| 00:03.27 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:13.18 | MrWup | heh heh heh |
| 00:13.22 | MrWup | just finished my own voicemail system |
| 00:13.24 | MrWup | how leet |
| 00:19.52 | rob0 | Allison does mine for me. I like it being announced by her. |
| 00:20.55 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 00:23.27 | *** join/#asterisk andrewc (n=andrewc@216.160.70.198) |
| 00:25.13 | *** join/#asterisk inv_arp[work] (i=Inv_Arp@c-71-229-122-61.hsd1.fl.comcast.net) |
| 00:26.54 | saftsack | http://www.rowetel.com/ucasterisk/images/pr1_on_bf1.jpg are the 4 red elements there on this pri board fuses? |
| 00:40.29 | *** join/#asterisk bintut (n=bintut@cm170.gamma177.maxonline.com.sg) |
| 00:40.34 | bintut | hello all.. |
| 00:41.45 | saftsack | hi |
| 00:45.41 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:46.59 | bintut | how can i set my asterisk when dialing 9 first for the dialout will give me a dialtone? |
| 00:48.56 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:50.55 | rue_mohr | bintut, do you want it to give them a line right away? |
| 00:51.35 | bintut | rue_mohr: no |
| 00:51.48 | rue_mohr | hmm, thats more trickey |
| 00:51.56 | rue_mohr | I'm not sure |
| 00:52.09 | bintut | rue_mohr: i just want that all extension/local phones will hear a dial tone after they dial 9 |
| 00:52.36 | rue_mohr | but you wan tto do that without passing it off right away after they press 9 |
| 00:52.38 | bintut | rue_mohr: that's how the traditional pbx works, right? |
| 00:52.50 | rue_mohr | no, they pass off when you press 9 |
| 00:53.41 | rue_mohr | ;Outside line |
| 00:53.41 | rue_mohr | exten => _9,1,Dial,Zap/7 |
| 00:53.44 | rue_mohr | I think |
| 00:53.51 | rue_mohr | thats what mine does |
| 00:54.05 | rue_mohr | I'm not using a group cause I only have 1 outgoing line |
| 00:54.17 | bintut | rue_mohr: my understanding for the traditional pbx was if an extension/local phone wants to dial out, he/she needs to dial 9 first then followed the external telephone number. but after a user presses the 9 button, he/she will hear a dial tone |
| 00:54.26 | rue_mohr | I dont want to talk about why I have a pbx with one outgoing line |
| 00:55.00 | bintut | rue_mohr: i also have one trunkline here. this is my home setup |
| 00:55.02 | rue_mohr | bintut, yes, but when you press 9 it passes you off to the pstn |
| 00:55.10 | bintut | rue_mohr: yes |
| 00:55.17 | rue_mohr | yea |
| 00:55.18 | rue_mohr | so |
| 00:55.43 | rue_mohr | zapata.conf: |
| 00:55.47 | bintut | in my [extensions] context, i have this line ==> exten => _9.,1,Dial(zap/g2/${EXTEN:1}) |
| 00:55.58 | rue_mohr | ;Next 2 channels are the FXO card |
| 00:55.58 | rue_mohr | context = telus |
| 00:55.58 | rue_mohr | signalling = fxs_ks |
| 00:55.58 | rue_mohr | group = 2 |
| 00:55.59 | rue_mohr | channel=>7-8 |
| 00:56.12 | rue_mohr | and in extensions.conf: |
| 00:56.50 | rue_mohr | [house] |
| 00:56.51 | rue_mohr | exten => s,1,Answer ; Answer the line |
| 00:56.51 | rue_mohr | exten => s,n,BackGround(h_inside) ; Play some instructions |
| 00:56.51 | rue_mohr | exten => _9,1,Dial,Zap/7 |
| 00:57.03 | rue_mohr | oops, sorry, wrong zapata set |
| 00:57.10 | rue_mohr | but you get the idea I'm sure |
| 00:57.20 | rob0 | ~pb |
| 00:57.33 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 00:57.34 | rue_mohr | (sorry) |
| 00:57.53 | rue_mohr | you have to admit, its pretty quiet right now |
| 00:58.27 | rue_mohr | bintut, does that make sense to you though |
| 00:58.40 | bintut | rue_mohr: i still can't get it.. |
| 00:59.17 | rue_mohr | bintut, tell me what you have done |
| 00:59.20 | bintut | rue_mohr: my problem is on how an extension user hears a dialtone after he/she presses 9 |
| 00:59.48 | rue_mohr | I dont understand |
| 01:00.02 | rue_mohr | you hear a dialtone, you press 9 you hear a dialtone |
| 01:00.12 | rue_mohr | first one is from pbx, second from pstn |
| 01:00.25 | *** join/#asterisk seanwg3 (n=seanwg3@216.249.37.157.ppp.northrock.bm) |
| 01:00.42 | rue_mohr | I dont do the first dialtone, I have it set up to start speeling a numbers directory |
| 01:01.23 | rue_mohr | you pick it up "bliiiinnnnggggg , press 9 to make an outside call, dial 201 for dan, dial 203 for phil...." |
| 01:01.38 | seanwg3 | anyway in asterisk to measure jitter or latency easily? while a call is up? |
| 01:01.57 | seanwg3 | i have a PIIII running Mythtv, now running asterisk |
| 01:01.57 | bintut | rue_mohr: when you pick up the phone, normally you will hear a dialtone.. that's normal.. but, if you are going to dial out to the pstn line, you have to press 9 then followed by the telephone number you are about to call.. now, what i want is this.. when i pickup the phone, i will hear a dialtone.. then when i press/dials 9 then i should hear another dialtone then dials the pstn number |
| 01:02.12 | seanwg3 | it works .. but calls seem to have allot of distoration at times |
| 01:02.27 | rue_mohr | bintut, |
| 01:02.51 | rue_mohr | in the context the subscriber gets the they pick up the line, put this in |
| 01:02.56 | rue_mohr | exten => _9,1,Dial,Zap/7 |
| 01:03.09 | rue_mohr | where Zap/7 is replaced by your outgoing line |
| 01:03.25 | rue_mohr | it will connect the user to the pstn AS SOON AS they press 9 |
| 01:03.36 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:03.40 | rue_mohr | (line or group) |
| 01:03.49 | bintut | but, what about the dial tone after they press 9? |
| 01:03.57 | rue_mohr | [anyone who knows more than I do, feel free to interrupt with corrections] |
| 01:04.04 | rue_mohr | bintut, they have it |
| 01:04.08 | rue_mohr | from the pstn |
| 01:04.22 | rue_mohr | the pstn dosn't recieve the 9, it was eaten by asterisk |
| 01:04.24 | bintut | rue_mohr: no, i can't hear it from my side.. |
| 01:04.39 | rue_mohr | bintut, |
| 01:04.59 | rue_mohr | how many phones do you have on your system |
| 01:05.16 | bintut | rue_mohr: remember, this is my home pbx and i'm also the user of this setup.. my analog phone which is connected to my fxs port is right beside me.. |
| 01:05.32 | rue_mohr | how many phones do you have on your system |
| 01:05.46 | bintut | rue_mohr: currently, 1 analog phone to fxs and i have 2 sip phones |
| 01:05.54 | rue_mohr | ok! |
| 01:05.59 | rue_mohr | do you have the sip phones working? |
| 01:06.09 | bintut | rue_mohr: yes.. |
| 01:06.11 | rue_mohr | ok! |
| 01:06.25 | rue_mohr | do this |
| 01:07.15 | rue_mohr | rewrite the extensions.conf for the analog phone context, make it immediatly connect to the sip phone |
| 01:08.28 | rue_mohr | exten => s,1,Answer |
| 01:08.28 | rue_mohr | exten => s,n,Dial,Zap/1|30 |
| 01:08.38 | rue_mohr | or whatever your sip phone is |
| 01:08.48 | rue_mohr | tell me if that works |
| 01:09.43 | rue_mohr | when you pick up the analog phone, the sip phone shoudl start ringing |
| 01:11.13 | *** join/#asterisk mceGEEK (n=nospam@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 01:11.30 | mceGEEK | hi .. any suggestions on voip providers |
| 01:11.42 | tzanger | MrWup: where are you going to be terminating? |
| 01:11.46 | tzanger | north america, europe, what |
| 01:11.55 | tzanger | that makes a bit of a difference, wouldn't you think |
| 01:11.57 | MrWup | uk |
| 01:12.07 | tzanger | er not MrWup mceGEEK sorry |
| 01:12.33 | mceGEEK | USA :) |
| 01:12.54 | tzanger | mceGEEK: then I'd suggest US-hosted providers with good reputations |
| 01:12.58 | tzanger | personally I like nufone and asterlink |
| 01:12.59 | mceGEEK | i have sunrocket apparently it has started stinking so have to get rid of it |
| 01:13.03 | tzanger | both are supporters of Asterisk |
| 01:13.12 | tzanger | rates are pretty good |
| 01:13.17 | bintut | rue_mohr: my current /etc/asterisk/{extensions,zapata}.conf ==> http://paste.debian.net/30080 |
| 01:13.30 | mceGEEK | tzanger: do you get DID |
| 01:13.48 | tzanger | no, but both have them for various locations |
| 01:13.53 | rue_mohr | bintut, hu!?! |
| 01:13.53 | tzanger | there are scads of US VOIP providers |
| 01:13.55 | bintut | rue_mohr: before i make changes |
| 01:13.59 | *** join/#asterisk Tomasu (n=moose@S0106000c765956b8.ed.shawcable.net) |
| 01:14.06 | rue_mohr | bintut, but that will just ... |
| 01:14.16 | bintut | rue_mohr: what do you think of my current configs? |
| 01:14.22 | rue_mohr | leave you witha busy signal wont it? |
| 01:14.29 | Qwell | tzanger: asterlink a supporter of asterisk? ...they stopped allowing iax2... O.o |
| 01:14.41 | Tomasu | anyone use freepbx here? (the #freepbx channel is a tad slow atm), I'm getting a "DB error: unknown error". |
| 01:14.49 | Qwell | ~freepbx |
| 01:15.02 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:15.02 | tzanger | Qwell: well both bkw and anthm know asterisk well and contract themselves out to fix problems with it |
| 01:15.09 | Qwell | tzanger: sure |
| 01:15.20 | tzanger | Qwell: and to give him his due, anthm knows his shit |
| 01:15.25 | tzanger | even if he believes asterisk is shit :-) |
| 01:15.37 | rue_mohr | bintut, |
| 01:15.41 | Tomasu | Qwell, notice how I mentioned I already joined. bound to be some poeple here awake that use it. |
| 01:15.48 | Qwell | Tomasu: nope, none |
| 01:15.49 | bintut | rue_mohr: i'm a newbie actually.. i'm just testing them out. honestly, i don't understand them all.. |
| 01:15.49 | rue_mohr | want me to adjust this nd you can give it a try? |
| 01:15.53 | Qwell | we burned them all at the stake |
| 01:16.01 | bintut | rue_mohr: yes, please..? :) |
| 01:16.04 | rue_mohr | bintut, I'm a little less new than you |
| 01:16.04 | tzanger | Qwell: what, newbies? |
| 01:16.07 | tzanger | dammit I missed it |
| 01:16.10 | Qwell | tzanger: freepbx users |
| 01:16.13 | tzanger | oh |
| 01:16.13 | rue_mohr | by about > < that much |
| 01:16.16 | tzanger | that'd leave an awful smell |
| 01:16.37 | Qwell | it did |
| 01:16.37 | mceGEEK | tzanger: does nufone charge per min for local calls or are local calls free |
| 01:16.37 | Qwell | still trying to wash it off my clothes |
| 01:16.38 | tzanger | mceGEEK: again, I don't know how they charge for DIDs |
| 01:16.45 | Qwell | mceGEEK: There really isn't a such thing as a "local call" with an ITSP |
| 01:16.49 | tzanger | check nufone.net or find shido6 or jerjer here |
| 01:17.34 | bintut | rue_mohr: actually, i still have 3 other problems on my list that i need to solve.. :( |
| 01:17.46 | rue_mohr | hold up |
| 01:19.32 | mceGEEK | ITSP .. interesting |
| 01:19.36 | rue_mohr | bintut, http://paste.debian.net/30081 |
| 01:19.53 | rue_mohr | try that, when you pick up the analog phone it should ring your sip phone right away |
| 01:20.00 | rue_mohr | I think |
| 01:21.10 | *** part/#asterisk seanwg3 (n=seanwg3@216.249.37.157.ppp.northrock.bm) |
| 01:21.47 | rue_mohr | hmm, I need to figure out how to fit all this junk into the wiring closet |
| 01:22.37 | tzanger | ok |
| 01:22.47 | rue_mohr | small channelbank wouldn't hurt |
| 01:22.51 | tzanger | there is something funny about seeing in my tv listings |
| 01:22.52 | rue_mohr | smaller |
| 01:23.01 | tzanger | "Deja Vu (rerun)" |
| 01:24.38 | rue_mohr | suppose I could tear out a few more shelfs |
| 01:27.43 | bintut | rue_mohr: i still don't get a dialtone after pressing 9.. :( |
| 01:28.16 | bintut | rue_mohr: and, when i picked up my analog phone, my sip phone didn't ring at all |
| 01:28.54 | rue_bed | no |
| 01:29.16 | rue_bed | that is supposed to ring your sip phone when you pick up the analog one |
| 01:29.26 | rue_bed | nothing at all with dialing 9 |
| 01:29.34 | rue_bed | you have to get that working first |
| 01:29.43 | rue_bed | before you start trying to make dialplans |
| 01:29.47 | bintut | yeah.. but again, my sip phone didn't ring |
| 01:29.58 | rue_bed | yea, THAT is something to fix |
| 01:30.02 | rue_bed | I dont know how |
| 01:30.21 | rue_bed | anyone know anything about sip phones? |
| 01:30.24 | bintut | ok.. thanks for your assistance |
| 01:30.35 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 01:30.36 | rue_bed | I can tell you how to do it with analog phones |
| 01:30.52 | bintut | rue_bed: i'm more interested with analog phones |
| 01:30.53 | rue_bed | I dont have any sip phones on my system yet |
| 01:31.12 | bintut | rue_bed: actually, sip softphone on my pda |
| 01:31.14 | rue_bed | bintut, now many fxs channels does your system have? |
| 01:31.35 | rue_bed | is it more or less than 1.5 |
| 01:31.37 | bintut | rue_bed: only 1 for now.. and i still have 1 extra slot |
| 01:31.43 | rue_bed | ah, k |
| 01:31.55 | rue_bed | that makes it hard to plug in more than 1 phone |
| 01:32.05 | bintut | i am using the digium's dev kit which comes with 1 fxo and 1 fxs |
| 01:32.20 | rue_bed | hmm, I'm using a T1 card and a channelbank |
| 01:32.44 | rue_bed | I suppose I have more roommates |
| 01:32.44 | bintut | cool! you may have so many phones at home? :) |
| 01:33.08 | rue_bed | hmm, 5 for now |
| 01:33.13 | rue_bed | soon to be 6 |
| 01:33.28 | rue_bed | and soon to be 2 fxo ports |
| 01:33.29 | bintut | ah.. :) |
| 01:34.30 | rue_bed | that leaves me 16 ports... |
| 01:34.31 | lee_is_me | anyone up for a polycom config question? |
| 01:34.47 | rue_bed | lee_is_me, I cant help |
| 01:34.48 | bintut | wow! |
| 01:35.24 | mceGEEK | any voicepulse users in here? |
| 01:35.25 | rue_bed | $500 for a T1 card, and a tiny bit extra for a channelbank off ebay |
| 01:35.46 | lee_is_me | rue_bed: was that really a "can't"?? |
| 01:35.59 | lee_is_me | lol |
| 01:36.03 | rue_bed | yea, I'm a newb |
| 01:36.16 | rue_bed | I know about > < that much |
| 01:36.24 | lee_is_me | ok, just struck me odd, lol |
| 01:37.04 | rue_bed | well if everyone did that, then you would know you were heard by them |
| 01:37.44 | mceGEEK | what voip providers are you guys using .. i already checked voicepulse, asterlink and nuphone |
| 01:38.13 | lee_is_me | axVoice |
| 01:38.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 01:38.39 | mceGEEK | lee_is_me: how is the service |
| 01:39.01 | lee_is_me | the server as in communication quality, etc is wonderful |
| 01:39.10 | lee_is_me | err, I mean to say "service" |
| 01:39.24 | lee_is_me | their customer service is pretty good to, but preferably by email |
| 01:39.39 | mceGEEK | cool |
| 01:50.29 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:51.15 | *** join/#asterisk joebob777as7 (n=corn13re@67-42-53-56.eugn.qwest.net) |
| 01:52.26 | joebob777as7 | i'm going to sign up with teliax to test my system out. I am running sip phones but it is asking me if I am using my own sip or iax device? can someone help |
| 01:52.28 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:53.14 | rudholm | Asterisk can connect to teliax via either SIP or IAX |
| 01:53.26 | rudholm | I'm using teliax on my Asterisk with IAX |
| 01:53.46 | joebob777as7 | rudholm, which should i choose? does it matter? |
| 01:54.33 | rudholm | either will work, and you can change it if you want later. a friend of mine who is far more experienced with Asterisk than I am recommended IAX, so I went with that and it's been fine. |
| 01:54.43 | rudholm | IAX seems a bit easier to configure |
| 01:55.13 | rudholm | I use SIP with BroadVoice and it seems a bit fussier (but most of that has to do with the fact that I'm registering to two BV accounts from one Asterisk) |
| 01:55.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:55.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:56.52 | rudholm | I'm having an odd call progress tone problem. I have an auto-attendant setup that doesn't supervise until the call is actually answered by a human (or voicemail, etc). |
| 01:57.12 | rudholm | callers from POTS lines don't hear ringback after they enter the desired extension |
| 01:57.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 01:57.47 | rudholm | if I answer/supervise the call before (or during) the Background command, then progress tones work fine |
| 01:58.56 | joebob777as7 | thank you rudholm! Did you not like teliax? |
| 01:59.13 | rudholm | oh, I like them. they're great. |
| 01:59.30 | joebob777as7 | rudholm, why are you using BV then? |
| 01:59.34 | rudholm | I use them for nearly all of my outbound call completion |
| 02:00.00 | rudholm | because BV doesn't charge for inbound calls and gives me additional numbers for 1.99$/month. |
| 02:00.16 | rudholm | BV's "BYOD Lite" plan is pretty cheap |
| 02:00.27 | MrWup | exten => s,1,Answer() exten => s,n,Goto(pstn-incoming,2,1) |
| 02:00.29 | joebob777as7 | rudholm, Wow! do you like their service? |
| 02:00.29 | MrWup | this works fine |
| 02:00.37 | MrWup | but i i put a playback between those two extensions |
| 02:00.42 | MrWup | it plays the file |
| 02:00.45 | MrWup | but then doesnt do the goto |
| 02:00.48 | MrWup | anyone know why? |
| 02:00.58 | joebob777as7 | rudholm, have you heard anything about RNK? |
| 02:01.27 | rudholm | BV's service is not nearly as good as Axvoice or Teliax, but BV is cheaper, so I use a combination of providers to end up with something that suits my needs best. |
| 02:01.32 | rudholm | no, haven't heard of RNK |
| 02:01.52 | Strom_C | hello |
| 02:01.53 | joebob777as7 | which is better axvoice or teliax rudholm ? |
| 02:02.02 | Strom_C | I've heard of RNK, but I didn't know they provided consumer service :) |
| 02:02.46 | joebob777as7 | Strom_C, how is their service? are they a big company? do you think they'll last? |
| 02:02.50 | rudholm | I'm using teliax because they have a 2 cents per minute plan with zero monthly fee and the let me set my calling number. |
| 02:03.17 | Strom_C | joebob777as7: well I know they've been around for some time, but I don't really know much else about them |
| 02:03.42 | joebob777as7 | Strom_C, ok thank you. |
| 02:03.45 | rudholm | joebob777as7: so with teliax, when I forward calls off my Asterisk (like, to my cell) I can preserve the original Calling Number if I send the call out for completion through Teliax. |
| 02:04.32 | rudholm | but as far as quality and customer service goes, both Axvoice and Teliax are very good. |
| 02:04.54 | MrWup | hmm this is weird |
| 02:04.58 | MrWup | maybe its something to do with my audio file |
| 02:05.01 | MrWup | but when i play beep |
| 02:05.06 | MrWup | it does the goto right away |
| 02:05.13 | MrWup | with my audio file it waits for ages after the end |
| 02:05.22 | MrWup | yet theres no silence at the end of my audio file |
| 02:05.26 | joebob777as7 | rudholm, thanks a bunch! |
| 02:06.51 | rudholm | Strom_C: I'm having this weird problem with call progress tones on unsupervised calls :-) |
| 02:07.10 | rudholm | Strom_C: was I right about that track ID? |
| 02:07.23 | joebob777as7 | Strom_C, do you know how long rnk has been around in terms of years? 1,2 5? |
| 02:07.25 | Strom_C | rudholm: no |
| 02:07.38 | Strom_C | joebob777as7: no; why don't you go read their webpage? |
| 02:08.16 | *** join/#asterisk remmo (n=junk@203.62.147.6) |
| 02:08.49 | rudholm | I'm listening to it now, and it sounds right |
| 02:08.58 | Strom_C | rudholm: it quite definitely feels like Vince Clarke, but it is /not/ Situation |
| 02:09.54 | rudholm | there are a bunch of remixes of Situation |
| 02:09.59 | rudholm | one of those, perhaps? |
| 02:10.03 | rudholm | were there vocals? |
| 02:10.24 | Strom_C | yeah, but I couldn't hear clearly enough |
| 02:10.27 | Strom_C | it could be "Don't Go" |
| 02:10.33 | lee_is_me | joebob777as7: haven't tried Teliax, but axVoice has been great for 9 months |
| 02:10.49 | joebob777as7 | lee_is_me, thanks! |
| 02:12.11 | rudholm | Strom_C: Have you heard State Farm? |
| 02:12.16 | Strom_C | yeah, I have the 12" |
| 02:12.19 | rudholm | oh right |
| 02:12.22 | rudholm | so it's not that |
| 02:12.36 | Strom_C | I heard lyrics that might have been "don't go" |
| 02:12.44 | rudholm | do you have Don't Go? |
| 02:12.47 | Strom_C | no |
| 02:12.52 | rudholm | oh, let me fix that.. |
| 02:13.04 | Strom_C | I thought it was saying "Don't Let Go" so it's probably "Don't Go" |
| 02:13.56 | Strom_C | turn your slimserver on :) |
| 02:16.12 | rudholm | heh |
| 02:16.16 | rudholm | ok |
| 02:16.29 | rudholm | give me a minute to update it |
| 02:20.15 | rudholm | Strom_C: ok, it's up |
| 02:20.51 | rudholm | Strom_C: hmm, database must be out of date, I see no Yaz(oo) |
| 02:20.56 | MrWup | whats the best way to simulate ringing? |
| 02:21.23 | rudholm | Strom_C: oh, nm, I was looking under "albums" instead of "artists" |
| 02:21.27 | rudholm | MrWup: what do you mean? |
| 02:21.43 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:21.49 | MrWup | someone calls, i answer, i simulate ringing while waitexten before going on to answer properly |
| 02:22.07 | Strom_C | rudholm: yes, this is it |
| 02:22.07 | MrWup | so people in my company have 3 seconds while most people think it is dialling, in order to enter their secret code |
| 02:22.24 | Strom_C | MrWup: Ringing() |
| 02:22.45 | MrWup | oh i did playtones(ring) |
| 02:22.48 | MrWup | which is better? |
| 02:22.58 | rudholm | Neither work. :-) |
| 02:23.01 | Strom_C | depends on if you're supervised yet |
| 02:23.09 | Strom_C | Ringing() will send an actual ISDN ALERTING message |
| 02:23.24 | Strom_C | Playtones() will just play tones inband |
| 02:23.42 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-85-185.hag.east.verizon.net) |
| 02:23.42 | rudholm | ...and they both result in silence! :-p |
| 02:24.23 | rudholm | Strom_C: you might like "Bad Connection" --it's about an operator-assisted telephone call |
| 02:24.27 | Strom_C | rudholm: I have no idea why your PBX is doing that |
| 02:25.14 | Strom_C | rudholm: do you have the 12" of 'Genetic Engineering'? |
| 02:25.22 | Strom_C | er, sorry, the 312mm ;) |
| 02:25.29 | rudholm | I don't think so |
| 02:25.37 | rudholm | it's such a perky track :) |
| 02:25.48 | Strom_C | i picked it up at amoeba today |
| 02:25.54 | rudholm | cool |
| 02:25.58 | rudholm | I should visit that place more |
| 02:26.08 | Strom_C | yeah, today was a bumper crop day |
| 02:26.11 | Strom_C | i bought too many records |
| 02:26.13 | rudholm | haha |
| 02:26.34 | Mavvie | anybody from Sydney, Australia here? |
| 02:26.49 | Mavvie | (except me) |
| 02:30.45 | Strom_C | Mavvie: I'm going to be connecting through Sydney on my way to Melbourne |
| 02:30.47 | Strom_C | does that count? |
| 02:30.53 | MrWup | module reload |
| 02:30.55 | MrWup | oops |
| 02:30.55 | MrWup | lol |
| 02:31.06 | Strom_C | ERROR: This is IRC, nub :) |
| 02:31.11 | rudholm | heh |
| 02:31.18 | Mavvie | Strom_C: close, but not really what I had in mind. |
| 02:31.28 | Strom_C | yeah, I figured :) |
| 02:31.33 | rudholm | Strom_C: hey, you paused it! you were almost to the telephone sounds |
| 02:31.37 | Mavvie | Strom_C: I was wondering if other asterisk users in Sydney were interested in meeting up. |
| 02:32.21 | Strom_C | rudholm: yeah, I took a break to listen to my 'genetic engineering' record |
| 02:43.20 | Strom_C | rudholm: are you playing with the phone? |
| 02:54.52 | triplep | anyone ever have festival only play one word for each AGI call? it's only playing the first word of the first line, first word of the second line, and so on |
| 02:59.06 | *** join/#asterisk wikityler (n=wikityle@S01060004e2f8b842.gv.shawcable.net) |
| 02:59.38 | rudholm | Strom_C: playing with which phone? |
| 03:00.01 | wikityler | what is the de facto web interface for asterisk? |
| 03:00.06 | rudholm | Strom_C: I didn't call you, if that's what you're asking |
| 03:00.09 | *** join/#asterisk cy303 (n=cy@208.69.76.73) |
| 03:00.15 | cy303 | ugh @ SIP behind NAT. |
| 03:00.34 | cy303 | externip is key.. anything else? |
| 03:00.46 | rudholm | nat=yes |
| 03:00.52 | cy303 | yeah, got that |
| 03:00.58 | rudholm | then you should be good |
| 03:01.03 | cy303 | calls are going through and external clients are registered |
| 03:01.07 | cy303 | but no audio either way |
| 03:01.26 | rudholm | what ports have you forwarded? |
| 03:01.38 | cy303 | 5060 and 10000-20000 for xlite |
| 03:01.51 | Strom_C | rudholm: I was referencing the song :) |
| 03:01.58 | cy303 | and 4569 for iax peers |
| 03:02.04 | rudholm | Strom_C: ah, yes, the operator |
| 03:02.17 | rudholm | cy303: that should do you |
| 03:02.23 | cy303 | hmmm, it's not :( |
| 03:02.35 | *** part/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 03:02.47 | RypPn | phones ringing, but no audio? |
| 03:02.53 | rudholm | cy303: UDP, right? |
| 03:04.57 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:04.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:05.59 | cy303 | yeah UDP |
| 03:07.43 | cy303 | heh |
| 03:08.18 | cy303 | nat=yes in sip.conf [general] context |
| 03:08.29 | cy303 | and externip |
| 03:08.32 | rudholm | Strom_C: you think running a later version of Asterisk might make a difference with my absent progress tones? |
| 03:08.42 | cy303 | externip = <my external IP> |
| 03:09.17 | rudholm | right |
| 03:09.37 | rudholm | I put nat=no in [general] and set it to yes in the individual sip profiles that needed it |
| 03:10.49 | rudholm | most of my sip clients are on the same VPN as the Asterisk box, so they are nat=no, but one or two are not on the VPN (and are behind someone's NAT) |
| 03:11.03 | cy303 | cool |
| 03:11.08 | cy303 | alright, done that... |
| 03:15.09 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.12 | MrWup | is it possible to originate a transfer from the manager interface? |
| 03:18.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 03:18.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:25.47 | Strom_C | rudholm: I'm not sure, honestly |
| 03:27.04 | rudholm | Strom_C: have you seen progress tones work on pre-supervised calls? |
| 03:27.20 | rudholm | I mean, obviously, Playback and Background work |
| 03:27.40 | Strom_C | rudholm: yes, i think so |
| 03:27.46 | Strom_C | lemme screw with mine tho |
| 03:27.50 | rudholm | cool |
| 03:32.36 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:34.22 | Strom_C | rudholm: who are you using for the inbound service? |
| 03:34.30 | rudholm | BV |
| 03:34.33 | Strom_C | oh ok |
| 03:34.45 | Strom_C | teliax seems to be overwriting my progress audio with ringing |
| 03:34.45 | Strom_C | the cunts |
| 03:34.46 | rudholm | all my inbound voip is BV |
| 03:34.52 | rudholm | ew |
| 03:34.55 | Strom_C | yeah |
| 03:34.56 | Strom_C | im gonna complain |
| 03:35.23 | rudholm | what if you want to send a busy, or a reorder? |
| 03:35.43 | Strom_C | lets see what happens if I do send Busy() |
| 03:36.13 | rudholm | BV is cool in some ways, and evil in others |
| 03:36.24 | Strom_C | yeah, they don't pass the BUSY back to my handset |
| 03:36.57 | rudholm | if they didn't have evil ToS, term commitments, and intermittent service quality/availability, they'd be great :) |
| 03:37.23 | Strom_C | rudholm: are you up for going out tonight? |
| 03:37.32 | rudholm | their support is actually good, and their prices are great, and they don't get in your way much (except for not allowing you to set CID) |
| 03:38.43 | Strom_C | let's play with broadvoice now... |
| 03:39.04 | rudholm | a friend of mine is was bored and is bringing over a DVD. I can't really do anything 'cause I have to be available for stuff going on at work |
| 03:39.15 | Strom_C | alright |
| 03:39.32 | Strom_C | I had a fun time there last week; this week they supposedly have a live band or something |
| 03:39.34 | rudholm | yeah, I'm curious if you can get progress tones through via BV |
| 03:39.39 | rudholm | neat |
| 03:41.12 | Strom_C | Progress() Wait(5) Playtones(ring) Wait(500) seems to work... |
| 03:41.25 | Strom_C | now to try sticking a sound file in there |
| 03:42.16 | Strom_C | "Baltimore *ring* *ring* |
| 03:43.01 | Strom_C | works with Background() also |
| 03:43.08 | Strom_C | which BV server are you registering to? |
| 03:44.44 | *** join/#asterisk techie (n=gus@antibala.net) |
| 03:50.31 | MrWup | how do i send an intercom call to one of my sip phones from asterisk? |
| 03:50.41 | Strom_C | what do you mean "intercom call"? |
| 03:50.50 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 03:51.35 | bintut | rue_bed: still there? |
| 03:52.35 | MrWup | well |
| 03:52.35 | *** join/#asterisk bmg505 (n=leon@196.209.183.105) |
| 03:52.35 | MrWup | i want to originate a call from one of my sip phone to voice mail, and have the sip phone auto answer the call |
| 03:52.36 | MrWup | so someone presses a button on their pc screen |
| 03:52.38 | MrWup | and their phone immediately starts playing their voicemail |
| 03:52.39 | bintut | what is the use of the "immediate=yes/no" in zapata.conf ? |
| 03:52.45 | Strom_C | bintut: PLAR |
| 03:53.00 | bintut | Strom_C: what do you mean by PLAR? |
| 03:53.08 | Strom_C | MrWup: use a combination of auto-answer and AMI |
| 03:53.14 | Strom_C | bintut: private line automatic ringdown |
| 03:53.20 | Strom_C | i.e. the cold war hotline to the Kremlin |
| 03:53.26 | Strom_C | pick it up, and it rings another phone |
| 03:53.33 | MrWup | Strom_C, yeah i gathered that |
| 03:53.37 | MrWup | not sure how to though |
| 03:53.46 | MrWup | i can turn auto answer on on my 9133is |
| 03:53.51 | MrWup | and i know how to originate a call with the AMI |
| 03:53.59 | Strom_C | MrWup: look up how to insert the correct SIP header into your setup message for your phone |
| 03:54.01 | MrWup | but im not sure what to send to cause my 9133is to autoanswer |
| 03:54.09 | Strom_C | AddHeader, IIRC |
| 03:57.08 | bintut | what is the purpose of adding an "underscore" before the 9 number? i.e. _9.,1,Dial(Zap/g2/${EXTEN:1}) |
| 03:58.07 | Strom_C | bintut: pattern matching |
| 03:58.14 | Strom_C | bintut: sounds like you should read the book |
| 03:58.15 | Strom_C | ~book |
| 03:58.31 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:58.38 | cy303 | anyone feel like helping me troubleshoot one-way sip audio? woot! I know you do |
| 03:59.17 | bintut | cy303: open your firewall's UDP 10000 and beyond.. |
| 04:00.38 | cy303 | bintut: I have 10000-20000 UDP forwarded to the asterisk server |
| 04:01.00 | Strom_C | cy303: asterisk is behind NAT? |
| 04:01.03 | cy303 | yes |
| 04:01.16 | Strom_C | have you set externip, localnet, etc in sip.conf? |
| 04:01.20 | cy303 | yes |
| 04:01.24 | bintut | maybe you need a sip router in between? |
| 04:01.34 | Strom_C | is the other endpoint behind a different NAT? |
| 04:01.43 | cy303 | yes |
| 04:01.49 | Strom_C | what are the settings on that NAT? |
| 04:02.01 | Strom_C | two NATs with a SIP connection always causes uber-headaches |
| 04:02.49 | bintut | how do i set my asterisk when dialing 9 first before the telephone number i wish to call is that i can hear a dialtone from my phone? |
| 04:02.58 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-116-61.columbus.res.rr.com) |
| 04:03.07 | cy303 | Strom_C: yeah, heh |
| 04:03.34 | cy303 | Strom_C: was just telling rudholm that I always fight with this but eventually get it working from setup to setup |
| 04:03.40 | cy303 | it's always a pain in the ass |
| 04:03.40 | jql | bintut: DISA |
| 04:03.45 | cy303 | but eventually works right |
| 04:03.58 | bintut | because, dialing 9 first will trigger outgoing calls but i want to set my asterisk in such a way that after i dial 9 i will hear a dialtone |
| 04:04.02 | bintut | jql: what's DISA? |
| 04:04.27 | jql | the thing you may be looking for |
| 04:04.57 | Strom_C | bintut: what kind of telephone are you using? |
| 04:05.16 | cy303 | just have one-way audio issues now |
| 04:05.21 | Strom_C | cy303: why not just save yourself the headache and put asterisk in front of the NAT? |
| 04:05.29 | bintut | Strom_C: 1 analog connected to my fxs port of my digium dev kit |
| 04:05.32 | cy303 | :P |
| 04:05.35 | cy303 | I suppose I could do that |
| 04:05.46 | Trentster | hey all, I am having a strange problem, I am sitting in australia, using a snom 320 phone to connect via sip to an asterisk server in another country and breaking out from there to call local numbers......The problem is I can make the call and we can hear each other for about 2 secs then it hangs up the call, from what i can tell by the logs it thinks I am hanging up.....other users do not have this problem......incoming calls |
| 04:05.47 | Trentster | <PROTECTED> |
| 04:05.50 | Strom_C | bintut: put "ignorepat => 9" in your phone's context |
| 04:05.58 | bintut | cy303: you need a sip router on that i think |
| 04:06.15 | bintut | Strom_C: yeah, i already have that.. but i want a "pause" |
| 04:06.23 | jql | Trentster: Wow, that's weird. Do you have a log trace of that? |
| 04:06.31 | cy303 | bintut: I have it up and running in a couple installments |
| 04:06.45 | cy303 | but, those settings don't apply to this setup apparently :/ |
| 04:06.46 | cy303 | annoying |
| 04:06.48 | Strom_C | bintut: you don't want to use DISA() for something as simple as "dial 9 first" |
| 04:07.07 | Trentster | jql, yes it allways says " == Spawn extension (from-internal, *97, 5) exited non-zero on 'SIP/6720-0892ff50' -- Executing Macro("SIP/6720-0892ff50", "hangupcall") in new stack" |
| 04:08.16 | Strom_C | bintut: you could do something like playtones(recall) and then waitexten() |
| 04:08.16 | bintut | Strom_C: basically what i want is: pick up the phone and i will hear a dial tone then dials 9 .. pause .. dial tone then dial telephone number for external call |
| 04:08.17 | Strom_C | yes I know |
| 04:08.17 | Strom_C | try that ^^^^^^^^^^^^ |
| 04:08.21 | jql | Trentster: Well, perhaps you could turn on 'sip debug' and get a full trace of the communication between the phone and asterisk. It'd be interesting to see what the hangup cause is |
| 04:08.51 | Trentster | jql, sip debug? whats the command to turn it on? |
| 04:09.00 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 04:09.01 | jql | sip debug # this should be enough |
| 04:09.17 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:09.23 | Trentster | jql, thanks lemme make a call and see whats up.... |
| 04:09.32 | bintut | Strom_C: where will i put the "playtones(recall)" ? is it after the "ignorepat => 9" ? |
| 04:10.04 | Strom_C | bintut: no |
| 04:10.50 | Strom_C | something like exten => 9,1,goto(outbound_stutter,s,1) |
| 04:10.51 | Strom_C | then |
| 04:11.05 | Strom_C | in your [outbound_stutter] context |
| 04:11.08 | Trentster | jql, weird, getting lots and lots of output, this seems to be common" Destroying call '11de204065c875163e7bd8c669b503fb@209.212.101.xx' |
| 04:11.53 | Strom_C | exten => s,1,progress() exten => s,n,playtones(recall) exten => s,n,waitexten(20) and then you include your outbound as direct 1+ dialing |
| 04:12.24 | jql | Trentster: Yeah, there will be about 300 lines of output for a single call. You'd want to put it in a pastebin, minus any possibly sensitive |
| 04:12.54 | Trentster | jql, seems to still be running even tho call is dropped and phone is down.... |
| 04:13.10 | jql | sip no debug # to disable |
| 04:13.21 | jql | or sip nodebug. I forget which is for which |
| 04:13.54 | Trentster | jql, is that pastebin.org ??? |
| 04:14.02 | jql | pastebin.ca is good |
| 04:14.37 | bintut | my extensions.conf and zapata.conf for your comments and suggestions |
| 04:14.52 | bintut | Strom_C: i'll apply that one now.. wait.. |
| 04:15.37 | bintut | http://paste.debian.net/30086 |
| 04:16.26 | *** join/#asterisk peanutb_ (n=paulb@c-71-231-82-231.hsd1.or.comcast.net) |
| 04:18.47 | Trentster | jql, http://www.pastebin.ca/554443 |
| 04:20.40 | Trentster | jql, this line is still being repeated all the time even though the call is over ten minutes ago " Destroying call '3a1533f71d51649f667705e16485669d@209.212.101.xx'" is this normal |
| 04:21.53 | jql | Trentster: Looks like a normal response to your phone's keep-alive messages |
| 04:26.51 | jql | Trentster: It looks like asterisk is ignoring the call accept for some reason |
| 04:26.58 | Trentster | jql, looks like the error occurs between line 345 and 420 |
| 04:29.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:29.31 | Trentster | jql, I have tried from my side 2 x different ISP, and a softphone as well, and I pick up the same problem, so the problem I am assuming is definately on the asterisk pabx side.....could it be a latency problem, that is dropping, cause i notice i have the highest latency compared to any of the other users... |
| 04:29.55 | jql | Trentster: I don't see why asterisk is ignoring the "ACK" messages on that call. Can you turn on "set debug 4" and "set verbose 4" and "sip debug" and give it another try? |
| 04:32.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:35.47 | Trentster | jql ok |
| 04:36.04 | Trentster | jql one sec |
| 04:39.39 | Trentster | jql, http://www.pastebin.ca/554465 |
| 04:41.12 | jql | Trentster: Can you look at /var/log/asterisk/full, or is that enabled? |
| 04:44.33 | Trentster | jql, i am looking at "/var/log/asterisk/full" what exactly am i looking for as hundreds of lines keep scrolling by all the time |
| 04:45.50 | jql | Trentster: yeah, you have quite a volume there. |
| 04:46.01 | Trentster | jql, do you want me to try and grep something from there? |
| 04:46.10 | jql | What I'm looking for is a reason why the ACK on Call-ID: 3c2a60a8493e-7gnq2vbwskte@ is being ignored by Asterisk |
| 04:46.38 | jql | the ACK arrives, and asterisk does nothing. I need some trace info on that. :) |
| 04:47.00 | Trentster | shall i grep "3c2a60a8493e-7gnq2vbwskte@" and then pastebin it? |
| 04:47.01 | cy303 | Strom_C: so I opted to just make the * server the DMZ for now and two way audio works dandy, however now my iaxregistry's are acting up.. request sent and auth. sent.. not displaying registered :/ |
| 04:47.26 | jql | well, I need the lines from the same second as each of those |
| 04:47.40 | jql | so, anything with the same timestamp as an ACK with that callid |
| 04:48.16 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 04:49.35 | cy303 | hmm guess I'm getting some packet loss... wtf |
| 04:53.22 | *** join/#asterisk akant2 (n=root@ip24-252-28-161.om.om.cox.net) |
| 04:53.58 | akant2 | Any ideas why my SIP clients cannot get asterisk 1.4 to react when trying to do a call transfer? |
| 04:54.47 | akant2 | I have everything set for defaults in features.conf |
| 04:54.53 | Trentster | jql, thanks just busy trying to cut and paste all relating to that call to pastebin.... |
| 04:54.56 | akant2 | asterisk IS in the media path |
| 04:55.08 | akant2 | Dial command has Tt options |
| 04:55.08 | jql | I know what you mean |
| 04:55.19 | jql | -rw-r--r-- 1 root root 13307733971 Jun 9 21:55 /var/log/asterisk/full |
| 04:56.09 | akant2 | DTMF mode set for the rfc |
| 04:56.23 | akant2 | but none of my clients can get asterisk wake up :) |
| 04:56.49 | Trentster | jql, http://www.pastebin.ca/554487 |
| 04:57.22 | Trentster | jql, hope that helps......or else i can zip the file and email it to you......thanks for your help btw ..much appreciated...!!! |
| 04:58.20 | jql | ok, looking |
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| 05:22.00 | bintut | brb |
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| 05:35.25 | cy303 | this is ridiculous! |
| 05:35.49 | cy303 | when I throw * on the DMZ two-way vox/SIP works dandy but I get packet loss |
| 05:35.51 | cy303 | what the hell |
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| 08:06.30 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 08:06.51 | Shaun2222 | anybody have or know where to get the latest sip firmware for teh polycom 550 phones? |
| 08:09.08 | jql | I could tell you, but then I'd have to shoot you |
| 08:10.06 | Shaun2222 | do you have it? |
| 08:10.27 | Shaun2222 | 2.1.0 has a reboot issue. |
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| 10:18.16 | Trentster | jql, hi, are you still around? |
| 10:18.52 | jql | yep |
| 10:22.48 | Trentster | jql, just banged through new 10k log to you via mail.... |
| 10:22.56 | jql | ooh, fun |
| 10:23.25 | jql | I know how to party on a saturday night! :) |
| 10:24.05 | Trentster | LOL.....join the club......except its sun night over here!!! |
| 10:24.29 | jql | pesky date line |
| 10:29.12 | Trentster | yeah, you must be west coast USA I presume..... |
| 10:29.20 | jql | yeah |
| 10:30.21 | Trentster | West coast Australia Myself....... |
| 10:32.00 | Trentster | Must be early Sunday Morning for you.........,, you been burning the candles from both sides it seems<grin> |
| 10:32.10 | jql | Trentster: is there more than a half-second ping time between the phone and your server? |
| 10:32.30 | Trentster | jql, yes.....about 700ms |
| 10:32.57 | jql | the call flow is surreal. that would explain it |
| 10:34.49 | Trentster | make that " time=574 ms" for ping and "sip show peers" shows a latency of OK (711 ms) |
| 10:35.21 | Trentster | elaborate on surreal, as in a salvidor dali? |
| 10:35.36 | jql | yeah, this problem is caused by the huge latency. I think this is a bug in asterisk |
| 10:35.45 | Trentster | hmmmm |
| 10:36.04 | jql | Trentster: It's like watching a movie where the audio is played 10 seconds before the on-screen action |
| 10:36.17 | tzafrir_laptop | JT, here? |
| 10:36.52 | Trentster | jql, damn......that could result in you not enjoying the movie...... |
| 10:37.21 | Trentster | jql, howcome it works when people call me, or when i phone someones sip extension directly? |
| 10:37.33 | Trentster | jql, any suggestions to remedy? |
| 10:37.57 | jql | the setting which controls that timer is compile-time, if I recall |
| 10:38.53 | jql | anyways, the phone itself has a retransmit timer which is also triggering. the phone is probably triggering the fault by sending the invite repeatedly the way it does |
| 10:39.29 | jql | hmm... the snom may have a setting for the "T1" timer, if you want to fiddle with that |
| 10:40.04 | jql | if you can make it 1000ms, that might mitigate the problem |
| 10:41.38 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-124-139.w90-36.abo.wanadoo.fr) |
| 10:41.41 | Trentster | jql, thanks.....at least its somewhere to look, instead of a dead end...... |
| 10:42.07 | Trentster | I had the same problem, using a softphone client as well tho... |
| 10:42.46 | Trentster | what should i be looking for in the snom firmware menu, would it be related to the sip registration or general options? |
| 10:43.19 | jql | it's a sip option |
| 10:43.45 | Trentster | ive got a " SIP T1 (ms): " under advanced settings ...currently its set to 500 |
| 10:44.00 | jql | yes, set that to 1000 |
| 10:44.26 | jql | it needs to be bigger than your ping time |
| 10:45.01 | Trentster | does "T" stand for Transmit? |
| 10:45.03 | *** join/#asterisk EvilGreen (n=andrewz@ppp85-140-139-63.pppoe.mtu-net.ru) |
| 10:45.20 | jql | no, that means Timer. there are several timers in the SIP spec, T1, T2, T3 |
| 10:45.29 | jql | T1 is the primary retransmit timer |
| 10:46.14 | Trentster | jql, I am busy rebooting the phone now, will retry the call....hope ...hope!!! |
| 10:49.40 | Trentster | jql, you know what.......!! |
| 10:49.45 | jql | what? |
| 10:49.56 | Trentster | you are a f....... Genius! |
| 10:50.00 | jql | awesome! |
| 10:50.29 | Trentster | yeah......thanks so much........I just retrieved some voice mail,,,,which i couldent do before.... |
| 10:50.37 | Trentster | now im gonna try a voice call. |
| 10:51.12 | jql | cool. that was an extraordinary problem. I always wondered why phones made that a configurable setting... :) |
| 10:51.39 | *** join/#asterisk angryuser (n=angryuse@df01t2-212-194-195-181.d4.club-internet.fr) |
| 10:52.03 | Trentster | yeah, call works 100% as well....... |
| 10:52.25 | angryuser | good day |
| 10:52.39 | Trentster | Never mind one beer, If you were here id buy you a 6 pack and a bottle Whisky! |
| 10:52.45 | jql | yay! |
| 10:53.23 | angryuser | i saw a zaptel 1.4.3 release, it is written that there alot of astribank fixes, where can i see them? |
| 10:54.03 | angryuser | i mean in details |
| 10:56.10 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net) |
| 10:56.14 | stoffell_h | angryuser, CHANGES file in the xpp dir maybe |
| 11:02.00 | angryuser | stoffell_h: yep |
| 11:03.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:05.42 | angryuser | what is the best way to interconnect asterisk servers?, let's say i have 5 servers all behind the nat, and a number of users for each , and i need people to call each other |
| 11:09.35 | Dovid | anyone here have expirience with php agi ? |
| 11:15.00 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 11:15.53 | tzafrir_laptop | angryuser, very recommended. e.g: PCM fixed. software-resettable firmware |
| 11:16.06 | tzafrir_laptop | Astribank BRI has been made usable |
| 11:16.46 | tzafrir_laptop | I'm now adapting bristuff to use zaptel 1.2.18 |
| 11:16.49 | angryuser | tzafrir_laptop: ok i will upgrade it today |
| 11:17.47 | tzafrir_laptop | just note that the old firmware is not yet software-resettable, and thus you'll still need to disconnect/connect the astribank to the power for the firmware reset |
| 11:35.55 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 11:54.53 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:59.19 | saftsack | does someone know why astfin.org isnt reachable? |
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| 12:12.58 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 12:16.54 | coppice | the blackfin was made into soup? :-\ |
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| 12:24.37 | *** part/#asterisk Gringo_ (n=root@213.219.147.192.adsl.dyn.edpnet.net) |
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| 12:29.02 | saftsack | coppice: you mean it isnt continued? |
| 12:29.34 | coppice | no. it must be just have a temporary problem. its actually quite active |
| 12:31.21 | coppice | it seems to be down from here too |
| 12:35.36 | saftsack | hmm. do you know a mirror where i can get astfin? i search for the best motherboard for building my own embedded pbx |
| 12:38.40 | tzafrir_laptop | anybody needs bristuff with zaptel 1.2.18? |
| 12:39.13 | coppice | if you look at blackfin.uclinux.org you will find related material, but I don't think there is an actual mirror |
| 12:41.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:41.13 | saftsack | i dont. but many people need bristuff with zaptel 1.4.X ^^ |
| 12:43.28 | tzafrir_laptop | there is a bristuff version for 1.4 |
| 12:43.43 | tzafrir_laptop | not for latest zaptel, but it should work |
| 12:44.29 | tzafrir_laptop | hmmm... that was fast |
| 12:44.51 | tzafrir_laptop | junghanns already have http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-g.tar.gz |
| 12:45.21 | tzafrir_laptop | the 1.4 version: http://www.junghanns.net/downloads/bristuff-0.4.0-test1.tar.gz |
| 12:47.09 | tzafrir_laptop | right. the -g version is still based on 1.2.17.1 |
| 12:47.11 | *** join/#asterisk rafonso (n=ronaldo@201-42-105-44.dsl.telesp.net.br) |
| 12:47.56 | rafonso | does anybody know or has any documentation aboud "DNS manager update" in Asterisk? |
| 12:48.06 | saftsack | i know from the test version but it is very unstable. but i have a patton gw here ;) |
| 12:48.45 | rafonso | ok, but what is it used for? |
| 12:50.26 | stoffell_h | damn, i just upgraded to 1y-f yesterday... lol |
| 13:02.38 | tzafrir_laptop | the only change is: fixed layer 1 problems of duoBRI cards introduced by 0.3.0-PRE-1y-f |
| 13:02.43 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 13:04.07 | stoffell_h | oh, okay, indeed, cool.. next upgrade can be postponed :p |
| 13:27.28 | Dovid | tzafrir: borker toker |
| 13:27.40 | Dovid | tzafrir: did u know of any sms gateways to Israel ? |
| 13:28.07 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 13:30.24 | Dovid | snook3r: borker toker |
| 13:30.32 | Dovid | snook3r: did u know of any sms gateways to Israel ? |
| 13:30.41 | tzafrir_laptop | Dovid, I don't know of any. But haven't researched |
| 13:32.08 | Dovid | thanks. |
| 13:32.27 | Dovid | i need a way of receiving alerts to my mobile phone here (in IL) from my servers |
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| 14:07.06 | SuPrSluG | ~pre |
| 14:07.17 | SuPrSluG | ~pre |
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| 14:32.31 | tzafrir_laptop | so now there's bristuff with zaptel 1.2.18 ... |
| 14:34.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:38.40 | SuPrSluG | is there any good documentation about configuring a pri line that is used for phone and data? |
| 14:39.53 | SuPrSluG | i've only dealt w/ analog lines to date. we'll possibly be upgrading to a pri shortly |
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| 14:52.24 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:01.14 | Bladerunner05 | Using tdm400p with 4fxo, both wirh asterisk 1.4.4 and asterisknow beta5, I can make a call, but can't answer any call, so I configure it correctly...... |
| 15:03.39 | [TK]D-Fender | pastebin your zapata.conf and extensions.conf |
| 15:03.40 | [TK]D-Fender | ~pb |
| 15:03.43 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:03.59 | *** join/#asterisk brussel_ (n=brussel@ppp-71-128-206-236.dsl.sndg02.pacbell.net) |
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| 15:11.36 | Bladerunner05 | <[TK]D-Fender> : those are generated by asterisknow beta5 http://www.pastebin.ca/556348 |
| 15:12.26 | Bladerunner05 | This is the same problen encountered using asterisk 1.4.4 with latest zaptel |
| 15:12.57 | tzafrir_laptop | 1.4.3 or 1.4.2.1? |
| 15:13.31 | Bladerunner05 | 1.4.2.1 |
| 15:13.51 | Bladerunner05 | but now I'm trying with *now beta5 updated today |
| 15:15.44 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:17.01 | [TK]D-Fender | blade what do you see when a call comes in? |
| 15:20.05 | tzafrir_laptop | Bladerunner05, what do you see in the CLI trace when a call comes in? |
| 15:20.21 | *** join/#asterisk mightnare (n=mike@p2036-ipad09motosinmat.mie.ocn.ne.jp) |
| 15:24.29 | *** join/#asterisk EvilGreen (n=andrewz@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 15:25.16 | Bladerunner05 | so now I'm using asterisknow beta5 |
| 15:25.28 | Bladerunner05 | in cli> when a call come in I don't see anything |
| 15:26.32 | Bladerunner05 | also enable sip set debug when a call come in now message was displayed |
| 15:27.45 | [TK]D-Fender | Bladerunner05, SIP debug is irrelevent to incoming zaptel calls. |
| 15:28.32 | Bladerunner05 | <[TK]D-Fender> : so I have to choose another board for the next time? |
| 15:30.02 | Bladerunner05 | How can I do too see why zaptel didn't answer the call? |
| 15:32.26 | Bladerunner05 | there is no digium man here ?= |
| 15:35.14 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:35.20 | Corydon76-home | It's early Sunday morning. They're all asleep or at church |
| 15:35.34 | Strom_M | or both |
| 15:35.36 | Bladerunner05 | eheheheh ;-) |
| 15:35.46 | Dovid | is there anyone here that is at VON in sweeden now |
| 15:35.48 | Dovid | ?* |
| 15:37.01 | Corydon76-home | Dovid: just you |
| 15:38.04 | Bladerunner05 | <@Corydon76-home> : can u help me to understand why zaptel didn't answer the call ? |
| 15:39.26 | Corydon76-home | Bladerunner05: given that you've provided next to no information, no. |
| 15:39.55 | Corydon76-home | Pastebin your zapata.conf and extensions.conf |
| 15:39.59 | Strom_M | Corydon76-home: HELP HELP I HAVE A PROBLEM AND IT DOESNT WORK FIX IT PLZ OMG HYPERVENTILATING |
| 15:40.47 | [TK]D-Fender | Corydon76-home, .... he already has.. |
| 15:40.49 | Corydon76-home | Strom_M: half an hour after waking up, you have a sense of humor? Lucky. |
| 15:41.00 | Strom_M | oh wait wait, i think i fixed it...you have to press "play" and "record" at the same time as it turns out |
| 15:43.15 | *** join/#asterisk EvilGreen (n=andrewz@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 15:44.22 | Corydon76-home | [TK]D-Fender: Sorry, an error has occurred. Reason: That is an invalid ID, or the post has expired. |
| 15:47.07 | [TK]D-Fender | Corydon76-home, So you're slow too ;) |
| 15:47.08 | *** join/#asterisk EvilGreen (n=andrewz@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 15:47.13 | Bladerunner05 | <@Corydon76-home> : those are my zapata.conf and extensions.con http://www.pastebin.ca/556408 |
| 15:47.27 | [TK]D-Fender | Corydon76-home, Still works for me... |
| 15:47.56 | [TK]D-Fender | Ok, off to see PotC 3 |
| 15:48.03 | [TK]D-Fender | later.... |
| 15:49.10 | *** part/#asterisk EvilGreen (n=andrewz@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 15:49.28 | *** join/#asterisk EvilGreen (n=andrewz@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 15:53.06 | tzafrir_laptop | Bladerunner05, can you pastebin a CLI trace from the time of an incoming call? make sure you set verbosity to at least 3 |
| 15:53.13 | tzafrir_laptop | core set verbose 3 |
| 15:56.05 | Bladerunner05 | <tzafrir_laptop> : I do that but when a call is coming no message was dispalyed |
| 15:56.55 | tzafrir_laptop | sanity check: connect a simple analog phone instead and call in. Does the phone ring? can you talk? |
| 15:57.14 | tzafrir_laptop | an FXO interface plays the role of an analog phone. |
| 15:59.08 | coppice | in daytime soap operas |
| 15:59.29 | Bladerunner05 | <tzafrir_laptop> a: ok go checking |
| 15:59.48 | Dovid | is there any variable in asterisk that will tell me if a call was bridged or not ? |
| 16:00.11 | mutilator | anyone wanna but a lightly used te110p? |
| 16:00.20 | ManxPower | Dovid: yes. you know where to find it too |
| 16:01.02 | Dovid | hehe. i am looking now. i just tried the lazy route ;) |
| 16:01.05 | *** join/#asterisk raptorrat (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 16:01.10 | iBuMp | morning |
| 16:01.59 | ManxPower | README.variables |
| 16:02.32 | Bladerunner05 | <tzafrir_laptop> : no ring tone.... so consider that I can call out |
| 16:04.26 | raptorrat | Would there be any cisco guru's in here today? I have a 2600 with a vwic that I want to use to send and recieve pots calls to a asterisk box. I have basic call setup and tear down working, however I'm losing audio going one way and I believe it is b/c the cisco device is trying to reinvite and I don't want that |
| 16:04.30 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 16:04.47 | raptorrat | I don't know how to do a noreinvite=no on a cisco |
| 16:05.10 | CrashSys | add it to that cisco's entry in sip.conf |
| 16:05.52 | raptorrat | CrashSys: I've added it to sip.conf and asterisk appears to be working 100% right |
| 16:06.03 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:06.03 | raptorrat | I don't think the cisco config is right and I have no idea how to configure cisco devices |
| 16:06.21 | raptorrat | I got asterisk sendin the calls in just a matter of minutes |
| 16:06.24 | CrashSys | I've never touched cisco stuff... cant help you much there |
| 16:06.31 | tzafrir_laptop | Bladerunner05, you're calling to the wrong numbr? |
| 16:06.49 | raptorrat | but teh 2600 doesn't register with asterisk so its never really going to be get the no reinvite |
| 16:07.07 | tzafrir_laptop | or the phone doesn't work? |
| 16:07.35 | CrashSys | Man... $450 for 500' 25-pair... doesn't even say if it's plenum :( |
| 16:07.39 | CrashSys | damn price of copper |
| 16:08.40 | coppice | i bet its not even oxygen free :-) |
| 16:08.52 | CrashSys | Trunk Cable... |
| 16:09.09 | CrashSys | Probably PVC Jacket... nothing special... |
| 16:10.56 | CrashSys | Half the places online are all quote-only on 25-pair... |
| 16:10.56 | CrashSys | unless it's cat-5 most of it is quote-only... |
| 16:10.56 | coppice | people just love paperwork :-) |
| 16:10.59 | Bladerunner05 | <tzafrir_laptop> : the number is correct and the phone works fine..... |
| 16:11.08 | *** join/#asterisk EvilGreen (n=Miranda@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 16:11.12 | SuPrSluG | anyone try the Open Source Line Echo Canceller? If so, does it work any better? |
| 16:12.23 | CrashSys | coppice: Know any online bulk cable sellers? The local supply house isn't open on sunday... |
| 16:12.36 | coppice | nope |
| 16:13.17 | coppice | SuPrSluG: any better than what? |
| 16:13.27 | SuPrSluG | zaptel |
| 16:13.43 | coppice | could it be much worse than those? |
| 16:13.52 | SuPrSluG | it's an add on to zaptel |
| 16:14.50 | Bladerunner05 | so the only think.... I'm attached the cable to the pstn line of an isdn controller |
| 16:15.14 | coppice | SuPrSluG: it is pretty good from an echo cancelling point of view. It hasn't really been optimised for speed, yet, which means it might take a bit more CPU |
| 16:16.45 | SuPrSluG | there are only a few users on the system. So it shouldn't matter. I'll give it a whirl. |
| 16:18.57 | coppice | this is a properly engineered EC, which gets through a wide range of tests |
| 16:19.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:19.51 | CrashSys | Anyone used sangoma's soft-echocan? |
| 16:20.11 | CrashSys | Interested to know what your opinion of it is... |
| 16:20.12 | coppice | you means octasic's? |
| 16:20.27 | CrashSys | Yeah, that sangoma gives away with their cards... |
| 16:20.35 | CrashSys | for use with up to 6-lines |
| 16:21.15 | Dovid | ManxPower: I am doing Noop(${DIALSTATUS}) and it is coming up blank :( |
| 16:24.32 | Dovid | nm. seems that it dosent work with the h extension..... |
| 16:24.49 | Dovid | i can understand y, but at the same time I want to see if the call was breidged |
| 16:25.00 | Dovid | bridged* |
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| 16:41.49 | *** join/#asterisk nephfl (n=no@wsip-70-184-144-158.ga.at.cox.net) |
| 16:42.14 | nephfl | hello, anyone here tried loading a tdm24xx card on trixbox? |
| 16:42.40 | CrashSys | I tried to get trixbox to work once with a TE205p... I found that installing slackware was the fix... |
| 16:42.47 | CrashSys | So I wont be of much help... sorry... |
| 16:42.55 | ReDNeQ | nephfl: im having the same problem |
| 16:43.18 | nephfl | yeah...ive tried everything...asterisknow will install it...but is too beta |
| 16:43.37 | ReDNeQ | i have the card working with a generic debian install |
| 16:43.44 | *** join/#asterisk sifusam (n=sifusam@cpe-24-167-119-83.satx.res.rr.com) |
| 16:43.45 | ReDNeQ | and i have had it working with a generic redhat fc7 install |
| 16:43.50 | ReDNeQ | but nogo with tb2.2 |
| 16:44.01 | Dovid | nephfl: try #trixbox |
| 16:44.02 | ReDNeQ | it loads a ztdummy and the card but wont configure |
| 16:44.05 | nephfl | i see...so a normal distro... |
| 16:44.21 | ReDNeQ | yeah i actually got fc7/freepbx working |
| 16:44.31 | bakermd | Man, all I gotta say is Asterisk is insaine. There's NOTHING you can't do with it |
| 16:44.35 | ReDNeQ | but switched to tb thinking that would be the better rout |
| 16:46.20 | nephfl | does freepbx have an extension editor that will let you reconfigure ivr? |
| 16:46.48 | Dovid | nephfl: i beleive so but it dosent learn how to work asterisk vanilla. u get more control |
| 16:47.03 | rob0 | nephfl: Maybe. But why not ask in the right channel? |
| 16:49.50 | tzafrir_laptop | ReDNeQ, it shouldn't load ztdummy, actually. Unnedded interrupts |
| 16:50.14 | tzafrir_laptop | nephfl, --^ |
| 16:50.55 | tzafrir_laptop | nephfl, "extension editor" as in vi? |
| 16:51.10 | tzafrir_laptop | or as in freepbx's one? |
| 16:51.25 | nephfl | as in freepbx one...like asterisknow |
| 16:55.40 | CrashSys | I like the pico extension editor |
| 16:55.42 | CrashSys | i'm lame :( |
| 16:56.03 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 16:56.20 | nephfl | yeah, i dont mind editing by hand but i wont be the one eventually administrating this system |
| 16:56.32 | *** part/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 16:57.58 | coppice | saftsack: astfin.org has risen from the grave |
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| 17:04.46 | tzafrir_laptop | nephfl, generally freepbx does not expose that |
| 17:05.22 | tzafrir_laptop | but unlike the asterisknow, in freepbx you can put your customizations in _custom files |
| 17:06.03 | tzafrir_laptop | asterisknow has to know about *everything* in your config |
| 17:09.04 | nephfl | do i need to remove kudzu before installing the tdm24xx card? |
| 17:11.45 | tzafrir_laptop | I don't think so |
| 17:11.58 | tzafrir_laptop | I think that this was once true. |
| 17:12.16 | ReDNeQ | nephfl |
| 17:12.24 | tzafrir_laptop | actually , kudzu is obsolete, and there should be no harm from removing it anyway |
| 17:12.29 | ReDNeQ | i think we need to grab the kernel-mdl versions of zaptel |
| 17:12.42 | tzafrir_laptop | udevd now does the real work |
| 17:13.15 | tzafrir_laptop | but what I wrote above is untested |
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| 17:13.43 | BUM92SK | hello |
| 17:14.12 | BUM92SK | who can help me im new at this i want to install asterisk on Fedora core 2 |
| 17:15.26 | tzafrir_laptop | BUM92SK, why FC2? |
| 17:15.33 | nephfl | anything special i should do with fc7 to install asterisk/freepbx? |
| 17:15.39 | tzafrir_laptop | that distro is no longer maintained |
| 17:16.02 | BUM92SK | becouse its on P3 |
| 17:16.02 | BUM92SK | only for test |
| 17:16.31 | tzafrir_laptop | so? you don't need an old distro for an old computer |
| 17:17.14 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:17.19 | BUM92SK | so if i make this i will install on bether machine |
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| 17:17.32 | hi365 | anyone familiar with crc4 errors? |
| 17:18.18 | EvilGreen | hi365: errors where? |
| 17:18.40 | hi365 | crc4 (carrier claims on my side) |
| 17:19.08 | EvilGreen | are they permanent or just from time2time ? |
| 17:19.37 | hi365 | permenant. no d-cah reset after a dissconect. the carrer claims its my card or something |
| 17:19.41 | BUM92SK | tzafrir_laptop |
| 17:19.48 | hi365 | d-can=d-chan |
| 17:20.04 | EvilGreen | check wiring & clocking |
| 17:20.20 | hi365 | tried 0 and 1 - no luck |
| 17:20.22 | EvilGreen | normally you should take clocking from your provider |
| 17:20.39 | hi365 | btw, its a snagoma a102d-x |
| 17:20.44 | hi365 | that 0, right? |
| 17:21.21 | EvilGreen | not sure, never configured it ;) |
| 17:21.25 | wunderkin | 1 |
| 17:21.42 | EvilGreen | 1 = external clock - right? |
| 17:22.10 | hi365 | i belive so. how do you get asterisk to reload the zaptel.conf - ztcfg? |
| 17:22.30 | wunderkin | you will need to rmmod, modprobe, and maybe ztcfg again after the change.. i forgot what all... i just do it all every time |
| 17:24.22 | bakermd | I'm trying to record a message... getting Jun 10 13:25:16 WARNING[3597] file.c: File RecordedMessage43241.ulaw does not exist in any format |
| 17:24.37 | wunderkin | bakermd, .ul |
| 17:24.46 | hi365 | the pri channel just wont reset! carier saiyes they see crc4 errors. i tried the system with another pri and all is fine... |
| 17:24.56 | bakermd | wunderkin:Trying... |
| 17:27.13 | *** join/#asterisk saftsack (n=oliver@p54A7F513.dip.t-dialin.net) |
| 17:27.45 | saftsack | tzafrir_laptop, hi, i tested the ec now and the voice quality is superb :) BUUUT ;) asterisk crashes after every call .... :( |
| 17:28.34 | saftsack | <PROTECTED> |
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| 17:41.42 | JerJer | http://digg.com/microsoft/Totally_Hilarious_VoIP_PBX_Promotion_by_Microsoft |
| 17:41.46 | bakermd | OK - so I am recording a file, and that seems to work fine - but when I do a playback I get " File CallBlast/RecordedMessage18066.ul does not exist in any format" |
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| 17:54.54 | wunderkin | bakermd, don't specify the file ext |
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| 18:51.20 | SuPrSluG | can't compile zaptel get error zaptel-base.c: In function `calc_fcs': |
| 18:51.34 | SuPrSluG | any ideas? 1.4.3 zaptel |
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| 18:59.39 | rob0 | Can you compile other things? What distro or OS is it? |
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| 19:13.49 | SuPrSluG | debian . yes i can compile 1.2X ok |
| 19:18.36 | Qwell | You're gonna have to pastebin more of the error |
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| 19:29.01 | tzanger | |
| 19:31.29 | nephfl | i recompiled my zaptel drivers, reran genzaptelconf..it configured properly, but now zaptel wont load at startup, anyone have any ideas? |
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| 19:35.33 | rob0 | nephfl: put a "modprobe $YOUR_ZAP_DRIVER" command somewhere in your OS's startup scripts. A specific answer isn't possible without details. |
| 19:36.31 | rob0 | (Um, modprobe(8) is Linux-specific, of course.) |
| 19:37.27 | nephfl | i am trying to load a tdm21421e... i dont get errors when i run modprobe wctdm24xxp but when i restart, i get failed at "waiting for zap to come online" |
| 19:39.53 | nephfl | should i just load modprobe with an init file? |
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| 19:45.34 | nephfl | dev/zaptel/ctl is missing |
| 19:49.13 | nephfl | it is up now...but startup isnt working |
| 19:50.56 | rob0 | Just get the modprobe done before * starts. |
| 19:51.20 | rob0 | If 2.6 kernel, udevd will create your devices. |
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| 19:56.54 | saftsack | tzafrir: hi. are you here? |
| 19:57.13 | nephfl | it just fails at waiting for zap to come online...is there a timeout i can reset or something |
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| 20:12.00 | zeeesh | hi |
| 20:12.30 | karlm | nephfl: what kinda card are you using with zap? or none? |
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| 20:16.26 | zeeesh | can i make call conference without using zap ? |
| 20:17.44 | zeeesh | i mean if i have about more 6 countries gateway .. can i make rooms for my users for call conferencing without using zap ? |
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| 20:37.10 | steliosk | zeeesh : use ztdummy |
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| 22:55.05 | Hmmhesays | holy shit it was a nice weekend |
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