omer (i=st@203.215.180.250) |
| 09:45.29 | _omer | hi |
| 09:45.59 | *** join/#asterisk LANmower (n=LANmower@ndn-165-114-43.telkomadsl.co.za) |
| 09:47.09 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.42) |
| 09:47.49 | _omer | I have configured my Asterisk to save CDR in MS-SQL....now I want to get the SIPPEERS from the same Database...any help??? |
| 09:48.24 | LANmower | hi all :) |
| 09:50.20 | SparFu1 | hi lan. |
| 09:50.30 | LANmower | I need help, I'm reading numbers into variables with read, but I need a way to detect timeouts. |
| 09:51.01 | LANmower | customers want, 'you have taken to long, try again' |
| 09:51.47 | _omer | any help on realtime SIP PEERS? |
| 09:53.24 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:54.57 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:00.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:00.34 | puzzled | morning |
| 10:02.57 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:05.05 | _omer | any help on realtime SIP PEERS? |
| 10:06.29 | lme | nope sorry |
| 10:07.19 | _omer | oh NO! :) |
| 10:09.19 | LANmower | _omer, what is wrong? |
| 10:10.44 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 10:10.47 | _omer | LANmower: My Asterisk is successfully connected to MS-SQL ...as It insert the CDR in MS-SQL....now I have created a table sip_conf and created a peer in it...have reloaded asterisk... |
| 10:11.01 | _omer | but unable to register by using this DB Peer... |
| 10:11.34 | _omer | have made the required changes in extconfig.conf as well to get the SIPPEERS and SIPUSERS |
| 10:12.11 | LANmower | are you aware that MS-sql is not the same as using postgres and mysql |
| 10:12.12 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 10:12.29 | LANmower | perhaps you should give those a go, theyre more reliable. |
| 10:12.56 | _omer | but I dont think so problem is in using MS-SQL .... |
| 10:13.14 | _omer | as asterisk just need to use ODBC and get the SIPpeers from given database and table |
| 10:13.29 | LANmower | also, where cdr is concerned, you shouldnt create the peers. |
| 10:13.46 | LANmower | asterisk creates them, all you ever do is insert the initial table. |
| 10:14.24 | _omer | I have created a table for SIPPEERS .... which is sip_conf and mentioned it in extconfig.conf |
| 10:15.47 | LANmower | _omer, previously, the only setup I had to do to get cdr working in asterisk was creating a cdr database and inserting its table, asterisk wrote to it fine using both postgres and mysql, I have no experience the other things though... |
| 10:16.00 | *** join/#asterisk koperniqs (n=koperniq@129.187.15.52) |
| 10:16.01 | LANmower | sorry. |
| 10:17.47 | _omer | MY asterisk is also working fine with MS-SQL .... I use unixODBC and FreeTDS .... |
| 10:17.52 | _omer | anyways....thanks :) |
| 10:17.57 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:20.34 | RoyK | hæpp |
| 10:24.18 | _omer | RoyK : :) |
| 10:24.37 | _omer | what are you looking at ... just help me out :p |
| 10:26.12 | kernelbee | hi, through app_queue, I wana block dialing when number is busy. anyone know how I can acheive this.. |
| 10:26.33 | Ikarus | Hrm, insane, our current telco is charging us 136 euro per month, the new one 16 euro |
| 10:27.27 | Ikarus | I asked my boss if I could get 50% of what I save as a raise |
| 10:27.30 | Ikarus | this is going to be fun |
| 10:27.49 | jahani2 | hi |
| 10:28.03 | jahani2 | this can work fine with asterisk ? like FXO card ? http://www.ges.fr/voip/product_info.php?products_id=213&osCsid=5a718500b8928a7a863b41543e169037 |
| 10:29.12 | Ikarus | jahani2: considering it supports SIP, I would expect so |
| 10:29.28 | jahani2 | ok thank you |
| 10:30.17 | kernelbee | jahani2: hopefully this gateway will work with asterisk, if comany have good support ;) |
| 10:31.25 | LANmower | so any help on my read problem? |
| 10:31.48 | *** join/#asterisk Abbas (i=Abbas@203.81.220.62) |
| 10:36.55 | RoyK | _omer: I beleive in a true, segreative unix world. mixing mickysoft into this is EVIL |
| 10:39.22 | Ikarus | ah yes, Unix, the radical islam of the software world ;) |
| 10:40.15 | *** join/#asterisk prh (n=paul@212.13.203.80) |
| 10:41.15 | jahani2 | LANmower not know if they have good support but i prefer buy this with 6 ports at 450euro than buy FXO card 2 ports at 350 euro :) |
| 10:44.02 | _omer | RoyK : :) still evil??? if asterisk is successfully connected to MS-SQL which means ODBC is working fine. |
| 10:47.58 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 10:48.00 | shadebob | hi |
| 10:49.56 | *** join/#asterisk d-tech (n=dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 10:51.55 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:54.03 | LANmower | jahani2, um... |
| 10:54.17 | LANmower | did you talk to me? or typo? |
| 10:55.17 | LANmower | _omer, saying that shows how little you know, every version of sql has different syntax, software has to be designed to specefically use certain ones in most cases. |
| 10:56.49 | Ikarus | Urgh, heart attack, I just compared a ready made solution vs asterisk |
| 10:56.59 | _omer | LANmower: you must know that SQL Syntax is same whether you talk about MYSQL or MS-SQL or MS_Access.... |
| 10:57.13 | *** join/#asterisk dummie (n=dummie@202.155.89.122) |
| 10:57.18 | iDunno | Ikarus: oh? and which one got more of your attention? |
| 10:57.50 | _omer | Software needs to be designed for specific ODBC..... |
| 10:57.51 | LANmower | _omer, no its not |
| 10:57.57 | _omer | ok U WIN!!! |
| 10:58.01 | dummie | why when i compile asterisk addons (cdr_addon_mysql), it comes up "-lmysqlclient" not found |
| 10:58.06 | dummie | can somebody help? |
| 10:58.06 | LANmower | there are only 4 commands that are the same |
| 10:58.09 | Ikarus | iDunno: the price of the ready made solution, starting for the same number of phones at 1200 euro excluding the actual phones, my solution is 1000 euro, including the phones |
| 10:58.15 | LANmower | the rest are dependant |
| 10:58.30 | RoyK | dummie: because you don't have the mysqlclient lib |
| 10:58.45 | iDunno | Ikarus: what phones are you planning on using? |
| 10:58.46 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 10:58.47 | LANmower | I know, I develop a lot of software in sql (even if its against my will) |
| 10:58.52 | Ikarus | iDunno: BudgeTone |
| 10:59.01 | iDunno | Ikarus: not the 101/102? |
| 10:59.05 | Ikarus | iDunno: the 101 |
| 10:59.14 | iDunno | Ikarus: they're a little "unreliable" in my experience ;) |
| 10:59.19 | Ikarus | iDunno: unreliable ? |
| 10:59.26 | iDunno | Ikarus: we've just replaced one of them with a Snom 190 :) |
| 10:59.41 | iDunno | Ikarus: don't *always* get a hang up from them, so sometimes get phantom calls. |
| 10:59.57 | Ikarus | iDunno: oh, flash problems |
| 11:00.16 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 11:00.18 | RaYmAn-Bx | tak |
| 11:00.24 | iDunno | Ikarus: also, they have the worlds *worst* set of buttons, the display is not entirely nice, and they really do feel cheap and nasty. |
| 11:00.30 | LANmower | odbc is also specific to microsoft products. |
| 11:01.11 | iDunno | Ikarus: but that's just my experience of them so far, I wouldn't use them in an office environment that's going to give them lots of use again... |
| 11:01.22 | iDunno | Ikarus: but for an occasional use phone, they're alright. |
| 11:01.37 | Ikarus | iDunno: I might use more expensive phones at the 2 places they are used alot, but this is a school |
| 11:01.55 | Ikarus | no one except the secretary, administration and ICT are supposed to use the phone alot |
| 11:02.22 | iDunno | Ikarus: ahh - for a school they're probably not wanting to do anything really taxing, so they should be fine :) |
| 11:04.52 | LANmower | _omer, its not about winning or losing, its about learning something. |
| 11:05.27 | Ikarus | iDunno: well, we also have no budget |
| 11:05.41 | iDunno | uh huh |
| 11:06.02 | Ikarus | Which is why any real phone is out of range (the next decent phone is twice as expensive) |
| 11:06.04 | iDunno | Ikarus: neither did we, hence the 4 BudgeTone 101s that are sitting in the London office ;) |
| 11:06.22 | iDunno | I do like the Snom 190 though - it's much more suited to office use :) |
| 11:06.39 | iDunno | but it's *lots* more expensive per phone. |
| 11:06.53 | _omer | I will definitely let you know when I make my asterisk working with MS-SQL in real time :) .... and then I'll refer you a book |
| 11:06.55 | Ikarus | iDunno: ah well, I don't have any issues with the BudgeTone, and with the extra savings I managed to get it is not a big issue to maybe upgrade 2 phones to Snom 190's |
| 11:07.05 | Ikarus | Although, can Snom 190's do Early Dial ? |
| 11:07.10 | iDunno | the budgetone would be nice if it was smaller - it's too big for the lack of functionality ;) |
| 11:07.27 | iDunno | Ikarus: not sure, never been worried about early dial ;) |
| 11:07.39 | LANmower | _omer, I'm not saying it doesnt, I'm saying its unlikely and you should try setting it up with something its kown to work with first, then doing the same thing on ms-sql to see if it works. |
| 11:07.44 | Ikarus | iDunno: agreed, it is a big phone |
| 11:07.55 | Ikarus | iDunno: I am using Early dial on the BudgeTones and have some special dial plan mods for it |
| 11:08.02 | iDunno | ahh - right. |
| 11:08.05 | LANmower | Would someone care to inform me how to drop the end of a string in the dialplan?! this is driving me crazy :) |
| 11:08.32 | LANmower | 32(drop the last number) |
| 11:08.38 | iDunno | I'm not using early dial on any of the phones, the phones do a 2 -> 5 second timeout (depending on the primary user), or they can hit send or # to lose the timeout |
| 11:08.43 | _omer | LANmower: why didn't you say that at first? :p |
| 11:16.06 | LANmower | _omer, I'm sorry, I'm a terrible scatterbrain :) |
| 11:16.40 | LANmower | I was chatting to you while taking a phone call, eating and trying to chop off the end bit of a string in a dialplan. |
| 11:16.54 | LANmower | wich seems to be no mean feat |
| 11:18.05 | _omer | :-D |
| 11:19.24 | mcnobody | LANmover: SetVar(temp=Foobar) |
| 11:19.25 | Ikarus | LANmower: so you are either female or a geek |
| 11:19.32 | mcnobody | LANMover: NoOp(${temp:0:$[${LEN(${temp})} - 1]}) |
| 11:19.35 | shadebob | make update with cvs still working? How I can make an update with svn? |
| 11:19.43 | LANmower | Ikarus, I'm an artist :) |
| 11:19.50 | LANmower | that explains it completely |
| 11:20.10 | LANmower | mcnobody, thats some decent script-fu :) |
| 11:20.36 | LANmower | sheesh, now I gotta try and figure out what it all means first... lol |
| 11:20.39 | Ikarus | What math do telcos use to make 21.95 * 4 equal 136.14 |
| 11:21.11 | LANmower | Ikarus, its likely the same math they use to charge overseas rates. |
| 11:21.45 | Ikarus | LANmower: ah, great, because my boss is asking questions |
| 11:22.05 | LANmower | lol :) |
| 11:22.24 | dummie | i've install asterisk-addons 1.2.0 for mysql odbc support |
| 11:22.42 | dummie | but still don't have the command "odbc" appeared |
| 11:22.48 | dummie | can somebody help please? |
| 11:23.11 | dummie | "odbc" command still doesn't appear in my asterisk CLI |
| 11:23.52 | LANmower | ${EXTEN:0:$[${LEN(${EXTEN})} -1]} somehow it makes more sence with EXTEN :) |
| 11:24.00 | LANmower | same thing, dunno why |
| 11:24.50 | mcnobody | But you didn't say you want strip the end of the extension.. You said string.. =) |
| 11:25.24 | LANmower | mcnobody, er, cause I figured it would come down to the same thing :) |
| 11:26.07 | LANmower | I'm only doing it like this because read doesnt report on timeouts, or I cant seem to make it do that. |
| 11:26.21 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 11:26.39 | LANmower | and now I'm trying to do WaitExten do the same thing as Read. |
| 11:26.55 | LANmower | so gotta cut off the '#' |
| 11:33.35 | p1tst0p | can anyone point me in the direction of a web based cdr viewer, my cdr is going into mysql. ( i thought there was a php cdr_mysql reporting thing somewhere.. ) |
| 11:34.55 | LANmower | p1tst0p, astcc |
| 11:35.00 | LANmower | on the digium ftp site |
| 11:35.21 | LANmower | or look for the official areski website, they have a good cdr viewer too |
| 11:35.37 | LANmower | its like the cdr bit of areski cut out. |
| 11:36.36 | LANmower | SetVar(AMTC,${EXTEN:0:$[${LEN(${EXTEN}) -1]}) sets AMTC to the number including the last. |
| 11:39.29 | LANmower | oh my bad |
| 11:39.30 | LANmower | wait |
| 11:39.41 | mcnobody | LANmower: It shoud be ..AMTC=${EX... |
| 11:39.45 | LANmower | yeah |
| 11:39.52 | LANmower | bash scripting kicking in there :) |
| 11:39.54 | mcnobody | That works for me.... |
| 11:40.32 | LANmower | my problem was the square bracket instead of the curly one. |
| 11:41.42 | LANmower | er I mean the missing bracket... |
| 11:43.35 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 11:46.04 | *** join/#asterisk gvag11 (n=g@84.254.11.90) |
| 11:46.32 | gvag11 | hi all |
| 11:47.23 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 11:47.29 | shanky | hi |
| 11:50.21 | *** join/#asterisk ful|work (n=fulgas@209.8.233.247) |
| 11:54.52 | LANmower | theoretically this should be correct: ${EXTEN:0:$[${LEN(${EXTEN})} -1]}... but it doesnt remove anything |
| 11:57.09 | *** join/#asterisk rob_- (n=robb@matrix.netsoc.tcd.ie) |
| 12:03.37 | *** join/#asterisk zoa (n=zoa@cust-5-0.dsl.versateladsl.be) |
| 12:04.13 | Zeeek | how about -2 ? |
| 12:05.34 | *** join/#asterisk eivindtr (n=wingnut-@194.248.208.94) |
| 12:05.53 | mcnobody | LANMower: which version of * you are using? |
| 12:06.01 | LANmower | 1.0.9 :) |
| 12:06.05 | LANmower | they dont have 1.2.0 |
| 12:06.26 | mcnobody | LANmower: I have 1.0.9-BRIstuffed-0.2.0-RC8n |
| 12:06.46 | mcnobody | LANmower: and that is working correctly |
| 12:06.58 | mcnobody | exten => 111,1,SetVar(AMTC=${EXTEN:0:$[${LEN(${EXTEN})} - 1]}) |
| 12:06.58 | mcnobody | exten => 111,2,NoOp(${AMTC}) |
| 12:06.58 | mcnobody | exten => 111,3,Hangup |
| 12:07.34 | mcnobody | LANmower: This prints '11' on console on NoOp. |
| 12:07.51 | *** join/#asterisk _foxfire_ (n=kvirc@cica-adm.fe.up.pt) |
| 12:07.52 | LANmower | is that the same line of text as mine!? |
| 12:08.11 | koperniqs | hi! i've got a problem with my snom360 not sending DTMF-tones when using g.729 |
| 12:08.18 | mcnobody | LANmower: It's copied from my dialplan. |
| 12:09.09 | mcnobody | LANmower: but it look likes just same... |
| 12:10.23 | mcnobody | LANmower: Try to print out the variable with NoOp before and after the SetVar row. |
| 12:12.07 | mcnobody | LANmower: And keep in mind that this row does _not_ change the variable EXTEN content. It just returns the EXTEN's content without last character. |
| 12:13.44 | LANmower | yeah |
| 12:14.12 | LANmower | this is mine, I'm gonna try yours now |
| 12:14.17 | LANmower | exten => _.#,1,SetVar(AMTC,${EXTEN:0:$[${LEN(${EXTEN})} -1]}) |
| 12:14.33 | *** part/#asterisk rob_- (n=robb@matrix.netsoc.tcd.ie) |
| 12:15.17 | mcnobody | LANmower: The ',' after AMTC shoud be '=' |
| 12:15.38 | LANmower | mcnobody does the same thing |
| 12:15.42 | mcnobody | ok |
| 12:15.57 | LANmower | its a function that takes two variables, it could also be a | |
| 12:16.21 | mcnobody | LANmower: but your way is not working for me.... |
| 12:16.28 | LANmower | lol |
| 12:16.44 | mcnobody | LANmower: If I change the ',' to '=' it's working... |
| 12:16.49 | LANmower | ler me try again, I may have forgotten to restart with yours. |
| 12:17.32 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 12:17.43 | mcnobody | LANmower: And if I change my '=' to ',' the variable AMTC would be empty... |
| 12:19.14 | LANmower | no its not empty |
| 12:19.18 | LANmower | its just the whole string |
| 12:19.44 | LANmower | I have about 50 setvars in that dialplan thats working with ',' |
| 12:19.48 | LANmower | bristuffed 0.2.0 |
| 12:19.58 | mcnobody | LANmower: No. You are setting AMTC somewhere before if it's not empty.. |
| 12:19.58 | LANmower | RC8o |
| 12:20.05 | LANmower | nope |
| 12:20.14 | mcnobody | LANmower: wierd.. |
| 12:20.22 | LANmower | I have one setvar for every of the other 50 too, all with ',' |
| 12:20.29 | LANmower | always did it that way |
| 12:20.38 | LANmower | can also use | |
| 12:21.06 | mcnobody | LANmower. Ok. But on my dialplan example if I change the = to , the AMTC is empty on NoOp command. |
| 12:21.23 | mcnobody | And I have 1.0.9-BRIstuffed-0.2.0-RC8n |
| 12:21.53 | LANmower | perhaps its the space I have after the , I dunno :) |
| 12:22.42 | LANmower | if you look at verbose 4, it turns the = into a , anyway |
| 12:23.16 | LANmower | oh no it doesnt actually... |
| 12:23.25 | LANmower | but it does the same thing here |
| 12:24.06 | mcnobody | Perhaps the string parser library is different... |
| 12:24.49 | LANmower | meybe so, I have been doing it like this since 1.0.2 |
| 12:29.40 | mcnobody | LANmower: The problem could be the # character... |
| 12:30.13 | mcnobody | LANmower: It makes my dialplan working very wierd... |
| 12:33.18 | mcnobody | LANmower: _.# seems to match everything in my *. |
| 12:33.49 | mutilator | pressing * in Voicemail() should drop you to exten => a correct? |
| 12:34.02 | mutilator | there any reason it wouldn;t? |
| 12:34.12 | LANmower | mcnobody, I think youre right |
| 12:34.26 | Ikarus | <PROTECTED> |
| 12:34.35 | LANmower | so problem unsolved then, I cant make asterisk read into variables and go to an extension on timeout. |
| 12:34.43 | LANmower | shit |
| 12:34.47 | mcnobody | LANmower: of course I'm =) |
| 12:35.22 | LANmower | this is like the one bloody thing you'd expect asterisk to be able to do... :/ |
| 12:37.23 | Ikarus | Asterisk can do anything |
| 12:38.07 | LANmower | wait a second... hmm. |
| 12:38.10 | LANmower | no |
| 12:38.18 | LANmower | yeah! |
| 12:38.23 | LANmower | what about settimeout and read |
| 12:38.33 | LANmower | let me see |
| 12:42.33 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.137) |
| 12:45.42 | LANmower | look at this |
| 12:46.14 | LANmower | responsetimeout doesnt go to t if theres a read, it just goes to the next extension. |
| 12:54.39 | Druken | LANmower: what exactly are you trying to do ? |
| 12:55.21 | oej | responsetimeout goes to t at timeout of next read action. Responsetimeout just sets a timer, does not read anything |
| 12:55.38 | LANmower | oej, responsetimeout affects the read command |
| 12:55.54 | oej | Yes, and background and all other commands that waits for a DTMF response |
| 12:56.03 | LANmower | it makes read time out if its in read, instead of going to t it then goes to the next extension. |
| 12:56.23 | oej | Ah, so the read app is bad. |
| 12:56.24 | LANmower | I have found a solution though. |
| 12:56.31 | Druken | why not just put a time limit on the read? |
| 12:56.49 | fugitivo | responsetimeout is obsolete, Set(Timeout(response)=xx) should be used instead for 1.2 |
| 12:56.55 | LANmower | Drunken because it doesnt go to an extension selectively based on timeouts, just to the next one. |
| 12:57.03 | LANmower | fugitivo, theire not on 1.2 yet |
| 12:57.12 | LANmower | anyway |
| 12:57.16 | LANmower | I found a solution |
| 12:57.34 | LANmower | check if the variable is empty, if it is go to timeout :) |
| 12:57.37 | Druken | glad you found your answer... cause you've confused the shit out of me |
| 12:57.43 | fugitivo | lol |
| 12:58.27 | LANmower | its basically just they didnt give read a 'you took too long to respond' functionality, it passes successfully even if you take too long, just with an empty var. |
| 12:58.47 | LANmower | so theres no go to the 't' extension for it. |
| 13:00.38 | Druken | well, i don't belive READ was intended to be used for menu's... |
| 13:00.43 | Druken | that's what background is for |
| 13:01.16 | LANmower | Drunken its not a menu, I'm reading numbers from the user for bank transfers. |
| 13:01.29 | fugitivo | LANmower: what do you need to do? |
| 13:01.54 | Druken | well, wouldn't you have an error checker in the first place then ? |
| 13:01.54 | LANmower | read a number terminated by hash, say you took too long and loop back to the read if he took too long. |
| 13:02.03 | Druken | if i was dealing with bank accounts, i know i would |
| 13:02.15 | LANmower | Drunken the error checker runs at the end, its a script made by the bankers. |
| 13:02.28 | fugitivo | LANmower: what's wrong with responsetimeout and digittimeout? |
| 13:02.31 | Ikarus | Druken: some bank account numbers don't have an error chec |
| 13:03.02 | RoyK | anyone that knows a good load balancer that supports SIP? |
| 13:03.05 | LANmower | fugitivo, that doesnt make it go somewhere on timeout when youre in read, it just goes further on in the dialplan, wich is the next number he has to enter. |
| 13:03.12 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 13:03.27 | LANmower | RoyK, the linux kernel ^^ |
| 13:03.32 | fugitivo | LANmower: it goes to the t extension |
| 13:03.43 | LANmower | fugitivo, not if youre in read, it just goes to the next one. |
| 13:03.49 | fugitivo | LANmower: digittimeout |
| 13:03.52 | RoyK | LANmower: no sip support there |
| 13:03.54 | LANmower | that too. |
| 13:03.58 | shadebob | I have a problem to communicate between 2 asterisk box. 1st box (server) have a tel [shadebo]. 2nd box (client) have register.... (it's ok with sip show registry) and a [shadebo-out] phone. When I launch call from 2nd to 1st I have a "Failed do authenticate user asterisk"? |
| 13:04.31 | LANmower | RoyK, if you can take sip calls in linux, that means it has sip support. what do you want to balance various sip calls or give sip higher priority than everything else? |
| 13:04.44 | shadebob | fromuser and username as well configured... I don't known why I have user asterisk? |
| 13:04.51 | RoyK | LANmower: please. rtfm. LVS does not support SIP |
| 13:05.39 | LANmower | you havent answered my question |
| 13:06.10 | Druken | oej: pressing * in voicemail() drops you do exten=a,1 |
| 13:06.22 | oej | yes, and? |
| 13:06.24 | fugitivo | LANmower: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout |
| 13:06.45 | LANmower | fugitivo, do you honestly thing I havent tried that? |
| 13:06.49 | *** join/#asterisk Laerte (n=io@195.47.232.200) |
| 13:06.50 | Laerte | hy |
| 13:06.53 | LANmower | thats the first thing I tried. |
| 13:06.58 | Druken | and you asked.... |
| 13:07.04 | Druken | just took me a while to respond |
| 13:07.30 | LANmower | lol drunken :) |
| 13:07.50 | Druken | LANmower, please look very carefully at my name |
| 13:07.58 | LANmower | lol |
| 13:08.07 | fugitivo | LANmower: if you want a timeout when user has to enter a number, and go to t extension if that number doesn't exists, that's what you need, if not, i don't understand what you need |
| 13:09.18 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 13:11.43 | LANmower | fugitivo, scroll up |
| 13:11.47 | LANmower | lol |
| 13:12.08 | fugitivo | my buffer is 10 lines |
| 13:12.25 | LANmower | LANmower fugitivo, theire not on 1.2 yet |
| 13:12.25 | LANmower | LANmower anyway |
| 13:12.25 | LANmower | LANmower I found a solution |
| 13:12.25 | LANmower | LANmower check if the variable is empty, if it is go to timeout :) |
| 13:12.41 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:13.23 | fugitivo | ok, if you want to do it that way |
| 13:14.02 | LANmower | thats exactly what you told me to do, if the var is empty go to t, the only way to do it it seems. |
| 13:14.13 | fugitivo | who? me? |
| 13:14.24 | fugitivo | i said to use digittimeout |
| 13:14.33 | LANmower | fugitivo LANmower: if you want a timeout when user has to enter a number, and go to t extension if that number doesn't exists, that's what you need, if not, i don't understand what you need |
| 13:14.59 | _foxfire_ | hi guys , got a basic isdn question i can't find the answer and i already did rtfm . |
| 13:15.02 | LANmower | your buffer really is 10 lines isnt it :) |
| 13:15.22 | fugitivo | yes, linux console, not X :) |
| 13:15.50 | LANmower | shift-pgup :) |
| 13:16.02 | fugitivo | LANmower: i changed consoles |
| 13:16.28 | LANmower | anyway this works pretty well, so I have no complaints. |
| 13:16.30 | fugitivo | LANmower: anyways, you can use digittimeout, without using variables, the function will do all de job |
| 13:17.06 | LANmower | fugitivo, if you use a digittimeout while in a read, it just times out the read. so if youre in priority 3, it goes to 4 and not t. |
| 13:17.10 | LANmower | thats the problem. |
| 13:17.29 | LANmower | thats why I'm checking weather the variable is empty and using a gotoif on it. |
| 13:17.32 | fugitivo | waht |
| 13:17.39 | fugitivo | you're using Read() or background() ? |
| 13:17.58 | _foxfire_ | i need to interconect our old PBX with asterisk , i would need at least 8 lines . I thought of connecting an ISND E1 PRI card with the old PBX , can i connect both cards ( the old PBX and asterisk ) with an crossover cable like i can do when i have and BRI ? |
| 13:18.01 | *** join/#asterisk coppice (n=chatzill@213.202.17.210.dyn.pacific.net.hk) |
| 13:18.03 | Druken | he's using READ |
| 13:18.03 | LANmower | I'm using read(), background plays a sound, read is something completely different. |
| 13:18.13 | fugitivo | LANmower: well, i didn't know that :) |
| 13:18.22 | LANmower | fugitivo, you feel my pain :) |
| 13:18.31 | fugitivo | ok, i just woke up, don't listen to me |
| 13:18.37 | LANmower | its ok now it works though. thx for the advice anyway. :) |
| 13:18.41 | Druken | fugitivo: e's just explaining what he needs very stupidly :) |
| 13:18.50 | fugitivo | hehe |
| 13:18.52 | Druken | that's my new word for today... |
| 13:19.31 | Druken | read was not designed to FORCE an answer, it was designed to ask, and let sleeping dogs lie |
| 13:20.05 | LANmower | Druken that's what background is for |
| 13:20.06 | LANmower | LANmower Drunken its not a menu, I'm reading numbers from the user for bank transfers. |
| 13:20.06 | LANmower | fugitivo LANmower: what do you need to do? |
| 13:20.13 | fugitivo | ok ok |
| 13:20.21 | LANmower | your memory is as long as your buffer :) |
| 13:21.22 | LANmower | Drunken you are correct, but its a shame asterisk doesnt have a better way of getting input from a user, this case should be common enough to require it. |
| 13:21.25 | Druken | no... bacground is for menus... |
| 13:21.46 | fugitivo | LANmower: you could make a patch for read to using timeouts |
| 13:21.56 | fugitivo | god, what did i write? |
| 13:22.23 | fugitivo | i think i'm going to bed again |
| 13:22.31 | LANmower | fugitivo, I would then have to convince my customers to run a patched asterisk for money transfers. |
| 13:22.43 | fugitivo | LANmower: what's the problem with that? |
| 13:23.00 | LANmower | fugitivo, theyre windows foo's they know nothing. |
| 13:23.28 | fugitivo | LANmower: they're using asterisk at this point, so adding a patch is not difficult |
| 13:23.56 | [TK]D-Fender | If you want to use "t" for reads, make a whole macro with single digit reads, no termination char, and expected return values. Can all be done in pure dial-plan rather easy. In fact you could bybass the "read alltogether and include EVERY SINGLE DIGIT with exten's...... |
| 13:23.59 | [TK]D-Fender | geez! |
| 13:24.56 | LANmower | [TK]D-Fender, sure or I can just add -> exten => s,3,GotoIf($["${CC}" = ""]?t|1:) |
| 13:25.13 | LANmower | lol, that end looks like a smiley |
| 13:25.41 | LANmower | [TK]D-Fender, honestly, I was considering that... |
| 13:26.19 | LANmower | no not really, but hey |
| 13:28.12 | [TK]D-Fender | you can remove the ":" at the end.. it isn't necessary |
| 13:28.33 | LANmower | I just thought of that, but its in there now so I'll leave it :) |
| 13:28.40 | LANmower | looks nice |
| 13:28.41 | [TK]D-Fender | "* .. the land of 20 ways to beat a dead horse" |
| 13:29.17 | [TK]D-Fender | and that doesn't ENTIRELY work... it will be blank if they hit "#" all by itself ;) |
| 13:29.23 | [TK]D-Fender | :O |
| 13:29.41 | LANmower | shit fender youre right |
| 13:29.49 | LANmower | there is an optional var in there at the end. |
| 13:30.01 | LANmower | that one will be required now |
| 13:30.21 | [TK]D-Fender | Read upon "read" :) |
| 13:32.02 | LANmower | well, I guess that one just doesnt get the check |
| 13:32.14 | LANmower | if they take too long on that, they can just bloody well start over. |
| 13:32.24 | [TK]D-Fender | In other circles that would considered "bad". |
| 13:32.46 | [TK]D-Fender | You are failing your users! REPENT! |
| 13:34.11 | [TK]D-Fender | You may yet be saved! (Call your bookie now 'cause the odds are wild!) |
| 13:34.14 | *** join/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
| 13:35.08 | LANmower | follow me brothers and sisters, salvation for just a small donation!!! the more the donation! the more the salvation! step right up!!! |
| 13:35.27 | LANmower | now thats entertainment :) |
| 13:36.08 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:36.50 | mtupper | I cant get * to start, getting this error: [chan_features.so]Nov 29 04:22:51 WARNING[11491]: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version ... I think it was from inadvertantly compiling and installing the zaptel and libpri (following the AMP installation) but I dont have any cards yet... |
| 13:39.44 | LANmower | anyone know much about festval? need to know how to make it read out numbers one by one, like 1 0 0 instead of 100. |
| 13:41.23 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 13:41.27 | [TK]D-Fender | Split up the figures... thats the only way. it reads what it sees. |
| 13:41.59 | LANmower | yeargh |
| 13:42.05 | _foxfire_ | lanmower : i remember it did so when you wrote 1 0 0 instead of 100 , you can also pass your file througt an perl script file |
| 13:42.09 | [TK]D-Fender | Now that you can check a variables length in a function you could just create a macro to split it up. |
| 13:42.40 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 13:42.53 | _foxfire_ | i stoped using festival because it is to slow ... |
| 13:43.19 | _foxfire_ | at least on my machine |
| 13:43.22 | LANmower | _foxfire_, I've managed to cache it quite well with my script |
| 13:43.46 | LANmower | any sentinces that stay the same it just keeps the audio files for |
| 13:44.02 | _foxfire_ | hmm , good thinking ;-) |
| 13:44.21 | LANmower | so just the two lines of dynamic text gets generated and deleted. |
| 13:44.31 | _foxfire_ | but what if the text is dynamic ? |
| 13:45.18 | _foxfire_ | last time i tried it took my machine about 4 secs to get the speech ready |
| 13:45.51 | _foxfire_ | maybe got somethink wrong configurated ... never was able to optimizing it |
| 13:46.55 | _omer | LANmower: you could check SayDigits(100) |
| 13:47.28 | mtupper | can anyone recommend a work around to this? its preventing me from starting up asterisk: WARNING[11491]: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version ..... WARNING[11491]: Loading module chan_features.so failed! |
| 13:49.15 | *** join/#asterisk fugitivo (n=ajf@209.13.248.48) |
| 13:49.59 | _foxfire_ | i am quite new to asterisk , if you don't need the modules don't load it |
| 13:50.54 | _foxfire_ | mtupper just add un noload chan_features.so to modules , but don't know if that works ;-) |
| 13:51.16 | _foxfire_ | and what stops working |
| 13:52.38 | mtupper | worth a try, thanks... I was thinking something along those lines, but I am not only a * newbie, but only been in Linux for about a month and feel like an american in Paris making an ass of himself... |
| 13:52.57 | h3x | thats better than a frenchman making an ass of himself in the us |
| 13:52.59 | *** join/#asterisk plshelpme (n=needhelp@85.104.146.39) |
| 13:53.08 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 13:55.00 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:55.30 | mtupper | another question... so does it matter where I put the statement in the modules.conf? Or can I just throw it in anywhere? |
| 13:59.58 | *** join/#asterisk gvag11 (n=g@84.254.11.90) |
| 14:00.02 | gvag11 | hi all |
| 14:02.05 | LANmower | heh, cool its working now, used saydigits rather for the numbers. |
| 14:02.14 | LANmower | easier than writing a script I guess. |
| 14:04.01 | koperniqs | <PROTECTED> |
| 14:04.12 | h3x | koperniqs: See the /topic |
| 14:04.21 | h3x | bugs.digium.com type in # 5780 |
| 14:04.45 | h3x | i know, its really really stupid that bug slipped into a release. |
| 14:04.47 | h3x | i didnt do it :P |
| 14:04.48 | gvag11 | spandsp works fine for up to 15-20 concurent faxes but for more i get pages cut short and the channels freeze. I am using Asterisk 1.2, TE205P and spandsp 0.0.2pre21. Any idea , please!!!!!1 |
| 14:05.57 | *** join/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 14:06.13 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 14:06.55 | koperniqs | h3x: gonna try that thanks |
| 14:07.44 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 14:08.23 | johnrage | i have problems compling zaptel-1.2.0. running suse linux pro 9 |
| 14:09.05 | johnrage | make *** [gendigits.o] error 127 |
| 14:09.25 | coppice | gvag11: I haven't tried load tests for some time. I could get about 100 channels on a dual xeon at one point, but I have no idea what it can do these days |
| 14:09.52 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 14:10.38 | MRH2 | dumb question how do u redcue verbosity on the cli |
| 14:10.51 | MikeJ[Laptop] | set verbose 0 |
| 14:11.01 | gvag11 | coppice: i can see that the CPU and memory load is like 4% (working on a XEON 3.0) but still problems. When you had 100 channels what version of Asterisk - spandsp you were using? |
| 14:11.27 | coppice | that was in the very early days of testing it |
| 14:11.28 | MRH2 | thanks |
| 14:12.06 | *** part/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
| 14:12.09 | gvag11 | coppice: so now things should be the same or better ... or not ??? |
| 14:13.48 | coppice | probably a bit worse, and more bells and whistles were added. i don't expect it to be too much worse, though. if you had a low cpu usage, it sounds like something gets congested and misses real time deadlines |
| 14:14.12 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 14:14.47 | gvag11 | coppice: the issue is that i share IRQ with the SATA controller, it could be that, right? |
| 14:15.09 | coppice | could be. digium card? |
| 14:15.41 | gvag11 | coppice: yes TE210P and the two spans are connected each other (E1 cross over) |
| 14:16.20 | gvag11 | coppice: if i was disabling all the modules that i don't use (almost everything) could that help you think?> |
| 14:17.41 | coppice | if you don't use them, they probably don't cause any harm |
| 14:19.11 | gvag11 | coppice: Sangoma boards could perfrom better you think? |
| 14:20.34 | LANmower | sangoma boards always do :) |
| 14:20.47 | LANmower | magic things those, they can fix problems you didnt know you had. |
| 14:22.28 | coppice | gvag11: I don't know the reason, but people who have both digium and sangoma always say the sangoma is much more tolerant of load. digium updated their firmware recently, and are supposed to be better. I have no direct experience |
| 14:23.58 | gvag11 | so before i abandon the project it could be a good idea to try Sangoma.... |
| 14:24.18 | coppice | if you have access to a sangoma, try it. |
| 14:24.46 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 14:30.10 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:30.49 | [TK]D-Fender | I've got the A104d (w/ echo canceller) at work, and their S518 ADSL card on my home server. |
| 14:30.56 | ^Howler | netstat -nap| egrep "(0\.0\.0\.0).*(0\.0\.0\.0).*LISTEN" | \ |
| 14:31.06 | gvag11 | tk: works fine? |
| 14:31.07 | ^Howler | bah...sorry |
| 14:31.15 | [TK]D-Fender | Both use the same ncurses based setup and the same driver. |
| 14:31.24 | [TK]D-Fender | works great. |
| 14:32.09 | [TK]D-Fender | I had horrible echo across the 2 TE405P's I've had and it absolutely vanished with the A104d. |
| 14:32.22 | [TK]D-Fender | I still have NASTY fax problems though :( |
| 14:32.37 | gvag11 | tk : like cut short pages for example? |
| 14:32.39 | [TK]D-Fender | SpanDSP is being super picky this last while |
| 14:33.02 | [TK]D-Fender | gvag11 : I've had short pages on both cards, and some that won't even train. |
| 14:33.10 | [TK]D-Fender | and give an immediate failure |
| 14:33.44 | *** join/#asterisk saftsack (n=oliver@p54A7E28E.dip.t-dialin.net) |
| 14:33.46 | saftsack | hi |
| 14:34.07 | [TK]D-Fender | I am getting my provider to re-examine which builds he's using and prepare for an upgrade to hopefully clear it up a bit. |
| 14:34.12 | saftsack | im searching for a 8 port isdn card. |
| 14:34.23 | gvag11 | tk : like cut short pages for example?? |
| 14:34.28 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 14:34.31 | gvag11 | any idea for other solution to use fax??? |
| 14:34.46 | LANmower | Spandsp isnt exactly what I call production-safe software |
| 14:34.49 | moonwick | yeah. vonage. |
| 14:34.54 | LANmower | use a fax machine on a seperate line. |
| 14:34.56 | LANmower | lol |
| 14:35.13 | saftsack | do you know a good card? |
| 14:35.26 | iDunno | fax machine, seperate analogue line, bog standard modem, hylafax ;) |
| 14:35.30 | LANmower | the Junghanns card works |
| 14:35.38 | LANmower | I' |
| 14:35.40 | LANmower | er |
| 14:35.43 | fugitivo | saftsack: my x100p works ok for faxes |
| 14:35.43 | Ikarus | a NT1 (BRI device) has a maximum of 2 channels, right ? |
| 14:35.49 | Ikarus | (2 B channels) |
| 14:35.52 | DrDeke | Ikarus: Yup |
| 14:35.56 | saftsack | fugitivo, :) cool |
| 14:36.04 | LANmower | I'm personally having echo problems with the junghanns octobri but that doesnt mean you will. |
| 14:36.05 | fugitivo | saftsack: but it's a crappy card |
| 14:36.08 | DrDeke | Ikarus: Unless you get a device with multiple BRI ports |
| 14:36.11 | gvag11 | yeah ok but i am talking about minimum 30 channels ... |
| 14:36.18 | fugitivo | saftsack: i don't use it on production systems |
| 14:36.20 | saftsack | i want a 8 port card, with is connected on 2 isdn lanes and on 6 isdn telephones |
| 14:36.31 | saftsack | junghanns is ok for this? |
| 14:36.53 | Ikarus | DrDeke: okay, then I really do have a ghost ISDN connection |
| 14:37.00 | DrDeke | ? |
| 14:37.29 | [TK]D-Fender | gvag11 : Yes, cut short pages. |
| 14:37.51 | saftsack | someone who has a junghanns card? |
| 14:38.45 | Ikarus | DrDeke: I have a so called combined ISDN connection (2 BRI's with one set of numbers) and a single ISDN connection, one of the combined BRI's how ever does not exist |
| 14:38.58 | DrDeke | Odd. |
| 14:39.01 | Ikarus | But we do pay for it |
| 14:39.08 | Ikarus | so it is an administrative snafu somewhere |
| 14:39.20 | gvag11 | coppice : any other idea regarding spandsp ? |
| 14:39.33 | DrDeke | Ikarus: Well, we are talking about a phone company here, aren't we? ;) |
| 14:39.49 | Ikarus | hehehe |
| 14:40.42 | DrDeke | i wonder how much ISDN BRI lines cost these days in the u.s. |
| 14:40.59 | Ikarus | No idea, here it is 22 euro a month at the old national telco |
| 14:41.03 | DrDeke | lol |
| 14:41.11 | DrDeke | It'll be more than that here, I can guarantee that :) |
| 14:41.14 | Ikarus | and via VoDSL technology 9 euro per line |
| 14:41.17 | DrDeke | wow |
| 14:41.46 | LostFrog | Phone lines here start at like $30ish. |
| 14:41.58 | Ikarus | 15 euro for an analog line here |
| 14:42.00 | DrDeke | lostfrog: isdn or pots? |
| 14:42.02 | Ikarus | (we have one) |
| 14:42.04 | LostFrog | pots |
| 14:42.09 | coppice | gvag11: I should probably do some load testing |
| 14:42.21 | DrDeke | yikes that's pretty expensive. do you have SBC there? |
| 14:42.32 | saftsack | are the junghannscards ok for providing internal isdn ports AND external on the same card? |
| 14:42.51 | gvag11 | coppice : any possible date to expect the results? |
| 14:43.23 | coppice | depends when I have time |
| 14:44.16 | gvag11 | that might be in a week for example? |
| 14:51.22 | saftsack | someone knows a price for the quadBRI PCI ISDN card? |
| 14:51.35 | Ikarus | saftsack: too expensive |
| 14:51.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:51.49 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 14:51.52 | saftsack | Ikarus, are there better cards? (production environment) |
| 14:52.02 | saftsack | :) |
| 14:52.13 | Ikarus | saftsack: for BRI, prolly not, beronet is also decent |
| 14:53.38 | saftsack | BN4S0 Quad Port S0 Card (TE/NT) |
| 14:53.50 | saftsack | this one is a good card, also in a production environment? |
| 14:53.59 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 14:56.50 | saftsack | Ikarus, are you german? |
| 14:57.03 | DrDeke | Dutch, not Deutsch :) |
| 14:57.10 | DrDeke | iirc |
| 14:57.18 | saftsack | wtf? ^ |
| 14:57.38 | DrDeke | I think he's Dutch (or at least is located in the Netherlands), as opposed to German. |
| 14:58.22 | saftsack | DrDeke, are you familliar with the beronet isdn cards? |
| 14:58.38 | DrDeke | I have not had the "pleasure" of dealing with anything ISDN-related in about 8 years. |
| 14:58.46 | saftsack | ok |
| 14:59.03 | koperniqs | saftsack: the jungahanns and beront cards are identical in their construction ... |
| 14:59.05 | Ikarus | saftsack: Dutch :) |
| 14:59.14 | saftsack | koperniqs, thank you :) |
| 14:59.49 | fugitivo | what function can i use to send a message to the cli? (extensions.conf) |
| 14:59.58 | fugitivo | verbose? |
| 15:00.23 | saftsack | ok that means if i have a 4 port hfc card i can provide 2 internal isdn busses for normal telephones and 2 external lines? |
| 15:02.57 | saftsack | fxs is for internal, right? |
| 15:02.57 | saftsack | also for plugging on faxes etc? |
| 15:02.57 | fugitivo | ~fxsfxo |
| 15:02.58 | jbot | from memory, fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 15:02.58 | DrDeke | Yeah, an fxs port connects to a telephone "station" |
| 15:03.02 | DrDeke | (s=station, o=office) |
| 15:03.09 | DrDeke | whoa, I didn't know we had a bot in here |
| 15:03.11 | DrDeke | (that kind of bot) |
| 15:03.20 | saftsack | :) |
| 15:03.45 | koperniqs | saftsack: as far as i remeber the config for the junghanns cards that might work |
| 15:03.48 | saftsack | i didnt get the bot. i have a fax here. i want to connect it to my asterisk computer. so i need a fxs card. right? |
| 15:03.57 | DrDeke | saft: Right |
| 15:03.58 | saftsack | koperniqs, ? |
| 15:04.02 | saftsack | DrDeke, thank you |
| 15:04.08 | DrDeke | np |
| 15:04.13 | saftsack | DrDeke, Wildcard TDM400P Quad-Port analog or ADSI Telefonen und POTS do you know this card? |
| 15:04.24 | DrDeke | I have a TDM400P with two FXS modules on it. |
| 15:04.31 | fugitivo | saftsack: a fax needs to receive dialtone, so it'll need an fxs port to provide dialtone |
| 15:04.41 | saftsack | ok and it works properly? |
| 15:04.52 | DrDeke | It seems to work very well. |
| 15:04.56 | saftsack | :) |
| 15:05.12 | DrDeke | I have read that the FXO modules do not work as well as the FXS, but I have not had any problems with the FXS modules. |
| 15:05.16 | saftsack | wow 1050eu for telephones .... ^^ |
| 15:05.48 | *** join/#asterisk junbug (n=junya@64.252.185.155) |
| 15:05.51 | steff | hi all, finally i have AVM C2 working for incoming call :-) with chan_capi-0.4.0_pre1, but for outgoing calls asterisk Killed when execute dial... any idea ? |
| 15:06.20 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 15:07.05 | m160858 | hello to all |
| 15:07.10 | saftsack | hi |
| 15:08.20 | m160858 | i need to know |
| 15:09.12 | m160858 | what kind of machine i need, to use 2 digium cards T210E and 2 digium cards of 24 FXS ports |
| 15:09.23 | m160858 | sorry by my english, again |
| 15:09.23 | DrDeke | that's a lot of line cards |
| 15:09.37 | m160858 | i know |
| 15:09.44 | m160858 | but, i can do it? |
| 15:10.14 | DrDeke | I will have to let someone else answer that one; my Asterisk machine has only 2 FXS ports ;) |
| 15:10.26 | m160858 | 4 cards of that kind |
| 15:10.28 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:10.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:10.30 | m160858 | hehehe |
| 15:10.55 | m160858 | i think that i want to create a frankenstein |
| 15:11.06 | m160858 | hehehe |
| 15:11.09 | DrDeke | It sounds like an interesting project ): |
| 15:11.11 | DrDeke | :) |
| 15:11.37 | m160858 | yes |
| 15:12.08 | m160858 | i need know, if i can do it |
| 15:13.02 | m160858 | any idea? |
| 15:13.12 | fugitivo | m160858: that's too many pci slots, you'll have irq problems maybe |
| 15:13.47 | shanky | just a question, If I have a asterisk connection to an external VoIP provider, but all my sip clients are going to local, do I need to have a public IP address or use nat in my asterisk server? |
| 15:13.59 | m160858 | maybe? |
| 15:14.02 | mog_work | you can nat through asterisk server |
| 15:14.08 | mog_work | you could get away with it |
| 15:14.14 | m160858 | so, you not sure of that |
| 15:14.25 | fugitivo | m160858: well, i'm sure you'll have irq problems |
| 15:14.32 | m160858 | ok |
| 15:14.44 | mog_work | maybe |
| 15:14.46 | [TK]D-Fender | shanky : point your sip clients to the internal and make sure to use "canreinvite=no" for your provider. That'll force * to proxy your conversations. |
| 15:14.50 | mog_work | we have a machine back in testing |
| 15:14.53 | mog_work | that has 6 |
| 15:14.54 | mog_work | cards |
| 15:14.58 | [TK]D-Fender | Nice and clean, but increases load a little |
| 15:15.10 | fugitivo | mog_work: without irq sharing? |
| 15:15.28 | m160858 | so, what would i do? |
| 15:15.33 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 15:15.35 | m160858 | my english it's very bad |
| 15:15.37 | fugitivo | m160858: use gateways |
| 15:15.49 | mog_work | yeah no irq problems |
| 15:15.59 | mog_work | or get a higher density card |
| 15:16.03 | mog_work | like the tdm2400p |
| 15:16.20 | fugitivo | mog_work: what motherboard? |
| 15:16.33 | mog_work | id have to go ask em |
| 15:16.38 | koperniqs | m160858: ever thougt of ata's like sipura SPA 2100 |
| 15:16.39 | mog_work | its an intel server mobo |
| 15:16.42 | mog_work | single proccessor |
| 15:16.44 | mog_work | but id have to ask |
| 15:16.54 | steff | x |
| 15:17.10 | fugitivo | mog_work: it'd be great if you can get that info :) |
| 15:17.11 | mog_work | but you can get a tdm2400p and have room to grow to 24 extensions |
| 15:17.27 | fugitivo | mog_work: he's going to use 2 tdm2400p of 24fxs |
| 15:17.53 | mog_work | groovy |
| 15:18.41 | m160858 | excuse me? |
| 15:18.47 | DrDeke | hehe |
| 15:18.58 | DrDeke | groovy means "cool" or something along those lines |
| 15:19.02 | m160858 | which is the solution for me? |
| 15:19.23 | DrDeke | m160858: In general, it is thought that having more than 2 Digium cards in a single machine is likely to cause IRQ trouble. |
| 15:19.37 | *** join/#asterisk mesfet (n=iw3grx@ppp-217-133-38-177.cust-adsl.tiscali.it) |
| 15:20.01 | DrDeke | m160858: So, instead of buying the 2 24FXS cards, you could buy two 24FXS SIP channel banks, which would connect to your Asterisk server via Ethernet. |
| 15:20.13 | DrDeke | Then you could have the two T210E cards in the Asterisk machine. |
| 15:20.21 | fugitivo | m160858: check the audiocodes gateways |
| 15:20.25 | mog_work | or one te410p |
| 15:20.35 | DrDeke | yeah |
| 15:20.51 | shanky | thanks |
| 15:21.14 | steff | so, anyone have an idea when* hang in executiong dial ? |
| 15:21.47 | mog_work | could you go more in depth? |
| 15:22.04 | *** part/#asterisk santiago (n=santiago@208.195.215.160) |
| 15:22.06 | steff | in have a Dial string: _0X.,1,Dial(CAPI/contr1:b${EXTEN},30) |
| 15:22.26 | mog_work | k |
| 15:22.29 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:22.45 | steff | when i compose a number in * log i have: Executing Dial("SIP/200-e7e6", "CAPI/contr1:b00472529881|30") in new stack |
| 15:22.57 | steff | and * hang |
| 15:23.08 | mog_work | i imagine its a bug in capi |
| 15:23.11 | mog_work | what you should do |
| 15:23.18 | mog_work | is go build asterisk unoptimized |
| 15:23.26 | mog_work | and then do this again |
| 15:23.34 | mog_work | and then grab a core |
| 15:23.42 | mog_work | probably send it to the capi people |
| 15:23.50 | *** join/#asterisk gvag11 (n=g@ppp19-adsl-132.ath.forthnet.gr) |
| 15:24.18 | steff | yes, but incoming call works well |
| 15:24.19 | m160858 | yeah, i see the audiocodes |
| 15:24.36 | steff | i rebuild.... |
| 15:25.10 | mog_work | hmm then it might be a bug in asterisk |
| 15:25.12 | DrDeke | Ok, I have a question: When you are looking at T1/E1 cards (like the TE210P), what does it even mean when a card says that it supports loopstart, groundstart, etc...? I mean, T1 signalling is digital isn't it? The T1 card on the other end is not going to try to initiate a call by shorting the line, or grounding one leg of it...? |
| 15:25.35 | mog_work | it can if the other end is a channel bank |
| 15:25.37 | tzanger | DrDeke: T1s just use the two robbed bits to simulate the analog conditions |
| 15:25.42 | mog_work | you can imitate it |
| 15:25.48 | mog_work | err simulate |
| 15:25.48 | DrDeke | Oh, ok. |
| 15:26.16 | DrDeke | So basically there is a "standard" digital sequence that a channel bank sends to indicate "I have just been (loop / ground) started on channel xx" ? |
| 15:26.30 | mog_work | yeah |
| 15:26.36 | m160858 | i can do it or not |
| 15:26.37 | DrDeke | Makes sense; thanks. |
| 15:26.59 | m160858 | i confused now |
| 15:27.03 | *** join/#asterisk bluedemon (n=shannonm@merlintechs.kvinet.com) |
| 15:27.54 | steff | is ztdummy must be be loaded with chan_capi ? |
| 15:27.55 | mog_work | do what m1 |
| 15:27.59 | funxion | has anyone played with the experimental T.38 pass through listed on the wiki? |
| 15:28.08 | mog_work | i dont know steff dont ahve a capi card |
| 15:28.25 | steff | mog_work: ok... rebuilding |
| 15:28.55 | DrDeke | m160858: It is best to have only 1 or 2 Digium cards in a single computer. So, instead of buying the 24FXS Digium cards, you could buy AudioCodes 24 port FXS to SIP gateways: http://www.audiocodes.com/Objects/MP_Series.pdf |
| 15:29.22 | m160858 | ok, i see it yet |
| 15:29.24 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 15:30.18 | mog_work | or a t1 card and a channel bank |
| 15:30.20 | mog_work | its up to you |
| 15:30.49 | m160858 | thanks to all |
| 15:30.53 | DrDeke | Good luck1 |
| 15:31.08 | m160858 | i'll need it |
| 15:31.10 | m160858 | heheheeh |
| 15:31.12 | DrDeke | :) |
| 15:32.49 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 15:33.36 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 15:37.58 | [TK]D-Fender | AudioCodes = pricey, bu there aren't any really cheap players in th high-density FXS-> SIP commodidty market. Sipura's stuff would be great for this |
| 15:38.03 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-241-139.dsl.irvnca.pacbell.net) |
| 15:38.19 | DrDeke | Does Sipura have any high density FX?->SIP equipment? |
| 15:38.26 | [TK]D-Fender | nope |
| 15:38.57 | DrDeke | So how would Sipura's stuff be ideal? Having 24 SPA-2002s sitting on a shelf seems.... well.... |
| 15:39.09 | [TK]D-Fender | they only did the 2-port tops before being bought by Linksys (who was bought by Cisco). It unlikely they would put out something much larger since it would compete with high-end products. |
| 15:39.18 | bluedemon | Does anyone know what "monitoring" problems MONITOR_CONSTANT_DELAY and MONITOR_DELAY in channels.c fixes? |
| 15:39.27 | [TK]D-Fender | Yeah, 2 port ATA's would be a bitt messy, but very inexpensive |
| 15:39.31 | bluedemon | i mrean channel.c |
| 15:39.35 | moonwick | DrDeke: it'd be the 21st century equivalent of http://slacker.com/photos/misc/pophell |
| 15:39.54 | DrDeke | HAHA nice |
| 15:39.58 | moonwick | yep :) |
| 15:40.01 | [TK]D-Fender | OMG |
| 15:40.32 | DrDeke | man that looks like my wiring closet! |
| 15:40.39 | [TK]D-Fender | Well... it COULD be done a lot cleaner if you mount all of them to a board with even spacing and sptaple down the wires. |
| 15:40.45 | DrDeke | true |
| 15:40.49 | [TK]D-Fender | But still.... INELEGENT would be a way to put it... |
| 15:41.16 | [TK]D-Fender | If you need 200 ports you MIGHT just want to save the trouble and jsut spend the cash :) |
| 15:41.35 | bluedemon | i wonder if they have crosstalk issues |
| 15:41.47 | [TK]D-Fender | T1-channel bank isn't great either. Saves a bit, but doesn't scal as easy. |
| 15:41.48 | funxion | I use quintum for high density fx |
| 15:42.09 | DrDeke | It seems to me that using T1 equipment for anything other than interfacing with the telephone company is probably a waste of money. |
| 15:42.34 | DrDeke | Oh, also, speaking of that: I know what T1s and E1s are, but what on earth is a J1? Something used in Japan? How many channels does it have? |
| 15:43.10 | koperniqs | cu |
| 15:43.43 | *** part/#asterisk koperniqs (n=koperniq@129.187.15.52) |
| 15:44.21 | sivana | bah.. I think there's a bug in voicemail |
| 15:46.23 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-228-2.rcmdva.fios.verizon.net) |
| 15:46.25 | sivana | major bug |
| 15:46.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:46.42 | moonwick | whazzat? |
| 15:49.41 | *** join/#asterisk christo (n=chris@135.196.140.252) |
| 15:51.02 | *** join/#asterisk touzas (n=touzas@213.60.220.36) |
| 15:51.15 | touzas | hello |
| 15:51.44 | touzas | somebody help me to develop iaxclient for windows in c# or c++ language?? |
| 15:52.20 | Nugget | start with "main() {" and keep typing until it works. |
| 15:52.26 | DrDeke | hehe |
| 15:52.27 | fugitivo | lol |
| 15:52.27 | mesfet | touzas: have you seen iaxclient.sf.net ? |
| 15:53.06 | bluedemon | first thing you need to do is start using a linux workstation :) |
| 15:54.26 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 15:54.41 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.137) |
| 15:55.01 | mog_work | j1 is a japanese t1 |
| 15:55.07 | mog_work | it is almost exactly like a t1 |
| 15:55.09 | DrDeke | mog_work: How many channels does it have? |
| 15:55.11 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 15:55.12 | DrDeke | Oh. |
| 15:55.12 | mog_work | 24 |
| 15:55.15 | DrDeke | Oh, ok. |
| 15:55.30 | mog_work | if you were going to use a digium card you set it to t1 mode |
| 15:55.37 | mog_work | its pretty straight forward |
| 15:55.39 | DrDeke | interesting |
| 15:55.52 | mog_work | japan just wanted to be different |
| 15:55.55 | DrDeke | hehe |
| 15:56.06 | mog_work | and box people out |
| 15:56.10 | DrDeke | Well, you know, the more something changes, the more it stays the same... |
| 15:56.11 | DrDeke | hehe |
| 15:56.20 | mog_work | yeah |
| 15:58.39 | ManxPower | ~docs |
| 15:58.40 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:58.42 | ManxPower | ~mailinglist |
| 15:58.43 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 15:58.58 | fugitivo | ~google |
| 15:58.59 | jbot | [google] a search engine found at http://www.google.com/ |
| 15:59.10 | fugitivo | that's all we need |
| 15:59.15 | fugitivo | to be happy |
| 16:03.49 | *** join/#asterisk freezer (n=leetiden@ACB1B028.ipt.aol.com) |
| 16:03.53 | freezer | hello |
| 16:05.45 | mog_work | hi |
| 16:06.07 | freezer | any germans here? |
| 16:06.26 | *** join/#asterisk bluedemon (n=shannonm@merlintechs.kvinet.com) |
| 16:06.46 | moonwick | nope, just us lowly members of the resistance. |
| 16:06.54 | mog_work | lol |
| 16:07.01 | moonwick | listen very carefully, for I shall say this only once! |
| 16:07.14 | coppice | good moaning |
| 16:07.26 | oej | Morning, coppice |
| 16:07.50 | coppice | seems oej doesn't get the reference :-) |
| 16:08.56 | touzas | thanks mesfet |
| 16:10.04 | mesfet | touzas: you're welcome |
| 16:10.33 | mesfet | touzas: there are several iax clients, i've just tried iaxcomm and works well |
| 16:11.25 | touzas | i have iaxclient ocx but i need iaxclient.dll and don`t find |
| 16:11.44 | mesfet | Dunno. |
| 16:12.13 | touzas | i make this in linux and all ok but i make in windows with cygwin and compile error |
| 16:13.08 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 16:14.07 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 16:14.12 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 16:16.17 | tzafrir_laptop | touzas, yet another reason to stick with linux |
| 16:16.28 | Katty | hi tzafrir_laptop (= |
| 16:16.35 | *** join/#asterisk saftsack (n=saftsack@p54A7F575.dip.t-dialin.net) |
| 16:16.35 | mesfet | touzas: I don't know windows environment... |
| 16:16.55 | saftsack | hi, a 2.6ghz computer is enough for a asterisk pc with a 4 port isdn card? |
| 16:17.05 | tzafrir_laptop | sure |
| 16:17.19 | zigman | saftsack a 1ghz would do it |
| 16:17.30 | tzafrir_laptop | that card doesn't take much CPU as-is |
| 16:17.37 | zigman | yeah |
| 16:17.51 | zigman | saftsack we are talking about the quadbri ? |
| 16:17.52 | tzafrir_laptop | It really depends what you need asterisk to do. |
| 16:19.15 | saftsack | zigman, yes |
| 16:19.24 | yxa | anyone knows how i can use Authenticate() by using Extension-PIN pairs? |
| 16:19.40 | saftsack | an 4 port fxs card is in the computer too |
| 16:20.45 | mesfet | A question about SIP/IAX and NAT: I've read that IAX2 uses only 4569 udp port: is it possible for 2 users in the same LAN to communicate with an asterisk PBX at the same time? |
| 16:21.17 | mesfet | The lan is NATted, and asterisk is in internet, not LAN. |
| 16:21.41 | tzafrir_laptop | mesfet: clients only? or servers as well? |
| 16:21.44 | saftsack | zigman, do you know it? |
| 16:21.57 | mesfet | tzafrir_laptop: clients only. |
| 16:22.04 | fugitivo | mesfet: you will not have problems with IAX2 in that scenario |
| 16:22.32 | mesfet | fugitivo: ok... very good. |
| 16:22.33 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 16:23.14 | fugitivo | mesfet: BitchX |
| 16:23.57 | mesfet | fugitivo: eh eh... thank you for the answer |
| 16:24.00 | mog_work | yeah mesfet you can terminate thousands of iax connections over same port |
| 16:24.14 | steff | ok after rebuild i have outgoing call working (i downgrade to chan_capi-0.3.5) |
| 16:24.16 | mog_work | and that is only a limit on hw |
| 16:24.22 | mog_work | ahh |
| 16:24.25 | mog_work | cool stef |
| 16:24.40 | steff | but (it must be :-) |
| 16:25.04 | steff | i have no sound in outgoing calls |
| 16:25.10 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.137) |
| 16:25.17 | saftsack | someone of you knows if 2.6ghz and 512mb mb ram are enough for a quadbri and a quadfxs card? |
| 16:25.21 | saftsack | -s |
| 16:25.33 | mog_work | more than enough |
| 16:25.49 | fugitivo | saftsack: how many simultaneous calls, are you going to use transcoding? |
| 16:25.49 | saftsack | thank you :) |
| 16:26.02 | saftsack | 4 simultanious calls |
| 16:26.37 | saftsack | what is transcoding? i think no, because the clients are connected with isdn and the pbx is connected with isdn too |
| 16:26.39 | fugitivo | saftsack: transcoding? how many phones? |
| 16:26.52 | saftsack | 6 phones |
| 16:26.59 | saftsack | on 2 s0 buses |
| 16:27.14 | saftsack | all isdn |
| 16:27.18 | fugitivo | ~transcoding |
| 16:27.23 | saftsack | ? |
| 16:27.26 | saftsack | what is transcoding? |
| 16:27.42 | moonwick | converting between two codecs |
| 16:27.47 | fugitivo | transcoding is making the conversion between one codec to another |
| 16:27.49 | saftsack | are the fxs cards prepared for transcoding? |
| 16:27.54 | fugitivo | that takes much more cpu power |
| 16:28.13 | fugitivo | asterisk is in charge of transcoding |
| 16:28.17 | saftsack | yes i know what transcoding is in general, but if i have just isdn there is no need to do this, right? |
| 16:28.20 | *** join/#asterisk evilrabbi (i=evilrabb@ip68-12-81-212.ok.ok.cox.net) |
| 16:28.22 | SparFu1 | What are recommended codes for blind transfer, disconnect, attended transfer etc.? The default codes clash with e.g. MeetMe which uses * for its menu. |
| 16:28.45 | fugitivo | saftsack: well, not generally if you plan to use g711 for your phones |
| 16:28.56 | evilrabbi | i'm getting " set_destination: Can't find address for host ' <ip_address>' " in the asterisk console when i try to receive a call. What could the issue be? |
| 16:29.10 | steff | i got it, it's RFC3389 |
| 16:29.15 | saftsack | fugitivo, is g711 a bad codec? |
| 16:29.19 | fugitivo | saftsack: no |
| 16:29.30 | saftsack | then its allright :) |
| 16:29.30 | evilrabbi | it will ring through just fine, but when i attempt to answer it I get that error |
| 16:29.59 | saftsack | thank you :) |
| 16:30.03 | fugitivo | saftsack: it's the codec used for regular phone calls, so if you use that codec for your phones, there'll be no transcoding and your cpu will be happy |
| 16:30.10 | saftsack | ok :) |
| 16:30.12 | saftsack | thanks |
| 16:30.25 | fugitivo | saftsack: anyways, with only 6 phones, your machine is ok for transcoding |
| 16:30.52 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 16:31.11 | steff | how i deactivate RFC3389 ? |
| 16:31.24 | evilrabbi | HEH |
| 16:31.27 | evilrabbi | ok i figured it out |
| 16:31.35 | evilrabbi | i had a space where i shouldnt |
| 16:31.55 | fugitivo | steff: deactivate? set dtmfmode= to anything else in your registration |
| 16:32.03 | steff | oki |
| 16:33.09 | yxa | anyone knows if its possible to use Authenticate() using a pin that is only specific to a number dialled? |
| 16:33.25 | steff | i have no extension with dtmfmode set |
| 16:33.42 | *** part/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 16:33.54 | lehel | hello |
| 16:34.04 | fugitivo | yxa: authenticate is limited to an unique password |
| 16:34.25 | fugitivo | steff: where do you want to deactivate it? |
| 16:34.29 | fugitivo | steff: a phone? |
| 16:34.57 | steff | i use a ip10S in the phone there was dtmfrelay=off |
| 16:35.14 | fugitivo | steff: is that phone registered to asterisk? |
| 16:35.15 | yxa | fugitivo yeah i know the limitation. i'm pretty sure it can be done by agi |
| 16:35.22 | fugitivo | yxa: yes |
| 16:35.35 | fugitivo | yxa: agi is the answer :) |
| 16:35.39 | yxa | fugitivo any docs/examples on this? |
| 16:35.47 | *** join/#asterisk junbug (n=junya@64.252.185.155) |
| 16:36.08 | bluedemon | Has anyone here had problems with a false echo while using the Monitor command? |
| 16:36.21 | fugitivo | yxa: not for authentication, but you're not the only person i see wanting to do that, check the wiki |
| 16:36.57 | yxa | fugitivo already did. nothing fruitful there... |
| 16:38.08 | fugitivo | yxa: you could pass callerid to the agi, validate the code using a database or anything you want, and then return an error code or not |
| 16:39.03 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:39.10 | fugitivo | it shouldn't be difficult to do |
| 16:39.28 | *** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 16:39.48 | Katty | hi Qwell (= |
| 16:40.14 | mog_work | morning qwell |
| 16:42.41 | Katty | mog_work: (= |
| 16:42.59 | steff | fugitivo: yes, incoming call work very well |
| 16:43.47 | steff | after a few seconds i ear sound from sip phone but not from the external side |
| 16:44.03 | lehel | what is to now about Asterisk Licensing? |
| 16:45.11 | mog_work | what lehel? |
| 16:46.39 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 16:47.43 | lehel | i'm not sure either what i'm asking, who needs to license it? |
| 16:48.17 | asterboy | looking for s100m or tdm cards. |
| 16:48.25 | tzafrir_laptop | everyone who distributes it needs a license. be that license GPL... |
| 16:48.29 | ManxPower | lehel, Asterisk is GPL. |
| 16:48.41 | tzafrir_laptop | Could you be more specific? |
| 16:49.12 | asterboy | anyone selling? |
| 16:49.26 | DrDeke | selling? selling what? |
| 16:49.35 | tzafrir_laptop | asterboy, I'm selling. For 20000$. Wanna buy? |
| 16:49.43 | asterboy | what currency? |
| 16:49.50 | tzafrir_laptop | US$ |
| 16:49.50 | ManxPower | tzafrir_laptop, that's pretty cheap for a soul 8-) |
| 16:50.01 | asterboy | do I get a cigarette with that? |
| 16:50.11 | asterboy | I usually smoke after I get... |
| 16:50.34 | tzafrir_laptop | ManxPower, I was talking about Asterisk copies. I never claimed to be explusive, though ;-) |
| 16:50.51 | tzafrir_laptop | exclusive, that is |
| 16:50.53 | asterboy | looking for S100M or TDM cards if anyone wants to sell. |
| 16:51.05 | Katty | asterboy: where did you get them? |
| 16:51.07 | Katty | asterboy: ebay? |
| 16:51.10 | DrDeke | I think that www.digium.com sells TDM cards ;) |
| 16:51.15 | Katty | asterboy: oh, i read that wrong. |
| 16:51.16 | asterboy | no, I'm looking to buy. |
| 16:51.19 | Katty | asterboy: nevermind (= |
| 16:51.21 | asterboy | lol |
| 16:51.38 | *** join/#asterisk Cresl1n (n=matt@m195e36d0.tmodns.net) |
| 16:52.12 | asterboy | not much on ebay save for some overpriced T1 cards and those $11 FXO Winmodem clones. |
| 16:53.08 | DrDeke | Yeah, I have been largely disappointed with eBay as a source of VoIP gear |
| 16:53.16 | *** join/#asterisk |cleric| (n=dacleric@p5482A579.dip0.t-ipconnect.de) |
| 16:53.46 | asterboy | atacomm has good prices...goodluck getting through to anyone. |
| 16:53.53 | asterboy | I wonder if they actually exist. |
| 16:53.57 | ManxPower | DrDeke, eBay is a good place to get stuff that's been around for a while and that is/was popular. i.e. channelbanks, echo cancelers, switches, etc |
| 16:54.22 | DrDeke | yeah |
| 16:54.51 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 16:54.56 | moonwick | You can get a good deal on palletloads of Cisco AGSes. |
| 16:55.53 | *** join/#asterisk ropeguru_work (n=ropeguru@141.152.37.26) |
| 16:55.55 | marcus2 | i picked up a cac channel bank for $300 |
| 16:56.06 | marcus2 | with a battery backup and snmp management |
| 16:56.45 | DrDeke | I think it's funny how atacomm has this obvious asterisk setup on their telephone number, which does not actually seem to work most of the time. |
| 16:56.55 | DrDeke | (One tends to get disconnected when one gets to the head of the queue) |
| 16:58.45 | DrDeke | I mean if you get the impression that that is how well (or poorly) VoIP works, you probably wouldn't want to buy any ;) |
| 16:58.55 | *** join/#asterisk dokhench (n=dochench@209.12.249.242) |
| 16:58.56 | *** part/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 16:59.06 | lehel | ManxPower: (1) copyright the software, and (2) offer you this license which gives you legal permission to copy, distribute and/or modify the software. this is GPL isn't? |
| 16:59.13 | *** join/#asterisk john867530 (n=john8675@12-218-49-92.client.mchsi.com) |
| 16:59.34 | dokhench | Have a question about * Business edition.. How exactly does it differ from 1.2? |
| 16:59.39 | mog_work | it is moderatly different |
| 16:59.42 | marcus2 | business edition? wtf is that |
| 16:59.52 | ManxPower | lehel, Asterisk is GPL. However since Digium does own the copyright to Asterisk, you can get a non-GPL license and a binary version |
| 16:59.54 | DrDeke | It is the edition that is in the business of giving its customers the business. |
| 16:59.54 | mog_work | it was based on the version of asterisk head that became 1.2 |
| 17:00.23 | dokhench | does it have any features that aren't avail in 1.2? |
| 17:00.26 | mog_work | no |
| 17:00.31 | mog_work | it is a feature stripped version |
| 17:00.43 | john867530 | how can I get the call length of a hungup call? |
| 17:00.47 | mog_work | for example it does not have chan_skinny |
| 17:01.13 | DrDeke | Here is the difference between asterisk and asterisk business edition: s. |
| 17:01.13 | DrDeke | <mog_work> it was based |
| 17:01.15 | DrDeke | s. |
| 17:01.15 | DrDeke | <mog_work> it was based |
| 17:01.16 | DrDeke | uh |
| 17:01.21 | marcus2 | what is the point of it? |
| 17:01.22 | DrDeke | That's not quite it. |
| 17:01.27 | DrDeke | http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE |
| 17:01.28 | DrDeke | there :) |
| 17:01.48 | mog_work | it has a warranty |
| 17:01.53 | mog_work | indmenification |
| 17:01.55 | mog_work | and support |
| 17:02.03 | dokhench | so * Business is basically for support. If you have a problem with something not working you can call digium up and have them fix it, right? |
| 17:02.34 | fugitivo | * business is like redhat enterprise |
| 17:02.39 | mog_work | that is one of the things you can do |
| 17:02.53 | fugitivo | not worth the price (for me) |
| 17:03.54 | dokhench | One oif the reasons I ask is, we had a customer who had "TDM PCI reset" errors on their box and it looks to be a hardware issue from what I gleaned off of google. However, if they had been running ABE would this have been something digium would have resolved? |
| 17:04.08 | lehel | ManxPower, a binary version? i'm not sure about this licensing, i'm able or not to install/configure an asterisk system at a company? |
| 17:04.30 | mog_work | hardware issues get support |
| 17:04.33 | mog_work | ed |
| 17:04.35 | asterboy | Yippeee! |
| 17:04.36 | mog_work | no matter what |
| 17:04.41 | DrDeke | lehel: You can use the regular asterisk software for any purpose you like. The only license restrictions have to do with if you wanted to change the software and sell or distribute the changed version. |
| 17:04.53 | fugitivo | dokhench: hardware is supported, even without asterisk business edition |
| 17:04.53 | DrDeke | (basically) |
| 17:04.55 | asterboy | just got an ATA 486 for $43 on ebay....joy joy |
| 17:05.10 | DrDeke | is the ata486 the 2fxs version or the 1fxs 1fxo? |
| 17:05.19 | dokhench | mog: only hardware purchased direct from digium, or all digium hardware even if bought through a reseller? |
| 17:05.19 | asterboy | just 1. :-< |
| 17:05.30 | asterboy | the 488 is the 2 port version |
| 17:05.31 | junbug | asterboy: bidding? |
| 17:05.33 | DrDeke | ohhh |
| 17:05.34 | DrDeke | ok |
| 17:05.35 | asterboy | won |
| 17:05.39 | DrDeke | still that's a good price |
| 17:05.44 | mog_work | any digium hardware gets full support from digium |
| 17:05.44 | junbug | asterboy: wow thats cheap |
| 17:05.56 | mog_work | we reccomend that you get rma from reseller or distrib as its faster |
| 17:06.01 | asterboy | yes, its cheaper than buying 1 S100M |
| 17:06.02 | mog_work | but we will do that for you too |
| 17:06.31 | DrDeke | Not to mention the TDM card into which to plug the S100M |
| 17:07.11 | DrDeke | Although I have to admit that I do get spoiled using TDM cards for FXS instead of ATAs... A TDM card in conjunction with Asterisk does an EXTREMELY good job of emulating a genuine telco POTS line (imho). |
| 17:07.20 | dokhench | also, question about the t cards from digium.. the 1 and 2 port cards don't have hardware echo cancellation, the 4 port does.. why is this? I was under the impression that only the analog pots lines introduced echo, and digital lines din't have the same issues of analog lines. |
| 17:07.23 | lehel | i understand DrDeke, thanks |
| 17:07.25 | asterboy | true...but I do have the TDM card with empty slots...just waiting for someone here to unload some modules. |
| 17:07.29 | DrDeke | yeah |
| 17:07.30 | steff | next, in the log: RFC3389 support incomplete. Turn off on client if possible |
| 17:07.48 | DrDeke | Yeah, I wish someone would "unload" an FXO module. I'd like to play around with that. |
| 17:07.55 | DrDeke | Although I guess that means I would have to give money to the phone company. |
| 17:08.16 | *** join/#asterisk KalD (n=Kal@c-24-16-117-210.hsd1.wa.comcast.net) |
| 17:08.19 | DrDeke | On the other hand, it wouldn't hurt to be able to call 911 from my apt. |
| 17:09.03 | asterboy | DrDeke: Get the FXO clone card on ebay for $11, although you need a non IRQ shared PCI slot. |
| 17:09.40 | marcus2 | get a pri card and a channel bank! |
| 17:09.42 | marcus2 | ;) |
| 17:09.45 | azzie | when call comes to * over SIP, how can I get the DID it comes to ? |
| 17:09.46 | DrDeke | hehehe |
| 17:10.13 | DrDeke | marcus2: Well, if I can sell enough cheesy services that I can provision through Asterisk (no questions asked voicemail boxes maybe?), my dream is to get enough money to be able to order a telco T1/PRI or something *grin* |
| 17:10.44 | DrDeke | asterboy: I heard that the non-TDM FXO cards really, really suck, though? |
| 17:11.14 | ropeguru_work | Does anyone here know how to get lists.digium.com to start delivering mail to a subscriber again?? I reconfigured my network last night and had to do some reboots on my router and other configs. Since then I am no longer getting emails from the lists. I tried sending password reminders and even subscribing to an additional list and I get nothing. Do not think it is my new setup as I am getting mail from multiple other sources. Sorry to be off to |
| 17:11.40 | mog_work | hey ropeguru |
| 17:11.46 | ropeguru_work | Hey mog_work.. |
| 17:11.53 | mog_work | i can get you hooked uop |
| 17:11.55 | DrDeke | ropeguru: It might just be backing off on sending you messages because of the timeouts it encountered when your network was down (if you run your own e-mail servers) |
| 17:11.57 | DrDeke | oh n/m |
| 17:11.58 | mog_work | just message me where you want it |
| 17:12.36 | dokhench | question about the t cards from digium.. the 1 and 2 port cards don't have hardware echo cancellation, the 4 port does.. why is this? I was under the impression that only the analog pots lines introduced echo, and digital lines didn't have the same issues of analog lines. |
| 17:13.10 | mog_work | one sec dokhench i got to answer call be right backl |
| 17:13.24 | dokhench | mog_work: sure thing :) |
| 17:13.27 | DrDeke | dokhench: Yeah, I really don't understand that either :) |
| 17:14.51 | ManxPower | lehel, you MUST read the GPL |
| 17:14.57 | JunK-Y | dokhench: u can have echocan of these cards. |
| 17:17.13 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:17.17 | *** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) |
| 17:17.21 | *** join/#asterisk SpaceBass (n=sp@static-71-251-228-2.rcmdva.fios.verizon.net) |
| 17:17.55 | mrtwister | hello friends,,, i need DID in ireland. like sipgate.co.uk. geographic is better (Dublin) but non-geographic ok too. what you can suggest? |
| 17:18.10 | SpaceBass | blueface.ie |
| 17:18.27 | SpaceBass | blueface.ie is what I used while on vacation in IReland |
| 17:19.00 | mog_work | okies im back |
| 17:19.21 | mog_work | the issue with pri is that it is not end to end |
| 17:19.37 | mog_work | somewhere down the line you are gonna hit 2 wire loop to someones telephone |
| 17:19.50 | mog_work | this is where the echo is introduced, that and latency |
| 17:19.56 | DrDeke | mog_work: Shouldn't the echo canceller at the call-ee's CO handle the echo? |
| 17:19.56 | mog_work | the echo can can cancel that |
| 17:20.05 | mog_work | yes it should be able to handle that |
| 17:20.09 | mog_work | but not all co's do it |
| 17:20.25 | mog_work | for example we had one customer whose co didnt echo cancel anything in the local area code |
| 17:20.32 | DrDeke | haha yikes |
| 17:20.47 | mog_work | but in most cases echo can board is for offloading cpu to the card |
| 17:20.59 | dokhench | so the echo can on the 4 port t card is more or less echo can insurance. |
| 17:21.01 | mog_work | so that you have more time for transcoding, or sticking another quad card in the box |
| 17:21.16 | mog_work | also it can do dtmf detection |
| 17:21.27 | mog_work | and in doing that asterisk doesnt have to watch the stream |
| 17:21.42 | mog_work | even less cpu |
| 17:21.49 | SpaceBass | mrtwister did you check out blueface.ie? |
| 17:22.10 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
| 17:22.21 | ropeguru_work | mog_work: Message was sent. |
| 17:22.26 | mrtwister | SpaceBass, no, thank you, i just not know avout voip in ireland |
| 17:23.31 | mog_work | yeah i am relaying it rope |
| 17:23.34 | mog_work | sorry for delay |
| 17:24.05 | ropeguru_work | No problem. I do not use IRC musch and wasn't sure I sent it correctly. I don't push people when they offer to help. :-) |
| 17:24.31 | asterboy | DrDeke: ya, I'm going to find out...from what I have read the clone cards are fine. |
| 17:24.39 | DrDeke | asterboy: Really... Interesting |
| 17:24.51 | DrDeke | asterboy: On the other hand, I could really go big and spring for an ISDN BRI. It would be a real blast from the past. |
| 17:25.00 | asterboy | DrDeke: lol |
| 17:25.02 | *** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:25.15 | asterboy | DrDeke: The cost of ISDN per month is outrageous. |
| 17:25.29 | dokhench | mog_work: thx for your help on answering my questions. is appreciated. :) |
| 17:25.53 | *** join/#asterisk rene- (n=rene-@dsl-200-95-25-242.prod-infinitum.com.mx) |
| 17:26.03 | DrDeke | jesus yeah it is |
| 17:26.13 | mog_work | no problem |
| 17:26.27 | DrDeke | I just got off the phone with SBC^H^H^H AT&T(shrug), and a flat rate residential ISDN BRI is now EIGHTY DOLLARS A MONTH not including taxes/fees/ |
| 17:26.30 | DrDeke | WTF?! |
| 17:26.37 | DrDeke | I was paying ~$40/mo for that only 8 years ago! |
| 17:26.44 | rene- | hello, my telco is telling me that they can provide me with DIDs in my analog trunk, an asterisk handle this? |
| 17:26.47 | asterboy | DrDeke: that is rediculous! |
| 17:26.49 | Himeko | might as well get a pri |
| 17:27.13 | mog_work | yes rene- |
| 17:27.14 | asterboy | and ISDN is a lower bandwidth and rate. |
| 17:27.17 | mog_work | wait |
| 17:27.19 | DrDeke | You know what, though... This is SBC we're talking about. They probably thought that when I said "residential ISDN," that I meant "business ISDN". |
| 17:27.22 | mog_work | what kind of analog trunk |
| 17:27.27 | dokhench | DrDeke: can you set outgoing clid and such with ISDN BRI? Just curious as to why would want ISDN. |
| 17:27.56 | DrDeke | dokhench: For one thing, I think that Asterisk can reliably detect call progress with ISDN, which I understand is flaky on POTS hardware. |
| 17:27.57 | Himeko | here they don't even offer residential isdn , if you want it you have to pay the business rate |
| 17:28.02 | DrDeke | Himeko: ugh |
| 17:28.18 | mog_work | dont do call progress on analog |
| 17:28.29 | dokhench | DrDeke: yeah, with my one pots line at home, it basically does silence detection to know whne the call is over with. |
| 17:28.32 | DrDeke | dokhench: Also, I only have four copper wires available at my location and two are already used for (naked) ADSL. So if I want more than one line without hassle, it has to be ISDN. |
| 17:28.36 | asterboy | IF they keep ISDN at those prices...it will become a thing of the past. |
| 17:28.43 | moonwick | it kinda already is :P |
| 17:28.52 | DrDeke | Honestly I think that ISDN BRI has always been either a thing of the future or the past in the usa :) |
| 17:28.53 | *** join/#asterisk DannyF (n=dannyf@c-604fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 17:29.20 | asterboy | wonder if you can emulate ISDN on a TCP/IP network. |
| 17:29.22 | many | call progress is flakey on ISDN? |
| 17:29.26 | DrDeke | It just skipped right over from being futuristic to a thing of the past though... It was never "current" |
| 17:29.28 | mog_work | no |
| 17:29.31 | Himeko | i had a bri when i lived in japan |
| 17:29.32 | mog_work | isdn it always works |
| 17:29.36 | DrDeke | many: No, call progress on analog/POTS is (in theory) flaky. |
| 17:29.40 | Himeko | 3600 yen/month |
| 17:29.41 | mog_work | tdmoe can work asterboy |
| 17:29.47 | mog_work | but thats what we have iax |
| 17:29.59 | many | ah |
| 17:30.05 | Himeko | that is prolly aroudn $30/m |
| 17:30.09 | asterboy | ya, I like IAX over sip for ease of use |
| 17:30.12 | DrDeke | nice :) |
| 17:30.31 | ropeguru_work | I think the IAX audio sounds better too.. |
| 17:30.47 | ropeguru_work | Not sure why that would be though. |
| 17:30.49 | mog_work | heh |
| 17:30.57 | rene- | mog: regular analog, by telmex. it doesnt have disconnect supervision tho |
| 17:31.01 | DrDeke | Maybe you have jitter on your network and the jitterbuffer helps. |
| 17:31.02 | asterboy | oh ya...don't forget the equipment costs with ISDN are far more expensive. |
| 17:31.16 | DrDeke | asterboy: Really? Some of these HFC isdn cards seem pretty cheap... |
| 17:31.26 | mog_work | doing dids over fxo ls is not supported to my knowledge |
| 17:31.26 | Himeko | at that time it was the fastest thing i could get for internet withough paying $300/m |
| 17:31.30 | asterboy | hmmm..not heard of those. |
| 17:31.32 | Himeko | er without |
| 17:31.45 | DrDeke | Himeko: That's the same reason I used to have it... Except that the next step up was 500$+/mo for me |
| 17:32.51 | asterboy | mog_work: ya, I do my did with VOIP. |
| 17:33.32 | *** join/#asterisk zoa (n=zoa@cust-2-0.dsl.versateladsl.be) |
| 17:33.34 | zoa | hey ho |
| 17:33.34 | mog_work | sorry that was dirrected at rene - earlier i think his telco wanted to give him fxo ls did which is not supported currently to my knowledge |
| 17:33.38 | mog_work | hey zoa |
| 17:33.45 | zoa | mog mog help me |
| 17:33.54 | zoa | my tdm card doesnt want to hangup |
| 17:33.56 | rene- | mog: you meant is not supported on loopstart or is not supported in any fxo |
| 17:34.11 | zoa | the channel disappears, but the card leaves the line open (i hangup on the sip end) |
| 17:34.22 | mog_work | oh noes |
| 17:34.40 | mog_work | oh so asterisk thinks call and zap are terminated? |
| 17:34.49 | zoa | yes |
| 17:34.53 | dokhench | polarity? |
| 17:34.56 | zoa | but in fact they are not |
| 17:35.04 | zoa | polarity is for the other way around i think |
| 17:35.05 | mog_work | rene- fxols,gs,and ks all work but not with did |
| 17:35.12 | zoa | for when you hangup on the pstn end |
| 17:35.18 | mog_work | does this happen every time |
| 17:35.23 | zoa | 100% of the time |
| 17:35.25 | mog_work | with every tdm card |
| 17:35.28 | mog_work | or just this card |
| 17:35.29 | zoa | yes |
| 17:35.33 | zoa | i tried another one already |
| 17:35.37 | mog_work | hrmmm |
| 17:35.46 | mog_work | you know the spec for line? |
| 17:36.04 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 17:36.20 | Blackthorn | Is there a reason I should run qualify=yes on a connection that is not behind nat? |
| 17:36.25 | zoa | normal belgian line |
| 17:36.37 | zoa | i normally never touch tdm lines though |
| 17:36.42 | zoa | so i could just have fucked up |
| 17:36.44 | DrDeke | Blackthorn: Yeah, because you are nosey and want to know if your users have their phones plugged in by typing 'sip show peers' ;) |
| 17:36.44 | zoa | but dont think so |
| 17:36.52 | DrDeke | Blackthorn: Oh wait, that's me ;) |
| 17:36.59 | mog_work | maybe zoa |
| 17:37.34 | *** join/#asterisk beavizd (n=beaviz@0x50a24527.abnxx9.adsl.tele.dk) |
| 17:37.42 | zoa | it really gives the impression its hanging up |
| 17:37.50 | zoa | maybe its because its on a via |
| 17:38.17 | zoa | any other suggestions ? |
| 17:38.23 | Blackthorn | I wonder if it improves relablity of the connection. because I have some units that just don't ever miss a call, while others seem to misse several each day. |
| 17:38.52 | Blackthorn | And the ones that are qualify seem to work better. for me anyway. so i just changed everything to qualify |
| 17:41.39 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:42.08 | zoa | aaaaaarghl im getting crazy here |
| 17:42.15 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 17:42.22 | PupenoL | How do you control the permissions used for the files created by asterisk ? particulary, voicemail ? |
| 17:42.50 | ManxPower | zoa, going? 8-) |
| 17:44.15 | zoa | ManxPower: HEEEEEEEEEEEEELP |
| 17:44.17 | zoa | :) |
| 17:45.23 | ManxPower | zoa, what is the problem? |
| 17:47.45 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
| 17:47.49 | justinu | join #asterisk-dev |
| 17:47.54 | zoa | when i ask asterisk to hangup the tdm, it says its ok, but it remains open |
| 17:48.34 | zoa | the channel disappears from the list, |
| 17:48.38 | zoa | but line is still open |
| 17:58.29 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 17:58.35 | *** join/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net) |
| 17:59.35 | *** join/#asterisk db48x (n=db48x@dsl093-026-247.hou1.dsl.speakeasy.net) |
| 18:00.20 | qw3rty | I am new to asterisk and voip. I want to set up an asterisk box at home to play with. What is the cheapest FXS device to get that will work with asterisk? |
| 18:00.56 | DrDeke | qw3rty: What you want is something along the lines of a Sipura SPA-2000 or 2002 |
| 18:01.14 | DrDeke | Although I'm not sure you can buy those new anymore without being a licensed lackey of Linksys. |
| 18:01.35 | qw3rty | so would ebay be the best bet? |
| 18:01.36 | junbug | qw3rty: i use grandstream 486 ... works fine about $65 these days |
| 18:01.46 | DrDeke | Or less if you get one on eBay perhaps |
| 18:02.07 | junbug | qw3rty: someone here just got one for $40 |
| 18:02.07 | db48x | has anyone here set up asterisk to work with SIPPhone/Gizmo? it fails to authenticate on INVITE when I dial a PSTN number |
| 18:02.20 | qw3rty | and if I get an FXS device I do not need the more expensive digium pci card correct? |
| 18:02.26 | DrDeke | Right. |
| 18:02.40 | DrDeke | Unless you want to hook up a pulse-dial phone. |
| 18:03.16 | SpaceBass | linksys pap2 from amazon is like $50... and totall hackable |
| 18:03.17 | SpaceBass | :) |
| 18:03.44 | SpaceBass | oops, guess that'd be an fxo |
| 18:04.00 | qw3rty | And as for VOIP companies, I like what Broadvoice offers... are the one of the better ones? |
| 18:04.08 | justinu | is there any way to buy an unlocked linksys rtp300? |
| 18:04.09 | DrDeke | ehh |
| 18:04.16 | SpaceBass | broadvoice is great |
| 18:04.20 | justinu | qw3rty: i've had good look with broadvoice |
| 18:04.23 | SpaceBass | esp the byod lite plan :) |
| 18:04.29 | justinu | luck |
| 18:04.47 | junbug | links sys pap2 fxo? |
| 18:04.52 | DrDeke | I have had a lot of trouble with incoming calls on Broadvoice. Even though I am registered with them, the caller gets a reorder (fast busy) signal, and Broadvoice never even attempts to send my Asterisk machine the call (verified through tcpdump). |
| 18:04.59 | LostFrog | I thought the pap2 was a FXS. |
| 18:05.10 | junbug | yeait is |
| 18:05.15 | SpaceBass | i guess it is |
| 18:05.18 | SpaceBass | i get the two confused |
| 18:05.35 | junbug | DrDeke: check extensions.conf |
| 18:06.03 | qw3rty | thanks for the info... i am sure I will be back soon :) |
| 18:06.13 | DrDeke | junbug: No offense but you didn't pay attention to what I said. Not one single TCP or UDP packet EVER makes it to my computer from Broadvoice. It is positively *NOT* a problem on my end. |
| 18:06.28 | *** part/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net) |
| 18:06.36 | justinu | it sounds like registration is timing out or something |
| 18:06.38 | junbug | DrDeke: u behind nat? |
| 18:06.46 | hypa7ia | DrDeke: have you tried with a softphone without NAT? |
| 18:06.48 | DrDeke | junbug: No NAT, totally public IP address, no blocked ports. |
| 18:06.52 | moonwick | DrDeke: you must be on that part of the internet that still uses IPX. |
| 18:06.53 | moonwick | :( |
| 18:06.55 | DrDeke | hypa7ia: I am not using NAT |
| 18:06.59 | LostFrog | lol.. IPX. |
| 18:07.00 | DrDeke | moonwick: Yeah, or NetBEUI ;) |
| 18:07.06 | justinu | try changing your defaultexpiry or something |
| 18:07.06 | moonwick | SNA 4eva! |
| 18:07.10 | hypa7ia | okay, but have you tried just connecting a softphone to BV |
| 18:07.12 | justinu | lol @ SNA |
| 18:07.14 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 18:07.16 | kippi | hi |
| 18:07.17 | justinu | twinax! |
| 18:07.24 | DrDeke | justinu: Broadvoice already asks my Asterisk to reregister every 30 seconds, which it does. |
| 18:07.31 | hypa7ia | DrDeke: okay, but have you tried just connecting a softphone to BV |
| 18:07.31 | kippi | http://www.skateeast.net/asterisk/call_flow_asterisk.htm how easy should it be to set that up? |
| 18:07.33 | justinu | DrDeke: weird |
| 18:07.34 | moonwick | heheh. I barely know what SNA is, I just know it's archaic. :) |
| 18:07.34 | junbug | DrDeke: hmm.... si iptables isnt blocking etc.... iptables -L -n |
| 18:07.45 | hypa7ia | moonwick: those are the huge ones |
| 18:07.51 | hypa7ia | like 1"x1" |
| 18:07.54 | justinu | SNA = IBM Mainframe and Minicomputer network protocol |
| 18:07.54 | DrDeke | junbug: no, there is nothing blocked on iptables on the asterisk machine, nor on my router, nor on my ISP's end |
| 18:08.14 | moonwick | ah, now those were the days. no accidentally kicking that plug out of the wall jack. |
| 18:08.27 | justinu | definitely not |
| 18:08.34 | junbug | DrDeke: u have any freonds that can scan u to make sure of that |
| 18:09.00 | DrDeke | junbug: This is one area where I do know what I am talking about. In addition, I have verified that all of the relevant ports are reachable on my machine from my computer at work. |
| 18:09.07 | justinu | moonwick: http://publib.boulder.ibm.com/iseries/v5r1/ic2924/index.htm?info/phyplan/cables/torder.htm |
| 18:09.15 | DrDeke | Additionally, it is in my ISP contract that they do not block ports. |
| 18:09.21 | DrDeke | And I set up all of my own equipment myself. |
| 18:09.28 | LostFrog | DrDeke: did you do the bv SIP dance? |
| 18:09.36 | DrDeke | LostFrog: what's that? :) |
| 18:09.47 | LostFrog | Oooh.. you didn't read the howto.. |
| 18:09.52 | *** join/#asterisk NDT (i=NDT@cpe-24-195-219-101.nycap.res.rr.com) |
| 18:09.55 | DrDeke | hypa7ia: No, I have not tried it with a softphone. Instead, I dumped Broadvoice, since I have not had this type of problem with any other provider, and also since BV doesn't let you set the outgoing CallerID anyawy ;) |
| 18:10.10 | justinu | yeah, that sucks |
| 18:10.13 | DrDeke | yeah |
| 18:10.16 | techie | yeah |
| 18:10.16 | LostFrog | If you call BV, you can change it. |
| 18:10.22 | DrDeke | LostFrog: change what? |
| 18:10.23 | LostFrog | But, no, you can't set it dynamically. |
| 18:10.25 | DrDeke | What howto? |
| 18:10.26 | DrDeke | What are you talking about? |
| 18:10.28 | LostFrog | CallerID. |
| 18:10.28 | DrDeke | Set what dynamically? |
| 18:10.32 | DrDeke | Ahhh, yes |
| 18:10.36 | hypa7ia | that sucks, thats a good reason to dump them |
| 18:10.39 | DrDeke | What I want is to be able to set it dynamically. |
| 18:11.41 | LostFrog | Yes, that would suck. |
| 18:11.57 | justinu | who else besides voicepulse allows you to set it? |
| 18:12.05 | justinu | via SIP that is |
| 18:12.07 | DrDeke | justinu: Off the top of my head, teliax, voipjet |
| 18:12.09 | SpaceBass | BV doesnt allow setting caller id |
| 18:12.09 | DrDeke | oh, I prefer IAX |
| 18:12.22 | justinu | i don't |
| 18:12.35 | DrDeke | I'm just saying that I don't know exactly how SIP works with these carriers, since I used IAX :) |
| 18:12.38 | DrDeke | No offense to SIP intended :) |
| 18:12.58 | NDT | Is there a way in extensions.conf to actually play a gsm file to the receiver of the call? Instead of the originator of the call? |
| 18:13.02 | hypa7ia | i haven't tried setting it with unlimitel |
| 18:13.07 | hypa7ia | but that's canadian :) |
| 18:13.14 | DrDeke | O Canada! |
| 18:13.23 | hypa7ia | indeedly |
| 18:13.32 | DrDeke | I may move to Canada in a few years. |
| 18:13.40 | DrDeke | Although that is really neither here nor there |
| 18:13.43 | file | mmm? |
| 18:14.45 | justinu | drdeke: no offense taken... let me tell you... their SIP offering usually sucks ass. |
| 18:14.51 | justinu | people can't figure out SIP |
| 18:14.53 | DrDeke | ugh, really? |
| 18:14.53 | DrDeke | Yeah |
| 18:15.06 | justinu | that's why we went with Level3 for SIP originataion/termination. |
| 18:15.16 | justinu | but I'm looking for something small time for my personal games |
| 18:15.20 | DrDeke | Right. |
| 18:15.32 | DrDeke | So why don't you want to use IAX then? :) |
| 18:15.49 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 18:15.57 | brad_mssw | justinu: use teliax |
| 18:16.13 | brad_mssw | justinu: lets you set outbound callerid, supports SIP and IAX |
| 18:16.22 | db48x | so, how do I go about figuring out why I can authorize an outgoing call with sipphone? |
| 18:16.51 | db48x | s/can/can't/ :P |
| 18:16.54 | NDT | I am trying to setup a way for asterisk to dial users and tell them they have a negative account balance etc..is there a way to have asterisk play a file to the receiver of the call instead of the person who dialed the number? |
| 18:17.07 | DrDeke | NDT: What you want is a callfile |
| 18:17.08 | DrDeke | I think |
| 18:17.15 | hypa7ia | that's one way to do it |
| 18:17.16 | junbug | NDT: u mean http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message |
| 18:17.19 | hypa7ia | yeah |
| 18:17.20 | DrDeke | yeah |
| 18:17.37 | NDT | ooo nice |
| 18:17.42 | NDT | 8) |
| 18:18.10 | *** join/#asterisk DannyF (n=dannyf@c-604fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 18:18.19 | db48x | I've used 'sip debug' to look at the sip headers, and I can see that they're sending a 401 response (unauthorized), and that asterisk then sends a digets auth header, but they just send another 401 back |
| 18:19.06 | db48x | digest |
| 18:21.57 | NDT | ahh this is definately nice...exactly what I was looking for...thanks 8) |
| 18:22.19 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:30.22 | justinu | db48x: sounds like your secret is wrong |
| 18:30.45 | justinu | brad_mssw: teliax seemed quite expensive last time I looked |
| 18:30.52 | justinu | i'll take another look. |
| 18:31.01 | db48x | yea, that's what I thought, but I've retyped it a couple of times to no effect |
| 18:31.08 | DrDeke | teliax doesn't seem expensive to me... they have several plans |
| 18:32.16 | *** join/#asterisk saftsack (n=saftsack@p54A7F8BB.dip.t-dialin.net) |
| 18:32.17 | saftsack | are there different drivers for the hfc chipset? |
| 18:34.02 | RoyK | visdn, isdn4linux and bristuff |
| 18:34.15 | RoyK | i'd recommend bristuff |
| 18:35.16 | *** join/#asterisk lorinc (n=ang@caracas-3954.adsl.interware.hu) |
| 18:36.12 | justinu | oh, i remember why I didn't like teliax |
| 18:36.18 | justinu | they don't offer unlimited inbound. |
| 18:36.31 | justinu | voicepulse does, for 11/mo vs. 10/mo |
| 18:36.58 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-21.dynamic.qsc.de) |
| 18:38.19 | DrDeke | justinu: The nice thing about voicepulse/teliax is that you can have high-usage inbound DIDs come in on voicepulse, low-usage inbound DIDs come in on teliax, then you just set the CallerID on outgoing calls to whatever you want... |
| 18:38.49 | db48x | justinu: any other ideas? |
| 18:43.13 | *** join/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 18:43.16 | killfill | hey |
| 18:43.22 | killfill | how do i play .gsm files?.. |
| 18:43.24 | killfill | im with ubuntu |
| 18:44.14 | iDunno | sox will probably do it. |
| 18:45.33 | killfill | hm.. |
| 18:48.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-170.dslextreme.com) |
| 18:50.20 | sellout_ | can someone give me a hand with blindfxr in features.conf |
| 18:50.27 | sellout_ | I can't seem to define a contect for it? |
| 18:50.49 | sellout_ | I get "Unable to find extension '' in context ''" whenever I hit my defined key |
| 18:51.29 | shimi | I hear a lot about SIP unable to work well through NAT, and I just saw a service provider that says: "No ports need to be open if you have a simple NAT router". How is that possible? |
| 18:51.48 | file | you can cheat slightly |
| 18:51.58 | file | and ignore everything inside of the SIP packets in regards to IPs and ports |
| 18:52.15 | file | and use the received IP and port in the header of the SIP messages and RTP packets |
| 18:52.15 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 18:52.32 | dippo | i have a dumb question: do you need a soundcard to use asterisk, and if so why? |
| 18:52.51 | shimi | dippo, asterisk is a server, it does not require a soundcard |
| 18:52.55 | file | dippo: not really, no |
| 18:53.08 | dippo | hm ok |
| 18:53.17 | BrianR___ | there's some hack logic in asterisk to allow SIP clients behind NAT to work most of the time. I'd suspect many SIP providers use similar hacks to get things going. |
| 18:53.17 | dippo | then perhaps that's not what my problem is |
| 18:53.30 | dippo | when I run asterisk it's dying at: Ouch ... error while writing audio data: : Broken pipe |
| 18:53.33 | dippo | Junk at the beginning 49443303 |
| 18:53.36 | dippo | Warning, flexibel rate not heavily tested! |
| 18:53.42 | sellout_ | is anyone else using 'blindxfr' in the latest 1.2 asterisk? |
| 18:53.45 | file | dippo: that's not the problem, the problem is above that |
| 18:53.46 | dippo | which I think is actually an mpg123 error |
| 18:53.47 | shimi | file> so you think their example will work? in their asterisk config they have a "cenreinvite=no". is that the trick you're talking about? |
| 18:54.06 | file | shimi: no, that just means a reinvite won't happen... ie: the two RTP streams won't go directly, they'll always go through asterisk |
| 18:54.18 | file | setting nat=yes is how you can get chan_sip and RTP to get around most NATs |
| 18:54.26 | dippo | oh i see it |
| 18:54.30 | dippo | it just wasn't displaying to stdout |
| 18:54.45 | shimi | funny, they don't tell me to do this. they do have a pedantic=off, though |
| 18:55.02 | shimi | my NAT is done by Cisco PIX firewall |
| 18:55.17 | shimi | I will of course open the relevant UDP port ranges... |
| 18:55.46 | shimi | why don't these VoIP providers just do IAX2 and that's it? :) |
| 18:55.56 | *** part/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 18:56.27 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 18:56.43 | DrDeke | many VoIP providers also offer IAX2 |
| 18:56.59 | h3x | shimi: coz nobody makes a high capacity iax2 voip gateway |
| 18:57.03 | syle | not large ones |
| 18:57.16 | DrDeke | true |
| 18:57.39 | Corydon-w | Well, why do only one protocol when the marketplace demands more than one? |
| 18:57.42 | shimi | any cheap one? :) |
| 18:57.50 | DrDeke | yes many cheap providers support IAX2 |
| 18:58.14 | shimi | I do _a lot_ of international phone calls |
| 18:58.33 | syle | where you from? |
| 18:58.45 | shimi | that's another problem... I am from Israel |
| 18:59.06 | shimi | basically I do calls to US, France, UK, etc |
| 18:59.18 | DrDeke | you could look at teliax.com, voipjet.com, gafachi.com... |
| 18:59.22 | DrDeke | (etc) |
| 18:59.38 | shimi | already been to voipjet and even asterisk'd it |
| 18:59.44 | *** join/#asterisk db48x (n=db48x@dsl093-026-247.hou1.dsl.speakeasy.net) |
| 18:59.45 | syle | Israel holy crap, is it a modern city where you live or really old type buildings? |
| 18:59.47 | DrDeke | didn't like it? |
| 18:59.49 | h3x | crappy sound quality? hehe |
| 18:59.51 | db48x | that sucked |
| 19:00.20 | shimi | I don't think there are non-modern city's here |
| 19:00.21 | syle | yeah voipjet sucks now |
| 19:00.33 | DrDeke | voipjet seems extremely sketchy, although i have had pretty good luck with them... |
| 19:00.35 | h3x | Hooray |
| 19:00.40 | shimi | the quality was ok |
| 19:00.46 | Corydon-w | syle: the reference to Israel is that their incumbent telecom charges way too much for international |
| 19:00.50 | syle | it seems to depend where your calling |
| 19:01.07 | shimi | but from time to time, I heard the other side bad |
| 19:01.12 | Corydon-w | That seems to be the case for most countries in the middle east |
| 19:01.18 | shimi | although he said he hears me perfectly, "as if I'm next to him" |
| 19:01.18 | DrDeke | hmm |
| 19:01.46 | shimi | I suspect it was our pretty busy line. I need to implement QoS... :) |
| 19:02.02 | h3x | maybe you need to turn on your echo can |
| 19:02.02 | syle | although i did have my xbox on still from last night on my call to kansas this morning so maybe that was another problem lol |
| 19:02.02 | h3x | heh |
| 19:02.52 | syle | someone mention echo? |
| 19:03.20 | Katty | echo |
| 19:03.38 | Corydon-w | Bad echo can is always heard on the opposite side of the call as to where the echo cancel is actually bad |
| 19:03.39 | syle | katty i am not convinced your female |
| 19:04.02 | Corydon-w | syle: I'm not convinced you're human. |
| 19:04.02 | h3x | true |
| 19:04.17 | twisted[asteria] | hahah |
| 19:04.18 | syle | owww someone is defensive |
| 19:04.55 | syle | i smell bs here hehe |
| 19:05.09 | shimi | what is better, teliax or broadvoice ? |
| 19:05.22 | iDunno | just for fun. |
| 19:05.40 | Katty | syle: i'm quite female, dear :) |
| 19:06.06 | Katty | syle: and a natural born one too. |
| 19:06.06 | syle | can i ask what you do with asterisk? |
| 19:06.20 | Katty | it manages our phone system.... |
| 19:06.20 | shimi | syle, http://www.asterisk.org/features |
| 19:06.21 | Katty | what else would i do with asterisk, syle? |
| 19:06.33 | syle | lots if your a programmer just asking |
| 19:06.41 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 19:06.48 | Katty | syle: there are plenty of programmers here, and several contributors. |
| 19:06.53 | Katty | syle: there's no reason to ask me directly :) |
| 19:07.21 | syle | you prefer i ask you indirectly? |
| 19:07.22 | syle | lol |
| 19:07.38 | syle | be a man! |
| 19:07.38 | Katty | twisted[asteria]: save me :< |
| 19:07.39 | iDunno | Katty: aww, but everyone knows you rock ;) |
| 19:07.40 | syle | j/k |
| 19:07.49 | *** join/#asterisk bluedemon (n=shannonm@merlintechs.kvinet.com) |
| 19:07.53 | Katty | iDunno: only in the mosh pit. |
| 19:07.56 | iDunno | (or a god damned know-it-all ;) |
| 19:08.04 | Katty | iDunno: actually, i can't go in the mosh pit. |
| 19:08.07 | iDunno | mosh pits are evil ;) |
| 19:08.09 | Katty | iDunno: i'd be injured. |
| 19:08.11 | iDunno | they hurt. |
| 19:08.14 | iDunno | awww |
| 19:10.08 | sellout_ | can I define the context that's used with the blindxfr in features.conf ? |
| 19:11.42 | iDunno | hmmm - do I want to go to the pub? |
| 19:11.50 | sellout_ | probably |
| 19:14.34 | iDunno | I think I will, in a bit, for double chocolate stout goodness. |
| 19:15.05 | sellout_ | where abouts in the uk are you? |
| 19:15.07 | hypa7ia | that sounds gross :/ |
| 19:15.13 | iDunno | Brighton. |
| 19:15.27 | sellout_ | only been to london, sheffield, and liverpool |
| 19:15.29 | sellout_ | nice place though |
| 19:16.15 | *** join/#asterisk Buana (n=thomasn@G62d4.g.pppool.de) |
| 19:16.32 | iDunno | :) |
| 19:17.02 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 19:18.08 | chapeaurouge | hmmm... how many asterisk process should be running at start ? |
| 19:18.13 | chapeaurouge | i thought only 1... |
| 19:19.10 | mog_work | it will start spawning threads |
| 19:19.13 | mog_work | dont worry about it |
| 19:19.29 | chapeaurouge | ok.. im reassured then. Because i put my * under Xen now |
| 19:19.38 | chapeaurouge | thought something was happening |
| 19:19.48 | chapeaurouge | ... |
| 19:21.34 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 19:22.06 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 19:22.38 | saftsack | is it hard to configre a quadBRI card? |
| 19:23.10 | iDunno | yes. or no. |
| 19:23.26 | perlmonkee | So the problem... |
| 19:23.41 | dippo | is there a way to connect to the asterisk console without running the actual daemon |
| 19:23.46 | dippo | i.e. if the daemon is already running forked in the background |
| 19:23.48 | perlmonkee | the reason I couldn't dial out to anything via iaxtel - the reason no changes seemed to do anything. |
| 19:23.52 | mog_work | asterisk -r |
| 19:23.53 | iDunno | dippo: erm - yes, -r |
| 19:24.00 | dippo | aha, thanks |
| 19:24.01 | perlmonkee | was that I had a demo setup - like I stated several times. |
| 19:24.18 | perlmonkee | and in extensions.conf, the [default] section only had include => demo - not include => local |
| 19:25.50 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 19:26.28 | [TK]D-Fender | silly include statments! |
| 19:26.49 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 19:28.30 | *** join/#asterisk toddf (n=toddf@adsl-70-233-152-247.dsl.okcyok.sbcglobal.net) |
| 19:29.35 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 19:30.15 | jeffik | Hello All: need some help getting sipura-1001 to register |
| 19:30.35 | DrDeke | ok |
| 19:30.39 | DrDeke | what's happening? |
| 19:31.31 | *** part/#asterisk ropeguru_work (n=ropeguru@141.152.37.26) |
| 19:31.32 | jeffik | it says regisration failed, i had it working, then i rebooted my modem, cable modem, and router, |
| 19:31.45 | jeffik | now it just sits in failed state |
| 19:31.46 | DrDeke | ok |
| 19:31.53 | DrDeke | What are you trying to register it with? An Asterisk machine that you run? |
| 19:32.19 | jeffik | AAH |
| 19:32.33 | jeffik | yes it's my server, asterisk at home |
| 19:32.36 | DrDeke | Are the sipura and the AAH machine both on your internal network? |
| 19:33.44 | *** join/#asterisk s0lid (i=somebody@203.167.27.182) |
| 19:34.14 | s0lid | hi guys im having a problem with queue system im using asterisk 1.0.10 and i configured the system through amportal 1.10.010 |
| 19:35.40 | jeffik | no, i am in toronto and my server is in chicago |
| 19:36.59 | hypa7ia | hey jeffik, have you checked out http://taug.ca |
| 19:37.05 | hypa7ia | it's tomorrow :) |
| 19:37.45 | jeffik | hypa7ia: is this Jim? |
| 19:37.55 | jeffik | I will be in chicago for 3 days |
| 19:38.00 | hypa7ia | i'm not jim, no |
| 19:38.08 | hypa7ia | heheh |
| 19:38.32 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 19:38.59 | jeffik | hypa79ia: ok who are you? i give up |
| 19:39.44 | jeffik | DrDrake: any thougths? |
| 19:39.55 | s0lid | can anyone help me???? |
| 19:40.23 | dippo | is this an appropriate place to ask questions about AMP? |
| 19:40.28 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:41.02 | SpaceBass | dippo can't hurt... but sometimes #amportal is better (if anyone is awake) |
| 19:41.32 | DrDeke | jeffik: sorry, i've got to run :( |
| 19:41.41 | dippo | ok.. well, I am adding a SIP extension in amportal, but i am a little confused as to what translates to the SIP user ID in those settings |
| 19:41.49 | dippo | it only asks for the display name and secret |
| 19:41.56 | dippo | among other things |
| 19:42.28 | SpaceBass | dippo using the latest version with users and devices? |
| 19:42.39 | dippo | of amportal? |
| 19:42.47 | SpaceBass | yeah |
| 19:43.04 | dippo | i am using AMP-1.10.010 |
| 19:43.30 | dippo | that's the most recent version isn't it? |
| 19:43.32 | SpaceBass | in the device section, Device ID is the sip user |
| 19:43.38 | SpaceBass | yeah, I think .010 is the latest |
| 19:43.42 | dippo | hm |
| 19:43.43 | SpaceBass | at least that is what I'm running |
| 19:43.45 | dippo | i don't see a device section at all |
| 19:43.53 | dippo | oh |
| 19:43.59 | dippo | in the extension options yo umean |
| 19:44.07 | dippo | okay, I don't have a Device ID though |
| 19:44.08 | SpaceBass | if you are in USERS then you won't see device ID |
| 19:44.29 | SpaceBass | if you click setup accross the top, then on the left you should have INCOMING CALLS, DEVICES, USERS, etc |
| 19:44.36 | dippo | yeah, i don't |
| 19:44.45 | dippo | i have incoming calls, extensions, ring groups, queues, etc |
| 19:44.47 | dippo | no devices or users |
| 19:45.11 | SpaceBass | ahhhh |
| 19:45.28 | SpaceBass | goto to MAINTENCE |
| 19:45.33 | SpaceBass | then CONFIG EDIT |
| 19:45.37 | dippo | no maintenance |
| 19:45.38 | dippo | :) |
| 19:45.54 | dippo | I feel like there's something I missed in the installation or something |
| 19:46.07 | SpaceBass | no... its a setting |
| 19:46.08 | dippo | does AMP edit stuff in /etc/asterisk/ or does it mostly store it all in mysql? |
| 19:46.40 | s0lid | maintenance is part of asterisk@home |
| 19:46.47 | mog_work | <PROTECTED> |
| 19:46.50 | mog_work | its nasty |
| 19:46.52 | SpaceBass | good point... thanks s0lid |
| 19:47.12 | SpaceBass | im looking for the setting |
| 19:47.33 | s0lid | where can i get that maintenance thing |
| 19:47.41 | s0lid | is it asterisk@home specific? |
| 19:47.45 | SpaceBass | it may be |
| 19:48.04 | s0lid | shees i'll just search it on there .tar.gz |
| 19:48.07 | *** join/#asterisk dardavoip (i=xT@84.22.47.50) |
| 19:48.19 | xT | hello |
| 19:48.20 | s0lid | anyway have anyone experienced with queue in AMP? |
| 19:49.03 | dippo | hmm |
| 19:49.06 | dippo | what's asterisk@home |
| 19:49.25 | dippo | i'm lost, now |
| 19:50.03 | SpaceBass | dippo there is a setting somewhere in the AMP configs that allows you to turn EXTENSIONS into DEVICES and USERS |
| 19:50.13 | dippo | ah |
| 19:50.15 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:50.20 | SpaceBass | however, you dont *need* that at all... I just cannot remember what the extensions screen looks like |
| 19:50.34 | dippo | hm |
| 19:51.05 | Katty | xT: the whole channel is asterisk support |
| 19:51.09 | Katty | xT: just ask your question :) |
| 19:51.12 | SpaceBass | dippo what are the first few sip fields under the extensions section? |
| 19:51.28 | xT | Katty i like to make one server |
| 19:51.31 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 19:51.33 | xT | for wholesale |
| 19:51.44 | xT | i like to buy hardware becouse i dont no how to install |
| 19:51.48 | dippo | there's "Edit extension" with display name, "extension options" with outbound CID, record incoming/outgoing, and then a whole slew of "device options" |
| 19:51.50 | xT | can you help me about htis |
| 19:51.54 | dippo | starting with "secret", "dtmfmode", etc |
| 19:51.59 | dippo | but nothing like username |
| 19:52.07 | dippo | there's "accountcode", i don't know what that is |
| 19:52.37 | xT | Katty |
| 19:52.43 | Katty | yes, xT? |
| 19:52.54 | xT | ? |
| 19:52.56 | SpaceBass | dippo I'm looking for screen shots of that screen |
| 19:53.00 | dippo | hold on i can get you one |
| 19:53.01 | Katty | xT: did you want something? |
| 19:53.08 | xT | yes |
| 19:53.14 | xT | asterisk hardware in europe |
| 19:53.22 | Katty | xT: i am not in europe. |
| 19:53.26 | Katty | xT: i am in the usa. |
| 19:53.29 | xT | :( |
| 19:53.37 | dippo | http://chris.quietlife.net/tmp/out.png |
| 19:53.37 | xT | ok in usa |
| 19:53.45 | SpaceBass | that was fast |
| 19:54.15 | dippo | xwd | convert - out.png; scp out.png etc :) |
| 19:55.13 | jeffik | All, anyone else give some help gettin sipura 1001 to register |
| 19:55.18 | SpaceBass | one sec... machine is dirt slow right now |
| 19:58.14 | Math` | is there any 6-port FXO cards available on the market? |
| 19:58.35 | mog_work | tdm2400p |
| 19:58.35 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 19:59.10 | Math` | great |
| 19:59.20 | dippo | i also added an iax trunk and it shows up in the Trunks section in AMP, however i don't see it reflected in /etc/asterisk/iax.conf |
| 19:59.22 | DrDeke | Yup, if you want 48 or fewer total FXO/FXS ports, one or two TDM2400s and the appropriate modules are probably a good solution |
| 19:59.27 | SpaceBass | dippo ok.. in that screen shot, 100 is your sip user |
| 19:59.34 | dippo | ok, i tried that |
| 19:59.41 | dippo | i think something bigger is wrong |
| 19:59.45 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
| 19:59.51 | dippo | i don't think the things I am setting up in AMP are even reflected in the actual asterisk instance |
| 19:59.57 | SpaceBass | what is the sip device ? |
| 20:00.05 | dippo | some of them are.. i see the voicemail i added for it in voicemail.conf |
| 20:00.12 | dippo | but there's nothing about the IAX trunk i added or the actual extension |
| 20:00.16 | dippo | it's a grandstream phone |
| 20:00.22 | *** part/#asterisk santiago (n=santiago@63.245.87.62) |
| 20:00.22 | SpaceBass | they would be in /etc/asterisk/extensions.conf and extensions_additional.conf |
| 20:00.51 | dippo | i've only got extensions.conf |
| 20:01.28 | RoyK | ip_nat_h323 3456 0 |
| 20:01.29 | RoyK | ip_conntrack_h323 147904 1 ip_nat_h323 |
| 20:01.32 | RoyK | ouch |
| 20:02.04 | RoyK | <retry>anyone here ever tried running a bunch of h.323 clients behind iptables nat?</retry> |
| 20:02.24 | DrDeke | nope |
| 20:02.42 | SpaceBass | dippo there is a point in the amp install- i think- where you have to generate the files... |
| 20:02.49 | SpaceBass | I've only installed AMP once and cannot remember at all |
| 20:03.14 | dippo | hm |
| 20:03.33 | dippo | well I just ran the /usr/src/AMP/install_amp |
| 20:03.37 | dippo | which appears to have mostly done all that |
| 20:04.24 | SpaceBass | do you see sip.conf and sip_additional.conf? |
| 20:04.34 | SpaceBass | (i was wrong, fyi, the sip stuff is in those) |
| 20:04.55 | dippo | only sip.conf |
| 20:05.14 | SpaceBass | hummm |
| 20:05.15 | dippo | you know, it might have been a permissions problem from the getgo |
| 20:05.19 | dippo | does the web-app create those files? |
| 20:05.30 | dippo | because at one point /etc/asterisk was not writeable by the web user |
| 20:05.54 | *** join/#asterisk arcy (n=arcanum@ppp40-adsl-168.ath.forthnet.gr) |
| 20:06.52 | dippo | i am gonna just try it again |
| 20:06.55 | dippo | installing |
| 20:08.01 | Math` | any where to buy the TDM2406E? its not even for sale on the digium store |
| 20:08.45 | dippo | that did the trick |
| 20:08.46 | dippo | weird |
| 20:12.32 | cnet2 | how do i check my voicemail? |
| 20:12.57 | DrDeke | Math`: http://www.voipsupply.com/index.php?cPath=99_300 |
| 20:14.08 | *** join/#asterisk Sam_F_Tails (n=E-102@pool-71-251-163-219.bflony.east.verizon.net) |
| 20:14.28 | Sam_F_Tails | 'Sup? |
| 20:16.17 | junbug | cnet2: set up an extension to check it .... go on the wiki explains all |
| 20:16.30 | junbug | ~wiki |
| 20:16.33 | Math` | uh, the RJ11 panel is 120$ |
| 20:17.52 | Sam_F_Tails | Wiki? |
| 20:17.54 | DrDeke | hm? |
| 20:17.57 | *** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
| 20:18.02 | Sam_F_Tails | Which wiki? |
| 20:18.21 | Math` | www.voip-info.org |
| 20:21.10 | s0lid | whats the agents.conf for? |
| 20:21.18 | asterboy | wow! |
| 20:21.27 | asterboy | Finally got through to Atacomm. |
| 20:21.37 | s0lid | is it just for queue system? what's the difference if i just added a normal local extension versus an agent |
| 20:21.50 | moonwick | heh. what, they figured a call from someone called 'asterboy' was a joke? |
| 20:21.58 | asterboy | lol |
| 20:22.15 | moonwick | see, if you were asterman... they'd feel honored to talk to you |
| 20:22.20 | asterboy | Polycomm IP 601 phone for $249 |
| 20:22.51 | Math` | "Third Lane Asterisk PBX Manager" <- is that a good management solutions for businesses? |
| 20:22.59 | asterboy | gotta grow some pubs before I'm asterman. :P |
| 20:24.16 | zigman | math url ? |
| 20:29.52 | dippo | okay this is really bizarre |
| 20:29.58 | dippo | if I delete /etc/asterisk |
| 20:30.03 | dippo | create the directory, and re-run install_amp |
| 20:30.09 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 20:30.14 | dippo | I see all the _additional.conf files and everything works for a time |
| 20:30.21 | dippo | and then periodically all those files are blown away and nothing works |
| 20:30.24 | dippo | i can't figure out what's going on |
| 20:31.05 | marcus2 | hm, gsm sounds pretty decent |
| 20:31.14 | marcus2 | is there another codec i should be using, tho, for lowish bandwidth? |
| 20:31.36 | DrDeke | GSM is close to as low as it gets, although I think that speex can be set somewhat lower. |
| 20:31.53 | marcus2 | ilbc! |
| 20:32.05 | DrDeke | ilbc is also good |
| 20:32.12 | mog_work | speex |
| 20:32.14 | moonwick | LPC10, people. |
| 20:32.15 | marcus2 | is there something that sounds better than gsm but uses less bandwidth than 711? |
| 20:32.55 | DrDeke | Well, G.726 can be set from 16 to 40kbps; one would hope that it sounds better than GSM at all these bitrates. |
| 20:33.08 | marcus2 | isnt 726 the one that is patent encumbered? |
| 20:33.13 | Math` | yeah like 729 too |
| 20:33.16 | DrDeke | I thought that was 729 |
| 20:33.18 | tzanger | ilbc is greater bandwidth and poorer quality in my experience |
| 20:33.31 | mog_work | 729 and 723 are not free |
| 20:33.33 | mog_work | others are |
| 20:33.36 | Math` | 726 is? |
| 20:33.40 | marcus2 | ah |
| 20:33.43 | DrDeke | 726 appears to be built into sterisk |
| 20:33.43 | Math` | ha |
| 20:33.45 | DrDeke | 16 (1 << 4) (0x10) audio g726 (G.726) |
| 20:33.51 | DrDeke | (from show audio codecS) |
| 20:33.55 | DrDeke | Man, I can't type today. |
| 20:34.09 | Math` | DrDeke: its not that bad, I make worse days |
| 20:34.44 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 20:35.01 | morale | anyone know where i can find some instructions on configuring a sipura spa-841 with asterisk? |
| 20:35.20 | marcus2 | so how about speex? |
| 20:36.28 | DrDeke | how about it? |
| 20:36.38 | Sam_F_Tails | THPOON |
| 20:37.17 | Sam_F_Tails | SEE! |
| 20:37.20 | Sam_F_Tails | SEE! |
| 20:37.30 | Sam_F_Tails | Doo doo crap, |
| 20:37.36 | Sam_F_Tails | space program. |
| 20:37.54 | marcus2 | how does it compare to gsm, quality wise |
| 20:38.43 | DrDeke | probably better at similar bitrates |
| 20:38.54 | freezer | hi |
| 20:38.55 | Sam_F_Tails | BATTERIES |
| 20:38.56 | DrDeke | Although these things are very subjective. |
| 20:38.59 | freezer | any germans here? |
| 20:39.05 | Sam_F_Tails | I AM AN EMAIL CHECKA! |
| 20:39.07 | mog_work | ze germans |
| 20:39.21 | Sam_F_Tails | cardboardian alloy! |
| 20:39.27 | Sam_F_Tails | HOW COME? |
| 20:39.46 | Sam_F_Tails | InSaNiTy! |
| 20:40.00 | DrDeke | http://www.speex.org/comparison.html |
| 20:40.36 | Sam_F_Tails | SPACE |
| 20:40.39 | Sam_F_Tails | CAPTAINFACE |
| 20:41.04 | Sam_F_Tails | suudsu |
| 20:41.20 | saftsack | someone knows an isdn telephone seller in germany? |
| 20:41.38 | Sam_F_Tails | yes |
| 20:42.03 | Sam_F_Tails | at the lemonade stand around the corner from the chocolate store |
| 20:42.11 | DrDeke | would someone get rid of this guy? |
| 20:42.19 | marcus2 | whats the diff between 'WB' and 'NB' on that comparison? |
| 20:42.24 | marcus2 | oh wideband/narrowband i guess? |
| 20:42.24 | Sam_F_Tails | NWB |
| 20:42.28 | DrDeke | WB is wideband and NB is narrowband |
| 20:42.28 | DrDeke | yaeh |
| 20:42.31 | saftsack | wtf lemonade and chocolate corner? |
| 20:42.35 | Sam_F_Tails | hhhhhhhhhhhhhhhhhhhCHEEZE! |
| 20:42.45 | Sam_F_Tails | poop shoot |
| 20:42.53 | DrDeke | Asterisk has a lot of trouble dealing with sampling rates other than 8KHz, so you will probably need to stick with narrowband, unfortunately. |
| 20:43.01 | freezer | query saftsack |
| 20:43.30 | Sam_F_Tails | rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr |
| 20:43.30 | Sam_F_Tails | rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr |
| 20:43.42 | Sam_F_Tails | PRETENDER OF THE GALAXIES |
| 20:44.18 | saftsack | maybe someone can kick him ^^ |
| 20:44.25 | SpaceBass | who? |
| 20:44.34 | saftsack | dunno |
| 20:44.40 | saftsack | Sam_F_Tails |
| 20:44.43 | arcy | he is probably on everyones ignore list ... |
| 20:44.53 | Sam_F_Tails | IGNORE YOUR MOM |
| 20:45.42 | Sam_F_Tails | ssassassassassassaSWITZERLAND! |
| 20:45.47 | Sam_F_Tails | ssa |
| 20:45.50 | Sam_F_Tails | ssa |
| 20:46.03 | Sam_F_Tails | sssssssssaaaaaaaaauuuuuuuuuuuuuuuuuuuuuuuuudulodmmuib |
| 20:46.07 | Sam_F_Tails | Bbbbbuummmmm |
| 20:46.07 | *** mode/#asterisk [+b *!*n=E-102@*.east.verizon.net] by kram |
| 20:46.24 | *** kick/#asterisk [Sam_F_Tails!n=mark@pdpc/sponsor/digium/kram] by kram (kram) |
| 20:46.55 | funxion | nice |
| 20:46.58 | morale | anyone know how to setup sipura spa-841 phones? |
| 20:49.48 | [TK]D-Fender | :) |
| 20:50.13 | saftsack | a one slot hfc card costs 15eu. why does a quadbri card costs more than 600eu? |
| 20:50.14 | [TK]D-Fender | morale : I have an SPA-941 which I presume is nearly identical. Whats your question? |
| 20:50.38 | morale | [TK]D-Fender: i can't get it to register with asterisk.. the line just keeps say "Not registered" |
| 20:50.39 | [TK]D-Fender | saftsack : sales volume, and density. |
| 20:50.45 | morale | i have setup the SIP account and everything |
| 20:51.02 | saftsack | [TK]D-Fender, ? |
| 20:51.03 | s0lid | where can i find howto on call transfer? |
| 20:51.15 | [TK]D-Fender | morale : Pastebin your SIP.CONF and pm it to me. |
| 20:52.14 | saftsack | [TK]D-Fender, what is volume and density? |
| 20:52.28 | [TK]D-Fender | saftsack : 1 port cards are common so there are a lot of sales and the price gets driven down. 4 port is a much rarer item and therefor the volum is lower and the price goes up. Then compound that with the question of qho would use it and jack it up a bit more |
| 20:52.56 | saftsack | ok |
| 20:52.57 | saftsack | got you |
| 20:52.58 | saftsack | thank you |
| 20:53.11 | [TK]D-Fender | sales voluem : how many the manufaturer sells. Density : the number of ports (4). Who needs a 4 port card? Business users (adds to price) |
| 20:53.21 | saftsack | yes thats right |
| 20:53.29 | [TK]D-Fender | yup supply & demand. |
| 20:53.42 | saftsack | so it would be cheaper to buy 4 singleportcards if i have enough pci slots? ^^ |
| 20:54.02 | [TK]D-Fender | I paid a lot for an ADSL line card because of limited production vs sales and for the rarity that anyone would even care to have one :) |
| 20:54.13 | [TK]D-Fender | saftsack : depends on your scenario I guess... |
| 20:54.23 | *** join/#asterisk freezer (n=leetiden@ACB76AFA.ipt.aol.com) |
| 20:54.29 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 20:54.32 | Chotaire | duh |
| 20:54.50 | *** join/#asterisk salog (n=salog@h081217002129.dyn.cm.kabsi.at) |
| 20:55.02 | [TK]D-Fender | saftsack : its the same question about why Sipura 2 port ATA's sell for +/- 80$USD (40$/port) whereas 24port gateways sell for just under 100$USD / port |
| 20:55.40 | saftsack | :> |
| 20:55.43 | [TK]D-Fender | If Sipura were to release a high-density low cost product it would disrupt the competition and there would be backlash. |
| 20:55.51 | salog | hi, i ve got a typical question: asterisk as a client behind a nat. how exactly do i use stun. to recieve incoming calls. |
| 20:56.05 | [TK]D-Fender | And it would interfere with its parents high-end sales, hence we are stuck for now :) |
| 20:56.42 | hypa7ia | [TK]D-Fender: you mean like this: http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1130279322217&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 20:56.59 | [TK]D-Fender | salog : You don't ned stun. Just forward the appropriate ports (TCP/UDP) and use the "localnet" and "externip" entries to tell * what your settings are |
| 20:57.15 | hypa7ia | err, that wasn't what i was thinking of :/ |
| 20:57.45 | [TK]D-Fender | hypa7ia : yeah that doesn't quite work :) LinksysOne is a vendor lock-in scheme and doesn't appear to be standard SIP :D |
| 20:57.54 | hypa7ia | oh, i know that |
| 20:58.02 | [TK]D-Fender | LOOKS pretty, but they'll fuck you in the end :) |
| 20:58.08 | [TK]D-Fender | (dammit) |
| 20:58.10 | salog | [TK]D-Fender, thx, will do it. but what if my ip wasnt static? mine didnt change for a over a year, but in general |
| 20:58.21 | hypa7ia | how very cisco of them :/ |
| 20:58.32 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:58.49 | freezer | is there an module available which submits the caller name instead auf the MSN(tele-number of the caller) to the ISDN phone? |
| 20:59.07 | *** join/#asterisk TommyTheKid (n=tommy@dsl-206-53-27-248.den.pcisys.net) |
| 20:59.25 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:59.43 | [TK]D-Fender | salog : get a dynamip DNS service like Dyndns.org, or CJB.net and get a program to update it. |
| 21:00.01 | saftsack | thats an interesting question. are you searching for a global adressbook? |
| 21:00.16 | [TK]D-Fender | And I guess you'll need to do something a little more if * won't let you use a FQDN |
| 21:00.17 | TommyTheKid | is there a way to offer a user non super user type access.. like a "safe console" where they can only do "show commands" or something.. or does anyone have any suggestions on monitoring :) |
| 21:00.49 | salog | [TK]D-Fender, so externip can also contain string not only ip-address |
| 21:00.49 | asterboy | Yippeee!!! It's Asterisk day. |
| 21:00.59 | asterboy | Just received my clone FXO card. |
| 21:01.05 | [TK]D-Fender | hypa7ia : not surprising is it? Their phone looked SOOOO good.... you just KNEW there had to be a price to pay :) I am so never getting into one of those situations. 3 chhers for commodity telecom! |
| 21:01.14 | asterboy | Now I'll see first hand how it works. |
| 21:01.16 | *** join/#asterisk LostFrog (n=reallyno@wsip-68-225-90-115.dc.dc.cox.net) |
| 21:01.28 | [TK]D-Fender | salog : Maybe, I'm not sure! Check the Wiki, and just give it a try! |
| 21:01.32 | freezer | saftsack: yes |
| 21:01.51 | salog | [TK]D-Fender, thx |
| 21:02.30 | [TK]D-Fender | np. |
| 21:02.41 | hypa7ia | [TK]D-Fender: hear hear. i |
| 21:02.43 | hypa7ia | err |
| 21:02.59 | hypa7ia | this is my last week working primarily with proprietary systems |
| 21:03.04 | hypa7ia | no more cisco for meeee |
| 21:03.21 | asterboy | Polycom IP 601 from AtaComm is out of stock until 1st week of December. |
| 21:03.23 | *** join/#asterisk backblue (n=moo@87.196.0.79) |
| 21:03.41 | asterboy | Worth the wait @ $249 |
| 21:05.44 | Chotaire | morning.. |
| 21:06.10 | Chotaire | anyone know an alternative software conference bridge that does SIP? (not app_meetme(2)) |
| 21:06.39 | asterboy | asterisk? |
| 21:07.48 | Math` | why don't you use meetme? |
| 21:09.56 | MikeJ[Laptop] | Chotaire, sip foundry has one |
| 21:10.14 | *** join/#asterisk sapo_original (n=Fenix@201.35.53.135) |
| 21:10.19 | sapo_original | hey there.... |
| 21:10.25 | sapo_original | i need a little help |
| 21:10.25 | backblue | anyone know eyebeam? |
| 21:10.38 | backblue | from counterpath? |
| 21:11.22 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 21:11.53 | backblue | there are alternatives opensource? |
| 21:12.09 | fugitivo | with video? |
| 21:12.12 | sapo_original | i need redirect a call received for a pstn line in my digium card, in a valid number, for a sip, how to make it? |
| 21:12.18 | saftsack | someone knows a good isdn phone? |
| 21:12.26 | [TK]D-Fender | backblue : not with video and all the features |
| 21:12.35 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 21:12.39 | sapo_original | fugitivo brazuca? |
| 21:12.52 | fugitivo | sapo_original: no, argentino |
| 21:12.55 | *** join/#asterisk MattB2 (n=Mattb@mail.tricycleinc.com) |
| 21:12.57 | sapo_original | =) |
| 21:13.05 | sapo_original | vc sabe como eu faço o q eu pedi acima |
| 21:13.06 | sapo_original | ? |
| 21:13.33 | MattB2 | hi all |
| 21:13.38 | fugitivo | sapo_original: yes |
| 21:14.04 | backblue | [TK]D-Fender: but i want video, the rest i dont care, it it is opensource implementation with G.726, i will code the rest. |
| 21:14.05 | sapo_original | pode me dar a dica? |
| 21:14.13 | MattB2 | I've got an analog polycom conf phone going via a grandstream ATA into asterisk. If I try and use DTMF during a call (ie to log into an external conferencing system) I can hear the DTMF tones being doubled |
| 21:14.15 | fugitivo | sapo_original: you want a sip device to ring when the call is received in your card? |
| 21:14.18 | MattB2 | Who should I be blaming?! |
| 21:14.24 | saftsack | is it profitable to buy a qaudbri card instead of four single cards (production environment) |
| 21:14.25 | sapo_original | fugitivo yes! |
| 21:14.29 | saftsack | ? |
| 21:14.50 | backblue | saftsack: expensive. buy four single cards insted. |
| 21:15.07 | backblue | from some compatible digium hardware |
| 21:15.15 | saftsack | digium? |
| 21:15.20 | backblue | they are a lot shipper then digium ones |
| 21:15.24 | MattB2 | my guess is that the ATA is sending not just the tone but also the SIP INFO and Asterisk is re-generating the DTMF tone from the SIP info. Short of mving over to inband DTMF, anyway of stopping Asterisk re-generating the tone? |
| 21:15.35 | fugitivo | sapo_original: just setup the rights contexts, then use the dial() function to dial the sip device |
| 21:15.36 | saftsack | i thought that i can buy every cologne chipset? |
| 21:16.59 | *** join/#asterisk veto_ (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 21:17.53 | fugitivo | sapo_original: for example, if calls from pstn go to the pstn-in context, you could set something like exten => s,1,Dial(SIP/xxxx,25,tr) |
| 21:19.10 | backblue | what codecs suports video? |
| 21:21.12 | klictel | hey all, has anyone ever tried to have an Avaya 4612 (H.323 phone) work with *? |
| 21:21.34 | sapo_original | fugitivo but, i have 2 lines... i need only one call a sip.. another, call another sip... how to make? |
| 21:21.35 | sivana | Corydon76-home: like really, are we leaving the 5887 bug in the software? |
| 21:21.50 | fugitivo | sapo_original: set 2 different contexts |
| 21:22.18 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-8.dialup.tiscali.it) |
| 21:22.21 | Gennaro | hi |
| 21:22.24 | fugitivo | sapo_original: set a context for each channel in your zapata.conf |
| 21:22.36 | Gennaro | someone can help me to start? |
| 21:22.46 | Gennaro | i have a probe like this |
| 21:22.51 | sapo_original | i can indicate for example, zap1-1,1,Dial( |
| 21:22.58 | Gennaro | i registred 2 ophone on * |
| 21:23.01 | sapo_original | i can indicate for example, zap1-1,1,Dial(SIP/xxxxx,25,tT) |
| 21:23.09 | DrDeke | So, how does one go about checking one's voicemail on an Asterisk system? |
| 21:23.10 | sapo_original | ? |
| 21:23.12 | Gennaro | i call |
| 21:23.16 | sapo_original | its works fugitivo? |
| 21:23.28 | fugitivo | sapo_original: no |
| 21:23.31 | Gennaro | and i speak i dont hear good |
| 21:23.34 | Gennaro | wy? |
| 21:24.13 | marcus2 | sweeet. another step in my asterisk/magix integration is completed |
| 21:24.42 | DrDeke | I mean, is there a way to let the user hit * or something during the announcement, or do you have to just have a separate extension send them to VoicemailMain? |
| 21:25.49 | [TK]D-Fender | DrDeke : the seperate extension is one way. You can also use "*" and create a context in your voicemail.conf and matching in extensions.conf to caputre it |
| 21:26.01 | Gennaro | if i call 600 |
| 21:26.09 | Gennaro | audio is good |
| 21:26.26 | Gennaro | i i try to change settings on the phone... |
| 21:26.26 | DrDeke | Oh, yeah, I guess I could do that ;) |
| 21:26.28 | [TK]D-Fender | but we don't fall down! |
| 21:26.37 | file | yay |
| 21:27.03 | [TK]D-Fender | DrDeke : what kind of phones do you use? |
| 21:27.07 | Gennaro | the problem is when 2 person speak in the same moment |
| 21:27.15 | Gennaro | ATcom 320 |
| 21:27.37 | DrDeke | tkd: all kinds ;)... I've figured out how I want to do it though; momentarily spaced out |
| 21:28.43 | Gennaro | tk what do think about? |
| 21:28.48 | [TK]D-Fender | Polycom's usually want to dial themselves, my SPA-941 is fully configurable whic I mapped to my dialplan, etc... |
| 21:29.15 | [TK]D-Fender | Gennaro : whats your question? |
| 21:29.21 | Gennaro | bo?!? |
| 21:29.28 | [TK]D-Fender | ? |
| 21:29.29 | *** join/#asterisk |cleric| (n=dacleric@p5482B58B.dip0.t-ipconnect.de) |
| 21:29.30 | Gennaro | i need a good configuration.. |
| 21:29.34 | [TK]D-Fender | for? |
| 21:29.42 | Gennaro | for a good audio |
| 21:29.48 | [TK]D-Fender | ? |
| 21:29.52 | Gennaro | i tried a lot .. |
| 21:29.55 | Gennaro | and nothing.. |
| 21:30.03 | [TK]D-Fender | what is "a good audio"? |
| 21:30.14 | Gennaro | like an analoge phone.. |
| 21:30.30 | [TK]D-Fender | Ok, try asking your question COMPLETELY from the start... |
| 21:30.41 | Gennaro | a normal phone that cost 10€ |
| 21:30.45 | Gennaro | or 10 $ |
| 21:30.50 | *** join/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 21:30.54 | MattB2 | anyone any ideas on the double DTMF thing? :( |
| 21:30.59 | *** part/#asterisk killfill (n=killfill@pc-200-74-44-159.asturias2.pc.metropolis-inter.com) |
| 21:31.24 | Gennaro | tk why i i call 600 |
| 21:31.30 | Gennaro | audio is good |
| 21:31.36 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 21:31.38 | [TK]D-Fender | Sorry Gennaro I can't understand what you're asking about at all... |
| 21:31.50 | Gennaro | !? |
| 21:32.45 | Gennaro | i see that jitter and audio frame can change, echo suppression, and silent can help... |
| 21:32.57 | Gennaro | but in general what is needed? |
| 21:33.40 | [TK]D-Fender | ok, maybe I can help you when I get home. BBIAB... |
| 21:35.15 | *** join/#asterisk _DAW (n=bob@adsl-222-51-184.msy.bellsouth.net) |
| 21:37.46 | Gennaro | someonce can help me with configuration of a phone? http://pastebin.ca/31730 |
| 21:38.16 | Gennaro | i have problems when 2 person speak in the same moment |
| 21:38.24 | Gennaro | what can i do?!? |
| 21:38.36 | fugitivo | what problems? |
| 21:39.04 | Gennaro | audio isn't continuous |
| 21:39.20 | Gennaro | specialy with g711u |
| 21:39.42 | fugitivo | calling from where to where? |
| 21:39.45 | saftsack | someone of you does a good isdn telephone? |
| 21:40.13 | Gennaro | form phone to phone on the same machine.. |
| 21:40.17 | Gennaro | the configuration is |
| 21:40.19 | fugitivo | Gennaro: inside your lan? |
| 21:40.24 | Gennaro | yes |
| 21:40.34 | fugitivo | the 2 phones are on the same machime? |
| 21:40.38 | Gennaro | yes |
| 21:40.46 | fugitivo | well, maybe that's the problem |
| 21:40.51 | fugitivo | try calling to another machine/phone |
| 21:41.23 | Gennaro | asterisk P4 2,4, [4050] ipphone, [4035] ipphone |
| 21:41.33 | Gennaro | p4 3com eth 10/100 |
| 21:41.47 | Gennaro | ip phone with 2 rj45 port |
| 21:42.05 | Gennaro | and another ipphone on it |
| 21:42.26 | Gennaro | i dont think about band.. |
| 21:42.37 | fugitivo | wait, you are using ip phones? |
| 21:42.39 | Gennaro | the strange think is that i can call 600 |
| 21:42.42 | Gennaro | without prob. |
| 21:42.56 | SpaceBass | FINALLY!!! HAHAHAH! finally found windows 2003 media that isn't scratche! |
| 21:42.59 | SpaceBass | sorry, just had to share with someone |
| 21:43.18 | fugitivo | Gennaro: i don't understand your scenario |
| 21:44.00 | Gennaro | i have and asterisk machine with p4 and fc4 on this machine i have 3com lan card |
| 21:44.23 | Gennaro | on this port i connect 1 phone with 2 rj45 port (like hub) |
| 21:44.46 | Gennaro | and on that phone i conncect another phone |
| 21:45.14 | Gennaro | so if i call from 2° phone i'arrive @ * passing by 1 phone.. |
| 21:45.26 | Gennaro | is more clear? |
| 21:45.56 | Gennaro | if i call from 2° phone to 600 audio maybe good |
| 21:46.21 | Gennaro | if i call phone to phone... not |
| 21:47.08 | Gennaro | and i tried in any way... |
| 21:47.23 | Gennaro | i tried to register as IAX2 and GSM |
| 21:47.37 | Gennaro | audio is continuos but 8k... |
| 21:47.48 | Gennaro | arghh |
| 21:47.54 | fugitivo | connect the 2nd phone to another place and try again |
| 21:48.17 | Gennaro | i need to buy a switch.. |
| 21:48.39 | Gennaro | i have a lot of phone like this on a Lan |
| 21:48.47 | Gennaro | and the prob. is the same.. |
| 21:48.56 | Gennaro | i tried to call assistance |
| 21:49.06 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 21:49.19 | Gennaro | and they said try to adjust jitter and audio frames.. |
| 21:49.23 | Gennaro | and codec |
| 21:49.37 | Gennaro | but i cant understand how.. |
| 21:49.44 | fugitivo | it looks that the problem is that |
| 21:50.00 | fugitivo | i'm sure that if you connect the 2nd phone to another port of the switch, it'll work |
| 21:50.19 | Gennaro | i'll try.. |
| 21:50.27 | Gennaro | fugitivo |
| 21:50.49 | Gennaro | can u help me to install a modem intel as x100p ? |
| 21:51.02 | Gennaro | i tried and doesnt work |
| 21:51.07 | Gennaro | do u know why? |
| 21:51.18 | moonwick | anyone know offhand how well meetme works under freebsd in 1.2.0? is it still kind of a hack? |
| 21:51.40 | fugitivo | Gennaro: what chipset? |
| 21:51.57 | Gennaro | 857? |
| 21:52.10 | *** join/#asterisk Brixius (i=Brixius@c-69-180-132-70.hsd1.mn.comcast.net) |
| 21:53.31 | saftsack | some germans here? |
| 21:53.38 | fugitivo | Gennaro: lspci | grep Communication |
| 21:53.57 | fugitivo | Gennaro: what do you get? |
| 21:54.20 | Gennaro | so .. |
| 21:54.29 | fugitivo | Gennaro: ? |
| 21:54.34 | Gennaro | i modprobe wcfxs |
| 21:54.36 | Gennaro | ok |
| 21:54.51 | Gennaro | ztcfg |
| 21:55.07 | fugitivo | Gennaro: type what i said before in your linux shell |
| 21:55.11 | fugitivo | Gennaro: what's the result? |
| 21:55.31 | Gennaro | opss |
| 21:55.34 | Gennaro | sorry |
| 21:55.55 | Gennaro | nothing |
| 21:56.02 | *** join/#asterisk Junbug (n=Darline@69.182.40.197) |
| 21:56.02 | Gennaro | say nothing |
| 21:56.25 | fugitivo | then 2 options 1) the chip is not x100p compatible, 2) the card wasn't detected by your linux |
| 21:57.03 | Gennaro | chipset should be compatible... for voip-info.org |
| 21:57.09 | Gennaro | 2? |
| 21:57.12 | fugitivo | type only "lspci" and see if you see the card in the list |
| 21:58.09 | Gennaro | 02:0a.0 Modem: Intel Corporation: Unknown device 1080 (rev 04) |
| 21:58.32 | fugitivo | hmm |
| 21:58.34 | fugitivo | weird |
| 21:58.54 | fugitivo | my x100p clone is detected as: Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 21:58.56 | Gennaro | if i understood with zapata i should us it.. |
| 21:59.30 | Gennaro | mmm |
| 21:59.48 | fugitivo | for x100p you should modprobe wcfxo not wcfxs |
| 21:59.52 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 22:00.14 | Gennaro | so i do.. |
| 22:00.18 | Gennaro | wcfxo |
| 22:00.28 | Gennaro | and no errors are displayes |
| 22:00.31 | Gennaro | and no errors are displayed |
| 22:00.32 | fugitivo | modprobe wcfxo |
| 22:00.40 | Gennaro | ok |
| 22:00.42 | Gennaro | i did |
| 22:00.45 | Gennaro | after |
| 22:00.47 | fugitivo | dmesg |
| 22:00.49 | Gennaro | ztcfg |
| 22:00.49 | fugitivo | what do you see? |
| 22:00.53 | Gennaro | nothing |
| 22:00.59 | fugitivo | did you run dmesg? |
| 22:01.06 | Gennaro | how? |
| 22:01.11 | fugitivo | dmesg |
| 22:01.22 | fugitivo | you need to learn some linux background, it'll be easy for you to detect problems |
| 22:01.40 | *** join/#asterisk liran_ (n=liran@80.178.0.198.adsl.012.net.il) |
| 22:02.14 | Gennaro | http://pastebin.ca/31734 |
| 22:02.36 | saftsack | is there a german asterisk channel? |
| 22:03.00 | moonwick | yeah, but the name is so long that it causes most people's IRC clients to crash |
| 22:03.03 | fugitivo | Gennaro: it's not being detected |
| 22:03.15 | fugitivo | moonwick: lol |
| 22:03.33 | moonwick | #derasteriskchatzung... |
| 22:03.37 | shido6 | anyone using em and get digits added to the dialstring? |
| 22:03.45 | fugitivo | Gennaro: i don't see the module being loaded neither |
| 22:03.46 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 22:03.48 | shido6 | or dropping digits at the end |
| 22:03.48 | *** join/#asterisk salog (n=salog@h081217002129.dyn.cm.kabsi.at) |
| 22:04.04 | fugitivo | shido6: no, what hardware are you using for em? |
| 22:04.10 | shido6 | qud card |
| 22:04.13 | shido6 | quad card |
| 22:04.18 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 22:04.18 | shido6 | and dialogic on the other end |
| 22:04.22 | shido6 | t1 crossover |
| 22:04.35 | shido6 | 1ft in between |
| 22:05.09 | shido6 | span=3,1,0,d4,ami |
| 22:05.09 | shido6 | e&m=49-72 |
| 22:05.17 | shido6 | and span=4,1,0,d4,ami |
| 22:05.17 | shido6 | e&m=73-96 |
| 22:05.26 | brc_ | alright, why doesn't this work |
| 22:05.27 | brc_ | 37.74.24.144 * 255.255.255.240 U 0 0 0 eth1 |
| 22:05.27 | brc_ | 192.168.100.0 * 255.255.255.0 U 0 0 0 eth0 |
| 22:05.27 | brc_ | 192.168.1.0 192.168.100.1 255.255.255.0 UG 0 0 0 eth0 |
| 22:05.27 | brc_ | default ip37-74-24-145. 0.0.0.0 UG 0 0 0 eth1 |
| 22:05.52 | fugitivo | oh hell |
| 22:05.53 | shido6 | usually default ip37-74-24-145. |
| 22:05.54 | shido6 | is an ip |
| 22:05.59 | brc_ | there's a vpn on 192.168.1.* in Novi Sad, with 192.168.100.1 being the gateway for that vpn |
| 22:06.00 | fugitivo | there goes my 10 lines buffer |
| 22:06.04 | shido6 | not a fqdm |
| 22:06.10 | salog | hi, i cant get sip from behind a router to work. if i ignore the fact that i am behind a nat, i can call outside, but cant recieve calls. if i set externip and localnet, i cant even get a sound when calling outside. which ports do i have to forward? do i need to use nat=yes. the sources on the net are in someway contradicting |
| 22:06.12 | brc_ | hold on |
| 22:06.16 | brc_ | everything works, except the vpn |
| 22:06.18 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 22:06.21 | brc_ | I can't ping 192.168.1.1 |
| 22:06.23 | fugitivo | brc_: route -n |
| 22:06.27 | Gennaro | fugitivo, what sould i do??! |
| 22:06.30 | fugitivo | brc_: so hostnames are resolved |
| 22:06.47 | fugitivo | Gennaro: check that you don't have any zaptel module loaded |
| 22:07.06 | fugitivo | Gennaro: lsmod |grep zaptel |
| 22:07.09 | brc_ | pretty much the same... |
| 22:07.09 | brc_ | 37.74.24 |
| 22:07.12 | brc_ | er |
| 22:07.17 | brc_ | 37.74.24.144 0.0.0.0 255.255.255.240 U 0 0 0 eth1 |
| 22:07.17 | brc_ | 192.168.100.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 |
| 22:07.17 | brc_ | 192.168.1.0 192.168.100.1 255.255.255.0 UG 0 0 0 eth0 |
| 22:07.18 | brc_ | 0.0.0.0 37.74.24.145 0.0.0.0 UG 0 0 0 eth1 |
| 22:07.47 | *** join/#asterisk jesster (i=jesster@jesster.org) |
| 22:07.52 | Gennaro | zaptel is loaded |
| 22:08.08 | jesster | has anyone been succesful creating ring tones for the Cisco 7940/7960 phones?..I've used the stuff off bkw's site but I want to make my own |
| 22:08.09 | Gennaro | rmmod? |
| 22:08.12 | Gennaro | rmod? |
| 22:08.18 | fugitivo | Gennaro: lsmod |grep wc |
| 22:08.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:08.34 | fugitivo | Gennaro: modprobe -r that modules |
| 22:08.37 | justinu | anyone familiar with the microbrower on the polycom 601? |
| 22:09.29 | Gennaro | http://pastebin.ca/31736 |
| 22:10.02 | fugitivo | Gennaro: why you have all those modules loaded? |
| 22:10.07 | fugitivo | Gennaro: modprobe -r all of them |
| 22:10.11 | Gennaro | i cant |
| 22:10.15 | Gennaro | are in use |
| 22:10.22 | Gennaro | i'm going to stop asterisk |
| 22:10.27 | salog | which ports do i forward to use asterisk as a sipclient behind a nat |
| 22:10.36 | fugitivo | Gennaro: start from the first, unload zaptel at the end |
| 22:10.54 | Gennaro | ah.. |
| 22:11.21 | *** join/#asterisk sandra78 (n=awefawe@200.106.101.60) |
| 22:12.58 | Gennaro | when i -r on all of them |
| 22:13.09 | Gennaro | i havent -r on zaptel |
| 22:13.27 | Gennaro | becouse is not more loaded |
| 22:13.40 | fugitivo | ok |
| 22:13.46 | Gennaro | before i -r on zaptel |
| 22:13.48 | fugitivo | so you don't have any module loaded now? |
| 22:13.49 | Gennaro | i see usb |
| 22:13.51 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-108-213.37-151.net24.it) |
| 22:13.53 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:13.57 | Gennaro | and i -r on it |
| 22:14.17 | Gennaro | <PROTECTED> |
| 22:14.20 | Gennaro | nothing |
| 22:14.28 | fugitivo | modprobe wcfxo |
| 22:14.48 | syzygybsd | I am trying to get a hybrid data/voice pri to work and have hdlc compiled as a module. When I run ztcfg -vvv i get "ZT_CHANCONFIG failed on channel 9: Function not implemented (38)" |
| 22:15.07 | syzygybsd | has anyone gotten hdlc to work or has an idea why this is failing for me? |
| 22:15.10 | sandra78 | hi |
| 22:15.13 | sandra78 | guys |
| 22:15.23 | saftsack | moonwick, whats the name of the german channel= |
| 22:15.24 | saftsack | ? |
| 22:15.32 | moonwick | I don't know of one, sorry |
| 22:15.33 | sandra78 | my extensions always rinning after the hangup |
| 22:15.39 | syzygybsd | #asterisk-de ? |
| 22:15.43 | saftsack | ok thanks |
| 22:15.44 | sandra78 | i have a x100 card |
| 22:15.50 | syzygybsd | nope |
| 22:16.07 | saftsack | syzygybsd, i will see in the next days |
| 22:16.15 | Gennaro | fugitivo, http://pastebin.ca/31738 |
| 22:16.16 | saftsack | someone of you having a isdn telephone? |
| 22:16.20 | Gennaro | errors.. |
| 22:16.30 | Gennaro | on an extension that i made.. |
| 22:17.11 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:17.15 | fugitivo | Gennaro: the extensions shouldn't be on that file |
| 22:17.31 | fugitivo | Gennaro: that goes to extensions.conf, please read the documentation |
| 22:17.32 | fugitivo | ~docs |
| 22:17.33 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:17.51 | saftsack | fugitivo, do you have an isdn telephone? |
| 22:18.05 | fugitivo | saftsack: no, i'm argentinian and poor |
| 22:18.16 | Gennaro | i killed lines.. |
| 22:18.49 | saftsack | ok |
| 22:19.19 | Gennaro | http://pastebin.ca/31739 |
| 22:19.33 | *** join/#asterisk keith80403_ (n=keith804@24-56-188-49.co.warpdriveonline.com) |
| 22:20.14 | sandra78 | :( |
| 22:20.41 | Gennaro | [root@asterisk etc]# modprobe wcfxo | ZT_CHANCONFIG failed on channel 1: No such device or address (6) | FATAL: Error running install command for wcfxo |
| 22:21.12 | sandra78 | my extensions always rinning after the pstn hangup |
| 22:21.15 | sandra78 | :( |
| 22:21.15 | fugitivo | Gennaro: are you sure the chip is compatible? |
| 22:21.25 | Gennaro | so i belive... |
| 22:22.04 | Gennaro | do u think that is not true?!? |
| 22:22.06 | RoyK | fugitivo: what chip? |
| 22:22.19 | *** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st) |
| 22:22.34 | fugitivo | Gennaro: gennaro's card chip |
| 22:22.47 | fugitivo | Gennaro: what chip is that? |
| 22:23.26 | Gennaro | i'm unpluging card just 4 minutes... so i can say the right thing.. |
| 22:23.29 | Gennaro | sorry.. |
| 22:23.39 | Gennaro | and realy tanx of your support.. |
| 22:24.02 | dippo | hm, weird. I am getting a syntax error when i make a ring group into a hunt group |
| 22:24.05 | dippo | ih apm |
| 22:24.54 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-121.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:26.33 | dippo | er, in |
| 22:29.41 | saftsack | dippo, are you german? |
| 22:29.58 | justinu | voicepulse is sucking today... |
| 22:30.16 | justinu | audio ncie and choppy |
| 22:30.51 | p1tst0p | what would be a good web (php) interface to manage asterisk ? |
| 22:31.09 | dippo | saftsack: heh, no |
| 22:31.12 | saftsack | ok |
| 22:31.17 | saftsack | but do you have an isdn telephone? |
| 22:31.20 | dippo | just a poor typist |
| 22:31.20 | dippo | no |
| 22:31.26 | saftsack | ok |
| 22:31.46 | dippo | i find it odd that the asterisk log gives you a syntax error but not a line number to find it |
| 22:32.13 | [TK]D-Fender | p1tst0p : GUI = Suck. ALL OF THEM. (Some less tahn others ;)) |
| 22:32.25 | fugitivo | yeah, look at windows |
| 22:32.26 | fugitivo | it sucks |
| 22:32.33 | [TK]D-Fender | INDEED! |
| 22:32.38 | [TK]D-Fender | Long live CDE!!!! |
| 22:32.40 | [TK]D-Fender | ;) |
| 22:32.43 | justinu | anything with overlapping windows sucks |
| 22:32.46 | justinu | screen forever |
| 22:33.08 | fugitivo | yeah |
| 22:33.09 | [TK]D-Fender | Yeah.. screen it is! |
| 22:33.11 | fugitivo | internet sucks too |
| 22:33.24 | fugitivo | it killed bbs era |
| 22:33.26 | justinu | plan9 was cool |
| 22:33.54 | [TK]D-Fender | I browse the w33b with only the most bleeding-edge releas of Links! |
| 22:34.05 | justinu | lynx? :) |
| 22:34.05 | p1tst0p | [TK]D-Fender, agreed.. |
| 22:34.17 | Rawplayer | DO |
| 22:34.28 | [TK]D-Fender | justinu : I don't need no stinkin' colour! LINKS, not LYNX! |
| 22:34.32 | justinu | lol |
| 22:34.34 | fugitivo | links is too gay |
| 22:34.36 | p1tst0p | [TK]D-Fender, but that wasnt the question. |
| 22:34.43 | fugitivo | lynx is 1337 |
| 22:34.48 | [TK]D-Fender | p1tst0p : Web gui's for * suck. Period |
| 22:35.11 | [TK]D-Fender | I'm stuck with one (for now) at work and wish only for the right to replace it with straight CLI |
| 22:35.21 | *** join/#asterisk dev-rad (n=jesse@adsl-149-23-23.mia.bellsouth.net) |
| 22:35.41 | [TK]D-Fender | fugitivo : I'll forgive your flashy preferences :) |
| 22:35.44 | dev-rad | anyone know is sunrocket will work with asterisk? |
| 22:35.49 | salog | hi, i cant get sip from behind a router to work. if i ignore the fact that i am behind a nat, i can call outside, but cant recieve calls. if i set externip and localnet, i cant even get a sound when calling outside. which ports do i have to forward? do i need to use nat=yes. the sources on the net are in someway contradicting |
| 22:36.12 | justinu | ok, teliax sucks. |
| 22:36.13 | moonwick | salog: shouldn't need to forward any ports |
| 22:36.13 | [TK]D-Fender | salog : make sure you set your phones to "canreinvite=no" |
| 22:36.17 | moonwick | but yes, nat=yes |
| 22:36.19 | justinu | they won't give me a number in 212 area code. |
| 22:36.41 | *** join/#asterisk CpuID (n=nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 22:36.44 | CpuID | kram, ping |
| 22:37.04 | CpuID | hmm anyone from digium here? |
| 22:37.06 | salog | [TK]D-Fender, i do have that... |
| 22:37.20 | dokhench | anyone give me a general place to start if i get a "TDM PCI master reset" on latest version digium tdm card? google had not much to say on the matter and bugs only had a little info from 2003. |
| 22:37.35 | [TK]D-Fender | p1tst0p : I've looked at AMP (the friendliest of the bunch, and makes somewhat readable code), and I use ScopServ at work (very clean code, but too much is hidden behind AGI and encrypted). |
| 22:38.27 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-28.dialup.tiscali.it) |
| 22:38.40 | Gennaro | hi again |
| 22:38.42 | Gennaro | my modem is an qtec 561mi 56k pci modem i fa82537ep 0427dgea44 exactly |
| 22:38.57 | Gennaro | can i use it as fxo? |
| 22:39.41 | Gennaro | fugitivo? |
| 22:40.11 | fugitivo | i don't think so |
| 22:40.14 | syzygybsd | it is about time i finally got hdlc to work |
| 22:40.31 | Gennaro | mm |
| 22:40.41 | *** join/#asterisk SparFu1 (n=player@tor/session/x-247c047f7eb06a38) |
| 22:40.49 | Gennaro | damn.. |
| 22:42.26 | Gennaro | where i can buy a goog clone that let me timing? |
| 22:43.18 | [TK]D-Fender | p1tst0p : I guess the question is : why would you want a gui for *? |
| 22:44.41 | p1tst0p | [TK]D-Fender, its not really for me.. its for a friend who has a company.. i just think if they had something they could use to manage it, it would eliviate them from comming to me all the time. |
| 22:45.24 | [TK]D-Fender | p1tst0p : Nice idea, they COULD do it, but ... ICK. Better that they learn the basics. What kind of setup is planned? |
| 22:45.34 | p1tst0p | so, usability, ease of use.. is the key.. for someone who can be just shown the interface.. and away they go |
| 22:46.15 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 22:46.27 | p1tst0p | [TK]D-Fender, its just a small office, 6 ip phones, 1xE1, maybe a couple of SIP lines.. a little IVR.. Voicemail.. thats about it |
| 22:46.47 | Gennaro | fugitivo can u see http://www.voip-info.org/wiki/view/X100P+clone? |
| 22:47.06 | Gennaro | i dont know if should try to do it! |
| 22:47.08 | [TK]D-Fender | hmmm.... there really IS so little to do for * though... est for a company that size... |
| 22:47.21 | p1tst0p | i know what your saying. |
| 22:47.32 | [TK]D-Fender | :/ |
| 22:47.44 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:47.47 | [TK]D-Fender | Are they all "technologically challenged"? |
| 22:47.54 | p1tst0p | not at all |
| 22:47.58 | p1tst0p | hehe |
| 22:47.59 | syzygybsd | aww, no messages show up for me now |
| 22:48.26 | [TK]D-Fender | basically you could make them a quick setup that is easily readble and maintainable.... |
| 22:48.41 | dev-rad | pltst0p have them try asterisk@home.. |
| 22:49.03 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 22:49.25 | p1tst0p | dev-rad yeh i thought of that, but i tend to stay away from those kind of things.. i prefer a good clean install that i know whats what.... ast@home uses AMP does it not ? |
| 22:49.31 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:49.48 | [TK]D-Fender | p1tst0p : yeah, it does. |
| 22:50.03 | dev-rad | ease of use has a price |
| 22:50.23 | [TK]D-Fender | basically a premade distro with zaptel precompiled and configred, etc. better of adding AMP over a natural install. |
| 22:50.41 | p1tst0p | i dunno, if they dont wanna do many changes i suppose i could just manage changes for them.. |
| 22:50.42 | [TK]D-Fender | dev-rad : YOUR VERY SOUL! |
| 22:50.54 | Gennaro | A modem with the following chipsets should work: Intel 537EP |
| 22:51.01 | dev-rad | lol |
| 22:51.03 | [TK]D-Fender | p1tst0p : Do they need anything special? |
| 22:51.04 | Gennaro | and shoul i sniff |
| 22:51.05 | Gennaro | pco |
| 22:51.07 | Gennaro | pci |
| 22:51.11 | Gennaro | to see vendor |
| 22:51.38 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 22:51.40 | [TK]D-Fender | Gennaro : maybe you should just buy a supported piece of hardware rather than fight over some questionable X100 clone..... |
| 22:51.42 | theblue | Hi all. |
| 22:52.06 | theblue | Does anyone know of any Free DID providers that give 202, 240, 703, 301, or any other MD/DC area area codes? |
| 22:52.39 | p1tst0p | [TK]D-Fender, not initially.. but you never know what the future holds...! |
| 22:53.18 | Gennaro | d-fender where i can buy a good card not too expensive that let timing for on old |
| 22:53.21 | Gennaro | ? |
| 22:53.36 | [TK]D-Fender | "timing for on old"? |
| 22:53.41 | [TK]D-Fender | ah MOH. |
| 22:53.50 | [TK]D-Fender | jUST USE ztdUMMY! |
| 22:54.11 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 22:54.14 | Gennaro | i havent 280 for developer card from digium.. |
| 22:54.18 | Gennaro | too expensive.. |
| 22:54.21 | Gennaro | for me.. |
| 22:54.46 | [TK]D-Fender | You don't need a card. Just use the ZTDUMMY module. |
| 22:55.11 | Gennaro | i understood, |
| 22:55.12 | fugitivo | Gennaro: $20 on ebay for an x100p clone |
| 22:55.24 | Gennaro | ebay.. |
| 22:55.30 | fugitivo | oh, you need only a timer? |
| 22:55.36 | Gennaro | i havent trust on ebay.. |
| 22:55.45 | SparFu1 | How can I make asterisk print a variable on the screen? E.g. in asterisk - r -vvv |
| 22:55.49 | Gennaro | i see that some card have timing.. |
| 22:55.52 | tzanger | Gennaro: I've purchased over US$100k of equipment for various companies through ebay... haven't been screwed yet |
| 22:56.31 | fugitivo | Gennaro: i've purchased several x100p clones on ebay |
| 22:56.39 | Gennaro | ok |
| 22:56.45 | Gennaro | i see |
| 22:57.13 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 22:57.16 | saftsack | night |
| 22:59.08 | Gennaro | http://iaxtalk.com/index.php?main_page=product_info&cPath=5&products_id=6 what do u think about? |
| 22:59.17 | Gennaro | can it work? |
| 23:00.03 | IronHelix | yeah but it says you dont get callerid |
| 23:00.19 | IronHelix | and that's no fun :( |
| 23:00.37 | p1tst0p | eheh |
| 23:01.14 | [TK]D-Fender | Poor cheap saps..... |
| 23:01.26 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 23:01.43 | IronHelix | besides if you just need timing you can use ztdummy |
| 23:01.47 | Gennaro | what should i buy?! |
| 23:02.04 | IronHelix | do you need timing or do you need an fxo port? |
| 23:03.08 | IronHelix | if you need an fxo port, buy this http://www.voipsupply.com/product_info.php?products_id=291 |
| 23:03.17 | IronHelix | Digium TDM01B |
| 23:03.32 | IronHelix | if you just need timing, buy nothing. Download the zaptel driver and compile it with ztdummy support |
| 23:04.02 | [TK]D-Fender | Or better still : http://www.voipsupply.com/product_info.php?products_id=203 |
| 23:04.12 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 23:04.13 | [TK]D-Fender | 1 FXS & FXO for 66% of the cost. |
| 23:04.33 | tzanger | [TK]D-Fender: yeah, but with a fucking wall wart and cable mess |
| 23:04.36 | [TK]D-Fender | And PABX agnostic... |
| 23:04.48 | tzanger | I'll stick to the PCI cards, thanks |
| 23:04.53 | [TK]D-Fender | tzanger : Imperfect worlds call for imperfect solutions! |
| 23:04.55 | [TK]D-Fender | ;) |
| 23:04.57 | docelm0 | YIPPIE! |
| 23:05.17 | IronHelix | hehe |
| 23:05.23 | tzanger | I like the TDM cards personally |
| 23:05.26 | IronHelix | yeah i prefer cards too |
| 23:05.32 | IronHelix | one less thing to configure |
| 23:05.36 | tzanger | I hate having stacks of plastic boxes everywhere, each iwth their own power supply and cabling |
| 23:05.43 | IronHelix | zaptel+asterisk instead of ATA+sip+asterisk |
| 23:05.45 | [TK]D-Fender | tzanger : I just ditched my TDM22B for an SPA-3000 & SPA-941 at par (+/-) and now don't have to have my * box worry about IRQ sharing, etc. |
| 23:06.00 | tzanger | yeah there is that aspect, but I've really never had issue with that |
| 23:06.06 | syzygybsd | I rebuilt a lot of my server, including asterisk, zaptel, and libpri, and now asterisk freezes when playing sounds to sip extensions. Anyone had issues like this? |
| 23:06.22 | tzanger | hell I even had a TDM card sharing IRQs with the network card, and the box was also the NAS for my house :-) |
| 23:06.25 | Gennaro | i sow .. |
| 23:06.26 | [TK]D-Fender | tzanger : and I wanted to recoup my $ :) |
| 23:06.45 | Gennaro | but i need to study asterisk |
| 23:06.52 | Gennaro | and i need card fxo |
| 23:07.19 | Gennaro | and i think i a lot expensive 150 € for fxo |
| 23:07.47 | tzanger | Gennaro: well if you find something more economical go for it |
| 23:07.53 | tzanger | that's pretty much what the market's decided though |
| 23:09.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:10.02 | kink0 | buenas noches |
| 23:10.33 | IronHelix | and besides, compared to most other telephony cards, digiums products are a steal |
| 23:10.41 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-61-219.cybersurf.com) |
| 23:10.51 | kink0 | finally I did my connection to a mobile phone <-> soundcard <-> asterisk... but I have not full tested yet |
| 23:11.01 | Corydon-w | Could someone tell me how defaulting all your users to have the same voicemail password is somehow a security issue in the Asterisk code? |
| 23:11.19 | IronHelix | because then anybody can get anybody elses voicemail |
| 23:11.22 | Corydon-w | Because I really don't understand that logic |
| 23:11.23 | malverian[work] | Yay.. chan_sip isn't crashing Asterisk on me anymore with 1.2.0 stable. |
| 23:11.26 | kink0 | if somebody can test and have a friend at spain mobile... this is ready. |
| 23:11.32 | msw | sounds like a flaw in the configuration |
| 23:11.33 | Corydon-w | IronHelix: yes, but how is that a bug in Asterisk? |
| 23:11.43 | msw | not in asterisk itself... |
| 23:11.59 | IronHelix | its not an asterisk flaw, its an asterisk administrator flaw |
| 23:12.02 | IronHelix | :) |
| 23:12.13 | Corydon-w | I've been telling this guy it's a bug with his local security policy, but he won't hear of it |
| 23:12.16 | distortion | is there a 4t1 isdn card someone would recommend for asterisk to use with a 1u dell 1750? |
| 23:12.46 | Corydon-w | Bug 5887 |
| 23:12.53 | IronHelix | dist- digium TE series |
| 23:13.16 | *** part/#asterisk jjuhlh (n=Irix@nat.kollegienet.dk) |
| 23:14.07 | tzanger | Corydon-w: it's not a bug in asterisk so much as it is just a very poor idea |
| 23:14.12 | Gennaro | goof nite |
| 23:14.46 | IronHelix | ahhh |
| 23:14.52 | IronHelix | what you described isn't the bug |
| 23:14.55 | IronHelix | there are two issues here |
| 23:15.12 | IronHelix | 1. his organization is using the same VM password everywhere, which is a VERY BAD idea |
| 23:15.53 | IronHelix | 2. VoiceMailMain() checks all contexts, instead of just default, which is contrary to documentation. This is a bug, either that it should restrict to @default, or that the documentation should be changed |
| 23:17.42 | IronHelix | so while his bug is valid, it certainly wasnt a MAJOR bug, and if this causes problems for his users, then he needs to re-think his organization's security. You shouldn't be able to access another user's files, let alone by accident |
| 23:18.21 | IronHelix | or at least change his dialplan to say VoiceMailMain(@default) |
| 23:19.16 | msw | IronHelix: vm in 1.2.0 can force the user to change their password when the password matches the extension, iirc |
| 23:19.38 | IronHelix | that wasnt the problem, the problem was that EVERYBODY was using the SAME password |
| 23:19.46 | IronHelix | like everybody's voicemail password is 1234 |
| 23:19.52 | IronHelix | but that is useful to know, thanks :) |
| 23:20.03 | shido6 | anyone using e&m? |
| 23:20.40 | kink0 | shido6, e&m mobile ? |
| 23:21.21 | dokhench | i don't quite get the (nod your head yes) comment in the config. |
| 23:21.42 | kink0 | shido6, I have connected my cell phone e&m to my soundcard, and then use asterisk opening the sound card and a little code to send AT commands to the mobile terminal to dial or hangup |
| 23:22.20 | syzygybsd | asterisk and my sip phones only seem to be passing back some control messages and no sound, any ideas? |
| 23:22.33 | dokhench | port 10000 to 20000 open? |
| 23:23.03 | fugitivo | kink0: errr, that's not e&m signalling |
| 23:23.19 | syzygybsd | local network, so yes |
| 23:23.37 | tzanger | Corydon-w: that's a bug |
| 23:23.39 | tzanger | your fix is bullshit |
| 23:23.40 | IronHelix | syz- if they have STUN, turn it off... that can cause issues |
| 23:23.50 | Corydon-w | tzanger: No, it's intentional |
| 23:23.55 | dev-rad | anyone know if sunrocket can be made to work with asterisk?? |
| 23:24.03 | Corydon-w | tzanger: if you read the code, there is no way it's not intentional |
| 23:24.04 | tzanger | Corydon-w: explain how it can possibly make things easier or better for a site admin |
| 23:24.19 | tzanger | there are lots of things in the code that are intentional and still wrong |
| 23:24.22 | dokhench | have a good one guys. beer 30 time. =) |
| 23:24.24 | tzanger | I am slowly but surely correcting them |
| 23:24.26 | syzygybsd | it was working until i recompiled everything, kernel, zaptel, libpri, asterisk |
| 23:24.36 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-121.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:24.43 | tzanger | executive decisions are made, that is Mark's job and he does pretty good at it. |
| 23:24.50 | Corydon-w | tzanger: if you can explain, at least better than this guy, how that's wrong, it's worth hearing |
| 23:25.00 | tzanger | Corydon-w: I did in the bugreport |
| 23:25.24 | tzanger | you just made all asterisk systems act different in their default behaviour |
| 23:25.30 | tzanger | it's called amplifying entropy |
| 23:25.35 | tzanger | and it's a terrible thing to do |
| 23:26.09 | tzanger | if an admin decides NOT to use @default, then they should EXPLICITLY tell asterisk what to use, not rely on asterisk's search-foo to magically find the correct context |
| 23:26.16 | tzanger | because you know what |
| 23:26.33 | tzanger | asterisk WILL find the wrong one given half a chance. Murphy was a son of a bitch, but he was dead on the money |
| 23:26.39 | Corydon-w | because you're not expecting it to do the right thing |
| 23:26.45 | IronHelix | heh |
| 23:27.04 | LostFrog | Has anyone tried using the Echo application without playing a sound first? |
| 23:27.20 | tzanger | code should be written such that it follows the path of least surprise |
| 23:27.30 | *** part/#asterisk SparFu1 (n=player@tor/session/x-247c047f7eb06a38) |
| 23:27.34 | tzanger | and trying ot find the first matching context out of a sea of infinite contexts is NOT the path of least surprise |
| 23:27.36 | IronHelix | i agree that it *should* search @default by default, and search all only if told to. I'd still rather have it do it the wrong way with correct documentation than the wrong way with the wrong documenation |
| 23:27.49 | Corydon-w | Actually, if you set up voicemail to run across multiple disks, and divide them up into different contexts this way, then what Asterisk is doing makes sense |
| 23:28.01 | fugitivo | is this #asterisk or #asterisk-dev, this bitchx client confuses me... |
| 23:28.07 | Corydon-w | i.e. different mounts for each context |
| 23:28.10 | theblue | #asterisk |
| 23:28.12 | IronHelix | this is #asterisk |
| 23:28.17 | tzanger | there is no reason in HELL to have voicemail across different disks, there's a goddamned limit of 100 voicemails! |
| 23:28.17 | IronHelix | (or is it?) |
| 23:28.24 | theblue | Does anyone know of any free DID providers? |
| 23:28.30 | Corydon-w | tzanger: how about space? |
| 23:28.32 | IronHelix | www.ipkall.com has free DID |
| 23:28.32 | tzanger | Corydon-w: and that is a VERY specific case of "right" :-) |
| 23:28.42 | mog_work | i have a question tzanger |
| 23:28.47 | tzanger | mog_work: shoot |
| 23:28.47 | mog_work | why in hell do you have 100 vm |
| 23:28.52 | mog_work | call people back damnit |
| 23:28.53 | tzanger | mog_work: I don't |
| 23:28.56 | IronHelix | hahahhaa |
| 23:29.03 | fugitivo | only 100 vm? |
| 23:29.13 | tzanger | mog_work: I'm saying each box has a limit of 100 messages, why in the blue fuck would you spread things acorss multiple disks if you have such a low limit |
| 23:29.17 | Corydon-w | mog_work: some boxes do, especially default mailboxes |
| 23:29.36 | tzanger | and really, who would use a separate mount ofr each context?! it works but any sane admin would be using LVM in the first place |
| 23:29.39 | Corydon-w | tzanger: because that limit can be expanded in voicemail.conf |
| 23:29.47 | tzanger | Corydon-w: I thought it was a hard limit? |
| 23:29.54 | Corydon-w | tzanger: Nope |
| 23:29.58 | mog_work | its just a default |
| 23:29.58 | IronHelix | i think it used to be at one time |
| 23:30.00 | IronHelix | but was then changed |
| 23:30.01 | mog_work | you can change it |
| 23:30.03 | Corydon-w | tzanger: it was at one time |
| 23:30.05 | tzanger | ahh okay |
| 23:30.11 | tzanger | so I'm not TOTALLY out to lunch :-) |
| 23:30.23 | Corydon-w | but now you can accumulate 1,000,000 vm's in an INBOX |
| 23:30.32 | tzanger | true enough |
| 23:30.33 | fugitivo | i think we should modify asterisk to only use DEFAULT context, everybody will be happy with that |
| 23:30.45 | mog_work | ???? |
| 23:30.47 | tzanger | but again your exmaple of it working "correctly" is a VERY specific example |
| 23:30.53 | tzanger | fugitivo: hahaha |
| 23:30.55 | IronHelix | yeah the path of least surprise is definately to use @default by default, and search all if told to |
| 23:30.58 | Corydon-w | What's wrong with a specific example? |
| 23:31.05 | tzanger | screw searching |
| 23:31.09 | Corydon-w | How would you specify to search all? |
| 23:31.11 | tzanger | @default if not specified, error out otherwise |
| 23:31.17 | tzanger | it's a configuration issue in any case |
| 23:31.44 | Corydon-w | I think we should leave the code alone, document how it works, and let the administrators make their own decisions |
| 23:31.48 | tzanger | relying on the system to just do the right thing is a very very poor thing ot do... sivana was banking on the documentation being right, but not specifying a context is still kind of poor form IMO |
| 23:32.04 | tzanger | Corydon-w: well the bug's open, let's have the discussion |
| 23:32.22 | Corydon-w | That's sane. Trying to force system administration decisions on people is never the right choice |
| 23:32.23 | tzanger | this is how things get better |
| 23:32.30 | tzanger | Corydon-w: absolutely. |
| 23:32.40 | Corydon-w | and this is a system administration decision |
| 23:32.52 | fugitivo | no boy, i don't want to read documentation, it should work out of the box |
| 23:32.58 | Corydon-w | So if you want a single context, then specify a single context |
| 23:33.01 | tzanger | and it's a case of Asterisk not following the path of least surprise |
| 23:33.07 | tzanger | so it's a code issue too |
| 23:33.08 | Corydon-w | If you don't, we'll let Asterisk find the right mailbox |
| 23:33.56 | tzanger | letting Asterisk "find" the riht mailbox is not following the path of least surprise |
| 23:34.05 | tzanger | if you set up a second mailbox in another context you can hit it by accident |
| 23:34.09 | Corydon-w | To you, perhaps. |
| 23:34.14 | Corydon-w | I think it's a marvelous idea |
| 23:34.19 | tzanger | and evne if the passwords don't match, you still have "wah, my voicemail don't work" issues |
| 23:34.32 | tzanger | and the admin then has to dig through and figure out what the fuck is wrong |
| 23:34.45 | fugitivo | ok, now i want to know what the hell you're guys talking about :) |
| 23:34.48 | Corydon-w | And of course, that would be due to the admin screwing up |
| 23:34.49 | tzanger | ideally the goddamned context should be specified at all times :-) |
| 23:35.08 | Corydon-w | Good, so specify the context at all times when you only want one context searched |
| 23:36.00 | tzanger | but unspecified should be @default IMO. if you want it to magically sniff the aether and your ESP hardware is working, you should be specifing @AUTO or |s for "I'm a crazy motherfucker who wants to let the system try and understand what I want instead of being a man and specifying it explicitly" :-) |
| 23:36.55 | fugitivo | asterisk searchs for the vm on another contexts if it can't find it under @default? |
| 23:36.57 | tzanger | I mean your documentation "fix" isn't that hot either... "all contexts will be searched for a matching mailbox" -- what is the criteria? |
| 23:37.01 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 23:37.07 | tzanger | all it does is cause more questions now :-( |
| 23:37.09 | m160858 | hello to all |
| 23:37.11 | m160858 | again |
| 23:37.20 | m160858 | i have problem with a fedora |
| 23:37.28 | m160858 | i want to install asterisk |
| 23:37.35 | Corydon-w | but that means I can't have an [AUTO] context. That also violates least surprise |
| 23:37.53 | tzanger | I really hope you don't take this personally, Corydon-w -- all I want is consistency and the best fucking PBX available... that's it |
| 23:38.08 | Corydon-w | Suddenly, someone's [AUTO] context stops working, because you've made it a special keyword. |
| 23:38.10 | tzanger | Corydon-w: I agree, which is why a flag to specify "I like taking chances" is the better option |
| 23:38.22 | tzanger | Corydon-w: totally agreed, it was just an example, not one I'd recommend either |
| 23:39.01 | Corydon-w | In the rewrite of voicemail, you're welcome to make that suggestion. In the meantime, we're going to keep the way it works. |
| 23:39.02 | tzanger | I managed to successfully lobby for consistency in the DIALSTATUS result code... it took some owrk but in the end consistency prevailed |
| 23:39.38 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 23:39.41 | Corydon-w | There's no good reason to change the way VoicemailMain works at this time |
| 23:39.46 | tzanger | Corydon-w: that makes sense... but can you edit the patch a little to give some indication of the criteria for "matching mailbox" ? is it alphabetic context search? creation date? modification date? inode number? filesystem-specific? |
| 23:40.05 | fiber0pti | Anyone have experience with polycom phones and the services button and getting idle images working? voip-info doesn't explain either well enough. |
| 23:40.12 | m160858 | but i try to install the cvs from yum |
| 23:40.26 | Corydon-w | tzanger: first one found in the list |
| 23:40.41 | m160858 | this my problem |
| 23:40.43 | m160858 | [root@dedicated src]# yum -y install cvs |
| 23:40.44 | m160858 | Gathering header information file(s) from server(s) |
| 23:40.44 | m160858 | Server: Fedora Core 1 - Base |
| 23:40.44 | m160858 | retrygrab() failed for: |
| 23:40.44 | m160858 | <PROTECTED> |
| 23:40.44 | m160858 | <PROTECTED> |
| 23:40.46 | m160858 | failover: out of servers to try |
| 23:40.48 | m160858 | Error getting file http://benny.primarydns.com/fedora/core/1/i386/os/headers/header.info |
| 23:40.50 | m160858 | [Errno 4] IOError: <urlopen error > |
| 23:40.54 | m160858 | some idea from this? |
| 23:40.56 | m160858 | sorry, but that |
| 23:41.00 | m160858 | many lines |
| 23:41.05 | m160858 | sorry |
| 23:41.37 | tzanger | "in the list" ? |
| 23:41.54 | tzanger | so first one found in voicemail.conf in a top-down search? |
| 23:42.03 | m160858 | i want to install cvs for can to install asterisk |
| 23:42.14 | tzanger | er first context found in top-down voicemail.conf search that has the specified mailbox? |
| 23:47.02 | Corydon-w | Yep, top-down in voicemail.conf |
| 23:48.08 | Supaplex | where are the cgi-bin files for astbill? this install says to copy them, but they don't appear to be in the tarball off the sf.net mirror ... |
| 23:49.23 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 23:50.57 | p1tst0p | is it possible to connect vonage (uk) with asterisk ? say a business package ( couple of lines ) |
| 23:51.07 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 23:51.10 | *** part/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 23:51.16 | sivana | I can't believe this |
| 23:51.37 | sivana | Corydon-w: you honestly believe that the current behaviour is the best approach? |
| 23:51.39 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 23:52.17 | sivana | or are you waiting for someone to write the correct patch? I can pay someone if you don't feel like it |
| 23:52.33 | *** join/#asterisk silug (n=steve@206.80.72.34) |
| 23:52.51 | Corydon-w | sivana: Yes, I honestly believe that leaving the current behavior as is is the best approach |
| 23:53.14 | Corydon-w | sivana: I am a very prolific patch writer, I don't need someone to write a patch |
| 23:54.08 | sivana | so if by chance two users pick the same password, no matter how remote the possibility, you think it's acceptable that VoicemailMain could give access to the wront user |
| 23:54.16 | sivana | wrong |
| 23:55.01 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 23:55.20 | Corydon-w | If they have the same mailbox number, yes. |
| 23:55.25 | moonwick | what's the argument in favor of leaving the current behavior? |
| 23:55.48 | sivana | make VoicemailMain consistent with the rest of * and make a search/all flag |
| 23:56.03 | sivana | or leave the security hole in the app |
| 23:56.27 | Corydon-w | moonwick: not changing existing behavior that was coded intentionally |
| 23:56.53 | Corydon-w | sivana: it's a security hole in your site config, because you insist on giving everybody the same password |
| 23:57.23 | Corydon-w | What kind of idiot gives everybody the same password on a system? |
| 23:57.24 | sivana | its a security to everyone who thinks not explicitly passing a context and has two or more mailboxes with the same password |
| 23:57.32 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 23:57.44 | sivana | Corydon-w: now we're into name calling? |
| 23:57.49 | sivana | wow |
| 23:58.01 | moonwick | why was it intentionally coded that way? sorry to ask, I don't feel like digging through scrollback |
| 23:58.18 | tzanger | Corydon-w: that's a VERY common procedure on large PBXes. What VoiceMailMain's lacking is the ability to force a user to change their password on the next access |
| 23:58.23 | tzanger | moonwick: yes |
| 23:58.29 | Corydon-w | tzanger: so add that |
| 23:58.41 | tzanger | Corydon-w: see the bugtracker. :-) |
| 23:58.48 | sivana | not to mention the rest of asterisk assumes [default] when a default is used |
| 23:58.49 | Corydon-w | tzanger: what's the bug number? |
| 23:59.08 | tzanger | 5882 or whatever |
| 23:59.13 | tzanger | I forget now :-) |
| 23:59.17 | sivana | Corydon-w: do you use the search "feature"? |
| 23:59.19 | sivana | 5887 |
| 23:59.32 | Corydon-w | 5882 is festival |
| 23:59.41 | Corydon-w | sivana: yes, actually |
