| 00:01.05 | *** join/#asterisk marc\cba (n=marc@cpc2-whit2-0-0-cust200.cdif.cable.ntl.com) |
| 00:03.13 | johngalt | I have a bunch of grandstream phones that I'd like to get paging functional on. where can I find info on setting that up? |
| 00:04.29 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:05.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:08.42 | *** join/#asterisk terinjokes (n=spader@wikinews/Terinjokes) |
| 00:08.46 | terinjokes | hello! |
| 00:10.12 | terinjokes | i'm having problems turning off the default asterisk scripts, whenever I change it (even following tutorials), asterisk no longer logs into the outgoing/incoming SIP server) |
| 00:12.51 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:26.08 | [hC] | [TK]D-Fender: ping! |
| 00:30.46 | SnrWup | hi |
| 00:30.46 | SnrWup | anyone alive in here? |
| 00:30.56 | SnrWup | anyone? |
| 00:30.56 | SnrWup | im having another problem with buggy asterisk |
| 00:30.56 | SnrWup | mixmonitor stops recording once a call is transfered |
| 00:30.56 | SnrWup | anyone alive to help? |
| 00:30.56 | JT | buggy |
| 00:30.56 | JT | more like |
| 00:30.56 | JT | configured wrong |
| 00:30.56 | SnrWup | nah its buggy |
| 00:30.56 | SnrWup | really buggy |
| 00:30.57 | JT | uhuh |
| 00:30.57 | SnrWup | like the mixmonitor thing |
| 00:30.57 | SnrWup | thats a blatant bug |
| 00:30.57 | JT | sounds like you may have configured it wrong |
| 00:30.57 | JT | have you analysed the problem? |
| 00:30.57 | SnrWup | and seems to have been a bug for a while now according to the web |
| 00:30.57 | SnrWup | jt well this is what i can ascertain |
| 00:30.58 | JT | SnrWup: sip? |
| 00:30.58 | SnrWup | when sip phones in the same context call each other and mixmonitor the calls, it is all ok |
| 00:30.58 | SnrWup | SIP1 calls SIP2, SIP2 trasnfers to SIP3... SIP2 drops out as a zombie |
| 00:30.58 | SnrWup | all fine |
| 00:30.58 | JT | SnrWup: do you do reinvites? |
| 00:30.59 | SnrWup | leaving sip1 and sip3 talking |
| 00:31.26 | SnrWup | sip1 calls pstn user, sip1 transfers to SIP2, sip1 then hangs up, mixmonitor stops and sip1 hasnt dropped out as a zombie |
| 00:31.33 | SnrWup | hasnt masqueraded as youd expect |
| 00:32.10 | SnrWup | it seems that when you're calling to a different context of the dialplan, mixmonitor dies when u transfer |
| 00:32.17 | JT | what is canreinvite= in sip.conf set to? |
| 00:32.18 | SnrWup | JT - reinvites? |
| 00:33.29 | SnrWup | that is not set |
| 00:33.43 | SnrWup | but it defaults to yes right? so that shouldnt cause a problem? |
| 00:34.01 | JT | umm, if the default is yes, you're in trouble |
| 00:34.10 | JT | try setting canreinvite=no and reload sip |
| 00:34.30 | SnrWup | that would solve the mixmonitor problem? |
| 00:34.40 | Greek-Boy | I need someone to explain this to me. I am starting a Wimax mobile operator company and getting interconnections to the other mobile companies (GSM/CDMA/etc). I will have my own asterisk servers in their premises. What is the best way to interface to them? |
| 00:34.42 | JT | maybe, please try it :) |
| 00:34.56 | SnrWup | JT - ok will do as soon as i get to the office tomorrow morning |
| 00:35.04 | SnrWup | thanks for making some sort of effort anyway |
| 00:35.20 | JT | SnrWup: a lot of people overlook that when recording calls |
| 00:35.26 | JT | it's not definitely your problem |
| 00:35.26 | SnrWup | its so hard to get any informed help with asterisk, been googling for days |
| 00:35.40 | SnrWup | what does it actually do? |
| 00:35.43 | JT | but if sip phones talk directly to each other, it is impossible to record them |
| 00:35.51 | JT | as the ast server doesn't have access to media |
| 00:36.20 | JT | Greek-Boy: don't you have some tech guys? |
| 00:36.25 | SnrWup | well |
| 00:36.33 | Greek-Boy | lol JT |
| 00:36.34 | Greek-Boy | i am the tech guy |
| 00:36.43 | JT | Greek-Boy: telco tech guys |
| 00:36.44 | Greek-Boy | but i dont know much about gsm inner workings |
| 00:36.51 | JT | are you talking data or voice interconnect? |
| 00:36.56 | SnrWup | if PSTN calls SIP1 and SIP1 transfers to SIP2, SIP1 drops out as zombie and mixmonitor works fine recording everything until SIP2 hangs up |
| 00:37.17 | Greek-Boy | I know that I can use gsm/cdma modules but I think I will need something else since i'm already in their network |
| 00:37.22 | Greek-Boy | voice or voip |
| 00:37.33 | SnrWup | thing is... if SIP1 calls PSTN and SIP1 transfers to SIP2, SIP1 just hangs up and doesnt drop out at a zombie which seems really weird |
| 00:37.37 | JT | same thing, the question was data or voice :) |
| 00:37.51 | JT | what phones SnrWup ? |
| 00:39.47 | SnrWup | aastra 9133i |
| 00:40.10 | SnrWup | thing is, its not a phone thing. asterisk is clearly not masquerading the channel which drops out after a transfer |
| 00:40.12 | JT | hrm ok |
| 00:40.22 | JT | time to do some sip debugging |
| 00:40.43 | JT | i don't know what you are refering to with "masquerading" in this context |
| 00:40.50 | SnrWup | cause normally the channel hangs up and name is changed to Transfered/SIP-105/ZOMBIE |
| 00:40.54 | Greek-Boy | how do telcos interconnect? |
| 00:41.02 | SnrWup | masquerading.. becoming ZOMBIE |
| 00:41.20 | SnrWup | thats what should happen to a channel which drops out after a transfer |
| 00:41.25 | SnrWup | and what normally happens |
| 00:41.37 | JT | SnrWup: what version of ast? |
| 00:41.44 | SnrWup | 1.4.1 |
| 00:41.54 | JT | Greek-Boy: generally in the first world: T1s or E1s and SS7 |
| 00:42.20 | JT | Greek-Boy: SS7 isn't something you could possibly get into at your knowledge level i'm thinking |
| 00:42.42 | JT | there's also DS-3 and E3s and up |
| 00:42.42 | Greek-Boy | JT i'm a fast learner - does asterisk support SS7? |
| 00:42.46 | JT | no |
| 00:43.07 | pipwerk | yes |
| 00:43.09 | JT | there's a couple of implementations for ast that cost money |
| 00:43.20 | SnrWup | im wondering if theres any software which i can use for a pbx which is more stable and less buggy than asterisk |
| 00:43.29 | JT | problem is they usually need to be certified before they will get permission to connect |
| 00:43.29 | SnrWup | but which still has all the same functionality |
| 00:43.31 | SnrWup | if not better |
| 00:43.33 | Greek-Boy | do the mobile companies also use this method? |
| 00:43.37 | JT | SnrWup: try 1.2.x or the latest 1.4 |
| 00:43.45 | JT | Greek-Boy: yes |
| 00:43.54 | SnrWup | the 1.4.4 changelog doesnt fix things |
| 00:44.02 | SnrWup | and 1.2.x is going to be worse |
| 00:44.12 | JT | Greek-Boy: if you connect as a customer you could just use an ordinary PRI |
| 00:44.21 | JT | SnrWup: what makes you say that? |
| 00:44.35 | SnrWup | because mixmonitor was only just implemented in 1.2 |
| 00:44.46 | SnrWup | its hardly going to be state of the art in that version |
| 00:44.58 | JT | SnrWup: your problem solving skills seem to include making a lot of assumptions, just give them a go |
| 00:45.08 | Greek-Boy | I would connect as another telco to them. I would rather have an ast server in their premises and let that do the ss7 from there so its IP on my own infrastrucute to me via fibre. |
| 00:45.08 | JT | 1.2.x is more stable thatn 1.4.x generally |
| 00:45.13 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 00:45.17 | JT | and if you're going 1.4.x, at least go the latest |
| 00:45.25 | Greek-Boy | asterisk or anything else for that matter that can the calls to arrive on IP |
| 00:45.35 | SnrWup | ive looked at the changelog, no fix for the issue |
| 00:45.39 | JT | Greek-Boy: asterisk isn't carrier grade |
| 00:45.50 | irule | what are the dependencies for asterisk-addons? I got *, libpri and zaptel ready but addons does not compile |
| 00:45.58 | JT | SnrWup: just try it, plenty of other people can transfer sip calls just fine |
| 00:46.02 | SnrWup | in fact this was even listed as "not fixable" in the asterisk dev forum |
| 00:46.13 | SnrWup | its not about transfering sip calls |
| 00:46.41 | SnrWup | its about transfering PSTN callers from SIP to SIP, which works *fine* except MixMonitor dies |
| 00:46.43 | *** join/#asterisk spatulamaan (n=ggilmore@70-58-64-35.tukw.qwest.net) |
| 00:46.49 | SnrWup | and its been documented as a bug |
| 00:46.57 | SnrWup | and closed as "not fixable" |
| 00:47.06 | Greek-Boy | JT: but why is it applied to carrier-grade applications then? |
| 00:47.34 | JT | SnrWup: tried Monitor instead of MixMonitor |
| 00:47.46 | JT | Greek-Boy: because it's cheap |
| 00:47.49 | JT | and to be honest |
| 00:47.58 | SnrWup | yeah ill give that a go |
| 00:48.14 | JT | there's very FEW applications of asterisk in an ACTUAL carrier grade application |
| 00:48.36 | SnrWup | what do the big corps use for their PBX systems? not asterisk surely? |
| 00:48.43 | JT | a lot of applications of asterisk at the higher and are in conjunction with other stuff like OpenSER, and proprietary products, and implement clustering |
| 00:48.47 | JT | SnrWup: some do |
| 00:48.47 | harlequin516 | Okay `sip debug` turns debug on. How do I turn it off? |
| 00:49.01 | JT | SnrWup: there are 2000 extension corporations running on asterisk clusters |
| 00:49.04 | harlequin516 | I thought it used to toggle on and off |
| 00:49.17 | JT | a pbx is not a carrier grade switch in any case |
| 00:49.21 | JT | sip no debug |
| 00:49.58 | SnrWup | ok another one to pick your brain while you're on |
| 00:50.16 | SnrWup | im using a tdm400p to connect to 4 POTS lines from my * box. |
| 00:50.17 | Greek-Boy | ok but asterisk is carrier-grade when it comes to value added services, right? |
| 00:50.34 | JT | Greek-Boy: sorry, that just sounded like marketing speak to me |
| 00:50.37 | harlequin516 | Thanks |
| 00:50.41 | JT | carrier grade is carrier grade |
| 00:50.52 | SnrWup | in the SIP phones internal context, i have a _9 so they can dial 9 to reach another context which plays them a dialtone |
| 00:51.13 | SnrWup | they type in the number they want to dial and it then does Dial(zap/g1/thenumber) |
| 00:51.20 | JT | iirc the ITU standard for carrier switch availability is 99.999985% |
| 00:51.22 | Greek-Boy | lol |
| 00:51.23 | SnrWup | the problem is this... |
| 00:51.23 | Greek-Boy | k |
| 00:51.27 | JT | which is 3 hours downtime in 20 years |
| 00:51.29 | JT | 2 hours |
| 00:52.11 | SnrWup | as soon as the dial command executes, there is silence, then the zap channel connects and u hear a bit of outside noise |
| 00:52.22 | SnrWup | then the zap channel starts ringing and the caller might pick up |
| 00:52.29 | *** join/#asterisk spatulamaan (n=ggilmore@70-58-64-35.tukw.qwest.net) |
| 00:52.30 | SnrWup | how can i avoid that silence? |
| 00:52.44 | Greek-Boy | I was thinking of clustering asterisk to form one big switch |
| 00:52.50 | SnrWup | the problem seems to be that asterisk regards the POTS line as being "answered" right after the dial command executes |
| 00:53.02 | SnrWup | rather than after the pstn user actually picks up |
| 00:53.03 | JT | SnrWup: you sure that silence isn't from the telco? |
| 00:53.10 | JT | SnrWup: yes that's normal |
| 00:53.12 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:53.16 | JT | analogue signalling is shit |
| 00:53.25 | JT | if you want everything to work nicely, get a digital circuit |
| 00:53.28 | SnrWup | JT - well... for example... if i put an r in the dial command, asterisk plays ringing after the dial starts |
| 00:53.28 | JT | is the simple answer |
| 00:53.37 | SnrWup | then after the ringing from asterisk i get silence |
| 00:53.44 | JT | never use r, it's useless shit imho |
| 00:53.45 | SnrWup | then after the silence i get the real ringing of the pots line |
| 00:53.51 | JT | false ringing indication |
| 00:54.21 | SnrWup | what would be great is if asterisk knew when the pstn user picks up, then i could just mask the whole shit with false ringing until they pick up |
| 00:54.26 | SnrWup | and it would appear normal |
| 00:54.41 | JT | SnrWup: you need answer supervision or digital phone lines |
| 00:54.55 | JT | a computer cannot easily tell when an analogue call is answered |
| 00:55.20 | SnrWup | JT - im in the UK with british telecom... as far as i know answer supervision is standard with polarity reversal for answering and hanging up |
| 00:55.33 | SnrWup | only thing is... answeronpolarityswitch=yes makes 0 difference |
| 00:55.44 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 00:55.47 | SnrWup | and when i phone up telco asking about answer supervision nobody seems to have a damn clue |
| 00:56.02 | JT | i see, these sort of problems aren't fun to play with |
| 00:56.13 | SnrWup | yeah, lots of late nights |
| 00:56.18 | JT | heh, a multimeter or oscilloscope might tell you if the line isn't polarity switching |
| 00:56.28 | JT | but... you're in the UK |
| 00:56.31 | JT | get digital |
| 00:56.34 | JT | for 4 lines |
| 00:56.40 | JT | why screw with analogye |
| 00:56.44 | JT | analogue |
| 00:56.48 | SnrWup | waste of £150 on my fxo card |
| 00:56.57 | SnrWup | and extra cash to connect pri to my * box |
| 00:57.04 | SnrWup | and extra cash to get pri rental |
| 00:57.05 | JT | or 2 * BRI |
| 00:57.22 | Greek-Boy | JT: know a carrier-grade product for what I'm trying to achieve? |
| 00:57.30 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 00:57.42 | JT | Greek-Boy: there's plenty, they all cost over $40k |
| 00:58.05 | Greek-Boy | thats ok |
| 00:58.24 | JT | you should find out what the technical requirements are first |
| 00:58.25 | Greek-Boy | whats the best one in your opinion? |
| 00:58.31 | JT | and you should really hire a telco guy |
| 00:58.37 | JT | you are getting in telco territory |
| 00:59.10 | Greek-Boy | ko |
| 00:59.12 | Greek-Boy | ok |
| 00:59.14 | Greek-Boy | will do that |
| 00:59.33 | Greek-Boy | but i still wanna see some specs of these products |
| 00:59.54 | JT | i don't know exactly what you have to connect to |
| 01:00.07 | JT | but you're probably looking for a Class 5 switch with SS7 capability |
| 01:00.45 | Greek-Boy | ok |
| 01:00.47 | Greek-Boy | thanks |
| 01:01.14 | Greek-Boy | and I should stick to offering PBX to my corporate clients then |
| 01:01.37 | JT | it's all a lot easier if you need less than a couple hundred voice channels and the telco allows you to connect with a standard ISUP T1/E1 PRI |
| 01:01.58 | JT | ast can do that easily, as long as you don't need to guarantee 5 nines of availability |
| 01:02.02 | SnrWup | I'm in the UK. Just to complicate things, we get an instantaneous polarity reversal (i.e. normal, reverse, back to normal all in an instant) with each incoming ring (with CID sent on first ring at least, and possibly on all rings, during the reversals). Then we get another instantaneous polarity reversal on hangup, followed by a continuous tone. |
| 01:02.09 | SnrWup | thats what one person says |
| 01:02.22 | JT | SnrWup: that is complicated :P |
| 01:02.42 | Greek-Boy | JT, can't u get the 5 nines by clustering? |
| 01:02.59 | SnrWup | might have to start hacking zap.c or something u think? |
| 01:03.10 | JT | not in my opinion |
| 01:03.25 | JT | there are very few solutions for hot failover of PRIs, none in asterisk |
| 01:03.57 | JT | terminating PRIs on a parallel dual system running in lock step would increase your reliability a lot |
| 01:04.05 | JT | as long as the software had been well tested |
| 01:04.13 | JT | NEC make a lock step X86 server |
| 01:04.15 | JT | it's awesome |
| 01:04.24 | JT | starts at USD$15k iirc |
| 01:04.42 | JT | SnrWup: quite possibly... so BRI is not an option? |
| 01:05.38 | Greek-Boy | interesting |
| 01:05.44 | Greek-Boy | thanks a lot for all your help JT |
| 01:05.44 | JT | to clarify, hot failover of PRIs, i meant losing no calls, there are plenty of solutions that lose all calls and take the PRI back up pretty fast |
| 01:05.47 | irule | http://pastebin.ca/493976 anybody has a clue on what is going on here? the only thing lefft to compile is addons, everything else is done correctly |
| 01:05.47 | JT | even for asterisk |
| 01:05.55 | SnrWup | jt - i dont even know what BRI is |
| 01:06.06 | JT | SnrWup: might also be known as ISDN2 |
| 01:06.10 | JT | basic rate interface |
| 01:06.16 | JT | 2 B channels, 1 16kbit/s D channel |
| 01:06.29 | JT | smaller digital circuit than a PRI |
| 01:07.04 | SnrWup | ah |
| 01:07.07 | SnrWup | ISDN2e |
| 01:07.09 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 01:07.14 | JT | right |
| 01:07.31 | SnrWup | would get better voice quality i guess too |
| 01:07.46 | JT | and have the same Q.931 signalling advantages of a PRI :) |
| 01:07.48 | Daejeo1 | hello guys. can i use NOKIA N80 with asterisk server? |
| 01:07.55 | Daejeo1 | oh JT |
| 01:08.14 | Daejeo1 | JT?:can i use NOKIA N80 with asterisk server? |
| 01:08.19 | JT | voice quality improvement would depend on how bad your current analogue lines sound |
| 01:08.28 | JT | Daejeo1: no idea, i think you can if it talks sip |
| 01:08.50 | SnrWup | thing is, im setting up the * PBX to replace our current analogue hardware pbx. so we'd need to somehow, convert to ISDN while stll allowing the analogue system to work |
| 01:08.51 | [hC] | anyone know if polycom's support DSCP? |
| 01:08.57 | SnrWup | before then changing over to asterisk |
| 01:09.09 | JT | what is dscp? |
| 01:09.40 | JT | SnrWup: so it's not replacing the current box atm? |
| 01:10.16 | SnrWup | nope, the company is running on its current phone system while i develop the new phone system. the idea was to simply flip over to asterisk one night with |
| 01:10.21 | SnrWup | without any real change |
| 01:10.34 | SnrWup | apart from new handsets and lots of new functionality |
| 01:10.36 | [hC] | JT: QoS bit |
| 01:10.37 | JT | SnrWup: well you could connect the analogue system to asterisk |
| 01:10.57 | JT | [hC]: not sure, i know it does some QoS stuff |
| 01:11.33 | SnrWup | would need FXS modules to do that |
| 01:12.00 | JT | true |
| 01:13.31 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:13.57 | SnrWup | btw hows this for price... |
| 01:14.07 | SnrWup | 500USD to install 2 ISDN2e channels |
| 01:14.29 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 01:14.30 | JT | i have no idea if that's a good price in the uk |
| 01:14.35 | JT | how much per month? |
| 01:14.50 | Sweeper | ouch >.< |
| 01:14.57 | Sweeper | that's what a t1 costs here :3 |
| 01:15.06 | JT | Sweeper: to install? |
| 01:16.00 | SnrWup | 180USD per quarter per ISDN2e channel |
| 01:16.31 | ThOr101 | Can asterisk via a Digium TDM22B distinguish a ring when the line is setup with distinctive ring? i.e. Can it only answer the phone when a certain ring is rung? Using the FXS of course? |
| 01:17.27 | Qwell | ThOr101: should be easy enough... look in the sample configs for the distinctive ring stuff |
| 01:17.44 | Sweeper | pero quarter.... |
| 01:17.53 | Sweeper | so $120 a month for both channels |
| 01:17.55 | ThOr101 | Excellent thanks. That was better than an absolutly not, are you crazy. |
| 01:18.04 | Sweeper | JT: per month |
| 01:18.37 | Sweeper | SnrWup: it sounds like you're getting your ass kicked for 2 freaking lines, but I dunno, ukland |
| 01:18.43 | *** join/#asterisk ari_stress (n=fajar@222.124.78.205) |
| 01:18.53 | SnrWup | yeah uk is insaney expensive |
| 01:19.06 | ThOr101 | I just bought a TDM22B for $110. Anybody want the contact? |
| 01:19.22 | ThOr101 | The guy has a bunch so he can configure it anyway you want as well. |
| 01:19.24 | Sweeper | I mean, wouldn't it be cheaper to just get 2 fxs'? |
| 01:19.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:21.17 | JT | SnrWup: USD$30/mo/line basically |
| 01:21.17 | JT | not awesome, but possibly acceptable |
| 01:21.17 | ari_stress | hi, do you know where can i debug my asterisk? i look into /var/log/asterisk/messages but there's not very detail info |
| 01:21.17 | JT | how much does analogue cost, SnrWup ? |
| 01:21.58 | JT | SnrWup: it's 4 lines |
| 01:22.02 | JT | Sweeper: it's 4 lines |
| 01:22.10 | JT | Sweeper: to replace 4 fxs |
| 01:22.24 | JT | get it right :) |
| 01:22.33 | SnrWup | similar for analogue actually |
| 01:22.55 | JT | i assume USD$180/quarter is for each BRI service (2 chans) ? |
| 01:22.58 | Qwell | ThOr101: sounds fake |
| 01:23.07 | SnrWup | yeah |
| 01:23.30 | JT | not bad |
| 01:23.34 | Qwell | ThOr101: I'd be interested in seeing an image of it |
| 01:23.36 | JT | similar to bri in australia |
| 01:23.41 | JT | from the main telco |
| 01:23.52 | SnrWup | say if i got 3 of those, 6 lines at 64k |
| 01:23.54 | JT | which is similar to analogue price :) |
| 01:24.05 | SnrWup | would be enough for us i reckon |
| 01:24.10 | JT | sorry, but i'm going to be back in half an hour or so |
| 01:24.12 | JT | talk then |
| 01:24.15 | JT | probably :) |
| 01:24.22 | SnrWup | just then to figure out the best way to connect the BRI to asterisk |
| 01:24.28 | SnrWup | some sort of card i assume |
| 01:25.41 | JT | cheap single bri HFC-s clones |
| 01:25.41 | JT | or multiport junghanns, beronet, or digium bri card |
| 01:25.41 | SnrWup | pci card is that? |
| 01:25.41 | JT | cost a lot more |
| 01:25.41 | SnrWup | ah |
| 01:25.41 | JT | yes |
| 01:25.41 | JT | gone |
| 01:25.41 | SnrWup | thanks |
| 01:25.41 | SnrWup | cu =] |
| 01:27.57 | *** join/#asterisk killown (n=killown@unaffiliated/killown) |
| 01:28.54 | killown | what system and hardware requetiment for run , execute asterisk? |
| 01:30.36 | SnrWup | all BRI has answer/disconnect supervision etc right? |
| 01:31.13 | *** join/#asterisk Tavitux (n=G@nat.netlp.com.ar) |
| 01:34.09 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-075db339e024a618) |
| 01:34.35 | *** join/#asterisk hachi (i=hachi@shego.kuiki.net) |
| 01:36.03 | killown | Domain/Realm: address IP of server and SIP Proxy: address IP of server what that want say? |
| 01:36.06 | ThOr101 | Qwell: I found the guy on e-bay, and he said he had a bunch, so I bought one on the side. http://cgi.ebay.com/Digium-Wildcard-TDM400P-with-all-4-modules-FXO-FXS-NEW_W0QQitemZ280116135681QQihZ018QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 01:36.31 | hachi | I'm trying to issue a 'show channels' command via the manager interface on asterisk, it gives me permission problems unless I set 'write = all'... I'm a bit puzzled by this setup of things, when I want to query something shouldn't that be more of a 'read = all' situation? |
| 01:36.40 | rikstah | Hi there, in 1.4 what has SetAccount changed to please? |
| 01:36.46 | hachi | what action am I doing anyways, where is the documentation on this stuff |
| 01:37.13 | SnrWup | which ISDN cards do you guys use with asterisk? |
| 01:37.41 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-ad8706f7998eb0c0) |
| 01:39.56 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-193.plaicmtc01.tx.dh.suddenlink.net) |
| 01:40.45 | ThOr101 | Qwell: I was leary at first, but he sold one previous, then sold the one I was looking at, and now has another up there. His e-mail address resolves back to a Voip company. He said they bought the cards for an asterisk solution, and then ended up going hosted. I paid via paypal so I could rescind the CC payment if everything went into the toilet. |
| 01:41.14 | ThOr101 | And then like the "Everyone is free to wear sunscreen" do one thing everyday that scares you. Well, this would be it. |
| 01:42.15 | hachi | is there a bot here that I can send my queries to? |
| 01:42.29 | hachi | otherwise, where is the asterisk manager interface docs? |
| 01:43.36 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:45.33 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 01:46.01 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:46.29 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 01:46.43 | BSD_tech | ok time to move to asterisk 1.4.5 |
| 01:46.50 | BSD_tech | they needs to release it |
| 01:51.09 | luckyone | ok, if I have a DID provider I wish to use to make calls for a specific pattern and receive calls for specific numbers defined as two contexts [provider-out] and [provider-in], can I create a [provider] context which links those two together? |
| 01:52.09 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 01:52.30 | luckyone | something like [provider] include => provider-out include => provider-in |
| 01:53.26 | luckyone | then have [provider] be the default context for a phone I wish to use those? |
| 01:53.45 | *** part/#asterisk hachi (i=hachi@shego.kuiki.net) |
| 01:53.55 | *** join/#asterisk Downchuck (n=downchuc@c-24-22-20-80.hsd1.mn.comcast.net) |
| 01:55.52 | Putzz | sure |
| 01:56.02 | luckyone | Putzz: was that to me? |
| 01:56.15 | Putzz | yes sir |
| 01:56.21 | luckyone | Putzz: I am trying to get my sip softphone to make my first real asterisk call! |
| 01:56.33 | Putzz | wow |
| 01:56.33 | Putzz | ok |
| 01:57.01 | Putzz | why not just make [provider-outgoing] for all outgoing calls |
| 01:57.09 | Putzz | and [provider-incoming] |
| 01:57.11 | Putzz | for all incoming numbers? |
| 01:57.13 | luckyone | ok, I have both of those |
| 01:57.26 | Putzz | ok.... |
| 01:57.29 | Putzz | so make the call |
| 01:57.31 | luckyone | they are just called [inbound] and [outbound] |
| 01:57.48 | Downchuck | i'm using openser and asterisk, and trying to figure out how to proxy the rtp data correctly. Any great suggestions? I'm behind a NAT at home, and it's killing the audio |
| 01:58.01 | luckyone | Putzz: I don't think I have the SIP device configured correctly |
| 01:58.14 | Downchuck | opened up 5060, 8k-20k on the openser/asterisk server |
| 01:58.55 | Putzz | luckyone: what "sip device" are you using? |
| 01:59.03 | luckyone | Twinkle on this machine |
| 01:59.29 | luckyone | Putzz: would you mind if I PM'd? |
| 02:00.01 | Putzz | luckyone: we are really all here to learn/share. why not share with everyone else? ;-) |
| 02:00.07 | luckyone | I don't want to spam channel if you are willing to look a sip.conf and ext.conf |
| 02:00.08 | luckyone | k |
| 02:00.10 | Downchuck | i know i'm here to share |
| 02:00.12 | Putzz | ~pb |
| 02:00.27 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:00.28 | luckyone | I am more than happy to! |
| 02:03.58 | *** join/#asterisk shodan- (n=shodan@ip141.99-113-216.pppoe4.joliette.intermonde.net) |
| 02:07.09 | luckyone | Putzz: check out http://pastebin.ca/494045 to view a pretty close representation to my ext.conf and sip.conf |
| 02:09.05 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 02:10.04 | Putzz | ok |
| 02:10.08 | Putzz | so whats the prob? |
| 02:10.23 | luckyone | my sip phone can't make a call |
| 02:10.34 | luckyone | not sure if it can receive a call.. |
| 02:10.39 | Downchuck | my sip phone makes calls but it sucks |
| 02:10.40 | Downchuck | how about that. |
| 02:10.54 | Putzz | luckyone: do you have to register to make outgoing calls? |
| 02:11.13 | luckyone | Thu 20:54:17 |
| 02:11.13 | luckyone | home: you have the following registrations |
| 02:11.13 | luckyone | <sip:homephone@192.168.1.107>;expires=3604 |
| 02:11.36 | luckyone | and .107 is my * server |
| 02:12.02 | luckyone | which can receive calls, verified with simple playback dialplan |
| 02:12.18 | Putzz | no username for homephone? |
| 02:12.27 | luckyone | homephone is the username |
| 02:12.28 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 02:12.43 | Putzz | context= default |
| 02:12.45 | Putzz | ! |
| 02:12.53 | luckyone | yeah, should that be outbound or inbound? |
| 02:12.56 | luckyone | that was my question |
| 02:13.15 | Putzz | since u want to make [provider] and include in and out |
| 02:13.23 | Putzz | put it to provider context |
| 02:13.31 | luckyone | is that what I *should* do? |
| 02:13.58 | luckyone | that is the problem, * will let me do whatever I want... I want to make sure I want the *right* things though! |
| 02:14.22 | Putzz | it is the right thing |
| 02:14.33 | Putzz | or u can point it to [provider-out] context |
| 02:14.35 | Putzz | if u want |
| 02:14.49 | luckyone | the one defined in sip.conf? |
| 02:15.15 | Putzz | well |
| 02:15.15 | Putzz | outbound |
| 02:15.15 | Putzz | context |
| 02:15.16 | Putzz | sry |
| 02:16.47 | luckyone | hmm, i set it to outbound |
| 02:16.51 | luckyone | still, no dice |
| 02:17.13 | *** join/#asterisk DaveCanoe (n=Dave@24.114.252.226) |
| 02:17.40 | Putzz | Oh boy |
| 02:17.50 | Putzz | pastebin me all your CLI output |
| 02:17.54 | Putzz | and verbose |
| 02:17.55 | Putzz | then |
| 02:18.46 | luckyone | wasn't matching the pattern, ringing now! |
| 02:18.57 | Putzz | ...ok |
| 02:18.58 | luckyone | stupid me, left the 1 off of the front |
| 02:19.06 | Putzz | heh |
| 02:21.39 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 02:23.54 | BSD_tech | writing dialplan sucks |
| 02:25.43 | Downchuck | I haven't made progress in near four days :-/ |
| 02:25.53 | Downchuck | Four, that's almost two days more than three |
| 02:26.09 | Downchuck | and three times two = 6.. and i think you all know where i'm going there |
| 02:29.00 | killown | what I need to have asterisk work for external telephone? |
| 02:29.25 | Downchuck | sip and/or card |
| 02:30.04 | killown | Downchuck, I need one provider voip? |
| 02:30.20 | Downchuck | if you don't have a direct connection to the phone system |
| 02:30.25 | Downchuck | then yes, you need a voip terminator |
| 02:31.02 | killown | ok thanks |
| 02:31.38 | *** join/#asterisk neuralwind (n=root@201.222.65.128) |
| 02:31.51 | Putzz | writing dialplan rulez |
| 02:31.57 | Putzz | it controls everything |
| 02:31.59 | Putzz | ;-) |
| 02:33.53 | neuralwind | i make install zaptel and now what is next? |
| 02:34.07 | neuralwind | i wanna see how it works the x100p |
| 02:34.35 | Putzz | x100p sucks |
| 02:34.59 | Putzz | but ok for beginners testing |
| 02:35.10 | neuralwind | so how do u test it |
| 02:35.23 | Putzz | did u install zaptel? |
| 02:35.26 | neuralwind | yes |
| 02:35.45 | Putzz | did u modprobe? |
| 02:35.58 | neuralwind | no |
| 02:36.00 | Putzz | modprobe zaptel |
| 02:36.00 | Putzz | modprobe wcfxo |
| 02:36.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:36.14 | Putzz | then configure your zaptel.conf |
| 02:37.10 | neuralwind | i booted with the wrong kernel i have to restart |
| 02:43.26 | *** join/#asterisk neuralwind (n=root@201.222.65.128) |
| 02:48.11 | *** part/#asterisk Tavitux (n=G@nat.netlp.com.ar) |
| 02:49.09 | Downchuck | Putzz: how's your asterisk over NAT? |
| 02:49.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:50.26 | Putzz | I dont use it behind nat |
| 02:50.41 | Downchuck | am i the only person who lives on wifi? |
| 02:50.49 | Downchuck | my asterisk isn't behind a nat, but i am |
| 02:51.06 | Putzz | I've used it with wifi |
| 02:51.10 | Putzz | and behind nat |
| 02:51.14 | Downchuck | did you have to dance for it? |
| 02:51.22 | Downchuck | i'm starting to think my asterisk is drunk |
| 02:51.27 | Downchuck | it's looking like more of a tilde |
| 02:51.29 | Downchuck | ~ |
| 02:52.04 | Downchuck | alright.. that's the best i can do.. I'll be here all week. |
| 02:52.20 | *** join/#asterisk Strom_M (n=strom@12.175.45.206) |
| 02:52.51 | *** join/#asterisk neuralwind (n=root@201.222.65.128) |
| 03:11.17 | JT | SnrWup: wow, that took a lot longer than expected |
| 03:11.26 | neuralwind | hey if u download a new kernel do u have to reinstall asterisk ? |
| 03:11.44 | JT | SnrWup: yes, answer/disconnect "supervision" is something that it just part of the signalling process over the D channel |
| 03:11.57 | JT | Q.931 messages are exchanged for all stages of a call |
| 03:12.04 | neuralwind | i mean if u install a new kernel do u have to reinstall asterisk? |
| 03:14.57 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:20.04 | blitzrage | neuralwind: you may have to recompile and install, yes |
| 03:20.17 | blitzrage | just go into the source directory and type, 'make install' |
| 03:21.47 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:28.41 | luckyone | really bad echo on my sound card, I am not sure what is causing that... |
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| 04:21.51 | killown | somebody can say me what necessary to have asterisk functioning to effect linkings for telephones public? |
| 04:22.31 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
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| 04:22.42 | JT | kiscokid: don't quite understand the question |
| 04:23.24 | kiscokid | I don't think I asked a question yet |
| 04:23.30 | asteriskguy | JT |
| 04:23.40 | asteriskguy | how's it going? |
| 04:23.43 | JT | killown: i mean |
| 04:23.49 | JT | kiscokid: sorry for confusion |
| 04:23.55 | JT | asteriskguy: not bad, you? |
| 04:23.59 | kiscokid | JT: np |
| 04:24.11 | asteriskguy | Pretty good |
| 04:24.20 | asteriskguy | our company is going crazy with projects |
| 04:24.26 | JT | sweet |
| 04:24.49 | asteriskguy | yeah, but I've got two weeks to figure out DUNDi clustering |
| 04:24.58 | JT | :/ |
| 04:25.03 | asteriskguy | either that or somekind of clustering with SER/OpenSER |
| 04:25.12 | asteriskguy | I don't know |
| 04:25.29 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 04:25.50 | asteriskguy | do you have any experience with setting up asterisk with sip trunk? |
| 04:26.08 | JT | what do you mean by sip trunk? |
| 04:26.16 | asteriskguy | sip provider |
| 04:26.22 | JT | yes |
| 04:26.31 | asteriskguy | Ok, question then |
| 04:26.45 | hansin321 | I have compiled latest Asterisk (along with Zaptel driver for my X101P FXO card). I usually have been ok with this before, but for some reason this time I can't seem to get * to listen for SIP (port 5060). I have a sip.conf set up. Any pointer or ideas of what I should check out? I understand this is limited information. Thanks. |
| 04:26.53 | neuralwind | i install asterisk and zaptel and also configure my zaptel.con file now how do i test it? |
| 04:26.56 | asteriskguy | you know in the zapata.conf you have to declare a context for incoming calls? |
| 04:27.10 | asteriskguy | context=whatever |
| 04:27.30 | asteriskguy | I'm not sure how I would do that with sip |
| 04:27.31 | JT | hansin321: check nothing else is listening on the port |
| 04:27.43 | JT | asteriskguy: same thing |
| 04:27.45 | JT | context= |
| 04:27.50 | neuralwind | i install asterisk and zaptel and also configure my zaptel.con file now how do i test my x100p card now? |
| 04:28.04 | asteriskguy | but you have to declare two parts right? user and peer? |
| 04:28.11 | asteriskguy | can you use friend? |
| 04:28.12 | JT | asteriskguy: or friend |
| 04:28.33 | asteriskguy | sounds simple |
| 04:28.34 | hansin321 | JT: There should be nothing. I am running on a bare-install server, and 'netstat -an' shows nothing. |
| 04:28.55 | JT | hansin321: errors on bringing up asterisk? |
| 04:29.20 | hansin321 | neuralwind: Try running the command 'ztcfg -vvv" to see at least if your card is being recognized. |
| 04:30.00 | neuralwind | hansin Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 04:30.42 | hansin321 | I don't have a modules.conf file, so is states no modules being loaded. Could I need certain modules? I know my kernel modules for my FXO card are loaded (separate modules I assume). |
| 04:31.35 | JT | hansin321: 1.4? |
| 04:31.48 | hansin321 | neuralwind: At least you know your card is being recognized. I can't help you much beyond that for now. I can grab you a link to on old article that deals with the configuration files that uses an FXO card. |
| 04:32.01 | neuralwind | ok |
| 04:32.05 | neuralwind | give it to me |
| 04:32.21 | JT | neuralwind: manners :) |
| 04:32.53 | neuralwind | ok thanks give it to me hansin |
| 04:32.55 | hansin321 | JT: yes. I have compiled * a couple times before and have always been able to get it so I can register a SIP softphone. But I am pretty new at this so I can't remember what I did. Ususally it just worked. |
| 04:33.14 | JT | hansin321: 1.4 doesn't load asterisk modules by default |
| 04:33.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:33.19 | hansin321 | neuralwind: I will. I need to google it, and then I will cut and paste it here. give me a sec... |
| 04:33.27 | JT | hansin321: either load chan_sip or enable autoload |
| 04:33.27 | neuralwind | k |
| 04:33.37 | hansin321 | how do |
| 04:33.44 | hansin321 | I enable auto load? |
| 04:34.11 | JT | modules.conf |
| 04:34.14 | JT | autoload => yes iirc |
| 04:34.30 | hansin321 | I guess I was under the assumption if you had a .conf file for something, it would load the modules needed for it :( |
| 04:35.35 | hansin321 | neuralwind: This link is quite dated, but I found it helpful. The configuration files (see link near end) are well commented: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 04:35.44 | JT | there are lots of default config files |
| 04:36.07 | hansin321 | JT: so from the command line I might just try load chan_sip fpr now? |
| 04:36.28 | JT | worth a go |
| 04:36.43 | hansin321 | JT: Thanks for the help. |
| 04:38.33 | hansin321 | JT: Awsome. It is now listen for SIP. I'll do a little homework on load modules. This appears to be the gist of the problem. It may affect other things also. |
| 04:39.11 | JT | no probs |
| 04:40.17 | hansin321 | neuralwind: what do you get back is you run a 'dmesg | grep -i wildcard'? |
| 04:48.21 | neuralwind | let me see |
| 04:49.12 | neuralwind | Found a Wildcard FXO: Wildcard X100P |
| 04:49.48 | neuralwind | hansin321 its a generic one with just one fxo port only |
| 04:50.34 | hansin321 | neuralwind: Ok, just wanted to make sure your kernel recognized your card (I could have assumed I guess if Asterisk did, but I am not an expert). |
| 04:50.48 | hansin321 | neuralwind: Anyway, at least the card is being recognized. |
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| 05:07.05 | JoelSolanki | hi all |
| 05:07.19 | JoelSolanki | is chanspy stable ? |
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| 05:17.45 | neuralwind | hansin321 u still there |
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| 07:33.20 | nitrus^ | does anyone know if a gentoo ebuild is available for 1.4? im trying to figure out the best configure options for building and i'd rather use an ebuild if there is one |
| 07:33.46 | JT | what sort of configure options? |
| 07:34.37 | nitrus^ | prefix, paths, and other options like optional packages |
| 07:34.51 | JT | not sure, i don't use gentoo |
| 07:35.11 | nitrus^ | do you have your ./configure stuff available? |
| 07:35.33 | DarKnesS_WolF | nitrus^: the default ./configure for asterisk is fine. |
| 07:35.42 | DarKnesS_WolF | it uses prefix=/usr |
| 07:35.48 | DarKnesS_WolF | so everything gose like it's a package |
| 07:36.08 | DarKnesS_WolF | i do always compile asterisk 1.4 ubuntu, debian, mandriva and all are almost the same setup |
| 07:36.42 | DarKnesS_WolF | only there is no init scripts for zaptel to debian based distro ;-) |
| 07:37.10 | nitrus^ | yeah |
| 07:37.18 | nitrus^ | i want to enable jabber gtalk |
| 07:37.23 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 07:38.38 | nitrus^ | do i just do --with-iksemel? |
| 07:38.39 | JT | DarKnesS_WolF: there isn't? |
| 07:38.39 | mightnare | it autodetects if iksemel libraries are installed |
| 07:38.39 | DarKnesS_WolF | JT: no just init script for asterisk |
| 07:38.48 | DarKnesS_WolF | JT: i'm talking about /asterisk/source/contrib/init/ |
| 07:38.57 | DarKnesS_WolF | u'll not find rc.debian.zaptel |
| 07:39.09 | DarKnesS_WolF | nitrus^: u can to compile iksemel 1st |
| 07:39.09 | JT | DarKnesS_WolF: look at the deb packaged ast, i'm sure it has a zaptel init script |
| 07:39.25 | DarKnesS_WolF | nitrus^: if u did then it should be enabled by default .. |
| 07:39.37 | DarKnesS_WolF | JT: i don't need too ;-) just put zaptel in ur /etc/modules and ur all done ;-) |
| 07:39.48 | JT | err |
| 07:39.52 | JT | what about ztcfg? |
| 07:39.58 | DarKnesS_WolF | nitrus^: after u compile and do ./configure for asterisk make menu and make sure that u have chan_gtalk |
| 07:40.08 | DarKnesS_WolF | JT: what i'll do with ztcfg ? |
| 07:40.15 | JT | err |
| 07:40.19 | JT | do you use zaptel? |
| 07:40.23 | JT | you MUST run ztcfg |
| 07:40.29 | DarKnesS_WolF | JT: i use TDM400P so waht i do put in /etc/modules are |
| 07:40.29 | DarKnesS_WolF | zaptel |
| 07:40.31 | DarKnesS_WolF | wctdm |
| 07:40.33 | JT | after loading the kernel driver for your card |
| 07:40.37 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 07:40.47 | DarKnesS_WolF | JT: really ?? i never do ! and it works perfect ! |
| 07:40.54 | JT | zaptel is unnecessary afaik, wctdm loads it |
| 07:40.58 | raidenz | Hi guys |
| 07:41.13 | DarKnesS_WolF | JT: yes ur right .. but i never needed to run ztcfg after i boot the machine |
| 07:41.29 | JT | maybe it's running and you don't know it |
| 07:41.40 | DarKnesS_WolF | JT: may be let us check on my machine at home :-) |
| 07:41.50 | JT | and s/ur/you're/ already :P |
| 07:41.51 | DarKnesS_WolF | i have asterisk and zaptel compiled and the TDM400P is up |
| 07:41.53 | raidenz | Does anyone know how in the dialplan to figure out how many active sip channels are going at that time? |
| 07:43.35 | DarKnesS_WolF | hmm JT i don'th ave it running but everything seems to be fine. |
| 07:43.52 | DarKnesS_WolF | JT: why do u need to run ztcfg the kernel loads the module ? |
| 07:44.04 | JT | it configures zaptel |
| 07:44.13 | JT | it doesn't stay running |
| 07:44.33 | JT | because how else does zaptel.conf get read? |
| 07:44.37 | DarKnesS_WolF | JT: why do i need to configure zaptel everything :-s? i have /etc/zaptel.conf and thats it |
| 07:44.42 | DarKnesS_WolF | hmmm |
| 07:45.20 | JT | what do you think reads it? |
| 07:45.23 | JT | ztcfg does |
| 07:45.30 | JT | read some zaptel documentation :) |
| 07:45.41 | DarKnesS_WolF | JT: will do :-) |
| 07:46.02 | DarKnesS_WolF | JT: may be when the driver loaded it do it .. but i never had anyproblems with any of my asterisk installations. |
| 07:46.18 | DarKnesS_WolF | i only run ztcfg -vvvvvvvvvvvvv to check if i did the /etc/zaptel.conf right or not :P |
| 07:59.15 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:00.38 | creativx | eh.. this might seem stupid; but how can I find out what kind of timer asterisk uses? im trunking iax2 |
| 08:00.49 | JT | zaptel |
| 08:01.50 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 08:03.21 | creativx | and where would I look for zaptel's existance |
| 08:04.03 | creativx | because I suddenly got the feeling I'm in lack of a timer |
| 08:04.26 | JT | if you don't have zap hardware, you need ztdummy |
| 08:05.09 | creativx | yeah |
| 08:05.13 | creativx | that I just read |
| 08:05.22 | creativx | and I sure havent got any ztdummy yet |
| 08:05.30 | creativx | im amazed nobody has complained, heh |
| 08:05.42 | JT | ? |
| 08:05.56 | creativx | FATAL: Module ztdummy not found. |
| 08:06.15 | creativx | so explain to me.. without ztdummy, how could I expect iax2 trunking to behave? |
| 08:06.46 | JT | it could break up and have other problems |
| 08:07.19 | creativx | based on number of concurrent calls, or just at total random |
| 08:07.43 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) |
| 08:07.48 | creativx | because we've been testing asterisk lightly the last 3-4 months with 10 users... and its worked like a charm |
| 08:10.05 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:12.03 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-52db56dbe807fd26) |
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| 08:46.11 | sebadel | hello, I'm the happy owner of a digium S800i; I bought it as a Development Kit but I can't ind any documentation on "how to configure it" (without the buggy web interface) |
| 08:46.16 | sebadel | can anyone help ? |
| 08:54.57 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 09:02.37 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:02.40 | Uatec | GREETINGS! |
| 09:03.22 | sebadel | anyone experience with Digium S800 appliance ? |
| 09:03.57 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
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| 09:30.07 | *** join/#asterisk tinrsh (n=claudiu@81.181.94.112) |
| 09:30.28 | tinrsh | 'mornin |
| 09:30.29 | *** join/#asterisk xuser (i=1000@unaffiliated/xuser) |
| 09:37.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:38.49 | yannj_fr | are you all in europe? |
| 09:39.31 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:40.30 | creativx | europe <3 |
| 09:40.33 | tinrsh | not all |
| 09:40.44 | tinrsh | I think :) |
| 09:42.12 | kippi617 | I have a main number that comes in and then goes to a group, is there away to get a cdr output when they ring in on the main number before it goes to the group? |
| 09:43.12 | penguinFunk | UK |
| 09:46.05 | yannj_fr | FR |
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| 09:48.26 | *** join/#asterisk shinao1 (n=shinao1@41.222.71.203) |
| 09:50.12 | yannj_fr | Is there a variable for technology, I mean to replace SIP/${EXTEN} by a generic param |
| 09:55.05 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
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| 10:14.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 10:25.26 | puzzled | hi |
| 10:28.06 | mvanbaak | hey |
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| 10:59.15 | skirmisha | guys |
| 10:59.23 | skirmisha | how can i make query to db of asterisk |
| 10:59.42 | mvanbaak | ${DB(key/value)} |
| 11:00.11 | skirmisha | if i want to retrieve info only |
| 11:00.28 | skirmisha | dbget? |
| 11:06.02 | skirmisha | ? |
| 11:08.40 | *** join/#asterisk raphxl (n=raph@202.22.228.42) |
| 11:08.40 | raphxl | hi |
| 11:08.40 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 11:11.24 | mvanbaak | skirmisha: use the DB dialplan function |
| 11:12.50 | skirmisha | let me think |
| 11:13.08 | raphxl | Hi all, I'm new to asterisk. what web frontend may I use for having a good webadmin and not breaking my already working asterisk config? |
| 11:15.27 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 11:18.30 | tinrsh | knows anyone how to create the voicemail recording files with 664 or 666 permissions ? |
| 11:22.53 | BrokenNoze | hi can anyone tell me why my features ain't working? I have put include +> featuremap into extensions.conf |
| 11:27.11 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 11:28.18 | Daejeo1 | i configured my box with voipjet, but i am unable to make a call |
| 11:28.34 | Daejeo1 | all circuits are busy |
| 11:28.38 | Daejeo1 | http://www.pastebin.ca/494690 |
| 11:28.43 | Daejeo1 | iax debug |
| 11:29.04 | Daejeo1 | whats wrong? |
| 11:32.06 | creativx | all your circuits are belong to us |
| 11:37.37 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 11:37.46 | puzzled | creativx: nice, I have that t-shirt |
| 11:38.03 | puzzled | ok, s/channels/base |
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| 11:56.04 | creativx | puzzled: hehe |
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| 11:59.51 | *** join/#asterisk Obergandhi83 (n=Obergand@P7986.p.pppool.de) |
| 12:01.08 | Obergandhi83 | hi @ all .. i've an error when i compile mISDN: /usr/src/mISDN/drivers/isdn/hardware/mISDN/l1oip_core.c:1571:55: error: macro "INIT_WORK" passed 3 arguments, but takes just 2 |
| 12:01.25 | Obergandhi83 | aby idea |
| 12:01.33 | Obergandhi83 | any |
| 12:01.45 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 12:02.15 | Daejeo1 | creativx: u belong to us |
| 12:02.45 | Daejeo1 | it means that all circuit belong to us |
| 12:02.46 | Daejeo1 | ok? |
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| 12:04.06 | cy303 | yo |
| 12:07.56 | BrokenNoze | does anyone know anything about the senddtmf manager api function? |
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| 12:49.01 | raphxl | where can I found the voicemail url I have to check for using with twinkle? i'm identified as user foo (1234) on server bar? |
| 12:49.44 | raphxl | it's on a stock asterisk server, just created some users and voicemail |
| 12:49.54 | matsk | Anyone that could help me with debugging the latest svn version of "configure" |
| 12:50.20 | [TK]D-Fender | raphxl: its your dialplan... what exten did YOU setupt to access VoicemailMain? |
| 12:51.38 | raphxl | from an analog phone, I use 500 and then my user number (1234) and a password |
| 12:51.46 | raphxl | (1234) |
| 12:53.20 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 12:53.33 | *** join/#asterisk djmarin (n=marin@64.9.33.130) |
| 12:53.39 | raphxl | [voicemail] |
| 12:53.40 | raphxl | ;Accueil boite vocales Asterisk |
| 12:53.40 | raphxl | exten => 500,1,Set(LANGUAGE()=fr) |
| 12:53.40 | raphxl | exten => 500,n,VoiceMailMain( ) |
| 12:54.08 | [TK]D-Fender | raphxl: Well then you dial 500. |
| 12:54.48 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:58.21 | djmarin | i have 2 Asterisk systems one in NJ and one in MD, both setup from what i was told the sane way. Email notification works on MD but not in NJ, if printed out both conf files and they both are the same. mail is running on both servers. any idea whats going on? |
| 12:59.29 | mvanbaak | run 'mailq' on the box in NJ |
| 12:59.40 | mvanbaak | look in the mailer logfiles |
| 12:59.44 | raphxl | [TK]D-Fender, ok but when I dial 500, it ask me for my voicemail number and then my password. Is there a way to get it directly? |
| 13:00.40 | djmarin | i got some domains not found |
| 13:00.45 | djmarin | with mailq |
| 13:01.20 | mvanbaak | raphxl: if your voicemailbox is the same as the CIDNUM you are dialing from you can do: VoicemailMain(${CALLERID(num)}) |
| 13:01.23 | [TK]D-Fender | raphxl: "show application voicemailmain" |
| 13:01.30 | creativx | JT: now I know what can happen without ztdummy |
| 13:06.45 | *** join/#asterisk Fieldy (i=Bp24R7OT@gentoo/contributor/Fieldy) |
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| 13:08.53 | *** part/#asterisk mrchicken (n=root@200.71.58.39) |
| 13:09.01 | gauriji | can any body help me to generate DTMF tone???? |
| 13:09.26 | mocker | Does anyone know if http://bugs.digium.com/view.php?id=3219 is part of standard meetme? |
| 13:10.30 | mvanbaak | mocker: "show application Meetme" |
| 13:10.58 | mocker | mvanbaak: I see the option 'i', but it doesn't say if that includes recording people's names when entering a bridge. |
| 13:12.04 | gauriji | can any one help me???? |
| 13:12.38 | intralanman | gauriji: push a key on your phone.... that should generate one ;) |
| 13:13.01 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 13:14.19 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 13:14.28 | stoffell | hm, I just configured a bunch of Polycom 430's with auto answer and distinctive ring tones, is there a way to disable the microphone for the phones that are paged? |
| 13:14.32 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 13:14.35 | Winkie | hey guys, having a little issue with a UK PRI, can dial in ok but trying to make an outgoing call immediately ends it |
| 13:14.45 | Winkie | also having a lot of issues with 'avoided initial deadlock' |
| 13:14.50 | Winkie | oh and a polycom 330 which won't register :) |
| 13:15.04 | stoffell | to make it seem more like a paging system like in a store ... |
| 13:15.10 | [TK]D-Fender | stoffell: IIRC app_page is supposed to drop them into the paging meetme muted... |
| 13:15.35 | gauriji | can u tel me plz how to generate DTMF tone in java applet??? |
| 13:15.40 | stoffell | [TK]D-Fender, okay; i see.. awesome.. thanks; gonna try it out :) |
| 13:15.45 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 13:15.58 | AndrewGearhart | morning folks. |
| 13:16.20 | [TK]D-Fender | gauriji: This has nothing to do with *. Go record some DTMF tones and have fun. |
| 13:16.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 13:17.12 | mocker | Or google 'dtmf wav' |
| 13:17.17 | *** join/#asterisk ctooley (n=ctooley@tx-71-2-123-118.dhcp.embarqhsd.net) |
| 13:17.20 | Winkie | so anyone got any suggestions on what i should do to debug this PRI problem? I have had no joy with the provider so far |
| 13:17.23 | mocker | First link has all of them. |
| 13:17.51 | gauriji | <[TK]D-Fender>:see this is not fun it not genearate signals in backend |
| 13:17.55 | mocker | Winkie: What does your PRI debug say? |
| 13:17.57 | gauriji | so can u help me?? |
| 13:18.11 | [TK]D-Fender | gauriji: I have no idea what you're talking about. |
| 13:18.28 | Winkie | mocker: what type of debugging would you like? |
| 13:19.05 | [TK]D-Fender | gauriji: This is not a java applet programming channel. this is #asterisk . |
| 13:19.23 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 13:19.27 | Assid | heya |
| 13:19.45 | Assid | anyone here using asterisk src on a default debian kernel? |
| 13:19.48 | [TK]D-Fender | gauriji: And your app has no direct relationship with *. |
| 13:19.56 | gauriji | <[TK]D-Fender>: ya my friend thats i knw but if any one knw then tell me abt dtmf tone |
| 13:20.10 | *** join/#asterisk chandresh (n=root@122.169.16.59) |
| 13:20.17 | chandresh | hello |
| 13:20.17 | [TK]D-Fender | gauriji: What ABOUT DTMF? |
| 13:20.25 | [TK]D-Fender | ROOOOOOOOT! |
| 13:20.31 | chandresh | does anybody know anything about slow playback in asterisk? |
| 13:20.53 | JT | creativx: bad audio? |
| 13:21.06 | JT | Winkie: pri intense debug -> pastebin.ca |
| 13:21.08 | chandresh | nope.. i just want to play a file at slow speed |
| 13:21.27 | chandresh | and i m using dictation application of asterisk |
| 13:21.28 | mvanbaak | Assid: yeah I am |
| 13:21.39 | Assid | no issues right ? |
| 13:21.47 | mvanbaak | nope |
| 13:21.53 | Assid | thinking whether or not to recompile my own kernel |
| 13:21.54 | [TK]D-Fender | chandresh: Use a system call to run it through SOX |
| 13:22.13 | gauriji | <mocker>: ya its play wave file but it not generate signals in backend |
| 13:22.44 | chandresh | ohk SOx does have facility o slow playback right? |
| 13:22.57 | chandresh | ohk SOx does have facility o slow playback right? |
| 13:22.58 | mvanbaak | Assid: I'm not recompiling anymore unless something needs a custom kernel |
| 13:24.48 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 13:25.24 | Winkie | JT: i'll get on it |
| 13:25.54 | *** join/#asterisk zm23 (n=chatzill@zombie.cc.columbia.edu) |
| 13:26.19 | Winkie | calling provider atm |
| 13:26.26 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:26.29 | Katty | morning lovables. |
| 13:27.26 | n0n0x | Hi! got a question for you. A third party company is developing a softphone for my company. I want to add g729 support. How this is done? |
| 13:27.43 | JT | pay patent royalties |
| 13:28.24 | Corydon76-home | Or wait until 2014 for the patent to expire |
| 13:28.51 | chandresh | <[TK]D-Fender>: do we have any other solution...? |
| 13:29.00 | creativx | JT: any good hints before I try to cram ztdummy into a fc4 @ dell poweredge850 box? |
| 13:29.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:29.32 | Katty | creativx: make a note to take breaks. |
| 13:29.45 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:29.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:29.48 | Katty | anthm: (= |
| 13:30.12 | creativx | Katty: i have no time for breaks. I want "It Just Works(tm)" |
| 13:30.39 | Katty | So much for wanting tips. |
| 13:30.52 | Katty | creativx: knock yourself out. |
| 13:31.41 | creativx | hehe:) |
| 13:31.55 | gauriji | <PROTECTED> |
| 13:32.06 | creativx | naa really.. I have time to take breaks, i just dont wanna dive into it without having a slight feeling it will work |
| 13:32.27 | creativx | and most of the zaptel install instructions seems slightly outdated |
| 13:32.27 | Winkie | hold music universally sucks |
| 13:32.33 | Winkie | this is old pop |
| 13:32.47 | creativx | my musiconhold is a beep |
| 13:32.50 | Winkie | IE music your parents think is modern :( |
| 13:32.57 | Winkie | mine is metal :o |
| 13:33.04 | Winkie | well at home |
| 13:33.29 | chandresh | gauriji: you should ask about DTMF in asterisk and how its generated |
| 13:33.37 | creativx | i dont have a home phone |
| 13:34.18 | Winkie | lucky you |
| 13:34.22 | punani | gauriji: why do you want to generate a dtmf tone? |
| 13:34.26 | Winkie | i have 2 and a mobile :o |
| 13:34.57 | chandresh | hello does anybody know how to create slow plaback in asterisk without using SOX? |
| 13:35.36 | cy303 | can anyone offer advice for creating a macro that will dial a few numbers and play a greeting on answer then drop them into a conference? :P |
| 13:35.58 | gauriji | <punani>: ya i generate DTMF tone bcoz i want to create java applet phone with DTMF tone |
| 13:36.17 | Winkie | cy303: use call files or manager originate i guess? |
| 13:36.26 | mvanbaak | I used to have Cradle of Filth as MOH but my boss forced me to remove it |
| 13:36.36 | cy303 | hmm |
| 13:36.54 | chandresh | hello does anybody know how to create slow plaback in asterisk without using SOX? |
| 13:37.16 | Corydon76-home | chandresh: it is impossible |
| 13:37.48 | chandresh | Corydon76-home: using SOX.. it is feasible......? |
| 13:37.59 | Corydon76-home | There's your answer. |
| 13:38.23 | Corydon76-home | Quit asking the same thing every 10 minutes. The answer is not going to change. |
| 13:38.26 | Katty | creativx: i'm puting 1.4 on a poweredge 2600 |
| 13:38.30 | Katty | creativx: it's gone fairly well. |
| 13:38.51 | *** join/#asterisk coppice (n=chatzill@96.199.17.210.dyn.pacific.net.hk) |
| 13:38.52 | chandresh | hello buddy.. i asked without using SOX? |
| 13:39.02 | Katty | coppice: (= |
| 13:39.14 | Qwell | gauriji, chandresh: Please don't message people without asking them first. |
| 13:39.17 | Katty | creativx: started awhile back, made my first phone call yesterday. |
| 13:39.24 | chandresh | Corydon76-home: And i m new to this channel.. give me time to know ur rules......:) |
| 13:39.25 | Katty | creativx: of course i have 44 pages of documentation to go along with it... |
| 13:39.40 | *** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com) |
| 13:40.06 | Katty | today i wanna get paging and voicemail/email thingy working |
| 13:40.17 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 13:40.22 | Assid | heya |
| 13:40.30 | rgsteele | Hey folks. I'm upgrading from 1.2.9 to 1.2.13, and I had a look at the changelog which doesn't seem to indicate any major configuration changes, but is it a safe assumption that I won't need to muck with the configs to upgrade from/to those minor versions? |
| 13:40.37 | Assid | hrmm.. tyring to think.. whether i hsould update the kernel from source |
| 13:40.44 | chandresh | Qwell: I ll take care buddy sorry..... |
| 13:40.47 | Katty | rgsteele: nah. |
| 13:40.55 | Katty | rgsteele: not a whole lot of difference there. |
| 13:41.05 | Katty | rgsteele: if anything, maybe some console commands. |
| 13:41.05 | rgsteele | Katty: Okay, I thought as much, but thanks for the vote of confidence! :) |
| 13:41.20 | Katty | rgsteele: 1.4 is a whole new beast tho ;) |
| 13:41.33 | Katty | and we're still going at it *grumble* |
| 13:42.05 | SuPrSluG | Katty:auto answer or overhead paging |
| 13:42.13 | Katty | SuPrSluG: autoanswer. |
| 13:42.20 | Katty | SuPrSluG: i configured my phones for it ahwile back. |
| 13:42.26 | Katty | SuPrSluG: but..uhh. |
| 13:42.37 | rgsteele | Katty: Yeah, I haven't jumped in to the deep end wrt 1.4 yet, heh. |
| 13:42.41 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:42.47 | Katty | rgsteele: it's been a fun ride (= |
| 13:42.52 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:42.55 | Katty | Zeeek: ! |
| 13:43.00 | SuPrSluG | Katty:I'm going to tackle that also |
| 13:43.07 | Katty | SuPrSluG: sweet. |
| 13:43.07 | rgsteele | Katty: I suppose that depends highly on your definition of fun, no? ;) |
| 13:43.10 | Zeeek | Katty |
| 13:43.19 | Katty | rgsteele: something like that. |
| 13:43.27 | Katty | Zeeek: :< |
| 13:43.32 | Mercestes | :) |
| 13:43.34 | Zeeek | to reach for the pringles |
| 13:43.39 | Katty | oh, k'then |
| 13:43.46 | Zeeek | don't stand in front of pringles! |
| 13:44.28 | Katty | my bad. |
| 13:44.28 | Zeeek | heh |
| 13:44.28 | Zeeek | hows the soy milk? |
| 13:44.28 | creativx | Katty: 44 pages that you wrote? |
| 13:44.28 | Katty | creativx: yes'r |
| 13:44.28 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 13:44.29 | creativx | Katty: im running on a PE850.. but I somehow magically forgot all about ztdummy timing |
| 13:44.29 | creativx | so now ive compiled the ztdummy and lsmod looks good |
| 13:44.31 | Katty | creativx: tho some of it is snippets from conf files. |
| 13:44.40 | creativx | and according to the interweb net i need a reboot to make udev changes effective |
| 13:44.43 | Zeeek | anyone interested in speech recognition? |
| 13:44.44 | Katty | creativx: and you can modprobe ztdummy? |
| 13:44.47 | creativx | so the question is, will hell break loose |
| 13:45.04 | Katty | if you're sitting on the hellmouth it will. |
| 13:45.05 | creativx | what output should modprobe give |
| 13:45.10 | Katty | none, usually |
| 13:45.14 | creativx | none given |
| 13:45.16 | Katty | it just shakes the system around a bit |
| 13:45.22 | Katty | and says GET THOSE DRIVERS RUNNING |
| 13:45.33 | creativx | well ok |
| 13:45.35 | Katty | you can probably do a ztcfg -vvvv to see some stuff |
| 13:45.45 | Katty | not sure if that works with ztdummy or not tho |
| 13:45.45 | creativx | 0 channels configured |
| 13:45.47 | creativx | it says |
| 13:45.50 | Katty | well yeah |
| 13:45.51 | creativx | which sounds correct |
| 13:45.56 | creativx | since i only want ztdummy |
| 13:45.57 | Katty | you don't have any /real/ channels |
| 13:46.03 | creativx | but i'm guana try to reboot the sucker then |
| 13:46.08 | Katty | this is just tea time with the teds. |
| 13:46.08 | Zeeek | where did the idea of adding more v's come from? It is really easier to parse than a single digit? |
| 13:46.09 | creativx | and pray to the IAX gods. |
| 13:46.12 | mvanbaak | creativx: try to see if you have a /proc/zaptel/ |
| 13:46.16 | mvanbaak | if you have, no need to reboot |
| 13:46.27 | creativx | i have |
| 13:46.30 | creativx | proc/zaptel/1 |
| 13:46.35 | mvanbaak | yeah |
| 13:46.44 | mvanbaak | if you cat it, it will say something about dummy |
| 13:47.00 | creativx | cat /proc/zaptel/1 |
| 13:47.00 | creativx | Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 13:47.01 | creativx | sweet moses! |
| 13:47.03 | Katty | mew. |
| 13:47.08 | creativx | great |
| 13:47.09 | mvanbaak | there you go |
| 13:47.09 | Katty | they need to makea mew command. |
| 13:47.14 | creativx | then restart asterisk then |
| 13:47.22 | mvanbaak | yeah |
| 13:47.23 | creativx | and she should not complain about trunking |
| 13:47.27 | Katty | and make sure to modprobe it before bringin asterisk back up |
| 13:47.29 | mvanbaak | but if you have the time you should reboot |
| 13:47.41 | mvanbaak | just to make sure ztdummy loads when the system boots |
| 13:47.49 | Katty | personally, i'd automate the modprobing and such. |
| 13:47.52 | Katty | but i'm lazy. |
| 13:48.36 | creativx | hmm |
| 13:48.41 | Katty | why do my co-workers have to bug me about how to reboot their phone? |
| 13:48.43 | creativx | i stopped asterisk, modprobed, started |
| 13:48.52 | Katty | i'd be like, oooh powerpack. |
| 13:48.53 | creativx | there shouldnt be any other configs to change? |
| 13:48.53 | Katty | *yank* |
| 13:49.05 | mvanbaak | creativx: nope |
| 13:49.09 | creativx | cause i got the same build_peer errors when doing a reload |
| 13:49.14 | creativx | missing zaptel timing asterisk says |
| 13:49.17 | Katty | zapata and zaptel.conf have to do with....card thingies |
| 13:49.18 | *** part/#asterisk shapr (i=shae@cs6625112-51.bham.res.rr.com) |
| 13:49.30 | creativx | yes ive never understood the need for these hardware thingies |
| 13:49.33 | mvanbaak | creativx: is chan_zap.so loaded ? |
| 13:49.50 | [TK]D-Fender | Katty: Mew. |
| 13:49.54 | Katty | creativx: well if you're just using sip and iax, there isn't |
| 13:50.05 | Katty | creativx: t1s, phone lines, etc. that gets a smidgen more tricky. |
| 13:50.17 | Katty | [TK]D-Fender: Mew. |
| 13:50.20 | creativx | mvanbaak: i forgot how to check for loaded modules :O |
| 13:50.31 | creativx | rofl nevermind |
| 13:50.33 | creativx | no mvanbaak its not loaded |
| 13:50.37 | creativx | *editing modules.conf* |
| 13:50.41 | mvanbaak | :) |
| 13:50.59 | Katty | [TK]D-Fender: i'm going to try to get autoanswer and voicemail -> email working today (= |
| 13:51.10 | creativx | does zap need to be loaded after other dependant modules? |
| 13:51.13 | creativx | or can i just put it.. at the end |
| 13:51.23 | Katty | [TK]D-Fender: is it possible to already use an email server we have here, or does it need to be on the * box? |
| 13:51.33 | creativx | ahgah.. wtf |
| 13:51.34 | mvanbaak | creativx: put it as one of the first |
| 13:51.45 | creativx | /usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory |
| 13:51.50 | creativx | i might have forgotten something it seems |
| 13:51.51 | creativx | :> |
| 13:51.55 | mvanbaak | because it needs to be loaded before the modules that need zaptel timing |
| 13:52.00 | mvanbaak | hhmm |
| 13:52.10 | mvanbaak | you compiled asterisk and afterwards zaptel ? |
| 13:52.14 | creativx | yeah |
| 13:52.20 | creativx | that messed it up didnt it |
| 13:52.22 | mvanbaak | you need to recompile asterisk then |
| 13:52.26 | creativx | oh crapit |
| 13:52.29 | [TK]D-Fender | Katty: You need an MTA on your * box IIRC, but it can be as dumb as you want and you can set it to forward through your other box if you want |
| 13:52.45 | mvanbaak | nullmailer ftw ! |
| 13:52.46 | creativx | Katty: we are sip to iax2 here actually.. so no need for hardware |
| 13:52.48 | Katty | i think that would be best. |
| 13:53.00 | creativx | mvanbaak: so how scared should that make me, recompiling asteriskque |
| 13:53.12 | Katty | i don't really /get/ any email programs besides exchange, cause i haven't worked with them. |
| 13:53.16 | mvanbaak | creativx: not at all |
| 13:53.20 | mvanbaak | creativx: just meet the reqs |
| 13:53.26 | mvanbaak | have a cup of coffee |
| 13:53.32 | mocker | Ugh, anyone have any tips for remote MWI? |
| 13:53.33 | creativx | stop now; make; wait |
| 13:53.38 | creativx | anything else I am forgetting? |
| 13:53.57 | djmarin | when i do a mailq it tells me the host is wrong. where can i edit this? |
| 13:54.00 | mvanbaak | and type after me: asterisk -rx 'stop now' && make && make install |
| 13:54.10 | creativx | let me get that cup of coffee first |
| 13:54.11 | creativx | =) |
| 13:54.12 | creativx | brb |
| 13:54.24 | mvanbaak | mocker: mount the voicemail dirs using nfs |
| 13:54.27 | mvanbaak | works great |
| 13:54.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:54.41 | mocker | mvanbaak: Ahhh, you cheater. :) |
| 13:54.47 | mvanbaak | I store voicemail on a netapp |
| 13:55.18 | Mercestes | Katty: I suggest nullmailer |
| 13:55.20 | creativx | 1.2.10 |
| 13:55.44 | gauriji | can anybody tel me plz how to generate DTMF tone in java applet??? |
| 13:55.50 | Katty | Mercestes: i, uh... |
| 13:55.54 | Katty | Mercestes: i wish i knew what that was. |
| 13:56.07 | Mercestes | Katty: You can read all about it on their webpage. google.com |
| 13:56.09 | Katty | Mercestes: and i wish i knew where in voicemail.conf i should be looking to do a script thingy |
| 13:56.13 | mvanbaak | Katty: nullmailer is a very very dumb MTA |
| 13:56.21 | Mercestes | Katty: You just put in an email address. :) |
| 13:56.25 | Zeeek | hot cheese break, anyone? |
| 13:56.25 | creativx | mvanbaak: if this goes to hell I will have a very long friday afternoon @ the office |
| 13:56.26 | creativx | =) |
| 13:56.35 | Mercestes | zeeek: ME! me! |
| 13:56.36 | Zeeek | s/break/bread |
| 13:56.43 | mvanbaak | creativx: gheh. sounds like my fridays ;) |
| 13:56.44 | mocker | creativx: Rule #1, never make big changes on Friday.:) |
| 13:56.50 | Mercestes | oh. |
| 13:56.51 | creativx | mocker: i like to tempt fate |
| 13:56.53 | Mercestes | you meant cheese bread |
| 13:56.54 | Zeeek | never even go to work on friday |
| 13:56.54 | creativx | heheh |
| 13:56.55 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.114) |
| 13:56.57 | creativx | i know |
| 13:57.05 | creativx | mondays and fridays are the worst |
| 13:57.06 | Zeeek | or never leave work |
| 13:57.07 | mvanbaak | Zeeek: nor mondays |
| 13:57.10 | creativx | no matter what I do, things break |
| 13:57.10 | Zeeek | right |
| 13:57.18 | Zeeek | Friday is the day for.... |
| 13:57.23 | mvanbaak | beer ! |
| 13:57.29 | Zeeek | ~AUC |
| 13:57.32 | jbot | methinks auc is Asterisk Users Conference, Friday at 12:30 PM Eastern Time |
| 13:57.32 | creativx | WAH |
| 13:57.33 | Katty | all of this stuff is commented out. |
| 13:57.37 | Katty | the body, subject, etc. |
| 13:57.40 | mocker | My boss didn't like my phone joke, "The nice thing about being a phone admin is if the system breaks you can't call me!" |
| 13:57.45 | creativx | WAHHH |
| 13:57.46 | creativx | chan_zap.so Zapata Telephony 0 |
| 13:57.46 | creativx | :D |
| 13:57.47 | Zeeek | heh |
| 13:57.50 | creativx | WEEKEND IT IS. |
| 13:58.00 | Katty | Mercestes: is all this commented stuff out normal? |
| 13:58.01 | mvanbaak | creativx: yay \o/ |
| 13:58.01 | creativx | haha mocker |
| 13:58.05 | creativx | than you mvanbaak :) |
| 13:58.07 | creativx | +k |
| 13:58.14 | mvanbaak | ur welcome |
| 13:58.19 | mvanbaak | now lets go and grab a beer |
| 13:58.20 | creativx | well |
| 13:58.20 | creativx | heh |
| 13:58.21 | creativx | almost |
| 13:58.26 | Katty | Mercestes: ah HA! "you can ovverride the default program to send e-mail if you wish |
| 13:58.28 | creativx | i forgot modules.conf |
| 13:59.16 | Katty | Mercestes: first, what's "MTA" stand for? |
| 13:59.19 | creativx | still getting Unable to support trunking on user '' without zaptel timing.. hum |
| 13:59.26 | creativx | Katty; mail transfer agent |
| 13:59.26 | mvanbaak | Mail Transfer Agent |
| 13:59.29 | [TK]D-Fender | Katty: Mail transfer Agent |
| 13:59.31 | Zeeek | While I get fatter, read this: http://x2z.eu |
| 13:59.53 | creativx | Mail Tit Agent. |
| 14:00.06 | mvanbaak | huh? tits? where? |
| 14:00.08 | creativx | mvanbaak: so i see chan_zap being loaded before chan_iax2, still it gives zaptel timing error |
| 14:00.26 | creativx | i might have jumped the "weekend" conclusion a bit fast there |
| 14:00.28 | Katty | [TK]D-Fender: it claims to be using sendmail. |
| 14:00.36 | Katty | [TK]D-Fender: i think. |
| 14:00.42 | mvanbaak | creativx: what does 'zap show channels' tell you ? |
| 14:00.43 | Katty | [TK]D-Fender: i'm actually not sure what i'm looking for in here. |
| 14:00.59 | creativx | mvanbaak; pseudo in context default |
| 14:01.00 | [TK]D-Fender | Katty: make sure the service is running, and that your firewall isn't blocking it |
| 14:01.11 | mvanbaak | hhmm, that should do the trick |
| 14:01.49 | creativx | there must be some problems with my tricks, obviously |
| 14:01.53 | Zeeek | Katty you want to receive mail or just send it? |
| 14:01.55 | creativx | :-) |
| 14:02.05 | Katty | Zeeek: outgoing only. |
| 14:02.11 | Zeeek | simple |
| 14:02.15 | mvanbaak | this reminds me why I stopped playing with it and just went with 'trunk=no' |
| 14:02.18 | Zeeek | if I could rememeber what I did |
| 14:02.33 | Katty | We have exchange here will all the accounts needed. |
| 14:02.39 | creativx | mvanbaak: hehe... and what happened then, no iax trunking |
| 14:02.41 | Katty | the firewall allows all outgoing traffic |
| 14:02.51 | Katty | i don't know how to check for services on the linux box tho |
| 14:02.53 | mvanbaak | creativx: indeed. it works great without 'trunk=yes' |
| 14:03.01 | Zeeek | yeah don't worry about fw. Just set up sendmail |
| 14:03.03 | mvanbaak | ok, a bit more overhead on the udp flow |
| 14:03.10 | creativx | mvanbaak: then why do i have trunk=yes :] |
| 14:03.14 | Katty | how do i make sure sendmail is running first? |
| 14:03.21 | mvanbaak | but we have 1gbit link to our ITSP |
| 14:03.23 | creativx | mvanbaak: i think i just c/p'ed from Voop |
| 14:03.30 | creativx | i have a 10mbit fiber to the interweb |
| 14:03.31 | [TK]D-Fender | creativx: You need a zaptel timing source to do IAX2 Trunking |
| 14:03.37 | Zeeek | Katty remind me where the startup scripts are in linux and I'll tell you |
| 14:03.41 | mvanbaak | creativx: voop works great without the 'trunk=yes' |
| 14:03.41 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:03.44 | mvanbaak | we use them as well |
| 14:03.49 | creativx | [TK]D-Fender: yes i have understood that now, however even with the pseudo channel loaded i still got the error |
| 14:03.53 | creativx | voop are cool peeps |
| 14:04.02 | mvanbaak | uhhuh |
| 14:04.10 | Katty | Zeeek: rc.local? |
| 14:04.11 | Zeeek | <PROTECTED> |
| 14:04.15 | Zeeek | wait a sec |
| 14:04.23 | creativx | i use their norwegian voiceprompts for our voicemail |
| 14:04.23 | Zeeek | sendmail is installed? |
| 14:04.25 | [TK]D-Fender | creativx: When you recompiled Zaptel did you uncomment ZTDUMMY so that it builds? |
| 14:04.27 | Katty | uhh |
| 14:04.31 | mvanbaak | Katty: depends on the linux distro |
| 14:04.34 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 14:04.36 | creativx | [TK]D-Fender: indeed i did, lsmod shows ztdummy loaded |
| 14:04.48 | creativx | .. and zaptel loaded |
| 14:04.49 | Katty | Zeeek: it claims to be in /usr/sbin/ and /usr/lib |
| 14:04.53 | Zeeek | see if which sendmail tells |
| 14:04.57 | Zeeek | ok |
| 14:05.02 | Zeeek | just a sec |
| 14:05.05 | Katty | Zeeek: yeah /usr/bin |
| 14:05.11 | Zeeek | I no nothing about linux and I set all this up myself |
| 14:05.26 | Zeeek | <PROTECTED> |
| 14:05.37 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:05.47 | mvanbaak | /usr/sbin/sendmail |
| 14:05.50 | AndrewGearhart | [TK]D-Fender: worked with the AsteriskGUI on the *appliance (and have opinions on it)? |
| 14:06.03 | Katty | Zeeek: nifty. |
| 14:06.10 | Katty | Zeeek: i hope there's an EASY conf file somewhere. |
| 14:06.18 | Katty | Zeeek: oh please oh please let there be a lil conf file. |
| 14:06.32 | Zeeek | no sendmial does NOT have an easy conf |
| 14:06.41 | mvanbaak | Katty: if it's the normal sendmail the config is huge |
| 14:06.46 | mvanbaak | and very hard to understand |
| 14:06.47 | Zeeek | it has an m4 macro to generate .cf from Mc |
| 14:07.00 | Zeeek | nahhhhh I did it |
| 14:07.03 | [TK]D-Fender | AndrewGearhart: GUI = ick (any), and the appliance is a dead-end I would never touch. |
| 14:07.05 | Zeeek | hold on |
| 14:07.14 | Katty | maybe i should just forward it to the other server? |
| 14:07.21 | Katty | well, no |
| 14:07.22 | Zeeek | Starting sendmail MTA daemon: /usr/sbin/sendmail -L sm-mta -bd -q25m |
| 14:07.24 | mvanbaak | Katty: that is the best way to do it |
| 14:07.27 | Katty | i want to learn how to use sendmail. |
| 14:07.29 | Zeeek | that starts the queue |
| 14:07.33 | Katty | because some of our clients don't have an email server |
| 14:07.35 | Zeeek | get the O'reilly book |
| 14:07.38 | creativx | mvanbaak: oh well im gonna try trunk=no then and see what happens:) |
| 14:07.47 | creativx | mvanbaak: since i cant get this ztdummy to time |
| 14:07.47 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 14:07.47 | mvanbaak | creativx: it worked for me |
| 14:07.49 | [TK]D-Fender | Katty: No you don't. you want to pay a pro for 1 hour to set it up for you.... its a MASSIVE topic./ |
| 14:07.50 | Zeeek | I hear Excange works great! <ducks> |
| 14:08.00 | Katty | Zeeek: ha :P |
| 14:08.03 | creativx | exchange <3 |
| 14:08.06 | Katty | [TK]D-Fender: yeah, but then i'll never learn how to do it )= |
| 14:08.15 | Katty | [TK]D-Fender: and that was the whole point of me setting up this second server. |
| 14:08.24 | [TK]D-Fender | Katty: Have another apple.... Eve ;) |
| 14:08.25 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 14:08.28 | Zeeek | Katty all you need to do is install sendmail and get the queue running, but um, beopfre that the .cf file needs ot be built |
| 14:08.36 | ZaVoid | anyone got any good expereince wth chaning ports ona linkysys pap2? |
| 14:08.41 | Zeeek | just a sec and I'll try to find all that |
| 14:08.51 | mvanbaak | ok |
| 14:08.55 | mvanbaak | I really need to run |
| 14:08.59 | Zeeek | cf conf whatevr |
| 14:09.02 | Katty | k |
| 14:09.07 | Zeeek | in sendmail it's .cf |
| 14:09.11 | Katty | how do i check to see if sendmail is running? |
| 14:09.12 | mvanbaak | I have an 'out of whisky' error here |
| 14:09.21 | mvanbaak | so should get to the store before 5 PM |
| 14:09.45 | Katty | like a list of services. |
| 14:09.47 | mvanbaak | creativx: good luck. |
| 14:09.55 | mvanbaak | Katty: ps afx |
| 14:10.00 | punani | try ps aux | grep sendmail |
| 14:10.15 | mvanbaak | bbl all ! |
| 14:10.16 | Zeeek | ok Katty.... |
| 14:10.26 | Zeeek | just a sec I'll make muy window bigger |
| 14:10.27 | Sweeper | ps e |
| 14:10.31 | Katty | sendmail is running |
| 14:10.43 | Katty | it has a process id attached to it :P |
| 14:10.52 | Zeeek | find your location equiv to this: /usr/share/sendmail/cf/m4 |
| 14:11.00 | guomi | Hi folks, i'm trying to get a TE212P card working on a Debian etch server but the initialisation of the echo cancellation DSP fails |
| 14:11.11 | guomi | anyone is familiar with that kind of problem ? |
| 14:11.35 | tzafrir_laptop | guomi, maybe, when you tell us how exactly it "fails" |
| 14:11.37 | Zeeek | Katty no wait, better yet |
| 14:11.42 | Katty | oh? |
| 14:11.53 | Zeeek | find this: /usr/share/sendmail/cf |
| 14:11.59 | guomi | the first error in dmesg is "Failed to open chip, code 00103017!" |
| 14:12.08 | Zeeek | then find submit.mc |
| 14:12.10 | guomi | full dmesg available at http://dev.camptocamp.com/~francois/dmesg.txt |
| 14:12.31 | creativx | mvanbaak: i guess i will find out more on monday with more concurrent calls |
| 14:12.41 | guomi | I'm using the latest releases of libpri, zaptel and asterisk |
| 14:13.18 | Katty | Zeeek: no good. |
| 14:13.31 | Katty | Zeeek: want to see the termianl? |
| 14:13.36 | Zeeek | Katty I'm looking at my configs and I only had to modify ONE line |
| 14:13.58 | guomi | by looking at the code, the error code comes from the oct6100 API and means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED |
| 14:14.01 | Katty | Zeeek: i have a sendmail.pp file |
| 14:14.01 | Zeeek | <PROTECTED> |
| 14:14.11 | Winkie | JT: i fixed my problem thanks, it was pridialplan options |
| 14:14.18 | Winkie | for some reason it seems really popular in the UK those two options |
| 14:14.26 | Zeeek | the above line is added to tell sendmail what its name is for the enveloppe |
| 14:14.30 | AndrewGearhart | [TK]D-Fender: a co-worker reopened the phone system can of worms... so I'm back on this again. if we're using VoIP phones... and analog lines (because I can't get a decently priced PRI line here) then I just need enough FXO ports to capture the number of lines we have... correct? |
| 14:14.36 | Zeeek | pipi? Hold it! |
| 14:14.51 | Katty | that's it, i'm pastebinning. |
| 14:14.55 | creativx | oh well |
| 14:14.57 | creativx | im out, weekend |
| 14:15.02 | creativx | thanks for the help mvanbaak and Katty |
| 14:15.09 | Katty | creativx: have a great weekend (= |
| 14:15.17 | [TK]D-Fender | AndrewGearhart: Correct. 1 simple PC + analog interface card w/ FXO ports |
| 14:15.18 | creativx | you 2 |
| 14:15.26 | Zeeek | Katty there should be a bunch of .mc and .cf files in a directory called /cf/cf off od sendmail |
| 14:15.46 | Zeeek | including generic ones for each distro |
| 14:16.01 | Zeeek | mine are in /usr/share/sendmail/cf/cf |
| 14:16.10 | Katty | Zeeek: pastebin.ca/494935 |
| 14:16.18 | Zeeek | looking at my original setup from march 2004 |
| 14:16.22 | Zeeek | k |
| 14:16.27 | AndrewGearhart | [TK]D-Fender: thanks. :) You like sangoma vs the rest of the world (including digium)? |
| 14:16.41 | AndrewGearhart | [TK]D-Fender: for the analog interface card w/ FXO ports I mean |
| 14:16.55 | Katty | Zeeek: is there any way to make sure asterisk is using sendmail? |
| 14:17.00 | Zeeek | aphrodite? Hmmmmmmmmmmmmmmm |
| 14:17.06 | Katty | yes. |
| 14:17.08 | [TK]D-Fender | AndrewGearhart: Yup. A200d or bust... |
| 14:17.14 | Zeeek | And I'm Adonnius? |
| 14:17.25 | Katty | Zeeek: i have this horrible feeling it might not be using sendmail. |
| 14:17.31 | Katty | Zeeek: but i'm overly paranoid. |
| 14:17.39 | Zeeek | well sendmail isn't running with config files |
| 14:17.46 | Zeeek | look for it in ps |
| 14:17.56 | Katty | ps? |
| 14:18.01 | Zeeek | as in ps -a |
| 14:18.04 | Katty | postscript :P |
| 14:18.13 | Katty | no |
| 14:18.33 | Zeeek | type sendmail -bp |
| 14:18.38 | Katty | what's that do? |
| 14:18.39 | Katty | the bp |
| 14:18.43 | Zeeek | shows the queue |
| 14:18.45 | Katty | k |
| 14:18.55 | Zeeek | no it does an rm -r / |
| 14:19.07 | Zeeek | what did sendmail say |
| 14:19.10 | Katty | nada |
| 14:19.15 | Katty | nothing |
| 14:19.22 | Zeeek | it ran? |
| 14:19.37 | Katty | it returned no error messages |
| 14:19.42 | Katty | sec |
| 14:19.49 | Zeeek | I'm gonna guess you need to make and install sendmail |
| 14:20.00 | Zeeek | there is another simpler client some people use with asterisk |
| 14:20.18 | Katty | i think Mercestes was recommending nullmailer. |
| 14:20.20 | AndrewGearhart | [TK]D-Fender: is the hardware echo cancellation worthwhile/necessary on the A200? |
| 14:20.29 | Katty | Zeeek: not sure what that is tho. |
| 14:20.36 | Zeeek | sendmail is very weird concerning the command switches. To get the version you have to go thru a whole scene |
| 14:20.56 | Zeeek | once it's running you never need to do anything to it thogh |
| 14:21.15 | [TK]D-Fender | AndrewGearhart: Zaptel has been very hit/miss and you really don't want to look back on a purchase and wish there was a way the graft your hair back in... |
| 14:21.20 | Katty | i can probably apt-get it, Zeeek |
| 14:21.33 | Katty | Zeeek: oh. |
| 14:21.39 | Zeeek | yeah |
| 14:21.40 | Katty | Zeeek: sendmail-cf is its own package. |
| 14:21.46 | Zeeek | there ya go |
| 14:22.03 | Katty | okay, sendmail has some dependencies i don't have. |
| 14:22.12 | Katty | sendmail-bin and rmail |
| 14:22.13 | Zeeek | if you use mail and hit return does it wait for a sunbject? |
| 14:22.26 | Katty | no, it says no mail for root |
| 14:22.39 | Zeeek | well it was polite enough to look at least |
| 14:22.53 | Katty | aye. |
| 14:22.54 | Zeeek | now try mail spam@spam.com |
| 14:23.05 | Katty | askign for subject. |
| 14:23.07 | Mercestes | subject: Add me to your list |
| 14:23.15 | Obergandhi83 | hi @ all .. i've an error when i compile mISDN: /usr/src/mISDN/drivers/isdn/hardware/mISDN/l1oip_core.c:1571:55: error: macro "INIT_WORK" passed 3 arguments, but takes just 2 |
| 14:23.15 | Zeeek | sure |
| 14:23.26 | Katty | now it's just sitting around and doing nothing lol |
| 14:23.30 | Zeeek | so Katty *something* is running |
| 14:23.35 | Mercestes | typey random stuff |
| 14:23.38 | Zeeek | yeah do about 3 ctrlC |
| 14:23.44 | Katty | k |
| 14:23.46 | Mercestes | Like, "I love mercestes for ever and ever --katty" |
| 14:23.53 | Mercestes | and then hit control d |
| 14:23.55 | Katty | weirdo. |
| 14:23.58 | Mercestes | ... |
| 14:24.00 | Zeeek | no no |
| 14:24.07 | Mercestes | :) |
| 14:24.21 | Katty | Zeeek: you want ps aux? |
| 14:24.29 | Zeeek | but Katty you see there is smething running that is capable of sending mail |
| 14:24.36 | Katty | xim? |
| 14:24.41 | Zeeek | Katty sure pastebin me |
| 14:24.42 | AndrewGearhart | [TK]D-Fender: isn't the digium hardware kinda what * was built for? (is that Zaptel?) |
| 14:24.59 | Katty | k, sec |
| 14:24.59 | Zeeek | include the root pw and ip adr |
| 14:25.12 | Zeeek | just so I know it's you |
| 14:27.09 | Katty | Zeeek: pastebin.ca/494961 |
| 14:27.10 | Zeeek | TWO HOURS from now, the Asterisk Users Conference is happening! Voice recognition is a reality. It used to be a realtor, but it changed into a reaity |
| 14:27.10 | [TK]D-Fender | AndrewGearhart: Zaptel is abstract and there are many makers of cards that are compatable with it. Sangoma's have (with me) proven themselves substantially better in stability & quality. |
| 14:27.10 | Katty | exim4 is in the list!!! |
| 14:27.10 | punani | that their digium counterparts? (sorry im new) |
| 14:27.10 | Zeeek | all that on a 286? Wowo, nice |
| 14:27.10 | punani | than, rather |
| 14:27.10 | AndrewGearhart | [TK]D-Fender: and regarding the echo cancellation hardware? |
| 14:27.12 | Zeeek | rasterisk, what asterisk with dreadlocks? |
| 14:27.24 | Katty | oh hush. |
| 14:27.26 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:27.33 | Katty | irule: (= |
| 14:27.39 | [TK]D-Fender | AndrewGearhart: Do a job right and do it the first time. Don't piss off your users or you'll end up stressed, bald, and witha company after your head/ass (in no particular order) |
| 14:27.52 | Zeeek | I don't see anything mail related oddly enough |
| 14:28.01 | Katty | exim4? |
| 14:28.27 | Zeeek | silly, that's for diaheria |
| 14:28.40 | Zeeek | take the 400+ for extreme cases |
| 14:28.46 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:28.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:29.12 | AndrewGearhart | [TK]D-Fender: so,... I'm guessing that means that you'd spend the extra on the echo cancellation hardware? |
| 14:29.25 | Katty | echo cancellation is dreamy., |
| 14:29.40 | Zeeek | [TK] never worries about spending company money! |
| 14:29.48 | Zeeek | Polycoms all round on the house! |
| 14:29.48 | Katty | Zeeek: maybe i'll just try leaving myself a voicemail. |
| 14:29.49 | punani | is echo cancellation inherent in most traditional digital pbx's or just a nice thing to have? |
| 14:30.17 | coppice | traditional PBXs have no need for echo cancellation |
| 14:30.21 | punani | how so |
| 14:30.21 | Zeeek | Katty, seriously, install sendmail and it'll basically all work |
| 14:30.27 | *** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it) |
| 14:30.39 | Zeeek | tradition is overrated |
| 14:30.46 | [TK]D-Fender | AndrewGearhart: You'll spend a LOT more money in time and suffereing before buying the proper card ANYWAYS, so why don't you spare yourself. |
| 14:30.57 | coppice | why would a tradition PBX have any echo to cancel? |
| 14:31.05 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 14:31.11 | punani | why would an asterisk based system? |
| 14:31.20 | punani | im just trying to understand the conversation :P |
| 14:31.22 | Zeeek | ah. |
| 14:31.26 | [TK]D-Fender | Zeeek: And Polycom is currently cheaper than Snam & Linksys at a basic level :) |
| 14:31.44 | punani | is it a sip/iax thing or zap too |
| 14:31.47 | Zeeek | That's no reason to go to the local bar and offer them out of a brown paper bag, though |
| 14:31.57 | coppice | asterisk inserts large delays. traditional telephony strictly minimises them |
| 14:32.23 | Zeeek | ah. |
| 14:32.26 | Zeeek | finally |
| 14:32.27 | [TK]D-Fender | coppice: Inserts delays.... and who says * isn't a "value added" product! ;) |
| 14:32.32 | punani | lol |
| 14:32.43 | Katty | hot dog, it's doing something. |
| 14:32.45 | Zeeek | "adding milliseconds to your life" |
| 14:32.54 | Katty | starting mail transport agent: sendmail!!! |
| 14:33.11 | Zeeek | "sendmail will now install .NET service pack. Please wait while we..." |
| 14:33.23 | Zeeek | MTA !!!!!!! W00t |
| 14:33.35 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-7de3837e770a9d99) |
| 14:33.38 | Katty | woo!!! accepting connects!! |
| 14:33.46 | coppice | asterisk - where voices float on the latent sea |
| 14:33.47 | Katty | i mean connections. |
| 14:33.48 | Zeeek | no you don't want that |
| 14:33.52 | Katty | no? |
| 14:33.56 | Katty | oh |
| 14:33.58 | Zeeek | you DONT waznt tconnects |
| 14:33.59 | Katty | open relay NOT good |
| 14:34.02 | Zeeek | but it won't hurt |
| 14:34.09 | Katty | good thing this is on internal lan (= |
| 14:34.14 | Zeeek | no you have a firewall so nothing will happen |
| 14:34.16 | MrWup | anyone here use mixmonitor? |
| 14:34.18 | Zeeek | yeah |
| 14:34.24 | irule | hi Katty |
| 14:34.39 | AndrewGearhart | anybody have recommendations on a softphone to use to be able to do a proof of concept on an * system? |
| 14:34.40 | Zeeek | so Katty now do you see sendmail running in ps -aux |
| 14:34.43 | Katty | yes |
| 14:34.46 | MrWup | anyone found that mixmonitor stops recording after certain transfers? |
| 14:34.49 | Zeeek | great |
| 14:35.28 | Zeeek | and you have configured the voicemail.conf stuff? |
| 14:35.28 | [TK]D-Fender | AndrewGearhart: on Windows : idefisk , on *nix : ekiga |
| 14:35.28 | MrWup | e.g. when a SIP phone calls a PSTN caller in a different context and then transfers the call to another sip phone? |
| 14:35.28 | Katty | sendmail -bp says recipient names must be specified. |
| 14:35.28 | Katty | Zeeek: i don' tthink so. |
| 14:35.28 | MrWup | mixmonitor simply stops recording when the first sip phone drops out? |
| 14:35.37 | Katty | Zeeek: the only thing i did was set attachments to yes, and added a user with pin, name, and email address. |
| 14:35.38 | Zeeek | that's wrong, it may be -Bp |
| 14:36.03 | Zeeek | show me the ONE LINE of sendmail in ps =? |
| 14:36.15 | Katty | bP |
| 14:37.07 | Katty | if i type "ps" sendmail doesn't come up. |
| 14:37.07 | Zeeek | Katty use man to see what the options are: mine is -bp Print a listing of the queue(s). |
| 14:37.18 | Zeeek | ok, just a sec |
| 14:37.52 | Zeeek | sendmail: Queue runner@00:25:00 for /var/spool/clien |
| 14:38.06 | Zeeek | you should see something like that |
| 14:38.24 | Katty | Zeeek: /var/spool/mqueue is empty Total requests: 0 |
| 14:38.26 | *** join/#asterisk cnbrk (n=canberk@teknopet.com) |
| 14:38.29 | cnbrk | hello guys |
| 14:38.41 | cnbrk | i have a problem with g729 codec |
| 14:38.46 | cnbrk | loader.c:325 __load_resource: Shared object "libpthread.so.0" not found, required by "codec_g729.so" |
| 14:38.53 | Zeeek | Katty that's GREAT! |
| 14:39.06 | Katty | yay, i have queues ^_^ |
| 14:39.20 | Zeeek | and in rc.sendmail you should see the line that starts the queue, something like |
| 14:39.22 | Zeeek | <PROTECTED> |
| 14:39.50 | MrWup | anymore for anymore step right up, answer a mixmonitor question |
| 14:39.50 | Zeeek | the only problem may be the need to masquerade |
| 14:39.52 | Zeeek | you know, like you do on Second Life? |
| 14:39.52 | MrWup | ill pay 50 USD to whoever solves it |
| 14:40.03 | Katty | rc.sendmail is a file? |
| 14:40.06 | [TK]D-Fender | cnbrk: What OS are you using? |
| 14:40.12 | MrWup | im serious |
| 14:40.15 | MrWup | paypal |
| 14:40.20 | Zeeek | <PROTECTED> |
| 14:40.30 | Katty | err, sec |
| 14:40.31 | Zeeek | but that varies from distro |
| 14:40.44 | cnbrk | freebsd |
| 14:40.47 | Zeeek | nayway, that's just when the sys boots |
| 14:40.57 | Zeeek | i love FreeBSD |
| 14:40.58 | [TK]D-Fender | MrWup: When 1 side hangs up, whats LEFT to record? |
| 14:40.59 | punani | [TK]D-Fender: would you say some handsets are worse for echo |
| 14:41.07 | punani | or is it more a server side issue |
| 14:41.21 | Zeeek | punani Poly and Linksys do a decent job for each suppr |
| 14:41.33 | MrWup | [TK]D-Fender, i dont understand the question |
| 14:41.42 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.114) |
| 14:41.46 | Zeeek | awwwww |
| 14:41.48 | punani | suppr? |
| 14:42.00 | Zeeek | oops Echo Suppression |
| 14:42.04 | [TK]D-Fender | punani: Yes, GrandStreams have been known to induce it in a nasty way previously. Byond that, not really. Sometimes a few ATA, but fixed with a minor tweak. |
| 14:42.05 | punani | ah right |
| 14:42.14 | BSD_tech | freebsd rules the world |
| 14:42.18 | [TK]D-Fender | punani: But in general SIP phones should not be the source of any echo. |
| 14:42.19 | Zeeek | Katty if the queue is running, go for it! Leave yourself a voicemail |
| 14:42.20 | BSD_tech | <PROTECTED> |
| 14:42.27 | Katty | mmk |
| 14:42.33 | Zeeek | FreebSD DOES rule! |
| 14:42.40 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:42.50 | Zeeek | more cheesebread anyone? It's not that great cold |
| 14:42.55 | Qwell[] | nuke it |
| 14:42.56 | Katty | actually, i'm gonna reboot first. |
| 14:42.57 | Zeeek | BEER! That'll help it a lot |
| 14:42.57 | [TK]D-Fender | Zeeek: Yes... better a king in the land of idiots than an idiot in the land of kings ;) |
| 14:43.00 | punani | ok cheers |
| 14:43.01 | Katty | and see if sendmail comes back up on its own. |
| 14:43.09 | cnbrk | so any opinions? |
| 14:43.15 | cnbrk | loader.c:325 __load_resource: Shared object "libpthread.so.0" not found, required by "codec_g729.so" |
| 14:43.20 | Zeeek | Katty, yes Exchange server needs you to reboot and also .NET service pack |
| 14:43.35 | Katty | dont' remind me about exchange |
| 14:43.38 | Qwell[] | cnbrk: This the Digium codec? |
| 14:43.44 | [TK]D-Fender | cnbrk: thats a POSIX threads library IINM but can't advise on how to correct for your setup |
| 14:43.45 | Katty | it's gotten to the point where i take it offline once a week to just back it up |
| 14:43.45 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:43.46 | Zeeek | Katty if it doesn't though it's pretty straightforward to start the queue |
| 14:44.02 | Katty | otherwise, it has this nasty habit of...not getting backedup through the Exchange Aware software. |
| 14:44.07 | cnbrk | this is not the digium codec |
| 14:44.18 | Qwell[] | cnbrk: Then you're pretty much screwed. :p |
| 14:44.18 | cnbrk | this is the cracked one, or the free one in other words |
| 14:44.34 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 14:44.40 | [TK]D-Fender | cnbrk: Words can barely express how little sympathy you are garnering for your compounded scenario :| |
| 14:44.41 | cnbrk | jeez |
| 14:44.41 | Zeeek | TK, I ressembel that idot remark! But what was it in reference to? |
| 14:44.52 | cnbrk | lol |
| 14:45.07 | Zeeek | compound scenario? You've been writing Bush speeches again? |
| 14:45.10 | Katty | ohoh |
| 14:45.12 | [TK]D-Fender | Zeeek: with regards to "freeBSD rules" ;) |
| 14:45.13 | cnbrk | lmao |
| 14:45.15 | Katty | i saw sendmail come up during boot stuffs |
| 14:45.21 | MrWup | [TK]D-Fender, ah you dont understand my problem |
| 14:45.23 | [TK]D-Fender | Zeeek: pwned |
| 14:45.30 | Zeeek | Hah you said it and it 's now recorded in the Anales of IRC |
| 14:45.32 | MrWup | [TK]D-Fender, 1 calls 2, 1 transfers 2 to 3. 1 hangs up |
| 14:45.41 | MrWup | leaving 1 and 3 in a conversation |
| 14:45.49 | MrWup | 1 and 3's call needs to be recorded |
| 14:45.52 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 14:45.54 | MrWup | yet only 1 and 2 is recorded |
| 14:45.59 | MrWup | because mixmonitor stops |
| 14:46.00 | Katty | yay, sendmail is back up |
| 14:46.07 | Zeeek | Kewl, Katty |
| 14:46.12 | [TK]D-Fender | Zeeek: No, I use big word, and they carry the full weight of their meaning with nothing hidden :) GWB is a MORON. |
| 14:46.14 | Zeeek | so we deserved the hugs? |
| 14:46.31 | Katty | voicemail time ^_^ |
| 14:46.41 | Zeeek | TK more a dangerous thug surrounded by other dangerous thugs |
| 14:47.04 | Zeeek | Katty give us a number, well call you and leave vmails :) |
| 14:47.17 | [TK]D-Fender | MrWup: Ah.... yeah, thats bounty-worthy |
| 14:47.25 | cnbrk | what is the difference between a icc and gcc codec |
| 14:47.31 | Zeeek | I'd rather have Adobe on my side than Rove |
| 14:47.32 | Katty | Zeeek: it's not quite working full just yet |
| 14:47.44 | Zeeek | but what does that say about me? |
| 14:47.46 | Katty | Zeeek: in fact, i have nothing handling incoming calls ;) |
| 14:47.55 | Katty | everything is outgoing and internal for now |
| 14:47.57 | Zeeek | Katty asterisk likes it that way |
| 14:48.00 | Zeeek | don't change |
| 14:48.20 | Zeeek | runnin a pbx with no incoming path is so much easier, maintenance wise |
| 14:48.29 | AndrewGearhart | ty [TK]D-Fender |
| 14:48.30 | [TK]D-Fender | Zeeek: Congress willing he'll be ousted along with Cheny simultaneously and Pelosi can be Lame-ducked if needed until the next election. |
| 14:48.34 | Zeeek | plus no support calls! |
| 14:49.04 | cnbrk | what is the difference between a icc and gcc codec???? |
| 14:49.05 | [TK]D-Fender | Zeeek: I figure she can't possibly do more damage than Bush/Cheney... they are trying too hard :) |
| 14:49.14 | Zeeek | TK this is my libertarian friends' scenario: just stop govt from doing anything at all |
| 14:49.30 | Zeeek | keep em all tied up |
| 14:49.37 | irule | I have a server already running with the same configuration, and this other server gives me this error message, I have already double checked for the phone cable to be correctly connected http://pastebin.ca/495003 |
| 14:49.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:49.53 | [TK]D-Fender | Zeeek: I could seriously live with the thought of Ron Paul or Mike Gravel in office... |
| 14:50.04 | Zeeek | vote for me instead |
| 14:50.26 | Zeeek | aren't you in Canada anyway? |
| 14:50.34 | [TK]D-Fender | Zeeek: Gravel preferred for being a Dem who really knows how less=more |
| 14:50.59 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 14:51.00 | [TK]D-Fender | Zeeek: Yes, but Bush has ruined everything south of the border vacation-wise for me! |
| 14:51.05 | Zeeek | never heard of him until just now |
| 14:51.21 | Zeeek | god he has tha pol look though |
| 14:51.34 | Katty | Zeeek: i left my self a voicemail |
| 14:51.38 | [TK]D-Fender | Zeeek: Mike Gravel is RIGHT up your alley. Also classifiable as a libertarian |
| 14:51.39 | Zeeek | "this one's only been dribven on Sundays by a little old lady" |
| 14:51.47 | Katty | SPAM! |
| 14:51.54 | Zeeek | I'm not a libertarian, but I have two friends that are |
| 14:51.59 | tzafrir_home | irule, you dial to Zap/g2 . But I don't see anything of the sort of group=2 in zapata.conf |
| 14:52.03 | Zeeek | Katty don't spam yourself |
| 14:52.07 | [TK]D-Fender | Zeeek: He single-handedly killed the Vietnam Draft |
| 14:52.08 | Katty | ooh it works it works |
| 14:52.08 | Katty | yay |
| 14:52.21 | vAd0r | I found this link to integrate asterisk w/ cisco call manager. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Will this allow me to make outbound calls through my work PBX instead of using a free voip one |
| 14:52.21 | [TK]D-Fender | Zeeek: Like I said... either I could live with. |
| 14:52.27 | Zeeek | who cares, I joined but didn't go over |
| 14:52.34 | Katty | Zeeek: now then. |
| 14:52.40 | Katty | Zeeek: i need two email address in there |
| 14:52.42 | [TK]D-Fender | Zeeek: Obama is "nice", but a little "green" for my liking. a good VP maybe |
| 14:52.50 | Zeeek | TK you and I could live with Godzilla compared to staus quo |
| 14:52.52 | irule | tzafrir_home should it be zap/g1 or something? |
| 14:53.03 | Katty | Zeeek: i can just tack it on at the end, right? |
| 14:53.08 | Katty | Zeeek: of voicemail.conf? |
| 14:53.17 | Zeeek | Obama has - check your dialplan, now on topic - a lot of charisma |
| 14:53.26 | [TK]D-Fender | Zeeek: :D http://www.youtube.com/watch?v=xAVwUb9hg4Y |
| 14:53.32 | Zeeek | Katty why 2 ? |
| 14:53.33 | AndrewGearhart | [TK]D-Fender: good place to find a great price on a sangoma A200D? |
| 14:53.52 | Katty | Zeeek: i want a notification to be send to my phonenumber@messaging.nextel.com address |
| 14:53.56 | Zeeek | there are two addresses in each line from the beeper days |
| 14:54.00 | Katty | Zeeek: in case i'm not at the office checking my regular email |
| 14:54.03 | [TK]D-Fender | AndrewGearhart: www.voipsupply.com , www.atacomm.com , www.telephonydepot.com |
| 14:54.08 | Katty | Zeeek: or maybe my gamil or something |
| 14:54.13 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 14:54.51 | Zeeek | Katty here's what we do: I sedn mail to my other serve. procmail send a mail to my cell provider and I get an SMS for each message plus the mail with attachment at any number of addresses I want |
| 14:54.58 | Zeeek | handle that stuff at a second server |
| 14:54.59 | AndrewGearhart | [TK]D-Fender: ty |
| 14:55.09 | *** part/#asterisk madeinny (n=Steve@70.88.255.133) |
| 14:55.19 | Zeeek | easy to filter, set up a special address for asterisk generated mail |
| 14:55.33 | vAd0r | I found this link to integrate asterisk w/ cisco call manager. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Will this allow me to make outbound calls through my work PBX instead of using a free voip one |
| 14:55.33 | [TK]D-Fender | Zeeek: bad link.. here : http://www.youtube.com/watch?v=ZpBkc2jK-6w |
| 14:55.51 | [TK]D-Fender | vAd0r: unload chan_echo.so |
| 14:56.12 | Zeeek | I'm so old I saw that in the movie theater! |
| 14:56.16 | nestAr | blah. |
| 14:56.17 | tzafrir_home | irule, what do you try to dial? if you have just channel 1, you can simply use Zap/1 |
| 14:56.17 | [TK]D-Fender | vAd0r: Go read ... THE BOOK |
| 14:56.22 | [TK]D-Fender | ~book |
| 14:56.24 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:56.34 | nestAr | i need a digium pci card, this dlink box sucks |
| 14:56.37 | Zeeek | also the Mel Brooks version |
| 14:56.45 | Zeeek | no, that was the MB version |
| 14:56.45 | punani | [TK]D-Fender: i had two potential cards lined up for our 5 pstn lines |
| 14:56.49 | punani | tdm800 or the Sangoma Remora A20003 |
| 14:56.56 | *** join/#asterisk madeinny (n=Steve@70.88.255.133) |
| 14:56.57 | irule | tzafrir_home thanks a lot! it works! I am just stating to understand all this lol |
| 14:57.30 | johann8384 | irule: if it makes sense....run away...lest you become one of us |
| 14:57.36 | Zeeek | TK "opposes a military confrontation with iran...". Ok, "Zeeek opposes walking in front of a truck on a freeway" |
| 14:57.44 | punani | am i on the right track? |
| 14:57.49 | Zeeek | like we have the resources for YAW |
| 14:58.11 | Zeeek | They'd have to draft 9 year olds |
| 14:58.42 | Katty | in the line 110,Dial(SIP/110,20wW) what does the wW mean? |
| 14:58.46 | Katty | i get 20 is the timeout |
| 14:58.53 | Zeeek | Monitor, no? |
| 14:58.55 | Katty | does that refer to recording? |
| 14:58.58 | Zeeek | ya |
| 14:59.04 | Katty | outgoing and incoming? |
| 14:59.06 | Qwell[] | that's the wrong syntax though |
| 14:59.17 | Zeeek | yeah missing a delimiter |
| 14:59.26 | Zeeek | 20,w) |
| 14:59.27 | Katty | i'm just gonna take it off for now |
| 14:59.36 | Zeeek | take what off? |
| 14:59.41 | Zeeek | oh. |
| 14:59.54 | Katty | the wW bit |
| 15:00.00 | Zeeek | so there a 110 registered? |
| 15:00.18 | Zeeek | ah |
| 15:00.51 | Zeeek | phone call |
| 15:01.01 | Katty | yeah |
| 15:01.03 | Katty | 110 is me phone |
| 15:01.28 | Katty | cause that's my shiny office number. |
| 15:01.37 | vAd0r | did trixbox sell out or something? |
| 15:01.55 | Qwell[] | vAd0r: it never sold..in |
| 15:02.07 | vAd0r | i mean it looks like they are a business now |
| 15:02.26 | Katty | oooh, my speakers are making noise |
| 15:02.30 | Katty | MY PHONE MADE NOISE |
| 15:02.51 | Katty | sweet |
| 15:02.53 | vAd0r | I found this link to integrate asterisk w/ cisco call manager. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Will this allow me to make outbound calls through my work PBX instead of using a free voip one |
| 15:03.22 | anonymouz666 | I don't know how to use cool edit to record waves, lol. |
| 15:03.31 | [TK]D-Fender | vAd0r: Welcome to months ago... Yes, Fonality "bought |
| 15:03.54 | [TK]D-Fender | " Trixbox. Its still free but now they are selling prepared servers w/ support for it. |
| 15:04.01 | Qwell[] | they paid one guy for the work of dozens |
| 15:04.12 | irule | johann8384 lol, hey guys can I gather a couple dialing patterns together within a single extension line? exten => _9001.&_9005.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) I saw a dialplan explanation for a sipura ata that put the things together with |, is it possible in *? thanks |
| 15:04.27 | [TK]D-Fender | irule: No. |
| 15:04.40 | irule | [TK]D-Fender thanks :) |
| 15:05.40 | vAd0r | does no one know the answer |
| 15:05.57 | vAd0r | or just not answering |
| 15:09.14 | [TK]D-Fender | vAd0r: You never said you were using CCM, or what you want * to do for you exactly. |
| 15:09.14 | vAd0r | ok ill reword |
| 15:09.14 | vAd0r | sry |
| 15:09.14 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 15:09.15 | [TK]D-Fender | vAd0r: Yes, rewording is sadly needed |
| 15:09.15 | vAd0r | I have asterisk at home and cisco call manager at work. I have sip phones at my house that connect to asterisk. The problem is the free voip providers all dont have local numbers for me to receive calls on. I want to send my calls through my work cisco phone system |
| 15:09.33 | vAd0r | and receive calls that route through my work to asterisk |
| 15:09.44 | vAd0r | i was wondering if that link would accomplish that |
| 15:11.15 | [TK]D-Fender | vAd0r: If you set up your CCM to allow use of a regular SIP end-point, sure. |
| 15:11.29 | Zeeek | Katty great newz |
| 15:11.42 | Zeeek | join the auc |
| 15:11.51 | vAd0r | im not sure how to do that |
| 15:11.56 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 15:12.03 | MrWup | no more bounty |
| 15:12.06 | MrWup | have found a workaround |
| 15:12.08 | irule | anybody have an example of vmauthenticate? thanks |
| 15:12.10 | vAd0r | do i add a gateway and the gatway would be my asterisk |
| 15:12.17 | MrWup | you people did yourselves out of 50 bucks |
| 15:12.22 | MrWup | bet youre all sorry now eh |
| 15:12.22 | MrWup | eh |
| 15:12.24 | MrWup | eh eh |
| 15:12.41 | [TK]D-Fender | vAd0r: They have PICTURES there... if you don't know how to configure your CCm or can't follow those instructions, go hire a Cisco tech. * can do its part of the job jsut fine. |
| 15:12.51 | ctaloi | hey all - anyone know what Non-Codec Capability (DTMF): 4 In the output of sip show channel foo means? |
| 15:13.00 | ctaloi | i'm having some dtmf issues.... |
| 15:13.27 | AndrewGearhart | [TK]D-Fender: any opinion on hardware platform for *? AMD/Intel? 32 vs 64bit? |
| 15:14.11 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 15:14.11 | vAd0r | I was just asking if that link would allow me to do what i want |
| 15:14.12 | [TK]D-Fender | AndrewGearhart: I've heard of numerous issues with 64-bit for the OS, but as base hardware running i386 I've heard little by way of issues. |
| 15:14.17 | vAd0r | would that accomplish the task |
| 15:14.22 | mocker | For soem reason ARI is dead slow on this system. |
| 15:14.34 | cnbrk | allrite guys |
| 15:14.39 | Zeeek | ~AUC |
| 15:14.51 | jbot | extra, extra, read all about it, auc is Asterisk Users Conference, Friday at 12:30 PM Eastern Time |
| 15:14.51 | Katty | Zeeek: it works it works it works ^_^ |
| 15:14.51 | cnbrk | now i installed the g729 codec |
| 15:14.52 | Zeeek | Katty join us out of gratitude |
| 15:14.53 | cnbrk | however i can't hear anything from one party |
| 15:14.56 | irule | AndrewGearhart Ive gone intel pentium d, _the book_ says intel rules |
| 15:15.00 | cnbrk | the other party can hear me... |
| 15:15.04 | [TK]D-Fender | vAd0r: Well that seems to say how to setup a SIP account on CCm so I'd guess the answer is "yes" |
| 15:15.25 | BSD_tech | ok 1.4.x how do you see how many channels of g729 are in use |
| 15:15.34 | [TK]D-Fender | AndrewGearhart: I've used both Intel & AMD just fine |
| 15:15.37 | cnbrk | aha i solved it |
| 15:15.38 | Qwell[] | cnbrk: you aren't going to get any help here with that codec |
| 15:15.44 | cnbrk | my asterisk is behind nat |
| 15:15.46 | BSD_tech | and is the a confrence this am |
| 15:15.56 | cnbrk | which ports should I forward to my asterisk server? |
| 15:16.01 | Zeeek | BSD_tech yup |
| 15:16.08 | vAd0r | it shows how to setup a SIP Trunk |
| 15:16.09 | BSD_tech | whats the url |
| 15:16.12 | [TK]D-Fender | cnbrk: ... |
| 15:16.14 | [TK]D-Fender | ~sipnat |
| 15:16.23 | jbot | [sipnat] for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:16.23 | BSD_tech | I want to tue in and call in |
| 15:16.23 | Zeeek | http://x2z.eu |
| 15:17.10 | irule | where is AsteriskTFOT.pdf second edition? :s |
| 15:18.13 | BSD_tech | zeek pm me |
| 15:18.15 | [TK]D-Fender | irule: Awaiting release to print. |
| 15:20.26 | *** join/#asterisk ecthelion_italy (n=ecthelio@host207-170-dynamic.2-87-r.retail.telecomitalia.it) |
| 15:21.20 | irule | [TK]D-Fender but wasnt there an open source version available in docbook or something? |
| 15:22.00 | [TK]D-Fender | irule: Its released under CC, it just isn't RELEASED. |
| 15:22.18 | *** join/#asterisk ccesario (n=ccesario@200-158-227-79.dsl.telesp.net.br) |
| 15:22.27 | [TK]D-Fender | irule: we'll get it when its "ready" |
| 15:22.49 | [TK]D-Fender | irule: Which is currently in the next short while (pre-summer from last I've heard) |
| 15:22.56 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 15:22.59 | Katty | who's the girl that did all the voice prompts? |
| 15:23.49 | Qwell[] | Allison |
| 15:23.51 | Qwell[] | Smith |
| 15:24.01 | Zeeek | folks for the conference, the number wont work until about 12:20 PM EDT |
| 15:24.49 | irule | Allison Smith's website http://theivrvoice.com/ |
| 15:25.06 | anonymouz666 | Qwell: she speaks portuguese? |
| 15:25.07 | anonymouz666 | :D |
| 15:26.16 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:27.44 | *** part/#asterisk madeinny (n=Steve@70.88.255.133) |
| 15:28.28 | Zeeek | Katty join us at 12:30 |
| 15:28.34 | Katty | who want when? |
| 15:28.50 | Katty | oh |
| 15:28.53 | Katty | conference for waht? |
| 15:28.55 | Zeeek | 12:30 pm EDT |
| 15:29.04 | Katty | i don't think i'll be awake then |
| 15:29.07 | Katty | oh, wait, it's friday |
| 15:29.08 | Katty | hmm |
| 15:29.13 | Zeeek | in one hour |
| 15:29.15 | Katty | are you accept iax connections? |
| 15:29.20 | Zeeek | no |
| 15:29.24 | Katty | )= |
| 15:29.27 | Zeeek | sip |
| 15:29.33 | Zeeek | no excuses |
| 15:29.42 | Katty | sip doesn't work nicely through firewalls |
| 15:30.04 | Zeeek | oh posh! |
| 15:30.06 | [TK]D-Fender | Katty: Works just fine, whatcha talkin' 'bout Willis? :) |
| 15:30.24 | Katty | k, well, help me set it up |
| 15:30.26 | Katty | and i'll come visit |
| 15:30.33 | Katty | mkay? |
| 15:30.39 | Zeeek | http://x2z.eu |
| 15:31.10 | Katty | oh |
| 15:31.15 | Katty | uhh, let me set it up real quick |
| 15:31.27 | Zeeek | use 2007200720 |
| 15:31.30 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 15:31.36 | Zeeek | use 2007200700 |
| 15:31.40 | Zeeek | use 2007200722 |
| 15:31.58 | Zeeek | any of above |
| 15:31.58 | Qwell[] | Zeeek: why not just open it up to everybody, without requiring a pin? |
| 15:32.00 | Katty | as the pin? |
| 15:32.04 | Zeeek | i cant |
| 15:32.05 | Katty | i don't lik pins. |
| 15:32.19 | Katty | i wanna test it first. |
| 15:32.24 | Katty | cause of this firewall thing |
| 15:32.29 | Zeeek | brb phone wife in SF |
| 15:33.02 | lee_is_me | has anyone tried the Sangoma with the onboard Octasic SoftEcho? Any thoughts? |
| 15:35.54 | Zeeek | so... Qwell you only need a PIN if you want to talk. Anyone can listen anonymously to the stream |
| 15:36.07 | Zeeek | However, the PIN allows me to see who is on so I can call on them to talk |
| 15:36.36 | Zeeek | anslo there's a java chat app, addmittedly a lame one, that people can use to exchange URLs or whatever |
| 15:36.57 | Zeeek | my typing becomesz very poor when I don't have enough beer |
| 15:38.49 | Brandon_W | ok |
| 15:38.51 | Brandon_W | I'm an idiot |
| 15:39.02 | Brandon_W | sip show peer got replaced with something new in 1.4 |
| 15:39.02 | Mercestes | No arguement here |
| 15:39.05 | Brandon_W | amen |
| 15:39.11 | Brandon_W | I'm not seeing it in help |
| 15:39.18 | Mercestes | probaby core sip show peer or something retarded like that. |
| 15:40.02 | angryuser | sip show peers i have 1.4 works there |
| 15:40.22 | angryuser | ah peer, ok ;) |
| 15:40.27 | Mercestes | Some doof-wad decided it'd be 133t to revamp the entire syntax to require a bunch more "geeky" words with no added functionality |
| 15:40.37 | Brandon_W | Asterisk SVN-trunk-r64657 |
| 15:40.49 | Brandon_W | I think I got the latest latest version that stripped that stuff out |
| 15:43.44 | Katty | is 2 email addresses the limitation of asterisk? |
| 15:44.34 | Mercestes | Katty, one is an email address and the other is a pager address. I think you can seperate them with semicolons tho |
| 15:44.35 | Mercestes | or commas, or...something. |
| 15:44.35 | Katty | oh! |
| 15:44.35 | Katty | so... |
| 15:44.38 | Katty | the position determines the message |
| 15:44.44 | Katty | and only two recipients are available |
| 15:44.47 | anonymouz666 | there is nothing like "invalid password" in asterisk-sounds? |
| 15:44.59 | Katty | anonymouz666: there's the voicemail one |
| 15:45.01 | Mercestes | you can send to more than 2 email addresses. |
| 15:45.06 | Katty | anonymouz666: "login incorrect" |
| 15:45.12 | cnbrk | here is an interesting question |
| 15:45.17 | anonymouz666 | I need "password invalid" |
| 15:45.24 | anonymouz666 | something related to password |
| 15:45.26 | Katty | Mercestes: myemailaddress@whoever.com |
| 15:45.28 | anonymouz666 | not login |
| 15:45.30 | Mercestes | anonymouz666, Call allison |
| 15:45.35 | Katty | Mercestes: my10digitphonenumber@messaging.nextel.com |
| 15:45.40 | cnbrk | asterisk does not support voice activity detection right. what happens if the trunk that i use forces asterisk to use it? |
| 15:45.53 | Mercestes | Katty, exactly so |
| 15:45.54 | anonymouz666 | Mercestes: how? where? what happened? |
| 15:45.57 | Katty | Mercestes: how do i add in a third? |
| 15:46.00 | mvanbaak | anonymouz666: it's better to do: 'login incorrect' |
| 15:46.03 | Katty | Mercestes: i tried, and nothing happened. |
| 15:46.09 | mvanbaak | anonymouz666: if you only do: password incorrect |
| 15:46.20 | mvanbaak | the one trying will know the username _IS_ correct |
| 15:46.20 | *** join/#asterisk TechBlazer (n=tim@70.88.27.130) |
| 15:46.22 | anonymouz666 | it's a lock |
| 15:46.26 | anonymouz666 | there is no login involved |
| 15:46.42 | Katty | then login incorrect can only refer to one thing anyway :P |
| 15:46.50 | mvanbaak | indeed ;) |
| 15:47.05 | Katty | anonymouz666: record one yourself. |
| 15:47.07 | *** join/#asterisk Strom_C (i=strom@nat/digium/x-fcaf1034551c91c0) |
| 15:47.10 | Katty | Strom_C: (= |
| 15:47.16 | mvanbaak | heya strom |
| 15:47.42 | anonymouz666 | Katty: my voice is so beautiful |
| 15:47.55 | Mercestes | Katty, sec |
| 15:48.05 | anonymouz666 | I am not native speaker of english |
| 15:48.15 | Katty | anonymouz666: well my certainly is :P |
| 15:48.31 | anonymouz666 | record to me then :D |
| 15:48.40 | Katty | whatcha want it to say? |
| 15:48.53 | anonymouz666 | Password invalid |
| 15:50.48 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:52.27 | Brandon_W | I think I figured out the sip show problem.... Could it be that the sip portion wasn't compiled in... MMmm... |
| 15:52.51 | cy303 | So am I just blind or is there no way to bypass the whole "record your name after the beep" before entering a MeetMe conference? |
| 15:53.52 | Zeeek | Here at #asterisk we take security very seriously. If you have any problems, please pastebin your root password and post the link here. Thanks you. |
| 15:53.58 | *** join/#asterisk johann8384 (n=johann83@206.80.75.240) |
| 15:54.20 | Mercestes | what good will that do? |
| 15:54.40 | Mercestes | my root password is H4r01d |
| 15:55.09 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:55.13 | anonymouz666 | l33th4x0r |
| 15:55.19 | Katty | anonymouz666: http://www.research.att.com/~ttsweb/tts/demo.php |
| 15:55.32 | Zeeek | ip addreszs that goes with that root pw? |
| 15:55.38 | anonymouz666 | Katty: I will check that, thanks |
| 15:55.46 | Katty | anonymouz666: Type Password In Valid |
| 15:55.50 | [TK]D-Fender | Zeeek: 127.0.0.1 |
| 15:55.54 | Katty | anonymouz666: choose Anjali |
| 15:55.58 | Katty | anonymouz666: download. |
| 15:56.14 | Zeeek | Let em try to log in. ssh 127.0.0.1 |
| 15:56.21 | Zeeek | It works! |
| 15:57.13 | anonymouz666 | Anjali? |
| 15:57.15 | mvanbaak | Zeeek: now type: rm -rf /bin |
| 15:57.16 | anonymouz666 | wtf |
| 15:57.36 | Zeeek | mvanbaak ohhhh it worked too |
| 15:58.01 | mvanbaak | find / -type l -exec rm -rf {} \; |
| 15:58.46 | Zeeek | stop trying to show off that's in ever linux for dummies edition since 1947 |
| 16:00.36 | *** join/#asterisk Ebola (n=Ebola@host86-137-4-175.range86-137.btcentralplus.com) |
| 16:03.41 | Katty | anonymouz666: http://www.brick.net/~izaah/talktokat.wvav |
| 16:03.41 | Katty | wav |
| 16:03.41 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 16:04.10 | Katty | Zeeek: (= |
| 16:04.16 | Zeeek | Press one for Spanish |
| 16:04.24 | Zeeek | Two for Polish |
| 16:04.42 | EmleyMoor | Tree for Irish <g> |
| 16:04.54 | Qwell[] | or for English, please press 42 |
| 16:04.54 | mocker | Can someone confirm that astctlpermissions in asterisk.conf is what sets the permissions on newly created files (i.e. voicemails)? |
| 16:05.45 | coppice | For Engrish, press å›› |
| 16:05.45 | LeddyHM | I pressed 1 and nothing happened |
| 16:05.45 | LeddyHM | your server must be broken |
| 16:07.01 | Zeeek | the next step is to HIT one |
| 16:08.47 | anonymouz666 | nice |
| 16:08.49 | anonymouz666 | :d |
| 16:08.51 | anonymouz666 | :P |
| 16:09.07 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 16:09.20 | anonymouz666 | I need portuguese voices too, that's almost impossible to find |
| 16:09.56 | Katty | http://www.brick.net/~izaah/screwedupprompt.wav |
| 16:10.53 | Katty | also, the above is NOT me |
| 16:10.56 | Katty | if anyone was wondering. |
| 16:11.44 | Katty | http://www.webcon.net/~izaah/audiorecordings.html <- that however, is. |
| 16:12.33 | Qwell[] | Katty: you...singing? |
| 16:12.36 | Qwell[] | that could be...scary |
| 16:12.41 | Katty | quite. |
| 16:18.05 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 16:18.27 | Zeeek | we're warming up the conference room folk |
| 16:18.39 | joe | if you have a foo.gsm file and you would like asterisk to call a specific number/extension say 4567 when an extention say 333 is dialed and play that foo.gsm file how would one do that? |
| 16:20.58 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:22.14 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-98a01739b05dfa91) |
| 16:22.14 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:22.56 | Zeeek | russellb u there? |
| 16:23.16 | Zeeek | you guys gonna hook up a meetme? |
| 16:26.10 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:26.40 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 16:26.56 | skirmisha | guys how can i connect on remote asterisk |
| 16:27.34 | gerphimum | asterisk -r |
| 16:28.09 | skirmisha | i am talking on remote server |
| 16:28.17 | tuan_modulis | ssh |
| 16:28.18 | skirmisha | can i tell ip or something |
| 16:28.24 | skirmisha | nope |
| 16:28.34 | skirmisha | i want to connect directly on asterisk |
| 16:28.46 | [TK]D-Fender | Joe : go read up on .call files and AMI Originate on the WIKI |
| 16:28.47 | skirmisha | shall i use asterisk manager interface |
| 16:28.48 | [TK]D-Fender | ~wikis |
| 16:29.00 | jbot | methinks wikis is http://www.voip-info.org |
| 16:29.22 | kaldemar | skirmisha: what do you want to do? connecting to asterisk doesn't tell that much. |
| 16:30.35 | joe | [TK]D-Fender: thank you much.. |
| 16:32.39 | skirmisha | i want to get info from remote asterisk server |
| 16:32.54 | skirmisha | want to do it either wirh php or perl |
| 16:34.51 | tuan_modulis | well u can execute external commands instead of using the CLI |
| 16:35.16 | tuan_modulis | so with say php, execute those commands |
| 16:35.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 16:35.41 | tuan_modulis | is this for monitoring purposes? |
| 16:36.13 | Strom_C | hi |
| 16:37.09 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 16:38.25 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 16:39.55 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 16:40.43 | [TK]D-Fender | skirmisha: AMI <- |
| 16:41.15 | *** join/#asterisk VJFROMGT (n=vijay_0@pool-71-125-253-94.nycmny.east.verizon.net) |
| 16:42.04 | VJFROMGT | I have 2 servers connected by iax2 , calls from a > b are ok but b > a drops after about 1 min |
| 16:42.58 | savaticus | AMI originate is my personal preference for call generation, you can talk to it with anything that can use sockets and perl has a few nice AMI modules through CPAN |
| 16:43.37 | skirmisha | tuan_modulis yes for monitoring |
| 16:43.57 | skirmisha | [TK]D-Fender AMI what is that |
| 16:44.05 | savaticus | I have all of my ast boxes being monitered through pelr IRC bots that I built |
| 16:44.17 | [TK]D-Fender | skirmisha: Asterisk Manager Interface. Go to the WIKI and get reading. |
| 16:44.18 | savaticus | Asterisk Manager Interface |
| 16:44.19 | [TK]D-Fender | ~wikis |
| 16:44.21 | jbot | from memory, wikis is http://www.voip-info.org |
| 16:44.21 | [TK]D-Fender | ~book |
| 16:44.25 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:44.34 | skirmisha | yes this is what i thought |
| 16:44.45 | skirmisha | on which port was listening |
| 16:44.47 | VJFROMGT | anyone familiar with iax2 ? I have 2 servers connected by iax2 , calls from a > b are ok but b > a drops after about 1 min |
| 16:44.51 | savaticus | 5038 |
| 16:44.53 | [TK]D-Fender | skirmisha: go READ |
| 16:44.57 | savaticus | lol |
| 16:45.07 | savaticus | RTFM rofl |
| 16:45.38 | [TK]D-Fender | savaticus: I figured I'd spare the expletives and at least just link the resources :) |
| 16:45.49 | savaticus | yeah sorry |
| 16:45.59 | savaticus | I didnt swear though |
| 16:46.09 | savaticus | my F stands for Freaking |
| 16:46.12 | savaticus | lol |
| 16:47.50 | savaticus | I love my IRC bots, I manage a net of 20 ast boxes and the bots really simplify things, I can have them tail logs to the chan, report call counts, and alert about important events |
| 16:48.18 | savaticus | its only about 600 lines of code as well. |
| 16:48.30 | [TK]D-Fender | savaticus: Sure, and the other letters really mean completely different things too ;) |
| 16:48.47 | DarKnesS_WolF | VJFROMGT: drops ? |
| 16:48.48 | savaticus | nope still Read THe Manual |
| 16:48.48 | DarKnesS_WolF | what do u ean ? |
| 16:48.50 | DarKnesS_WolF | mean * |
| 16:48.58 | VJFROMGT | call drops |
| 16:49.35 | savaticus | VJ: ie loses audio or call disconnects> |
| 16:49.36 | [TK]D-Fender | VJFROMGT: You have provided very poor details, no configs or CLI output, NOTHING. |
| 16:49.48 | VJFROMGT | disconnect |
| 16:49.56 | [TK]D-Fender | VJFROMGT: God helps those who help themselves... we are even LESS forgiving. |
| 16:49.59 | savaticus | and the cli ouput at point of drop? |
| 16:50.04 | VJFROMGT | i was not able to see anything in cli |
| 16:50.10 | savaticus | ? |
| 16:50.21 | savaticus | not even that the call ended? |
| 16:50.23 | VJFROMGT | cli was not showing anything weird,, |
| 16:50.28 | VJFROMGT | i will paste cli brb |
| 16:51.12 | DarKnesS_WolF | VJFROMGT: use pastebin.ca |
| 16:54.58 | VJFROMGT | Dark ,, http://pastebin.ca/495195 |
| 16:59.18 | DarKnesS_WolF | VJFROMGT: hmmm |
| 16:59.21 | DarKnesS_WolF | paste ur iax.conf ? |
| 16:59.35 | VJFROMGT | 1 sec |
| 16:59.57 | boch | what hw do you recommend for 50-100 simultaneous calls ? should i get a core 2 duo or maybe an amd64 ? |
| 17:01.24 | VJFROMGT | here you go dark http://pastebin.ca/495204 |
| 17:02.08 | [TK]D-Fender | boch: Depends what kind of work * actually has to do. |
| 17:02.52 | boch | only calls, it would be an interview system |
| 17:03.02 | boch | with transcoding |
| 17:03.18 | Strom_C | which codecs? |
| 17:03.33 | boch | gsm -> g729 |
| 17:03.37 | Strom_C | ew |
| 17:03.47 | boch | i cant find a softphone runing g729, do you ? |
| 17:03.51 | Strom_C | that'll sound like total crap, but whatever |
| 17:04.12 | boch | ulaw g729, dont know, but it wont be g729 native |
| 17:04.19 | [TK]D-Fender | boch: Why softphones, and yes, X-Pro / eyeBeam do G.729 just fine |
| 17:04.31 | boch | for free ? |
| 17:04.43 | Strom_C | get the digium tc400b, and any good server-class machine with a fast processor and a decent amount of RAM will do fine |
| 17:04.50 | [TK]D-Fender | boch: See, now you're just WHINING :) |
| 17:05.17 | [TK]D-Fender | boch: * is NOT a "free-ride". VoIP is not a "free-ride" |
| 17:05.25 | boch | hehe |
| 17:05.28 | [TK]D-Fender | boch: There is a cost to certain things. |
| 17:06.01 | [TK]D-Fender | boch: And for that many simultaneous calls I WOULD recommend the transcoder card. |
| 17:06.33 | boch | ok thanks dudes |
| 17:08.12 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 17:08.14 | zeeesh | hi |
| 17:11.11 | VJFROMGT | Darkness Source server http://pastebin.ca/495218 destination server http://pastebin.ca/495221 |
| 17:15.49 | tuan_modulis | is there a way to dynamically change the timeout of a call initiated by the Dial app? |
| 17:15.58 | tuan_modulis | like during the call |
| 17:16.30 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 17:17.59 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 17:18.26 | [TK]D-Fender | tuan_modulis: Nope. |
| 17:18.37 | *** join/#asterisk punani (n=m@87.127.7.210) |
| 17:19.59 | tuan_modulis | oh well :( |
| 17:22.04 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 17:23.13 | stoffell | [TK]D-Fender, i'm outta here, just wanted to let you know the app_page works perfect with the polycom's for paging. cheers ;) |
| 17:24.09 | [TK]D-Fender | stoffell: np |
| 17:26.32 | flujan | hi guys... I am using asterisk with X-lite... but it is not working with NAT. When I put the nat=yes on the sip.conf file, the softphone register... but I have no audio from both sides of the line. |
| 17:26.48 | flujan | I research a bit about NAT and there is a lot of configurations... |
| 17:27.01 | flujan | did you guys could point me a specific documentation that works? |
| 17:27.13 | [TK]D-Fender | ~sipnat |
| 17:27.23 | jbot | i heard sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:27.34 | [TK]D-Fender | flujan: Many things need to be done depending on WHO is behind NAT |
| 17:27.49 | LeddyHM | nats are bad, mmmmkay |
| 17:27.58 | *** join/#asterisk shapr (i=shae@cs6625112-51.bham.res.rr.com) |
| 17:28.20 | flujan | [TK]D-Fender, the server with asterisk is behind NAT and the softphone is also behind a NAT. |
| 17:28.36 | flujan | both sides have a router. |
| 17:28.37 | Qwell[] | [TK]D-Fender: hey |
| 17:28.49 | [TK]D-Fender | Qwell[]: y0 |
| 17:29.09 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 17:29.13 | Qwell[] | [TK]D-Fender: polycom.xml - what happens if you omit voIpProt.server.1.address? |
| 17:29.14 | shapr | Hi, I've started asterisk with debug, and I get REGISTER attempt 1 to username@hostname.com \n [May 18 12:27:32] NOTICE[4472]: chan_sip.c:12157 handle_response_peerpoke: Peer 'hostname' is now Reachable. (166ms / 2000ms). Does that mean I should be able to call through asterisk now? |
| 17:29.19 | *** join/#asterisk hongyee (n=hongyee@124.82.94.127) |
| 17:29.26 | shido6 | :) |
| 17:30.08 | [TK]D-Fender | flujan: you need to specify under [general] "nat=yes" , "canreinvite=no" , "localnet=" , "externip=" (or EXTERNHOST + EXTERNREFRESH). For the phone's entry you'll just need "nat=yes" |
| 17:30.39 | shapr | Also, can someone suggest softphones that should work with asterisk? Hopefully something that's already a debian/unstable package? |
| 17:30.42 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:30.45 | [TK]D-Fender | Qwell[]: That is not a standard polycom config file name format, and it would depend wherelse a server may be defined |
| 17:30.46 | Qwell[] | and reg.1.server.1.address |
| 17:31.13 | [TK]D-Fender | Qwell[]: I usually specify my server in sip.cf and jsut specif the user/pass in the "phoneXXX.cfg" for each phone |
| 17:31.26 | Qwell[] | what happens if you omit it though? |
| 17:31.40 | flujan | thanks [TK]D-Fender I will check this options now... :) |
| 17:32.12 | Sweeper | hey, what are those short sms-only numbers called that all those ringtones/googlesearch/whatever services get called? |
| 17:32.18 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 17:32.21 | Sweeper | and how do you go about getting one? |
| 17:33.50 | [[blah]asfd | I am trying to set queue_prio in a macro, but it does not seem to be working. all of my calls coming in do not get the prio set unless i do it right from the line before the queue it's self. here is how i do it: http://pastebin.ca/495252. can anyone suggest a better way to make this work? or point out what I am doing wrong? |
| 17:33.59 | *** join/#asterisk jeffik (n=Valued@CPE00e04cb010bc-CM000f9f1e3842.cpe.net.cable.rogers.com) |
| 17:35.43 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:35.55 | *** join/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 17:36.14 | *** join/#asterisk saftsack (n=oliver@p54a7f27e.dip.t-dialin.net) |
| 17:36.53 | saftsack | hi, if i originate a call over the manager api i get no voice in this call. is this a common bug/configuration error? |
| 17:37.06 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:37.21 | Sweeper | saftsack: no |
| 17:37.39 | [TK]D-Fender | QwellAs in your phone's configs (however many you have) don't specify ANY server amongst them? |
| 17:37.40 | savaticus | it means you prolly didnt properly spec the channels |
| 17:38.11 | saftsack | <PROTECTED> |
| 17:38.13 | Qwell[] | [TK]D-Fender: yeah |
| 17:38.52 | [TK]D-Fender | Qwell[]: Next I presum that it'll fallback to DHCP supplied server (No, NOT Opt 66), and barring even that... Dunno :) |
| 17:39.08 | Sweeper | saftsack: what is this "Local" business? |
| 17:39.20 | Qwell[] | dhcp supplied server? |
| 17:39.26 | [TK]D-Fender | Qwell[]: Figuring that since we're talking about config files, that the phone itself is not manually overrided on the server |
| 17:39.28 | blitzrage | Local is a magical channel that solves all problems ;) |
| 17:39.37 | Sweeper | blitzrage: ;) |
| 17:39.44 | [TK]D-Fender | Qwell[]: Yes, you can pass all sorts of crazy infor through DHCP :) |
| 17:40.44 | [TK]D-Fender | blitzrage: Except recursion ;) |
| 17:40.44 | saftsack | got it from here: |
| 17:40.44 | saftsack | http://asterisk-java.org/ |
| 17:41.07 | savaticus | hey guys when you have a sip behind a firewall but in the DMZ, it registers but the firewall nats the outbound, I set port=5060 in sip.conf but it is ignored |
| 17:41.19 | savaticus | how do you overide the device? |
| 17:41.27 | *** join/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 17:41.42 | savaticus | what the device is reporting anyway |
| 17:42.01 | Sweeper | saftsack: never used it. can you do a regular call to it? |
| 17:42.30 | saftsack | yes. same result |
| 17:42.45 | savaticus | Local/ is used to send a call out through your root context. |
| 17:43.06 | savaticus | hence the @to jump context from there. |
| 17:43.27 | savaticus | the main benefit being you get Dial()'s return codes |
| 17:43.47 | savaticus | whereas in Goto you wouldnt |
| 17:43.48 | saftsack | <PROTECTED> |
| 17:44.07 | [TK]D-Fender | I'm doing a YUM update just now and have like 100's of packages to update (running CentOS 4.4 here). I've been running it regularly, and had added RPMForge to the list (using EL4 as repo). |
| 17:44.09 | ChrisHardie | I'm back hoping someone might have some insight on a particular queue config I'm trying to accomplish. |
| 17:44.43 | saftsack | the other phone rings, i am able to hear a ring tone, BUT if the other party releases the hook i can hear nothing :( |
| 17:44.48 | [TK]D-Fender | I have since disabled it for fear that it might pollute things past the point where I needed it for a handful of packages, but want to be sure of whats going on before I risk nuking my box. |
| 17:45.22 | [TK]D-Fender | 414 to be precise |
| 17:45.42 | ChrisHardie | The first comment on http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf from John Lawler sums up the trouble I'm encountering. |
| 17:45.55 | ChrisHardie | Has anyone had any luck making this kind of "circular call distribution" work? |
| 17:47.37 | savaticus | I love yum update 'specialy when it grabs a new kernal and horks zaptel lol. |
| 17:51.07 | ChrisHardie | Any thoughts on whether the only way to solve the above with multiple/cascading queues? |
| 17:52.21 | saftsack | hasnt anybody an idea? |
| 17:52.25 | *** join/#asterisk onecentld (n=sales@ns2.tube-e.net) |
| 17:52.54 | onecentld | anymore having problems with dtmf with metaswitch |
| 17:53.57 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:53.57 | blitzrage | with 1.4? Maybe try dtmfmode=rfc2833compensate |
| 17:54.22 | onecentld | tried it but it did not help |
| 17:54.36 | onecentld | calls go through no problems |
| 17:54.43 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:54.59 | onecentld | but if you get to an ivr it doesn't accept any keys |
| 17:55.42 | shapr | Yay, I got kiax working with asterisk, I am now a happy asterisk user! w00! |
| 17:56.40 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:56.44 | *** part/#asterisk shapr (i=shae@cs6625112-51.bham.res.rr.com) |
| 17:57.41 | Sweeper | boing |
| 17:57.59 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 17:58.08 | coppice | arbus |
| 17:58.24 | *** part/#asterisk tbic (n=tbic@207.148.218.162) |
| 18:00.01 | anonymouz666 | fokker 100 |
| 18:01.28 | macTijn | "boeing", "airbus" |
| 18:01.40 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 18:01.42 | harlequin516 | Okay so why Asterisk over Sipx? |
| 18:01.48 | macTijn | (and the nicest fokker is the Fokker 70 :) |
| 18:02.07 | Katty | afternoon |
| 18:02.37 | *** part/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 18:02.49 | harlequin516 | Anyone know what features compare and don't? |
| 18:02.54 | harlequin516 | Hi Katty |
| 18:04.17 | *** join/#asterisk lorinc (n=ang@pool-5422.adsl.interware.hu) |
| 18:05.16 | Sweeper | and how do you go about getting one? |
| 18:05.32 | Sweeper | hey, what are those short sms-only numbers called that all those ringtones/googlesearch/whatever services get called? |
| 18:06.07 | harlequin516 | I can't get sound to work with Kiax. I think it requires oss, which I have just tried compiling into my kernel. |
| 18:06.52 | harlequin516 | Sweeper: Yeah I am curious too. I want to know how they can bill for their service. Where are the specs for these services available? |
| 18:07.09 | coppice | I don't really understand how small makers like Fokker can survive,,,,oh, wait..... |
| 18:07.33 | harlequin516 | I also can't belive that people pay $2.99 for stupid crap like that. |
| 18:07.41 | Qwell[] | harlequin516: $2.99 hell |
| 18:07.46 | Qwell[] | more like $19.99/month |
| 18:07.59 | harlequin516 | Monthly? |
| 18:08.02 | [TK]D-Fender | harlequin516: sipx only talks SIP, and doesnt' have a fraction of the call control that * does. No AGI, AMI, don't believe they have queues, and the list goes on and on. |
| 18:08.21 | Qwell[] | yes, if you send a message to one of those "ringtone" things...you are signed up for a monthly service :) |
| 18:08.25 | harlequin516 | Oh, thanks [TK]D-Fender |
| 18:08.30 | Qwell[] | and it is NOT cheap |
| 18:08.38 | Sweeper | I just want to know how they get those short sms numbers |
| 18:08.41 | Qwell[] | there is a joke one, that's like $1.99 PER DAY |
| 18:08.42 | harlequin516 | Scamilicious... |
| 18:09.05 | harlequin516 | Yeah I want to be in that business. |
| 18:09.11 | Qwell[] | No you don't |
| 18:09.27 | Qwell[] | They have like a 90% chargeback rate :p |
| 18:09.32 | [TK]D-Fender | harlequin516: SIPX also doesn't directly support hardware interfaces. Oh you want to use your LINES with it? Buy a SIP gateway and forget PCI cards. Immediately that means inbound fax = super trouble under their scenario |
| 18:09.36 | harlequin516 | I want to be rich biootch |
| 18:09.56 | harlequin516 | Qwell[]: Ewww that's bad, but I bet they still maker a killing |
| 18:10.23 | harlequin516 | Oh I guess its spelled b-e-e-o-t-c-h. |
| 18:11.21 | harlequin516 | [TK]D-Fender: Okay Okay.. I have regiained my one true faith in Asterisk... |
| 18:12.45 | [TK]D-Fender | harlequin516: Depends on your needs :) |
| 18:12.45 | irule | is it possible to create a password protected context? to avoid Authenticate() on erach exten |
| 18:12.45 | [TK]D-Fender | harlequin516: If you don't have any... SipX may be for you! ;) |
| 18:12.45 | harlequin516 | Now if only I could find a open source iax java phone. |
| 18:12.46 | [TK]D-Fender | irule: Use a catchall in one context that sees if it matches another. if so, authenticate, THEN forward it on, otherwise forward to a 3rd context |
| 18:12.46 | harlequin516 | Oh I am writing an intereactive agi script. |
| 18:13.50 | [TK]D-Fender | harlequin516: http://www.hem.za.org/jiaxclient/ |
| 18:14.16 | *** join/#asterisk Luxvero (i=c9268edb@gateway/web/cgi-irc/ircatwork.com/x-c48735e1dd122599) |
| 18:14.50 | harlequin516 | Yeah i read that one, but it appears that jiaxclient uses jni to iaxclient library |
| 18:14.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:15.06 | harlequin516 | I don't know why you just can't implement IAX in java. |
| 18:17.24 | harlequin516 | Also their demo won't run without root privilege, as it tries to install the iaxclient shared library. |
| 18:17.33 | Katty | for the autoanswer /paging thingy, does that use a meetme conference room? |
| 18:17.36 | coppice | nobody is stopping you :-) |
| 18:17.38 | Katty | or is there a good page that shows how to set that up |
| 18:18.27 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 18:19.54 | Luxvero | Hi everyone... I am reading some manuals, tutorials and etc, they explain the mechanics of * but I can't find a nice reference to understand most weird syntax like : something => _.123,.3,$hdgasd(sdf){#sdfsdjf}/_*78346238#234234${$2343}(SIP/111,25,Ttr) lol.... does anyone know of some syntax reference ? |
| 18:22.16 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:23.12 | [TK]D-Fender | Katty: Yes, it uses MeetMe. Didn't you already get this working before? |
| 18:24.18 | [TK]D-Fender | Luxvero: I'm guessing thats a scrambled sample that gets "decoded" in run-time to hid what its really doing from you. Or its just plain wrong/corrupted |
| 18:25.14 | Luxvero | oh no, sorry, i didn't mention i just typed random text as example |
| 18:25.39 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 18:26.18 | Zeeek | Katty |
| 18:26.23 | Zeeek | you have a great voice! |
| 18:26.39 | Zeeek | I need to leave now. |
| 18:26.45 | Zeeek | goodbye cruel world |
| 18:26.57 | Katty | oh |
| 18:26.59 | Katty | bye now |
| 18:27.06 | Katty | [TK]D-Fender: i did. |
| 18:27.09 | Zeeek | dinner time, bedtime |
| 18:27.09 | Katty | [TK]D-Fender: but i wiped that server. |
| 18:27.15 | Zeeek | bye |
| 18:27.16 | Katty | [TK]D-Fender: and only remeber a bit. |
| 18:27.18 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:27.22 | Katty | [TK]D-Fender: also, i didn't document anything /sob |
| 18:27.41 | [TK]D-Fender | Katty: And didn't back up your configs? |
| 18:27.51 | Luxvero | [TK]D-Fender, I was not asking for a particular help, just asking if there's some decent manual that will explain how to create these 'paterns'. |
| 18:27.56 | Katty | [TK]D-Fender: they told me to stop working on the server at that time. |
| 18:28.08 | Katty | [TK]D-Fender: *sigh* |
| 18:28.09 | Katty | [TK]D-Fender: no. |
| 18:28.13 | PioneerVM2 | anyone know how to invoke voice mail from AGI? |
| 18:28.33 | [TK]D-Fender | Luxvero: They are not "real" they are either corrupted or encoded by some other process. |
| 18:29.13 | Luxvero | yes, I just typed random text as example |
| 18:29.33 | [TK]D-Fender | Katty: NOT good. You don't just wipe a system like that and start from scratch for * configs. back upt extensions, voicemail,sip, zaptel, zapata at least... |
| 18:30.01 | [TK]D-Fender | Katty: You'll need to redo you SIP heder change, and make a really log Page call like before. |
| 18:30.13 | [TK]D-Fender | Katty: Time to start reinventing the wheel! ;) |
| 18:32.17 | Katty | [TK]D-Fender: well i did document the phone side of things |
| 18:32.30 | Katty | [TK]D-Fender: but not what was done in meetme, or extensions |
| 18:32.31 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 18:32.40 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:32.48 | Katty | [TK]D-Fender: the ftp part thingy |
| 18:32.55 | [TK]D-Fender | Katty: So you're all upgraded to SIP 2.1.1 on all your phones and nice & neatly reprovisioned? |
| 18:33.03 | Katty | yup |
| 18:33.26 | [TK]D-Fender | Katty: cool.... so just a few more hours to go! |
| 18:33.45 | Katty | i dunno |
| 18:33.47 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 18:33.54 | Katty | i'm just not /in the mood/ to do anything overly complicated right now. |
| 18:34.10 | Katty | especially after what a certain someone said over lunch )= |
| 18:35.23 | Katty | [TK]D-Fender: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config <- that the right page? |
| 18:36.10 | [TK]D-Fender | Katty: yup, one of them. merge in the one for "page" and that should cover it |
| 18:36.59 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 18:37.24 | Katty | [TK]D-Fender: if i recall correctly... i just added the sipheader. |
| 18:37.42 | Katty | and then dialed all the SIP/extensions i wanted. |
| 18:38.36 | Katty | but that wouldn't be meetme. |
| 18:39.00 | syzygyBSD | morning katty |
| 18:39.04 | Katty | yeah, too complicated for now. |
| 18:39.06 | Katty | hey syzygyBSD (= |
| 18:39.50 | syzygyBSD | what is complicated? |
| 18:39.59 | Katty | make a group of phones do auto answer. |
| 18:40.09 | syzygyBSD | what kinda phones? |
| 18:40.16 | Katty | polycom 501s |
| 18:40.26 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 18:41.13 | syzygyBSD | configured using tftp? |
| 18:41.26 | [TK]D-Fender | Katty: No, you don't need to configure MeetmMe for this, you just need a Zaptel timing source handy and it gets used transparently to you |
| 18:41.54 | Katty | syzygyBSD: well, ftp, but yeah. i dont' wanna get into this right now tho. |
| 18:42.24 | syzygyBSD | its friday, you can fuck everything up for the whole weekend! |
| 18:42.31 | bkruse | :D |
| 18:42.51 | PioneerVM2 | anyone know how to call VoiceMail function from AGI? |
| 18:43.03 | syzygyBSD | exten => 830,1,SIPAddHeader(Alert-Info: RANR) |
| 18:43.09 | syzygyBSD | like that katty? |
| 18:43.31 | Mad|Cow | anyone know how to configure meetme so it uses mysql rather than the text file? I tried adding a line to the extconfig file... but I must not have the syntax correct. Any ideas? |
| 18:43.51 | Katty | syzygyBSD: yeah, like that. but it's gotta use meetme somehow so that multiple people can be called and hear, etc. |
| 18:44.18 | syzygyBSD | :) I can do it |
| 18:44.29 | Katty | syzygyBSD: i'm sure you can, dear (= |
| 18:44.31 | syzygyBSD | macro-diallocal |
| 18:44.58 | syzygyBSD | wait, just an intercom? |
| 18:45.11 | Katty | yep |
| 18:45.13 | Katty | that's it |
| 18:45.21 | Katty | dail ext, 5 phones pick up and lbahlbahlbha |
| 18:45.27 | Katty | hangup |
| 18:49.41 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 18:49.41 | The_X | hi folks, is there any free software/module I can use to track some call stats? |
| 18:49.54 | Katty | The_X: what sort of call stats? |
| 18:50.02 | The_X | how many inbound/outbound calls |
| 18:50.05 | Katty | The_X: sure. |
| 18:50.06 | The_X | call waiting info |
| 18:50.08 | The_X | queues |
| 18:50.12 | Katty | The_X: look for the master.csv file |
| 18:50.16 | Katty | The_X: that has ALL the info. |
| 18:50.28 | Katty | The_X: from there, you can put it into other software. |
| 18:50.39 | Katty | The_X: make columns and pretty reports. |
| 18:50.43 | Katty | The_X: like in open office |
| 18:50.53 | Katty | The_X: or excel...or even access (gag) if you're feeling brave. |
| 18:51.07 | Katty | The_X: OR you could get really fance and have that dumped into sql, then write a php thingy around it. |
| 18:51.16 | Katty | The_X: and do lots of queries and generated pages. |
| 18:51.27 | Katty | The_X: select * where date=this |
| 18:52.14 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-b4b450ea8ec0aa19) |
| 18:52.16 | The_X | thanks |
| 18:53.06 | Katty | The_X: there might be some mod/plugin/software somewhere...but, meh. |
| 18:53.07 | Katty | The_X: why setup a new software program when you can just import/export |
| 18:56.11 | The_X | I don't see any master.csv file |
| 18:56.53 | kaldemar | The_X: /var/log/asterisk/cdr-csv/Master.csv |
| 18:57.03 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 18:57.10 | *** join/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com) |
| 18:57.22 | psi0n | hey guys |
| 18:58.02 | Katty | hewwo. |
| 18:58.27 | psi0n | any asterisk-gurus who have experience with dtmf on cell phones? |
| 18:59.01 | *** join/#asterisk watchy (n=watchy@spectra3.spectra-tech.net) |
| 18:59.05 | Strom_M | well, i have experience with dtmf on mobile phones, but im not a guru per se ;) |
| 18:59.23 | watchy | anyone here ever config a sangoma a200? |
| 18:59.52 | *** join/#asterisk JerJer[mobile] (n=jj@45.110.5.229) |
| 19:00.31 | psi0n | well my problem is whatever i try i cant use IVR thru cell phones |
| 19:00.48 | psi0n | since it doesnt recognize the dial commands |
| 19:01.15 | Strom_M | calling from moblie -> PSTN -> (trunk) -> Asterisk ? |
| 19:01.32 | psi0n | yup |
| 19:01.39 | *** join/#asterisk matsk (n=mk@h110n2fls32o882.telia.com) |
| 19:01.40 | JerJer[mobile] | so i am doing SIP/${EXTEN}@domain_name.com - domain_name.com has a valid DNS SRV record, but Asterisk refuses to cooperate |
| 19:01.54 | JerJer[mobile] | i get a no such host message |
| 19:01.59 | Katty | JerJer[mobile]: how about force feed it an ip address |
| 19:02.09 | JerJer[mobile] | we want the DNS SRV |
| 19:02.12 | Katty | k |
| 19:02.43 | JerJer[mobile] | perhaps do I need to use a peer to make dns srv query get fired?? |
| 19:02.59 | Katty | i would walk lightly and carry a big stick. |
| 19:03.36 | watchy | anyone here ever config a sangoma a200? |
| 19:03.38 | *** join/#asterisk tonycarstens (n=oper@206.135.21.162) |
| 19:04.47 | tonycarstens | can anyone help me with automon? |
| 19:04.52 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 19:05.28 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 19:05.32 | kaldemar | JerJer[mobile]: do you have srvlookup=yes in sip.conf? |
| 19:05.53 | syzygyBSD | JerJer[mobile]: ^ |
| 19:06.01 | Qwell[] | JerJer[mobile]: ^^ |
| 19:06.48 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:09.28 | JerJer[mobile] | kaldemar: yes - in general |
| 19:09.40 | psi0n | is DTMF on cell phones a common problem? |
| 19:11.47 | Strom_M | psi0n: what kind of trunk is being used to connect the asterisk box with the PSTN? |
| 19:11.48 | psi0n | SIP |
| 19:11.48 | Strom_M | what DTMF mode are you using on the sip trunk? |
| 19:12.07 | *** join/#asterisk Chiko_Medina (n=kristian@server.oficinasdigitales.com) |
| 19:12.18 | psi0n | info, but i've tried all of em |
| 19:12.25 | Strom_M | which codec? |
| 19:12.34 | Strom_M | does it work with wireline phones? |
| 19:12.42 | Strom_M | does it work with a different kind of trunking method? |
| 19:12.50 | psi0n | alaw, which seems to be the only codec supported by the cell phones |
| 19:12.53 | Strom_M | what is your provider set to? |
| 19:13.47 | psi0n | the IVR works fine from another VoIP phone, altho i dont have the opportunity to try it with an ordinary phone |
| 19:14.03 | Strom_M | you dont have a wireline phone handy? |
| 19:14.13 | Strom_M | that's not a good idea ;) |
| 19:14.24 | psi0n | no, only VoIP and cell phones where im at currently |
| 19:14.29 | Strom_M | LAME |
| 19:14.59 | Strom_M | can you use the voip phone to hit the PSTN and then come back in over the same trunk as your cellphone is? |
| 19:15.27 | Strom_M | also, is the asterisk box behind NAT and/or a firewall? |
| 19:16.05 | *** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:16.19 | psi0n | im not sure i understand the 1st question. as to the other, the PBX has its own public IP |
| 19:16.28 | Strom_M | well |
| 19:16.44 | Strom_M | this other voip phone you're using - how is it connecting to the asterisk box? |
| 19:17.42 | psi0n | my own voip phone connects to my own pbx, however the pbx in question is another pbx located elsewhere |
| 19:18.36 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 19:18.38 | psi0n | so from my voip phone -> my pbx -> trunk -> ... -> trunk -> other pbx |
| 19:19.21 | Strom_M | hm |
| 19:19.21 | Strom_M | and that works? |
| 19:19.21 | Strom_M | same SIP trunk going into the PBX in question? |
| 19:19.21 | psi0n | works just fine |
| 19:19.28 | psi0n | yup |
| 19:19.43 | Strom_M | what does it sound like if you call your desk phone directly from your cellphone and dial at yourself from the cellphone? |
| 19:20.45 | Katty | is cron the prefered way to schedule something? |
| 19:21.01 | Katty | like copy a to b at c:de |
| 19:21.12 | Qwell[] | Katty: yeah |
| 19:21.31 | psi0n | hmm, the tones sound just fine, some abit longer than the rest tho |
| 19:21.31 | Katty | Qwell[]: can it do execute this /script/ at this time? |
| 19:21.32 | Katty | Qwell[]: or just individual commands |
| 19:21.40 | Qwell[] | you can execute scripts |
| 19:21.41 | Strom_M | psi0n: what happens if you try a different trunking method to that PBX? |
| 19:21.49 | Katty | mm, goodly. |
| 19:22.11 | psi0n | i dont have that option. we only have a sip trunk towards the provider |
| 19:23.27 | psi0n | do you think the problem could be the SIP trunk? |
| 19:23.37 | Strom_M | i dont know. thats why im having you test things so we can narrow it down |
| 19:25.21 | Katty | Qwell[]: so lets say on my server, i want to back up /usr/src and /etc/asterisk, and a few other lil conf files. |
| 19:25.31 | Katty | Qwell[]: this thing technically has a tape drive, but i hate tapes. |
| 19:25.41 | Katty | Qwell[]: i was thinking about mapping a network drive, copying stuff over, and then dismounting it |
| 19:26.02 | Katty | Qwell[]: do you see any potential problems with this theory, or can you think of a better way |
| 19:28.45 | Qwell[] | not really |
| 19:29.56 | Katty | kk |
| 19:31.11 | Katty | Qwell[]: does linux handle most external hard drives okay? |
| 19:31.24 | Qwell[] | yeah |
| 19:31.39 | Katty | we've got a maxtor one touch that's coming back from rma (for the second time now) |
| 19:31.49 | Katty | i might try using that. |
| 19:31.56 | Strom_M | and you're willing to back things up on that? :) |
| 19:32.06 | Katty | gets it off site. |
| 19:32.16 | Katty | i don't have anything else at the moment. |
| 19:32.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:32.47 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:35.01 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 19:35.41 | psi0n | probably a dumb question, but if i have phones that dont support IAX, but connect them with SIP to a PBX, which in turn uses an IAX trunk to the provider, will that work? |
| 19:35.47 | nowork | hello, any one know any good company provide SIP softphone with my own logo? thanks/ or maybe some free? |
| 19:36.23 | sandorp | x-lite, maybe? |
| 19:36.53 | sandorp | nowork: look at www.counterpath.com |
| 19:37.48 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 19:38.43 | Katty | i hate when an isp tries to tell me bs. |
| 19:38.50 | Katty | expecting me to a dumb little girl and not know any better. |
| 19:39.39 | Qwell[] | Katty: no, that's just truly what they believe |
| 19:39.43 | Strom_M | psi0n: yes |
| 19:39.43 | Qwell[] | they're the idiots :) |
| 19:39.44 | *** join/#asterisk Jabroni (n=asskick@red-corp-201.143.40.212.telnor.net) |
| 19:40.03 | Qwell[] | "Your modem isn't synced? Try rebooting your computer." |
| 19:40.13 | Qwell[] | Katty: ^^ what they really think is the problem |
| 19:40.53 | Katty | or course, this is the same telco that wants to charge me twice as much for a pri than a t1 |
| 19:40.56 | Jabroni | Good afternoon everyone... i got a quick question.. anyone knows if there is a plan to support the new chan_mobile driver on asterisk 1.2.x ? |
| 19:41.03 | Qwell[] | that, or they give me the same line, because I sound like a girl on the phone or something :P |
| 19:41.03 | Katty | why, i haven't the slightest idea. |
| 19:41.05 | Qwell[] | Jabroni: no |
| 19:41.06 | psi0n | Strom_M: does it have any negative effects? like extra overhead, like when converting codecs? |
| 19:41.25 | Qwell[] | Jabroni: nor 1.4.x |
| 19:41.31 | Strom_M | psi0n: not really |
| 19:41.46 | Strom_M | protocol translation isnt as intensive as transcoding |
| 19:41.52 | Jabroni | i saw it was already added to the svn branch as addon driver |
| 19:42.00 | Qwell[] | Jabroni: to trunk only |
| 19:42.11 | psi0n | Strom_M: would you recommend it instead of pure SIP? |
| 19:42.21 | tonycarstens | when making an outbound call through zap do i have to configure * to know to use the next line if the first one is in use? |
| 19:42.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:42.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:42.35 | Qwell[] | tonycarstens: setup zap groups |
| 19:42.43 | Qwell[] | then just dial the group instead of a single line |
| 19:43.17 | onecentld | Anybody have alot of experience with VoIPSwitch??? |
| 19:44.14 | mog | <PROTECTED> |
| 19:46.11 | Jabroni | Qwell[] still that chan driver could still get backported to 1.2.x right ? (considering someone gets the time to backport the source to be 1.2.x compatible) |
| 19:46.32 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 19:46.59 | Qwell[] | Jabroni: somebody could, sure, but it won't be in any releases |
| 19:48.17 | Katty | mog: (= |
| 19:50.04 | mog | ? |
| 19:53.27 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 19:53.58 | *** part/#asterisk flujan (n=flujan@200.160.115.20) |
| 19:54.11 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 19:54.36 | tonycarstens | qwell: i put the four extensions under a group = 1 |
| 19:54.50 | tonycarstens | would this be the right dial() string |
| 19:54.55 | tonycarstens | Dial(Zap/g1/${EXTEN},120,r) |
| 19:55.02 | *** join/#asterisk ixela (i=ixela@nat/digium/x-d478c9c849fb5b5b) |
| 19:55.18 | Qwell[] | tonycarstens: yep or G1, depending on which order you want it to check the channels |
| 19:55.33 | Qwell[] | one would go from 1-4, the other would go from 4-1 |
| 19:56.02 | tonycarstens | i would want it to go from 1 to 2 to 3 to 4 |
| 19:56.07 | tonycarstens | so g1 is correct right? |
| 19:56.14 | Qwell[] | I don't know :d |
| 19:56.19 | Qwell[] | I can never remember which is which |
| 19:56.56 | Strom_M | g1 is correct |
| 19:57.04 | Strom_M | but don't use the "r" flag |
| 19:57.27 | tonycarstens | what does the 'r' flag do? |
| 19:57.29 | Strom_M | not unless you have a REALLY REALLY REALLY GOOD REASON to do |
| 19:57.39 | Strom_M | it wipes out your call progres information |
| 19:57.44 | tonycarstens | ok |
| 19:57.51 | tonycarstens | thank you |
| 19:59.06 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 19:59.27 | joat | anyone have an opinion on chan_skype? |
| 19:59.31 | Strom_M | garbage |
| 19:59.41 | joat | anything better? |
| 19:59.44 | Strom_M | yeah |
| 19:59.46 | Strom_M | asterisk :) |
| 20:00.03 | joat | that's only half an answer |
| 20:00.24 | Strom_M | skype itself is utter crap. I wouldnt touch it with a ten foot insulated pole ;) |
| 20:00.36 | joat | ah (heh) |
| 20:01.23 | joat | i can't avoid the issue though (i.e., can't talk the customer out of it) |
| 20:01.32 | joat | however, i'm hoping to bridge the two |
| 20:01.36 | *** join/#asterisk sasch (n=sasch@83.225.137.26) |
| 20:02.39 | Strom_M | well, skype is completely proprietary |
| 20:02.49 | Strom_M | anything you do is going to be kludge-o-riffic |
| 20:04.56 | joat | true... but so's the backend that i'm building... basically, he wanted an ivr to monitor various system/network stats |
| 20:05.30 | Strom_M | and this uses skype why? |
| 20:05.46 | joat | he likes his skype wifi phone |
| 20:06.05 | joat | yeah, it's a toy |
| 20:06.08 | joat | but it's his toy |
| 20:06.18 | joat | and he's paying for it |
| 20:06.47 | *** part/#asterisk TechBlazer (n=tim@70.88.27.130) |
| 20:07.48 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 20:07.51 | robin_sz | hi |
| 20:08.12 | robin_sz | so, one of my phoens has blocked again |
| 20:08.26 | robin_sz | cant make outgoing or rx incoming calls |
| 20:08.35 | robin_sz | have rebooted phone, no change |
| 20:09.03 | robin_sz | so, before I reboot * (which always cures it) what to check to identify exactly what the problem is? |
| 20:10.30 | tonycarstens | still having some probs with zap group |
| 20:10.37 | anonymouz666 | enough for today |
| 20:10.40 | tonycarstens | in zapata.conf i have |
| 20:10.41 | anonymouz666 | time to beer |
| 20:10.45 | tonycarstens | group = 1 |
| 20:10.53 | tonycarstens | channel => 1-4 |
| 20:11.56 | Katty | byebye (= |
| 20:14.06 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:14.44 | psi0n | i just got back from the phone and it turns out our voip provider dont support iax.. |
| 20:15.05 | Mercestes | robin_sz, what does the CLI say?? |
| 20:15.25 | psi0n | this cell phone DTMF problem is driving me nuts |
| 20:17.53 | robin_sz | Mercestes, it says CLI> |
| 20:18.15 | Mercestes | and? |
| 20:18.27 | Mercestes | what does the CLI say when you try to dial the phone?? |
| 20:18.35 | robin_sz | ahh, wait. checking |
| 20:19.29 | robin_sz | <PROTECTED> |
| 20:19.42 | robin_sz | thats what it says |
| 20:19.48 | robin_sz | its worng, of course |
| 20:21.54 | robin_sz | and, sip show channels doesnt reveal an active SIP channel related to that phone |
| 20:22.24 | robin_sz | so, now what? |
| 20:22.52 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 20:24.22 | robin_sz | just reboot and hope it doesnt happen again for a few hours? |
| 20:25.00 | robin_sz | not sure, but I think it might happen when a call is transfered to another peer |
| 20:25.50 | sandorp | is there a way to make a remote caller sound louder? I'm using x-lite and I've turned up the volume on the PC but the remote side still sounds faint |
| 20:29.06 | deeperror | sandorp: internet connection? |
| 20:29.16 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 20:30.50 | sandorp | deeperror: the remote caller is regular phone line |
| 20:30.56 | robin_sz | Mercestes, so, clues? |
| 20:31.25 | deeperror | so pstn - FXO - SIP phones? |
| 20:31.27 | sandorp | yes |
| 20:31.39 | *** join/#asterisk Divious1 (i=chatzill@65.112.134.160) |
| 20:31.40 | deeperror | how does it sound between sip phones? |
| 20:31.45 | Divious1 | hello everyone |
| 20:31.45 | sandorp | great |
| 20:32.09 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:32.20 | Mercestes | robin_sz, where is your usage level set? |
| 20:32.43 | Divious1 | i have a question |
| 20:33.03 | *** join/#asterisk cnbrk (n=canberk@212.156.213.131) |
| 20:33.07 | cnbrk | hello!! |
| 20:33.25 | cnbrk | this is i think the first time i join this channel with my asterisk server working without any problems!! |
| 20:33.26 | Divious1 | I have a box with fedora core 1 which I can't update and I have asterisk 1.0 on it I want to update it, what files should I back up? |
| 20:33.54 | psi0n | gratz cnbrk |
| 20:33.59 | cnbrk | Divious1, your *.conf files i think |
| 20:34.22 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:34.45 | Divious1 | the ones in /etc/asterisk and /etc/zaptel.conf |
| 20:35.01 | Divious1 | I have some databases but I'm not sure where does would be located |
| 20:35.22 | deeperror | mysql? |
| 20:35.51 | robin_sz | Mercestes, dunno, but 1 is the correct level |
| 20:35.57 | *** join/#asterisk bbryant (i=brett@nat/digium/x-c1e65bfb43e8656d) |
| 20:36.06 | sandorp | Divious1: you could use "mysqldump --opt -u root database_name > db.dump" to backup the database |
| 20:36.26 | sandorp | well, dump the database and then save the dump file |
| 20:36.31 | cnbrk | asterisk is the best open source software i've ever seen once you get it to work! |
| 20:36.38 | Mercestes | robin_sz, Maybe you are using a state check and your state isn't being updated correctly?? |
| 20:36.44 | robin_sz | err |
| 20:36.48 | deeperror | you can also tar up /var/lib/mysql/* |
| 20:36.52 | Divious1 | yea sql sorry |
| 20:36.52 | robin_sz | a what? |
| 20:37.47 | cnbrk | by the way guys, if you need to get a voip gateway, go with linksys spa400, it rocks! |
| 20:37.53 | cnbrk | cheap price, high quality |
| 20:38.04 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 20:38.12 | *** part/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 20:38.41 | deeperror | sandorp: may need to check how your signalling is setup on that card |
| 20:38.47 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 20:39.42 | sandorp | deeperror: in zapata.conf signalling=fxs_ks |
| 20:40.18 | robin_sz | Mercestes, where wold \i check that ? the state check thing? |
| 20:40.27 | deeperror | sandorp: have you played around with rxgain settings? |
| 20:40.37 | deeperror | test that and see how that works? |
| 20:40.55 | sandorp | deeperror: nope, where is that set? |
| 20:41.08 | Mercestes | sip show states or something of that nature |
| 20:41.11 | deeperror | under your signalling |
| 20:41.27 | deeperror | before channel =>1 |
| 20:42.49 | sandorp | ok, it is set to 0.0 and I now see in the wiki that is can be adjusted up or down ... I assuume up means louder |
| 20:43.02 | robin_sz | Mercestes, I have sip show inuse |
| 20:43.14 | robin_sz | elaine 1 1 |
| 20:43.33 | robin_sz | in use, 1, limit 1 |
| 20:43.35 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 20:43.52 | robin_sz | thats under peer name |
| 20:43.59 | deeperror | sandorp: try 8.0 |
| 20:44.03 | robin_sz | but elaine 0 1 |
| 20:44.06 | robin_sz | under user name |
| 20:44.23 | robin_sz | so the PEER has an in use count, but the user does not |
| 20:45.05 | robin_sz | I think its related to blind transfers |
| 20:48.53 | Mercestes | robin_sz, could be. That seems to be your problem tho |
| 20:49.13 | robin_sz | so ... |
| 20:49.17 | robin_sz | what to do ? |
| 20:49.36 | robin_sz | just set a cron task to reboot * every few hours? |
| 20:50.10 | Mercestes | that's what I'd do |
| 20:51.57 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 20:52.55 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 20:53.50 | *** join/#asterisk thoughtpolice (n=austin@c75-111-146-192.plaicmtc01.tx.dh.suddenlink.net) |
| 20:55.36 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net) |
| 20:59.11 | *** join/#asterisk cervi2 (n=cervicek@rhlx09.hs-esslingen.de) |
| 21:01.25 | cervi2 | blitzrage, just want to thank you for support regarding my feature-proposal :-) |
| 21:02.53 | *** join/#asterisk WindBack (n=jorge@host73.201-253-149.telecom.net.ar) |
| 21:05.50 | WindBack | I have a question. If I want to connect asterisk with a DataBase... Which is the best solution??? I am trying to use AGI and Python.. Is this a good solution?? |
| 21:07.40 | deeperror | that would work |
| 21:08.06 | deeperror | i believe there are some database cmd's that can be used in the dialplan |
| 21:08.19 | deeperror | but may need installed or configured |
| 21:08.36 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-caebbe3805f16fc2) |
| 21:09.33 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:10.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:14.51 | WindBack | deeperror, yes, I know that it would work, but I want to know if this is the general form?? |
| 21:17.20 | Luxvero | hello, can someone please point me a detailed reference to asterisk? These how-to's teach nothing |
| 21:17.37 | [TK]D-Fender | Luxvero, : |
| 21:17.39 | [TK]D-Fender | ~book |
| 21:17.50 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:18.25 | Luxvero | great! thanks! |
| 21:18.49 | [TK]D-Fender | Luxvero, http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:19.17 | blitzrage | cervi2: which proposal what that? :) |
| 21:19.20 | Luxvero | haha, nice bot, I wonder what goes if I PM him random ~keywords |
| 21:19.30 | [TK]D-Fender | Luxvero, The 2nd link is a pretty decent 1st guide writting recent by JerJer here, and edited by myself |
| 21:20.06 | [TK]D-Fender | blitzrage, You don't want to hear their praise! |
| 21:20.15 | blitzrage | [TK]D-Fender: nope... I just want ! ! ! |
| 21:20.19 | [TK]D-Fender | :D |
| 21:20.22 | blitzrage | :D |
| 21:20.33 | Luxvero | nice, thanks Defenderm, let me that prior to the book |
| 21:24.47 | pipwerk | the book is reaally useful if you want to understand what is going on |
| 21:26.26 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 21:27.09 | irule | why will * ignore all keys I press using xten softphone? it works great with other phones :s |
| 21:28.14 | cervi2 | blitzrage , http://bugs.digium.com/view.php?id=8971 |
| 21:28.14 | blitzrage | pipwerk: 2nd edition is gonna be a big improvment over the first edition too :) |
| 21:28.14 | pipwerk | i hear so, yes |
| 21:28.16 | *** join/#asterisk Divious1 (i=chatzill@65.112.134.160) |
| 21:28.33 | pipwerk | so worth buying when it hits the shelves |
| 21:28.39 | *** join/#asterisk imesper (n=ian@189-19-59-76.dsl.telesp.net.br) |
| 21:28.49 | blitzrage | pipwerk: I hope so :) |
| 21:28.54 | blitzrage | cervi2: ahhhh yes -- that bug |
| 21:28.59 | blitzrage | cervi2: glad it's moving forward |
| 21:29.06 | *** part/#asterisk WindBack (n=jorge@host73.201-253-149.telecom.net.ar) |
| 21:29.14 | cervi2 | blitzrage, again thank you :-) |
| 21:29.24 | cervi2 | what are the next steps? |
| 21:30.02 | blitzrage | cervi2: test, then commit by a developer |
| 21:30.41 | blitzrage | ideally you get a few people to test and write on the bug tracker, "works for me!" |
| 21:31.05 | imesper | Hi, I am having a problem with incoming call in my IVR, when the caller send dtmf too fast it's lock the channel, no one hear a thing, and the channel keeps locked until I kill asterisk and start again, Using asterisk 1.4 branch 65123 |
| 21:31.18 | imesper | Anyone has a clue what is going on? |
| 21:34.00 | cervi2 | "branch 65123", do you mean it's from CVS? |
| 21:34.13 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 21:34.22 | *** part/#asterisk flashnet (i=flashnet@admin.mojamreza.org) |
| 21:34.30 | imesper | Yes from svn |
| 21:34.31 | *** join/#asterisk flashnet (i=flashnet@admin.mojamreza.org) |
| 21:35.32 | cervi2 | oh, maybe it's better to ask on asterisk-dev |
| 21:35.42 | imesper | ok |
| 21:35.53 | cervi2 | or do you have the same problem on stable-versions? |
| 21:36.51 | imesper | I don't know cause I always used SVN Branch |
| 21:37.42 | *** join/#asterisk Downchuck (n=downchuc@c-24-22-20-80.hsd1.mn.comcast.net) |
| 21:38.40 | cervi2 | trunk? |
| 21:38.45 | robin_sz | so, what to do about this "blocked" phone then????? |
| 21:39.00 | imesper | No, not trunk Branch-1.4 |
| 21:39.19 | robin_sz | setting a cron job to reboot * every couple of hours seems like a kludge |
| 21:39.28 | robin_sz | is that what people usually do? |
| 21:41.17 | imesper | SIP/123-0829f250 (None) Up AppDial((Outgoing Line)) |
| 21:41.17 | imesper | Zap/27-1 i@bratechIVR:2 Up Dial(SIP/123|10) |
| 21:41.26 | Downchuck | did this patch go anywhere: http://bugs.digium.com/view.php?id=7252 |
| 21:41.27 | Downchuck | ? |
| 21:41.33 | Mercestes | robin_sz, not if they want to keep their jobs |
| 21:41.37 | imesper | This is a stuck channel, zap and sip |
| 21:43.24 | [TK]D-Fender | imesper, you can't "sof hangup" it? |
| 21:44.00 | imesper | No |
| 21:44.09 | imesper | I have to kill asterisk |
| 21:45.56 | Downchuck | is there an rtpbindaddr setting? |
| 21:46.06 | Downchuck | I want my bindaddr to be 127.0.0.1, but asterisk is not sending out with externip |
| 21:47.03 | *** part/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 21:48.09 | *** join/#asterisk viler (i=1000@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 21:51.28 | pipwerk | Downchuck: you'll have to look into your sip config then... rtp is negotiated by sip |
| 21:55.54 | Downchuck | ok, got it working, just gave up and did bindaddr of the external IP in sip.conf |
| 21:55.57 | *** join/#asterisk DaveCanoe (n=Dave@24.114.252.226) |
| 21:56.22 | Downchuck | can asterisk register with sipphone (gizmo project) easily? i tried their copy and paste, but not to success |
| 21:59.09 | robin_sz | Mercestes, so what IS the answer then? |
| 22:01.18 | pipwerk | Downchuck: look into the 'register =>' sip.conf setting |
| 22:01.21 | [TK]D-Fender | Downchuck, : |
| 22:01.24 | [TK]D-Fender | ~sitnat |
| 22:01.28 | Downchuck | pipwerk: :P |
| 22:01.33 | Downchuck | ~sipnat |
| 22:01.35 | jbot | [sipnat] for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:01.44 | [TK]D-Fender | Downchuck, and bindaddr should be 0.0.0.0 |
| 22:02.02 | Downchuck | i have multiple IPs on the machine |
| 22:02.05 | Downchuck | and only one has open ports |
| 22:02.30 | [TK]D-Fender | Downchuck, multiple IP's on the SAME subnet? |
| 22:03.30 | robin_sz | sigh ... |
| 22:03.53 | neverblue | what is the mailbox number? |
| 22:04.03 | Downchuck | [TK]D-Fender: yes. setting bindaddr to the IP i want worked out alright |
| 22:04.07 | Downchuck | 0.0.0.0 is for binding to all ips |
| 22:04.10 | neverblue | mailbox => ... |
| 22:04.16 | robin_sz | so ... every now and then ... I get a "blocked" phone .... |
| 22:04.26 | robin_sz | <PROTECTED> |
| 22:04.33 | robin_sz | but no call is in progress |
| 22:04.41 | neverblue | does it have to match with something in my context? |
| 22:04.44 | robin_sz | sip show inuse reveals? |
| 22:04.55 | robin_sz | elaine 1 1 |
| 22:04.56 | [TK]D-Fender | robin_sz, Clearly a counter bug. Wait it out or downgrade |
| 22:05.10 | robin_sz | for peers, but 0 1 for user |
| 22:05.35 | robin_sz | [TK]D-Fender, its been around since 1.2.13, this is 1.2.17 |
| 22:06.02 | *** join/#asterisk evoxy (n=rbw@83.212.57.46) |
| 22:06.03 | *** join/#asterisk sharp (i=sharp@gateway/tor/x-dc4f8d3e7800025f) |
| 22:06.16 | robin_sz | [TK]D-Fender, I think its provoked by blind transfers |
| 22:06.55 | evoxy | what are examples of what asterisk can do? |
| 22:07.23 | robin_sz | evoxy, read the book yet? |
| 22:07.45 | evoxy | no i just want to know what it does |
| 22:07.50 | robin_sz | !book |
| 22:07.56 | robin_sz | !the book |
| 22:07.58 | robin_sz | bah |
| 22:08.10 | neverblue | ~ |
| 22:08.16 | robin_sz | ~book |
| 22:08.18 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:08.29 | robin_sz | evoxy, that ^^ |
| 22:08.57 | irule | I put different phones @ different contects according to the trpes of dialings they are allowed, but now some of them can dial others and some can not dial others, so it looks like a mess but I organized it so neatly, what should I do to make sure all phones can dial each other? thanks |
| 22:09.45 | [TK]D-Fender | evoxy, It is a telephony toolkit that you can use for all sorts of things. Most popularly is as a big-featured PBX. |
| 22:10.19 | irule | is this valid in sip.conf? context=default,office,home,babes |
| 22:10.19 | neverblue | can only explain to me what that mailbox is? |
| 22:10.26 | neverblue | can it just be any value I want? |
| 22:10.27 | [TK]D-Fender | evoxy, In that respect it can take calls in from a variety of sources, process them, and send them out other channels, etc. |
| 22:10.35 | [TK]D-Fender | irule, No. |
| 22:10.56 | [TK]D-Fender | irule, Devices only have 1 context. |
| 22:10.56 | irule | [TK]D-Fender thanks lol |
| 22:11.35 | [TK]D-Fender | irule, To give a device access to multiple contexts, you make a "container" context and INCLUDE the others in it. |
| 22:11.46 | irule | how about include => home inside of [office]? that did not work so well |
| 22:12.45 | robin_sz | irule, the easiest way to think about it is this: |
| 22:12.59 | robin_sz | think of "destinations" as classes |
| 22:13.00 | neverblue | also, I have no voicemail listed in my asterisk cli |
| 22:13.25 | [TK]D-Fender | irule, Only reason it wouldn't work well is the contents you used. Try to do it the way you think it should ben, then show us so we can point out where you have gone wrong with it |
| 22:13.44 | neverblue | does that mean I dont have voicemail setup? |
| 22:13.46 | robin_sz | then think of classes of clients ... |
| 22:13.59 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:14.03 | diclophis-work | hello all |
| 22:14.04 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 22:14.09 | diclophis-work | anyone have any recomendations for TTS? |
| 22:14.15 | robin_sz | then just make up contexts for the different client classes, and include in each one the appropriate destination classes |
| 22:14.18 | [TK]D-Fender | neverblue, You aren't making any sense at all. Start over, be clear, and pastebin configs & output where appropriate. |
| 22:14.47 | neverblue | i cannot type voicemail <command> in my * cli |
| 22:14.56 | neverblue | so does that mean its not installed? |
| 22:15.14 | irule | http://pastebin.ca/495758 please excuse my spanish lol search for bodega oficina and saladejuntas to see how I included everything |
| 22:15.20 | robin_sz | irule, for example, say, in your organisation you have some publically accessibel phones ... that people can use |
| 22:15.58 | robin_sz | so ... the "public" context might not include the following destination groups: |
| 22:16.11 | [TK]D-Fender | irule, not EVERYTHING , ok? And please do EVERYONE (yourself included) and remove all the commented lines |
| 22:16.22 | robin_sz | "secretaries" "publicPhones" "localPSTN" |
| 22:16.34 | [TK]D-Fender | irule, permanently |
| 22:16.42 | neverblue | hmm, i thought that was clear |
| 22:16.46 | Mercestes | robin_sz, D-Fender's suggestion is probably the most practical. |
| 22:16.51 | *** join/#asterisk Obergandhi83 (n=Obergand@P3a14.p.pppool.de) |
| 22:16.56 | Mercestes | s/practical/realistic/ |
| 22:17.01 | robin_sz | Mercestes, not exactly, but thanks anyway |
| 22:17.04 | Mercestes | maybe you could script something to do a sip reload or something |
| 22:17.04 | [TK]D-Fender | neverblue, You have 99% uselss crap in there. Filter it out for "sanity" |
| 22:17.21 | Mercestes | or you could patch the code |
| 22:17.25 | neverblue | Fender u need to understand that some people dont know as much as you |
| 22:17.28 | robin_sz | sip relaod, I'll try that |
| 22:17.52 | robin_sz | Mercestes, im a user, not a developer |
| 22:18.05 | Downchuck | could someone explain to me, in layman's terms, why the caller ID voicemail hack works on major cellphone networks? |
| 22:18.10 | Downchuck | i'm slightly troubled by that. |
| 22:18.34 | robin_sz | Mercestes, a sip reload didn't cure it |
| 22:18.40 | Mercestes | Downchuck, because they only look at CID. that is why your voicemail box has a password |
| 22:18.54 | Downchuck | Mercestes: yes.. but it doesn't ask the password if the CID matches |
| 22:19.03 | Mercestes | Downchuck, It should. |
| 22:19.15 | Mercestes | Downchuck, that's an option you can set, or have set. |
| 22:20.12 | evoxy | yes, i'm asking of the unofficial meaning of those |
| 22:20.19 | evoxy | like what kind of cool things can it do? |
| 22:20.20 | irule | [TK]D-Fender for popular request, I have removed all commented lines :) I start including stuff at line 272 http://pastebin.ca/495778 |
| 22:20.22 | Obergandhi83 | hi @ all .. i've an error when i compile mISDN: /usr/src/mISDN/drivers/isdn/hardware/mISDN/l1oip_core.c:1571:55: error: macro "INIT_WORK" passed 3 arguments, but takes just 2 |
| 22:20.24 | Obergandhi83 | has anyone a solution? |
| 22:20.44 | [TK]D-Fender | neverblue, I wasn't saying you don't know what you're doing or that I'm "so much better" or any such business. I merely am stating that that many comments make it completely unreadable |
| 22:21.20 | neverblue | i have no idea what your talking about (as usual) |
| 22:21.26 | neverblue | i was talking about the * cli |
| 22:21.27 | Downchuck | Mercestes: found it "skip password" |
| 22:21.40 | [TK]D-Fender | neverblue, ok... whats the problem in your pastebin? |
| 22:21.42 | neverblue | so I am not sure what you need to be readable/unreadable |
| 22:21.56 | neverblue | i have nothing to pastebin |
| 22:22.10 | neverblue | when I connect to *, I have no voicemail commands |
| 22:22.22 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 22:22.33 | [TK]D-Fender | neverblue, can you give an example and pastbin the failure |
| 22:22.34 | neverblue | did I not add support for it? or is there no voicemail commands in *? |
| 22:22.50 | neverblue | sudo asterisk -r |
| 22:22.54 | neverblue | then type help |
| 22:23.01 | Mercestes | [TK]D-Fender, he wants to type "voicemail" in his CLI and make it do shit. and since that doesn't work, he assumes that means he doesn't have voicemail installed |
| 22:23.09 | [TK]D-Fender | neverblue, Please TRY an actual command and show the failure |
| 22:23.21 | neverblue | Fender do what I said |
| 22:23.29 | neverblue | and see if there are voicemail commands listed |
| 22:23.30 | Downchuck | Mercestes: I can imagine how that meeting went at the telcos; I'm certain there isn't a person I know with 'skip-password' turned off. |
| 22:23.32 | neverblue | in the "help" |
| 22:23.56 | neverblue | *CLI> voicemail list users |
| 22:23.56 | neverblue | No such command 'voicemail' (type 'help' for help) |
| 22:24.02 | neverblue | there, is that what you are asking for |
| 22:24.04 | [TK]D-Fender | neverblue, please forget what you think you should see, and jsut try to DO something and show the faile in a pastebin. |
| 22:24.08 | neverblue | does that help you understand? |
| 22:24.45 | neverblue | or do you want me to paste those two lines in pastebin :) |
| 22:25.07 | [TK]D-Fender | neverblue, try "load app_voicemail.so" |
| 22:25.19 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 22:25.24 | irule | I wrote an answer here yesterday, how to you like it? http://readthefuckingmanual.net/error/579/ |
| 22:25.27 | neverblue | Module 'app_voicemail.so' already exists |
| 22:28.24 | [TK]D-Fender | neverblue, rty to reload it |
| 22:28.51 | neverblue | * entirely? |
| 22:28.53 | [TK]D-Fender | neverblue, and that "answer".... I don't see the point fo what you think you're trying to teach someone... |
| 22:28.55 | neverblue | or just that .so |
| 22:29.03 | [TK]D-Fender | neverblue, "reload app_voicemail.so" |
| 22:29.10 | neverblue | yeah, doesnt do anything |
| 22:29.36 | [TK]D-Fender | irule, rather |
| 22:29.40 | Mercestes | :) |
| 22:30.06 | neverblue | i need a drink |
| 22:30.15 | neverblue | im stopping at a pub on the way home |
| 22:30.35 | irule | Mercestes is that smile for my good answer? |
| 22:30.45 | Mercestes | irule, no. |
| 22:31.03 | [TK]D-Fender | irule, Actually NONE of those pieces of "info" in your answer add up at all. |
| 22:31.04 | irule | Mercestes did you read it? |
| 22:31.13 | Mercestes | irule, Yea. |
| 22:31.14 | irule | really? |
| 22:31.22 | [TK]D-Fender | neverblue, what ver of * are you using? |
| 22:31.51 | Mercestes | irule, yea, I really read it. |
| 22:32.14 | irule | so how was it? am I still clueless |
| 22:32.15 | irule | ? |
| 22:32.24 | Mercestes | it blew a little. |
| 22:32.32 | neverblue | 1.2.16 |
| 22:32.33 | irule | lol ok |
| 22:32.42 | Mercestes | 1: autofallthrough won't happen unless you call a dial with a timeout. |
| 22:33.16 | irule | oh :s |
| 22:33.20 | Mercestes | 2: autofallthrough really does hang up on you, not, sorta hangs up on you, and * complains about a "noanswer" not being available. |
| 22:33.35 | Mercestes | I mean, maybe if you let it ring for an hour or two but I've never tested it to it's fullest potential. |
| 22:33.50 | Mercestes | it's well beyond the standard 15 of most telcos |
| 22:34.18 | [TK]D-Fender | neverblue, and that "ou did NOT tell asterisk what to do in case there is s-NOANSWER defined into the corresponding calls dialplan,"... WTH?! Who says ANYTHING about how you structure your dialplan or how it might be named/numbered? |
| 22:34.31 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 22:35.00 | *** part/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 22:35.14 | neverblue | ok, your obviously annoyed |
| 22:35.34 | neverblue | my question was related only to my * cli |
| 22:35.48 | neverblue | if that has something to do with any .conf, then ask me for it |
| 22:35.54 | [TK]D-Fender | neverblue, Sorry... poor target again! |
| 22:36.01 | [TK]D-Fender | irule, that last one was for you :) |
| 22:36.01 | neverblue | how am I suppose to know what your thinking? |
| 22:36.08 | [TK]D-Fender | neverblue, sorry again! |
| 22:36.17 | [TK]D-Fender | never.. so what ver are you on? |
| 22:36.18 | neverblue | have a good weekend ppl |
| 22:36.18 | Mercestes | neverblue, I can tell you what he's thinking. |
| 22:36.32 | Mercestes | >:) |
| 22:36.39 | [TK]D-Fender | neverblue, I've apologized twice now for a bad autocomplete on my part |
| 22:36.59 | [TK]D-Fender | neverblue, I jsut had 1 simple question that could help with your issue |
| 22:37.09 | neverblue | scroll up |
| 22:37.11 | neverblue | i answered |
| 22:37.22 | [TK]D-Fender | neverblue, Ah, sorry |
| 22:37.25 | neverblue | np |
| 22:37.26 | Mercestes | <neverblue> 1.2.16 |
| 22:37.27 | [TK]D-Fender | neverblue, blind today. |
| 22:37.32 | neverblue | im done anyways |
| 22:37.53 | LeddyHM | fight, fight |
| 22:37.53 | [TK]D-Fender | neverblue, Ok, well the commands you should be using are "show voicemail users" |
| 22:38.07 | neverblue | ah |
| 22:38.09 | neverblue | i see |
| 22:38.10 | neverblue | thanks |
| 22:38.13 | Mercestes | aw. |
| 22:38.17 | [TK]D-Fender | neverblue, the primary command starting with "voicemail" is only 1.4+ |
| 22:38.24 | Mercestes | you fed the troll. damnit. |
| 22:38.45 | [TK]D-Fender | Mercestes, Wasn't trolling. He was mistaken and in need of a quick tip. |
| 22:39.05 | Mercestes | [TK]D-Fender, I know. I'm slow to forgive sometimes tho. :) |
| 22:39.10 | [TK]D-Fender | Mercestes, Holding that back from him lead to a LOT of whining for NOTHING. |
| 22:39.10 | Mercestes | and I was enjoying watching his struggle. |
| 22:39.18 | Mercestes | not for nothing..I enjoyed it |
| 22:39.37 | *** join/#asterisk N9URK (n=icechat5@cpe-065-184-160-244.ec.res.rr.com) |
| 22:39.39 | [TK]D-Fender | Mercestes, I do when the answer is an inch or two away and they aren't screaming too much. |
| 22:39.47 | [TK]D-Fender | Mercestes, and I've already shown the way to start. |
| 22:39.50 | N9URK | HI, I am going to install ztdummy but after reading the page on voip-info.org I see that it says you have to have USB UHCI which I don't have. I then read on Asterisk zaprtc. and there is some mention that it won't work on Kernel mod 2.6 or something. Where can I find better info on installing Ztdummy or Asterisk zaprtc.? Thanks! |
| 22:39.54 | Downchuck | What is "it", neverblue? |
| 22:39.58 | neverblue | dick |
| 22:40.41 | [TK]D-Fender | N9URK, What OS are you on? |
| 22:40.43 | Mercestes | neverblue, don't be mean, he was just asking a simple question |
| 22:40.51 | N9URK | Ubuntu 6.10 |
| 22:41.02 | neverblue | who? |
| 22:41.07 | Downchuck | Mercestes: no, i was being an ass |
| 22:41.11 | [TK]D-Fender | N9URK, ZTDUMMY will work fine on 1.6 without any special mods |
| 22:41.18 | Mercestes | neverblue, Downchuck. |
| 22:41.22 | neverblue | my auto-complete didnt work |
| 22:41.29 | N9URK | even though I don't have the USB_UHCI? |
| 22:41.45 | N9URK | I have usbcore 134912 3 ohci_hcd,ehci_hcd |
| 22:41.50 | [TK]D-Fender | N9URK, Not needed. ZTDUMMY uses Kernel 2.6's internal timer |
| 22:41.51 | neverblue | Mercestes, i think i really need to stop listening to ANYthing you say |
| 22:42.01 | neverblue | in my own best interest |
| 22:42.03 | N9URK | [TK]D-Fender: ok cool. thanks for the help! |
| 22:42.10 | Mercestes | neverblue, please. do. :) |
| 22:42.13 | [TK]D-Fender | Mercestes, See... he CAN learn ;) |
| 22:42.20 | Mercestes | [TK]D-Fender, obviously. |
| 22:42.22 | Downchuck | I know very little about phone systems.. thus the question.. This office is using some plain phone system, with a few T1s. I've got a machine in the office, with asterisk.. |
| 22:42.31 | Downchuck | Can I do anything about connecting the two that doesn't involve the telco? |
| 22:42.32 | [TK]D-Fender | neverblue, go in peace child! |
| 22:42.35 | Downchuck | slash hardware |
| 22:42.38 | neverblue | i shall |
| 22:42.58 | [TK]D-Fender | Downchuck, You should be specific as to what you want to DO with it. |
| 22:43.11 | Downchuck | [TK]D-Fender: i'd like them to be able to dial an extension and reach the asterisk system |
| 22:43.12 | Mercestes | Downchuck, Um...Could buy a T1 card and plug it into yoru phone system as a trunk maybe. |
| 22:43.33 | Downchuck | Mercestes: thats the natural option.. anything without hardware? |
| 22:43.42 | Mercestes | Downchuck, chan_telepathy ? |
| 22:44.03 | Mercestes | Unless the phone system speaks TCP/IP you need an interface card that accomodates the phone system |
| 22:44.23 | Downchuck | darn phone system |
| 22:44.28 | Mercestes | silly phones |
| 22:44.40 | Mercestes | do you have an old modem handy? |
| 22:44.50 | [TK]D-Fender | Downchuck, and what will * be doing for you? |
| 22:44.54 | Downchuck | the office is about a thousand miles out of the way |
| 22:45.06 | N9URK | [TK]D-Fender: is this the best way to install the mod for ztdummy? Do you have a better suggestion? apt-get install linux-headers-`uname -r` asterisk-bristuff build-essential zaptel zaptel-source |
| 22:45.17 | N9URK | (on ubu 6.10 |
| 22:45.18 | N9URK | ) |
| 22:45.29 | Downchuck | [TK]D-Fender: the IT team is not emm.. supporting.. |
| 22:45.37 | [TK]D-Fender | N9URK, I don't know anything about the APT setup for *. I only compile from source. |
| 22:45.39 | Downchuck | * would be acting as its own system |
| 22:45.42 | Mercestes | N9URK, wow, that's complex. I just emerge zaptel |
| 22:46.02 | N9URK | Mercestes: wait until Gentoo starts giving you troubles! |
| 22:46.14 | Mercestes | N9URK, hasn't happened yet |
| 22:46.22 | Mercestes | Gentoo doesn't have "troubles" it has "users" |
| 22:46.31 | [TK]D-Fender | PEBKAC |
| 22:46.33 | Mercestes | who do stupid things to it. ....I'm one of them sometimes |
| 22:47.01 | N9URK | Mercestes: Yer very lucky. I sure hope it doesn't. That's the funnies thing I have ever heard though. Portage is mostly broke (but #asterisk is not the place to discuss it though) |
| 22:47.09 | N9URK | *funniest |
| 22:47.24 | Mercestes | portage is mostly broke? |
| 22:47.53 | [TK]D-Fender | Mercestes, wait a little more and it'll be entirely broke ;) |
| 22:48.01 | Mercestes | [TK]D-Fender, heh, troll. :P |
| 22:48.06 | Obergandhi83 | hey ist grad jmd da der sich nen bisschen mit misdn auskennt? ich habe ein problem beim compelieren |
| 22:48.29 | [TK]D-Fender | Mercestes, Nah... |
| 22:48.29 | Mercestes | Obergandhi83, Try #asstricks for german asterisk support, we only do english here. |
| 22:48.36 | N9URK | no sprechen sie der heilegen Deutsch |
| 22:48.51 | Obergandhi83 | ok thx |
| 22:48.56 | Mercestes | NP |
| 22:48.58 | *** join/#asterisk ArkyLady (n=arkylady@dsl-205-234-114-108.madisoncounty.net) |
| 22:49.20 | Mercestes | >.> what? |
| 22:49.29 | *** part/#asterisk ArkyLady (n=arkylady@dsl-205-234-114-108.madisoncounty.net) |
| 22:51.45 | N9URK | [TK]D-Fender: what os you have * on? |
| 22:52.12 | [TK]D-Fender | N9URK, I run CentOS and Slackware primarily |
| 22:52.37 | [TK]D-Fender | N9URK, and always compile from the FTP release versions |
| 22:53.18 | *** join/#asterisk killown (n=killown@unaffiliated/killown) |
| 22:53.34 | N9URK | [TK]D-Fender: what do you think of Trixbox? (couldn't get it going with the VMware image, so I said forget it and am just upgrading * to 1.4) I compilied 1.4.4 from source |
| 22:54.03 | Obergandhi83 | ihave got a problem when i compile misdn ... is here a misdn freak? |
| 22:54.48 | [TK]D-Fender | N9URK, I like comtrol over my PBX thank you... Trixbox is for people who know nothing about nothing to admin a PBX to waht little capacity their limited knowledge & imagination allows |
| 22:55.01 | N9URK | [TK]D-Fender: OMFG, that is hilarious. You are so right! |
| 22:55.44 | N9URK | [TK]D-Fender: I went into #trixbox and seemed like no one really knew how to help! They didn't seem to have a clue |
| 22:56.44 | N9URK | [TK]D-Fender: they decided that it was my phone, then they said that since it wouldn't let me dial *43 I had a dial context issue (and I hadn't yet installed that module). Funny. I am deleting the trixbox vps |
| 22:57.56 | N9URK | May 18 18:57:19 WARNING[11125]: loader.c:554 load_modules: Loading module res_clioriginate.so failed! |
| 22:57.57 | N9URK | Anyone know what I screwed up here? I am getting this error when I try to start * for the first time: May 18 18:57:19 WARNING[11125]: loader.c:554 load_modules: Loading module res_clioriginate.so failed! |
| 23:03.22 | Mercestes | <PROTECTED> |
| 23:05.34 | Downchuck | ltrs |
| 23:05.35 | Downchuck | thx4hlpz |
| 23:06.16 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 23:06.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:10.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 23:11.13 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:12.49 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 23:13.00 | *** join/#asterisk killown (n=killown@unaffiliated/killown) |
| 23:14.20 | lee_is_me | Hi all, I'm getting ===> ZT_CHANCONFIG failed on channel 2: No such device or address (6) with a TDM02B. Any suggestions? |
| 23:28.57 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:29.02 | lee_is_me | anyone? |
| 23:31.34 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 23:31.36 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 23:32.28 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 23:32.31 | hrmphh | grep: /include/linux/autoconf.h: No such file or directory |
| 23:32.39 | hrmphh | anyone get that when trying to make menuconfig on zaptel-1.4.2.1? |
| 23:32.59 | hrmphh | err make menuselect rather |
| 23:34.54 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:34.54 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 23:35.59 | hrmphh | looks like a bug in the src |
| 23:35.59 | hrmphh | hrmph |
| 23:36.42 | kiscokid | bluelinq try different codecs |
| 23:37.05 | lee_is_me | Hi all, I'm getting ===> ZT_CHANCONFIG failed on channel 2: No such device or address (6) with a TDM02B. Any suggestions? |
| 23:37.34 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 23:37.49 | lee_is_me | I've googled around and seen this questions asked several times but without any answers |
| 23:39.17 | lee_is_me | $50.00 U.S. to first person who can help me solve this problem |
| 23:40.51 | [TK]D-Fender | lee_is_me, pastebin "dmesg" |
| 23:41.08 | lee_is_me | sure. hold 1 |
| 23:41.25 | [TK]D-Fender | lee_is_me, and what OS are you on? |
| 23:41.30 | lee_is_me | centos 4.4 |
| 23:43.53 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 23:47.33 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:47.33 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 23:49.00 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 23:52.18 | sandorp | for some reason, asterisk appears to be occassionally dropping the first digit dialed thereby sending me to the wrong destination; is this an asterisk bug or a problem in my dialplan? it sometimes connects me to the right destination |
| 23:52.43 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:53.11 | DocHolliday | docelm0, you around? |
| 23:53.48 | osiris250 | lee_is_me: do you have the zaptel kernel mods loaded? |
| 23:58.26 | lee_is_me | osiris250: I do. TDKD-Fender is helping me |
| 23:59.56 | *** join/#asterisk N9URK (n=icechat5@cpe-065-184-160-244.ec.res.rr.com) |
| 23:59.58 | N9URK | How does the auth and register statements in extensions.conf differ in 1.4 from 1.2? |