| 00:02.34 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-213-14.dsl.hstntx.swbell.net) |
| 00:08.26 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 00:11.50 | katsuodo | decided to use enter-ext-of-person.gsm with background() however, callers complain it goes so quickly. How can I control the duration? |
| 00:11.54 | *** join/#asterisk KingDavid_ (n=CHRIS@ool-43555029.dyn.optonline.net) |
| 00:12.23 | mosty | background lasts exactly as long as the sound file it's playing |
| 00:13.02 | katsuodo | can not control it because all the callers hear is of-person. |
| 00:13.17 | KingDavid_ | hello everyone |
| 00:13.22 | katsuodo | hello |
| 00:13.28 | mosty | katsuodo, paste your dialplan on a pastebin site |
| 00:14.01 | HarryR | katsuodo, you can put a Wait(howmanysecondsminimum) after the background |
| 00:14.14 | HarryR | but if something happens before then it'll get cut short |
| 00:14.23 | HarryR | ime I just use Playfile instead |
| 00:14.46 | katsuodo | playfile? |
| 00:15.52 | HarryR | playback sorry, i'm worse for ware due to many many beers |
| 00:16.26 | HarryR | and rarely use the asterisk dialplan anyway, nowdays most stuff is done through agi for my stuff |
| 00:16.35 | *** part/#asterisk brad[] (n=brad[]@vianet-gw-atm0.brad-x.com) |
| 00:16.50 | HarryR | katsuodo, what's causing the Background audio to be cut short? |
| 00:17.35 | katsuodo | not sure just regular pots line |
| 00:17.36 | KingDavid_ | Hi guys,I have a weird problem, the register rejects some ATAs and some it allows... it is very weird |
| 00:18.17 | KingDavid_ | I get error 407 server rejected sip call |
| 00:18.19 | mosty | kingdavid: using the same account details? |
| 00:18.28 | katsuodo | substitute background for playback? Yes, looking to use perl soon |
| 00:18.30 | KingDavid_ | exact same account details! |
| 00:18.49 | HarryR | kaldemar, I presume the call gets connected and ringing or something is played instead |
| 00:18.58 | KingDavid_ | exact same details on the linksys pap2 boxes |
| 00:18.58 | mosty | KingDavid_, then the ATA's that don't work aren't configured properly, or don't support your network setup or something |
| 00:19.30 | KingDavid_ | no chance... I have reset the atas that dont work to factory defaults |
| 00:20.19 | KingDavid_ | I have been tinkering for 2 days already... the one thing that bugs me is what can possibly account for that |
| 00:20.43 | mosty | KingDavid_, what kind of ata's work, and what kind dont? |
| 00:20.43 | KingDavid_ | in fact, 2 ATAs work, and the rest dont |
| 00:21.05 | KingDavid_ | they are al the same brand... linksys pap2s |
| 00:21.41 | mosty | all the same firmware version? and not locked to a specific voip provider? |
| 00:21.50 | KingDavid_ | it is 5 atas all together, plus xlite from my computer...xlite doesn't work either |
| 00:22.01 | KingDavid_ | and they all register! |
| 00:22.34 | HarryR | then it looks like a problem with asterisk rejecting the number you dialed |
| 00:22.39 | HarryR | or something odd like that |
| 00:22.46 | KingDavid_ | mmm... did not check the firmware version... |
| 00:23.43 | KingDavid_ | mmm you know what... you got a point, when I call the USA using a SIP provider it goes through |
| 00:23.56 | katsuodo | Harryr can I build phone trees using playback? |
| 00:24.01 | KingDavid_ | but when I use the h323 provider it doesnt |
| 00:24.17 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 00:24.22 | kraptv | Any of you guys have a Polycom/Tandberg H323 videoconferencing room I can call and test? |
| 00:24.38 | HarryR | katsuodo, yes |
| 00:24.49 | HarryR | but background works better |
| 00:24.59 | HarryR | because it lets them dial an extension while it's playing |
| 00:25.02 | KingDavid_ | but wait... what bugs me is that the h323 provider works with the 2 ATAs that work |
| 00:25.04 | katsuodo | right |
| 00:25.19 | katsuodo | And this is what I am after |
| 00:25.32 | HarryR | but you can look at the playback documentation, it should return the digit entered and you can get it to cut short if somebody enters a digit |
| 00:25.45 | KingDavid_ | kraptv: I have a neat videoconference solution for you |
| 00:27.24 | katsuodo | that is was playback does correct? if a caller disrupts playback asterisk will stop playing the (enter-ext-of-person) |
| 00:27.54 | HarryR | not quite |
| 00:28.05 | HarryR | background does what you want and I presume what you've been using |
| 00:28.21 | katsuodo | yes |
| 00:28.22 | HarryR | playback will just give you the first single digit |
| 00:28.35 | HarryR | which is evil to program around |
| 00:28.46 | katsuodo | absolutely |
| 00:29.01 | katsuodo | it is better to use background |
| 00:29.06 | HarryR | for your use, yes |
| 00:29.38 | HarryR | my only suggestion to using playback was to keep the sound being played, then do whatever processing you need to do |
| 00:29.51 | HarryR | to stop background audio from being cut short by, for example a call connecting |
| 00:30.00 | katsuodo | understood |
| 00:32.12 | katsuodo | anyone have any issues with ata's by mediatrix? |
| 00:33.10 | HarryR | hrm rings a bell, we probably sell them |
| 00:33.43 | HarryR | shame it's the weekend or i'd haul one of the testers on here :D |
| 00:33.53 | katsuodo | looking to use them instead of audiocodes |
| 00:34.08 | HarryR | audiocodes? |
| 00:36.03 | katsuodo | audiocode voip gateways |
| 00:36.12 | katsuodo | for analog phones |
| 00:36.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:39.58 | Lisa696 | I need any program client for conect with asterisk???? |
| 00:44.21 | HarryR | uh |
| 00:44.35 | HarryR | Lisa696, xlite, eyebeam, brei or whatever counterpath's latest stuff is? |
| 00:44.38 | HarryR | sjphone? |
| 00:49.18 | _charly_ | zoiper |
| 00:49.31 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:49.56 | _charly_ | btw, is there a version of zoiper for 64bit linux anywhere? |
| 00:51.12 | nestAr | so what do i do when, chan_zap tells me that both sides think they are the CPE, and both sides think they are the Network? |
| 00:51.37 | Alowishus | _charly_: does there need to be? won't the 32 bit version run? |
| 00:52.12 | HarryR | _charly_, zoiper should work fine in 32bit mode with 32bit userland libs |
| 00:52.14 | Mercestes | nestAr, YOu find out if you are the CPE or network and set yoru zaptel stuff appropriately |
| 00:52.24 | _charly_ | Alowishus: i have 32bit userland libs installed on my system, but zoiper 32bit crashes with "Floating point exception" |
| 00:52.32 | Alowishus | interesting |
| 00:52.32 | HarryR | oo |
| 00:52.41 | HarryR | qemu |
| 00:52.41 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:52.43 | HarryR | :D |
| 00:52.57 | nestAr | Mercestes: one would assume I have a 50% chance of being right on either one, but neither seems to work |
| 00:53.02 | HarryR | it's better than I can anyway, I can't even run zoiper |
| 00:53.11 | Mercestes | nestAr, and you'ev guess twice? |
| 00:53.14 | HarryR | open source it so I can run it on linux/ppc :D |
| 00:53.17 | nestAr | neither worked |
| 00:53.35 | Mercestes | nestAr, is your zap channel coming from a Telco?? |
| 00:53.39 | nestAr | yes |
| 00:53.49 | Mercestes | Then it sounds like the Telco side is in loopback mode. |
| 00:53.52 | Mercestes | so your seeing yourself. |
| 00:54.03 | nestAr | good point |
| 00:54.11 | nestAr | never thought about it like that. |
| 00:56.03 | Mercestes | :) |
| 00:56.12 | Mercestes | Just remember I'm for hire. lol |
| 00:56.39 | nestAr | so what you're saying mister obvious... is he could still be a good contractor </bobandtom> |
| 00:57.18 | nestAr | Mercestes: if it was looped, shouldn't that show up in zttool? |
| 00:57.21 | _charly_ | has anyone ever asked to release the free version of zoiper as open source? |
| 00:59.43 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:03.20 | katsuodo | <PROTECTED> |
| 01:13.32 | dlynes_laptop | katsuodo: sure you can |
| 01:13.41 | dlynes_laptop | katsuodo: an extension is just an entry in your dialplan |
| 01:16.49 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:23.26 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:24.11 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:25.11 | *** join/#asterisk coppice (n=chatzill@142.204.17.210.dyn.pacific.net.hk) |
| 01:25.11 | katsuodo | dlynes I am getting a little confused because I understand ip phones needing extensions because it has to be registered |
| 01:25.19 | katsuodo | analog phones gets an extension depending on channel/port it is assigned to |
| 01:25.38 | nestAr | phone company says that they just don't see my equipment |
| 01:26.01 | dlynes_laptop | katsuodo: you're referring to channel numbers and sip peers, not extensions |
| 01:26.33 | katsuodo | trying for clarity |
| 01:26.35 | dlynes_laptop | katsuodo: any zaptel channel and/or sip peer and/or iax2 peer and/or SCCP peer and/or ... can have an extension |
| 01:26.57 | dlynes_laptop | katsuodo: an extension is just a dialplan number you can dial to reach that telephony device |
| 01:27.26 | dlynes_laptop | katsuodo: and then asterisk does a look up in the dialplan to determine what technology and peername to call |
| 01:27.48 | katsuodo | right but here the analog phone is plugged into a port on the card |
| 01:28.17 | katsuodo | there is not ata device |
| 01:28.21 | dlynes_laptop | katsuodo: eg.: exten => 201,1,Dial(Zap/1) would ring the analog phone plugged into port 1 on the zaptel card, when someone dials extension 201 |
| 01:28.33 | katsuodo | right |
| 01:28.43 | dlynes_laptop | katsuodo: is it clear now? |
| 01:29.03 | katsuodo | yes, |
| 01:32.53 | katsuodo | two phones plugged into some dohickey which plugs into the port can have two seperate extensions |
| 01:34.41 | *** join/#asterisk blq (n=Bl@dslb-088-064-154-082.pools.arcor-ip.net) |
| 01:36.18 | Mercestes | nestAr, No, but if you run a loopback test on it the test should be successful.l |
| 01:36.27 | Mercestes | nestAr, if you are not looped back the test should fail |
| 01:37.05 | nestAr | they ran a loopback on their side. I could see it on the Adtran box, but nothing on my Asterisk box. |
| 01:37.21 | Mercestes | katsuodo, IF you are referring to a splitter plugged into an FXS device, then no, two analog phones plugged into a single zap port cannot have individual extensions. |
| 01:38.17 | Mercestes | nestAr, I am pretty sure an Adtran puts out FXS so there would not be settings for pri_net or pri_cpe. |
| 01:38.48 | Mercestes | nestAr, So unless the adtran puts out PRI then your settings are likely wrong. |
| 01:40.30 | nestAr | they are telling me it's supposed to be PRI |
| 01:40.34 | Mercestes | katsuodo, Strom_C could probably patch zaptel to incorporate monkeys with little pieces of wire to outrun the ring signal and wire up the correct phone based on the extension....but he's about the only he who could do it. |
| 01:40.39 | nestAr | NI-2 |
| 01:41.00 | nestAr | unless they are throwing PRI around when they mean something else |
| 01:41.24 | Mercestes | The adtran could be in loopback mode. |
| 01:42.07 | Mercestes | in my experience tho you feed an adtran a PRI (from the telco) and the adtran splits it out into analog lines. |
| 01:42.53 | Mercestes | Do you have a T1 card or an FXO card? |
| 01:42.58 | nestAr | that's what we had, this morning.. they came and replaced one adtran for another, and told me to plug my pri card in to it, but they say they aren't seeing my d channel |
| 01:43.04 | nestAr | PRI card, TE120P |
| 01:43.20 | nestAr | zttool shows no alarms, and "OK" |
| 01:43.29 | katsuodo | yes it is a splitter, the other analog phone would need its' on fxs module? |
| 01:44.10 | nestAr | i would like nothing more than to find out they are screwing with me.. they have been nothing but a pain. |
| 01:44.23 | nestAr | what else is new, telcos' |
| 01:44.32 | katsuodo | mercestes, I could not think of the name ie. doohickey |
| 01:44.35 | Mercestes | Is there another box between the telco and the adtran? |
| 01:45.34 | Mercestes | katsuodo, right, if you are referring to a splitter than no. The "device" in this instance is one of hte ports on the FXS card. If you want individual extensions then you wiill have to use individual ports. |
| 01:45.56 | katsuodo | gotcha |
| 01:46.18 | nestAr | yeah, just a standard smart jack box.. |
| 01:46.20 | nestAr | 4 slot deal |
| 01:46.38 | Mercestes | nestAr, Any reason you can run from the smartjack straight to your T1 card without the adtran? |
| 01:47.00 | nestAr | my internet wouldn't work |
| 01:47.09 | katsuodo | Mercestes what about an ata? |
| 01:47.13 | Mercestes | So what you have is a partial T1. |
| 01:47.22 | Mercestes | nestAr, or a multimode T1 rather. |
| 01:47.33 | nestAr | yeah, it's dynamic |
| 01:47.41 | nestAr | i get a full 1.5 data if no one is on the phone |
| 01:47.58 | Mercestes | katsuodo, You'd have to dial it as a SIP device but yea, each ATA usually has two fxs ports. It sill wouldn't work without a splitter. |
| 01:48.04 | Mercestes | s/without/with/ |
| 01:48.18 | Mercestes | nestAr, Yea, that won't work. |
| 01:48.45 | Mercestes | nestAr, unless the Adtran magically provides T1 timing to asterisk when it's using it's b-channels for data traffic. |
| 01:49.16 | nestAr | that's the way i understood it.. i don't know.. there is a lot in this life i don't know. |
| 01:49.32 | Mercestes | So your adtran would have to fake a T1 signal to asterisk when it's using the T1 for data so asterisk thinks the PRI is still there and dynamical establish timing and deliver a call when a call comes in. |
| 01:49.46 | Mercestes | It sounds questionable to me, honestly. |
| 01:49.52 | katsuodo | jbot no splitter got it, so for every analog phone you need a fxs port always |
| 01:49.54 | katsuodo | ? |
| 01:50.05 | Mercestes | katsuodo, Yup. |
| 01:50.14 | Mercestes | katsuodo, if you want to ring them individually. |
| 01:50.21 | katsuodo | I do |
| 01:50.25 | Mercestes | Then yes. |
| 01:50.48 | katsuodo | an office has 10 analog phones and do not want to purchase ip phones |
| 01:51.20 | nestAr | Mercestes: yes, it has to me all along.. |
| 01:51.27 | katsuodo | but want the feature of an Asterisk IP PBX with an integrated T1 |
| 01:51.36 | Mercestes | katsuodo, Well, I don't think you wana run 5-10 ATAs so you might want to invest in one of those big FXS cards. |
| 01:51.46 | katsuodo | right |
| 01:51.57 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 01:52.12 | Mercestes | Well, your not going to get all the features unless you invest in some IP phones, honestly. |
| 01:52.40 | katsuodo | yeah working to explain this |
| 01:52.44 | Mercestes | nestAr, pastebin your zapata.conf and your zaptel.conf and I'll have a look. |
| 01:53.28 | Mercestes | katsuodo, Polycoms are only like $250 a piece I believe. |
| 01:54.40 | katsuodo | was looking at 501's & 601's if I remember correctly |
| 01:56.03 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:56.08 | nestAr | http://www.pastebin.ca/743073 |
| 01:56.12 | nestAr | i don't have much in there |
| 01:57.07 | nestAr | the reason for 2-23 is that they are supplying a analog line on 1 for my alarm |
| 01:58.42 | katsuodo | jbot, Mercestes, Harryr and others thanks a lot, chau |
| 01:59.35 | *** part/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 02:00.58 | Mercestes | nestAr, looks right to me. |
| 02:01.34 | nestAr | the channel 1 does work.. and it doesn't look like they have the other lines configured on the block |
| 02:01.45 | nestAr | my old key system works on line 1, but none else. |
| 02:02.13 | Mercestes | any reason you can't FXS the alarm? |
| 02:02.33 | Mercestes | I've found that deviating from the 1-23 setup tends to make things unhappy. |
| 02:02.55 | Mercestes | I never know if I'm supposed to config 1-22 with dchan 23 or 2-23 in zapata.conf and 1-22 in zaptel.conf, etc. |
| 02:06.08 | nestAr | I just figured it would be more reliable for the alarm to be off on it's own. |
| 02:06.21 | nestAr | if it were me, i wouldn't even plug the thing in.. it's got cellular too |
| 02:07.32 | Mercestes | Shouldn't the Telco be providing you with a Dchannel? |
| 02:07.53 | nestAr | d channel is 24, according to install tech. |
| 02:07.54 | k31th | morning |
| 02:08.07 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 02:08.21 | Mercestes | yea, but you said the Telco said they weren't seeing your Dchan. I thought they were providing it to you?? |
| 02:09.03 | nestAr | well, they said the d channel is down |
| 02:10.05 | nestAr | they are insisting that it's my config |
| 02:10.14 | nestAr | and i don't think it's my config. |
| 02:10.19 | nestAr | but it could be |
| 02:10.24 | Mercestes | The Dchan should be up between the smartjack and the adtran. |
| 02:10.56 | nestAr | they see that, that seems to be working |
| 02:11.04 | nestAr | my one analog line and the internet works |
| 02:11.25 | nestAr | i assumed that the adtran provided a virtual d channel to asterisk |
| 02:11.27 | nestAr | or something |
| 02:12.16 | nestAr | i really don't know, i used to play with T1's a lot back in the day, but it's been a few years... and i don't know if they are lying to me, or if they've made some sort of technological advances since i've been out of the industry. |
| 02:14.56 | Mercestes | I dunno, it seems to me they have some freaky setup and the adtran is supposed to be faking a T1 to asterisk. |
| 02:15.11 | Mercestes | you might want to change 2-23 to 1-23 and plug the T1 in directly and see if it works then. |
| 02:15.39 | nestAr | k |
| 02:15.45 | nestAr | i will try that |
| 02:15.55 | Mercestes | lemme know what that does. Should answer if it's your config or not. |
| 02:22.27 | *** join/#asterisk wunderkin (n=wunderki@ip72-208-16-104.ph.ph.cox.net) |
| 02:30.18 | *** join/#asterisk gardo (n=gardo@121.97.178.82) |
| 02:32.55 | _Sam-- | what directory within /lib/modules/$uname -a are the zaptel modules installed? |
| 02:33.16 | _Sam-- | or thats not where they go by default unless you configure with --prefix? |
| 02:37.26 | HarryR`Sleep | _Sam--, find /lib/modules -name 'zt*' |
| 02:37.49 | nestAr | Active, Provisioned, Down |
| 02:38.10 | nestAr | is what it showed, zttool showed it OK, but nothing else.. |
| 02:38.22 | nestAr | got rid of the "we think we're the cpe" error.. |
| 02:38.32 | nestAr | but didn't do anything else worthwhile. |
| 02:39.12 | _Sam-- | HarryR`Sleep: im just wondering where, by default, make install installs them. |
| 02:39.27 | _Sam-- | i have them in a few places within /lib/modules. |
| 02:39.34 | nestAr | but when it's plugged into the adtran, it shows "Provisioned, Up, Active" |
| 02:40.09 | HarryR`Sleep | for the kernel modules it'll be /lib/modules/`uname -r`/ something zt* |
| 02:40.23 | nestAr | man, i wish i knew what the password to get into the Adtran was. |
| 02:40.51 | _Sam-- | yes, but within /lib/modules/uname, i have them in /extra /misc and /zaptel. |
| 02:41.00 | _Sam-- | i dont know which, by default, is the place they are installed into. |
| 02:41.01 | _Sam-- | if any. |
| 02:41.03 | Mercestes | nestAr, Did you do a zaptel restart to try and bump the PRI back online? |
| 02:41.12 | nestAr | yea, i rebooted too |
| 02:41.14 | HarryR`Sleep | _Sam--, just do a fine.. |
| 02:41.19 | _Sam-- | i have. |
| 02:41.20 | HarryR`Sleep | tbh I don't know much about zaptel |
| 02:41.21 | _Sam-- | and an updatedb |
| 02:41.30 | _Sam-- | i cant find the new ones i compiled after make install. |
| 02:41.32 | nestAr | i actually halted the system while i switched the cables around |
| 02:41.34 | Mercestes | nestAr, and you were configured 1-23 cpe? |
| 02:41.39 | HarryR`Sleep | _Sam--, look at the output of make install |
| 02:41.44 | _Sam-- | maybe the old ones were of the same size, so it didnt install the new ones? |
| 02:41.50 | HarryR`Sleep | you'll see where it was installed to |
| 02:41.54 | nestAr | yes sir, and i tried 2-23 after that, just out of curiosity. |
| 02:41.59 | nestAr | but 1-23 first |
| 02:42.07 | _Sam-- | make install sucks when you run screen for scrollback buffer |
| 02:42.08 | Mercestes | weird. |
| 02:42.09 | nestAr | before i shut down, so it booted back up to 1-23 |
| 02:42.22 | Mercestes | and the 1 analog and internet works?? |
| 02:42.42 | nestAr | yep. |
| 02:42.53 | nestAr | that' what we're chatting across |
| 02:43.02 | nestAr | i am on screen, just detach when i logged off |
| 02:45.40 | nestAr | what pisses me off the most about all this.. is not that it doesn't work.. is that they wouldn't install a 2nd t1 so i could test, they would only reconfig the existing line. |
| 02:46.14 | nestAr | so i couldn't test, now we have no phones.. and it's the weekend and not much support available from the phone company. |
| 02:46.26 | nestAr | so monday, i will open with likely no phones |
| 02:48.20 | _Sam-- | the situation is you have a PRI and some kind of zaptel card, and cant get it working? |
| 02:48.59 | _Sam-- | what is the signalling? |
| 02:49.21 | nestAr | pri_cpe |
| 02:49.34 | nestAr | esf, b8zs |
| 02:49.36 | nestAr | ni-2 |
| 02:50.09 | nestAr | http://www.pastebin.ca/743073 |
| 02:50.47 | _Sam-- | the light on the zaptel card is red or green? |
| 02:50.52 | nestAr | green |
| 02:52.19 | _Sam-- | and you've tried channel 1-23 |
| 02:52.20 | _Sam-- | ? |
| 02:52.38 | _Sam-- | channel => 2-23 |
| 02:52.45 | _Sam-- | shouldnt matter, though. |
| 02:52.54 | nestAr | yes sir. |
| 02:53.53 | _Sam-- | its been a few years since i used my PRI (still have the zap card in the machine, but now use some other sip gateways)...but this worked for MY pri here in the USA: http://www.pastebin.ca/743105 |
| 02:54.11 | _Sam-- | using digium single port PRI card , maybe te110p or some such |
| 02:58.10 | _Sam-- | if you have internet on the same T1 as this connection, what makes you think channels 2-23 are for phone? |
| 02:58.35 | _Sam-- | you have some sort of fractional T1? some for phone channels, some for internet? |
| 03:00.07 | _Sam-- | i think mercestes hit on that earlier, in reading some scrollback buffer. |
| 03:00.18 | _Sam-- | but reading his comments, i tend to echo that sentiment. |
| 03:00.32 | _Sam-- | if you use the t1 for data and voice, then probably not all 2-23 channels are for voice. |
| 03:01.27 | nestAr | they are telling me it's supposed to be 2-23 |
| 03:01.32 | nestAr | but i don't know.. |
| 03:01.42 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 03:01.51 | Mercestes | It appears the adtran reassigns channels 2-23 for data and then resignals them as PRI when phone calls happen through some kind of magical voodoo. |
| 03:02.15 | nestAr | they emailed me a build out sheet and it shows only channels 2-11 or maybe it was 12. |
| 03:02.25 | Mercestes | So theoretically the Adrtan would supply a T1 signal to asterisk to appease it while the T1 itself was being dynamically used for voice and data traffic. |
| 03:02.26 | nestAr | i tried changing it to 11 and 12 |
| 03:02.28 | *** join/#asterisk blq (n=Bl@dslb-088-064-154-082.pools.arcor-ip.net) |
| 03:02.30 | _Sam-- | what is the model of the adtran? |
| 03:02.34 | nestAr | but neither of those seemed to make a difference.. |
| 03:02.38 | nestAr | hold on. let me check. |
| 03:03.10 | nestAr | adtran total access 616 |
| 03:03.35 | Mercestes | Now, more realistically would be a Cisco IAD setup where 1 meg of Internet was being used on b-chans 8-23 and channels 1-7 were voice DS0s with a dchan on 24. |
| 03:04.09 | Mercestes | 1 meg of INternet on DSOs 8-23 and chanels 1-7 were voice B-chans rather. |
| 03:04.20 | Mercestes | that would make *more* sense. |
| 03:04.29 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 03:04.37 | _Sam-- | nestAr : you'll have phones on monday. |
| 03:04.49 | Mercestes | This "oh, hell, I'll dump some IP packets on these DSOs for awhile and bring up a b-chan later" thing doesn't sit well with me. |
| 03:05.17 | _Sam-- | this may or may not help: http://www.voip-info.org/wiki/view/adtran |
| 03:05.32 | _Sam-- | "Edit your /etc/asterisk/zapata.conf and add the following lines (at the bottom): |
| 03:05.33 | _Sam-- | " |
| 03:06.48 | nestAr | hrmm, now sure.. |
| 03:06.57 | nestAr | not* |
| 03:07.07 | nestAr | i am going to go home.. it's getting late.. |
| 03:07.20 | nestAr | thank you guys for your help |
| 03:07.24 | nestAr | i'll be back tomm. |
| 03:07.26 | _Sam-- | hopefully you can ssh to the box |
| 03:10.12 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.76) |
| 03:15.34 | *** join/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 03:15.44 | *** part/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 03:15.51 | *** join/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 03:16.50 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:17.11 | KingDavid_ | Help, I can't figure a reason why in this server, h323 calls terminate in 2 of my PAP2s and not in another 3 pap2s |
| 03:17.55 | DaneM | Hi, all. I have an odd problem. I have 2 zaptel channels on fxo (going to the phone company) and 2 on fxs (going to handsets). The fxo channels are working fine, as far as I can tell, but only one of the fxs channels will give a dialtone, or take incoming calls. The odd thing is that it all worked fine yesterday, and I don't know what I've changed that broke it. :-( Any ideas? |
| 03:18.11 | KingDavid_ | furthermore, when I use h323, the error on the pap2 is '407 ' |
| 03:19.46 | KingDavid_ | even more puzzling, the h323 calls terminate well when I use DIDs |
| 03:24.33 | *** part/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 03:25.04 | *** join/#asterisk Sweeper (i=sweeper@208.64.37.147) |
| 03:25.13 | Sweeper | anyone been able to get * working on colinux? |
| 03:30.12 | Nugget | what a strange idea (colinux) |
| 03:30.15 | Nugget | I'd never heard of it |
| 03:31.30 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:42.28 | Sweeper | mm, apparently there are people doing it |
| 03:46.06 | Nugget | how strange. |
| 03:46.25 | Nugget | I'd be tempted to play with it except I don't have a Windows machine. |
| 03:47.41 | Sweeper | my work lappy is windows, so this is good |
| 03:48.38 | Nugget | Aside from my two asterisk servers I've managed to completely eliminate Windows and Linux from my life. It's quite liberating. |
| 03:49.10 | Nugget | I find I have to spend a lot less time thinking about my software. |
| 03:49.26 | Nugget | and more time thinking about more productive things |
| 03:50.33 | KingDavid_ | Nugget: If you don't use windows nor linux, what os do you use? |
| 03:50.40 | Nugget | we're all OS X and FreeBSD |
| 03:50.52 | Nugget | OS X on the desktops and FreeBSD in the server room |
| 03:51.18 | Nugget | and one OS X server to wrangle the desktops |
| 03:51.47 | KingDavid_ | I can't find not using linux liberating |
| 03:51.55 | KingDavid_ | :) |
| 03:51.58 | Nugget | I find Linux to be a bothersome headache. |
| 03:52.03 | *** join/#asterisk Runlvl (n=juan@190.19.94.95) |
| 03:52.09 | Nugget | It's like Unix but shittier and less predictable |
| 03:52.21 | Runlvl | Asterisk Latin America, Spanish community: http://www.asterisk-la.org |
| 03:52.23 | KingDavid_ | I though using linux was liberating :) |
| 03:52.55 | Nugget | Linux folks seem to think that "new and clever" is better than "boring and stable", which can be envigorating I suppose but in practice is a total pain in the ass. |
| 03:53.02 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
| 03:53.22 | Nugget | I don't really care if my server OS makes a good media center, or if you can run world of warcraft on it in WINE. |
| 03:53.36 | Nugget | but all the effort to make those other things possible makes it a poor platform for a server |
| 03:53.58 | Sweeper | I'd use freebsd if it support more hardware :/ I use it on servers if I don't need zaptel |
| 03:54.07 | Sweeper | *supported |
| 03:54.15 | Nugget | Yeah, FreeBSD isn't very good if you need support for stuff like $5 tape drives. |
| 03:54.28 | Nugget | I'm sure this is of critical importance for everyone who has exactly $5 worth of data. |
| 03:54.29 | Sweeper | or laptops....or recent video cards.... |
| 03:54.44 | Runlvl | Nugget, that is the reason, beacause i dont you freebsd :( |
| 03:55.03 | Nugget | hey, FreeBSD had USB support for about a year before Linux did. :) |
| 03:55.06 | Runlvl | Freebsd doenst support my fuc%&( laptop |
| 03:55.06 | Sweeper | or tdm interfaces..... |
| 03:55.32 | Nugget | yeah, it's zaptel that keeps my Asterisk servers on Linux. |
| 03:55.35 | Sweeper | netbsd supports lots of stuff, but is a horrid desktop os :v |
| 03:55.37 | Nugget | as soon as we got the PRI I was stuck |
| 03:56.44 | Nugget | I use Slackware because it's the least Linuxy Linux. |
| 03:56.48 | Runlvl | Any bsd have support for laptops :( |
| 03:57.01 | Nugget | stays out of my way, doesn't make bad assumptions about what I have installed |
| 03:57.14 | Sweeper | netbsd will support most of your hardware, Runlvl |
| 03:57.19 | Nugget | my girlfriend swears that FreeBSD was better for her than Linux on her dell laptop. |
| 03:57.28 | Nugget | specifically for wireless support |
| 03:57.37 | Nugget | I dunno though, my laptops are all macs. |
| 03:58.11 | Runlvl | Sweeper, freebsd support all.. except my sound card.. i need to use oss on that.. and really sucks |
| 03:58.35 | Runlvl | Sweeper, i will check that, thanks dude |
| 04:01.14 | *** join/#asterisk kavelot (i=x@201-68-27-8.dsl.telesp.net.br) |
| 04:01.52 | kavelot | is it possible for a SIP phone to use, like, ulaw for sending and gsm for receiving? |
| 04:02.44 | Sweeper | kavelot: doubtful |
| 04:03.10 | kavelot | i ask this because I can hear (on the normal phone to which I called) what I say on the voip phone... but not vice-versa |
| 04:03.19 | kavelot | althought, for incomming calls, both works |
| 04:03.20 | kavelot | any hints? |
| 04:03.24 | Sweeper | NAT issue |
| 04:03.30 | kavelot | hmm... |
| 04:03.37 | kavelot | would it work both directions for incomming calls? |
| 04:03.50 | Sweeper | depends on where the NAT is |
| 04:03.58 | Sweeper | but one-wa audio is almost always a NA |
| 04:04.01 | Sweeper | NAt problem |
| 04:04.03 | kavelot | hmmmm... |
| 04:04.05 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 04:04.13 | kavelot | my config is SIP phone -> asterisk -> internet (voip provider) |
| 04:04.23 | kavelot | I'll try setting nat to yes |
| 04:04.32 | kavelot | thanks |
| 04:04.35 | Sweeper | n/p |
| 04:06.35 | *** join/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com) |
| 04:10.08 | peanut- | I don't have enter-ext-of-person.gsm, is that not in the asterisk package? |
| 04:11.05 | Mercestes | peanut-, try an emerge -av asterisk-sounds |
| 04:11.29 | Mercestes | Unless your not on gentoo. then....go google how to install stuff on your OS. |
| 04:11.30 | peanut- | I don't think FBSD would like that |
| 04:11.36 | peanut- | is it in maybe asterisk-addons?> |
| 04:11.44 | Mercestes | In soviet russia, freebsd does not like you! |
| 04:11.56 | peanut- | freebsd doesn't like me here either |
| 04:12.56 | Mercestes | peanut-, Umm. ...no FreeBSD not having emerge does not mvoe that sound file into asterisk-addons. It's still in asterisk-sounds. |
| 04:12.56 | Mercestes | I think you just have to find it in the FTP site adn download it . |
| 04:13.02 | Sweeper | or pkg_add -r asterisk-sounds :P |
| 04:13.32 | Mercestes | or apt-get asterisk-sounds |
| 04:13.35 | kavelot | hm, I tried with nat yes and never, none worked... I also checked that ulaw is being both-ways... any other ideas? |
| 04:13.41 | Mercestes | or yum install asterisk-sounsd |
| 04:13.45 | Mercestes | or rpm -e asterisk-sounds |
| 04:14.18 | Mercestes | I think if the entirity of the *nix community wasn't a bunch of whiney kids that don't know how to share their toys, it'd make an excellent OS. |
| 04:15.00 | Sweeper | BSD and linux are completely different codebases -.- |
| 04:15.03 | Mercestes | But instead we have about 700 different 'flavors' (read, arguements between developers =/) and about 50 ways to install things. |
| 04:15.29 | Mercestes | Sweeper, by "completely different code bases" you mean "the developers started fighting a long time ago." |
| 04:16.02 | Sweeper | no, linux was a terminal emulator :P |
| 04:16.03 | peanut- | Mercestes: seeing as I had no indea what the is in asterisk-addons, it was woth a try.. |
| 04:16.15 | Mercestes | kavelot, One way audio is almost always Nat or firewall. Any firewalls involved? |
| 04:16.37 | kavelot | no... besides, it works both-ways for incomming calls |
| 04:16.51 | kavelot | this just happens for outgoing calls |
| 04:16.59 | Mercestes | peanut-, I think asterisk-sounds is just a gzip in the ftp site. You just download it and untar it in /var/lib/asterisk/sounds |
| 04:17.40 | Mercestes | peanut-, gentoo just makes it easy by adding a for-pretend package that does that simple step for me because I'm a refreshing mix of lazy and retarded. |
| 04:18.20 | Mercestes | kavelot, Still means either nat or a firewall. |
| 04:18.36 | peanut- | I hate gentoo |
| 04:18.40 | peanut- | and most linux in general |
| 04:18.42 | Sweeper | it hates you :P |
| 04:18.45 | Mercestes | I'm sorry. |
| 04:18.48 | Mercestes | what do you hate about gentoo? |
| 04:19.01 | Mercestes | Aside from the fact that it was a different code base last week? |
| 04:19.03 | kavelot | does my sip.conf helps? http://rafb.net/p/mofAHD69.html |
| 04:19.05 | Sweeper | the pretentious users, obviously \o |
| 04:19.19 | kavelot | I don't know what I could try to solve this besides setting nat=yes and nat=never |
| 04:19.21 | kavelot | (none worked) |
| 04:19.35 | Mercestes | Sweeper, lmao |
| 04:19.55 | Sweeper | hmm |
| 04:20.09 | Sweeper | see, I doubt the problem is between sip phone and asterisk |
| 04:20.10 | Mercestes | kavelot, Is there nat involved?? |
| 04:20.36 | Sweeper | kavelot: did you set sip on the sip phone or on the gvt entry? |
| 04:20.45 | Sweeper | nat, I mean |
| 04:20.52 | kavelot | on gvt entry |
| 04:21.04 | kavelot | the config Internet (me) -> asterisk -> Internet (voip provider), no NAT for that |
| 04:21.36 | Sweeper | kavelot: try on sip phone and gvt |
| 04:24.02 | kavelot | seems PhonerLite doesn't have this function... gonna try that later, thanks for now :) |
| 04:24.21 | peanut- | woo, more sounds! |
| 04:27.23 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 04:28.27 | *** join/#asterisk BBHoss (n=hoss@216.186.235.254) |
| 04:33.20 | Mercestes | :) |
| 04:33.30 | Mercestes | your jubilee inspires me. |
| 04:41.07 | *** join/#asterisk angom (n=Angel@201.170.9.172) |
| 04:46.24 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:53.00 | *** join/#asterisk coppice (n=chatzill@229.193.17.210.dyn.pacific.net.hk) |
| 05:07.17 | peanut- | -rw-r--r-- 1 root wheel 2046 Oct 20 04:23 what-are-you-wearing.gsm |
| 05:07.17 | peanut- | what? |
| 05:16.26 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:21.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 05:23.04 | coppice | its part of a suite of recorded phrases for those running automated telephone sex services |
| 05:23.48 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 05:28.19 | BBHoss | heh |
| 05:42.37 | *** join/#asterisk ltd (n=z@202.55.146.72) |
| 05:56.52 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:57.30 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:08.24 | *** join/#asterisk blq (n=Bl@dslb-088-064-154-082.pools.arcor-ip.net) |
| 06:13.43 | *** join/#asterisk neax (n=newdle@203-114-176-86.dsl.sta.inspire.net.nz) |
| 06:19.25 | peanut- | is there a way for a caller to call into asterisk then get dropped to a line to make an outgoing call? |
| 06:21.13 | kiscokid | DISA |
| 06:21.25 | kiscokid | http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA |
| 06:21.48 | peanut- | sweet, thanks |
| 06:26.13 | peanut- | exten => 700,1,VoiceMailMain() I have this as an extension in incoming, put in extension and pass, says invalid. is this generally accessable from outside? |
| 06:37.01 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 06:50.53 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 07:11.08 | *** join/#asterisk blq (i=Bl@dslb-088-067-026-009.pools.arcor-ip.net) |
| 07:14.47 | *** join/#asterisk blq (n=Bl@dslb-088-067-031-134.pools.arcor-ip.net) |
| 07:32.39 | *** join/#asterisk af_ (n=getsmart@81-174-44-210.dynamic.ngi.it) |
| 07:41.42 | *** join/#asterisk blq (n=Bl@dslb-088-067-024-251.pools.arcor-ip.net) |
| 07:43.44 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:51.18 | *** join/#asterisk sergey (n=sergey@gw4-131.iks.ru) |
| 07:54.38 | sergey | Hi any idea how change codec during quality SIP channels (for inst: good channel use g729 but when channel have big trafic use g723 when good channel back - use 729 again)? |
| 07:56.13 | *** join/#asterisk KE6IGZ (i=dave@cpe-66-75-70-232.socal.res.rr.com) |
| 07:57.35 | *** join/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 08:00.05 | *** part/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 08:04.56 | peanut- | anyone know why my voicepulse to germany is choppy like it's crap bandwidth? |
| 08:06.45 | peanut- | SIP works fine.. |
| 08:14.21 | peanut- | but yea, when I have my international dialing set to IAX instead of SIP it's horrible and choppy, but IAX2 for US calls works great |
| 08:22.17 | *** join/#asterisk knarfly (n=kanrfly@c-75-74-155-198.hsd1.fl.comcast.net) |
| 08:22.37 | knarfly | 8-) |
| 08:31.46 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 08:32.30 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 08:34.17 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 08:50.23 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:57.18 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:05.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:05.36 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 09:18.22 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 09:23.02 | *** join/#asterisk BBHoss (n=jacob@user-24-214-214-54.knology.net) |
| 09:24.51 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 09:25.22 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:27.24 | *** join/#asterisk s0lid (n=jlq@210.213.198.98) |
| 09:28.29 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 09:33.20 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 09:35.09 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 09:36.58 | *** join/#asterisk bantu (n=Miranda@p54A315D0.dip0.t-ipconnect.de) |
| 09:47.36 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 10:13.15 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 10:20.40 | *** join/#asterisk StevenElvisda_ (n=Steven_E@202.47.107.60) |
| 10:28.23 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:29.29 | roxlu | hi |
| 10:30.14 | knarfly | anyone using freebsd with asterisk? |
| 10:40.54 | *** join/#asterisk qdk (n=qdk@88.151.41.234) |
| 10:41.12 | roxlu | knarfly: yes |
| 10:41.31 | roxlu | I've installed it yesterday and I'm testing it |
| 10:43.17 | peanut- | knarfly: I am |
| 10:46.50 | roxlu | is there a way to use asterisk and some 'device' to receive free sms messages? (or other kind of messages which can be processed by a script?)\ |
| 10:53.50 | *** join/#asterisk qdk_ (n=qdk@88.151.41.234) |
| 11:01.38 | *** join/#asterisk beeew (n=chatzill@c-24-7-43-146.hsd1.ca.comcast.net) |
| 11:02.16 | beeew | hi guys. i just installed asterisk, without anything else (library, zaptel, addon) and nw i'm trying to do 'agi debug' from the CLI.. |
| 11:02.32 | beeew | it's not workng..did i have to compile something else for AGI? |
| 11:06.46 | tzafrir_home | beeew, agi comes from the module res_agi.so |
| 11:07.05 | tzafrir_home | do you have "agi" commands in the CLI? |
| 11:07.19 | tzafrir_home | if not, try: module load re_agi.so |
| 11:07.21 | beeew | crap, i thought agi was standard.. |
| 11:07.58 | tzafrir_home | most of asterisk's features are in modules. The standard configuration is to "just load all modules" in modules.ocnf |
| 11:08.23 | tzafrir_home | that is "autoload = yes" |
| 11:08.28 | beeew | yay that worked! |
| 11:08.39 | tzafrir_home | not optimal, but helps if you don't know your way around |
| 11:08.41 | beeew | do loading these modules cost me RAM? |
| 11:09.12 | tzafrir_home | RAM is not really the issue. load time and chances for bugs is probably more of an issue |
| 11:09.38 | tzafrir_home | tae a look at the sample modules.conf . It should be in /etc/asterisk/modules.conf |
| 11:09.53 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 11:11.12 | beeew | thanks..looks like i'm going to be investigating what each of these modules do..(i didn't like waiting so long with my previous trixbox) |
| 11:12.26 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 11:12.48 | roxlu | how can I check if asterisk i connected with my voip-subscription? |
| 11:18.37 | roxlu | I'm looking at some example configs and often I see lines like: exten => s,1,Dial(SIP/cisco1)... what is that "s" ? |
| 11:19.24 | beeew | tzafrir_home that's weird, i dont have such a file modules.conf |
| 11:20.06 | tzafrir_home | so the defaults were used, and by default practically no module is loaded |
| 11:20.16 | tzafrir_home | get the one from the samples directory |
| 11:20.50 | beeew | i did find / -name modules.conf and it returned nothing.. |
| 11:22.01 | beeew | ak, there's no samples directory, cause i dont think i installed the samples thing |
| 11:22.04 | beeew | brb.. |
| 11:22.58 | roxlu | someone? |
| 11:23.35 | tzafrir_home | roxlu, sip show subscriptions |
| 11:23.44 | roxlu | thanks ! |
| 11:23.57 | roxlu | tzafrir_home: ... but what is that "s", is that a special flag in Asterisk? |
| 11:26.19 | *** join/#asterisk GaryH (n=chatzill@2001:8b0:127:0:21b:21ff:fe02:dfd4) |
| 11:29.03 | tzafrir_home | roxlu, 's' is an extension. It is indeed some sort of special name, but it relates to where you are in the dialplan, and not to the action you take |
| 11:29.17 | roxlu | hmm okay |
| 11:29.29 | tzafrir_home | The ection take in that sample command is "Dial(SIP/cisco1)" |
| 11:29.38 | roxlu | but that: sip show subscriptions command is not working in the CLI |
| 11:37.41 | roxlu | tzafrir_home: how can I check if my register is working? |
| 11:38.52 | *** join/#asterisk trekroadie (n=chrislro@c-76-114-144-30.hsd1.md.comcast.net) |
| 11:39.18 | tzafrir_home | roxlu, maybe I had a typo. Use tab completion |
| 11:39.33 | trekroadie | any one using the new Asterisk appliance that can help with configuration |
| 11:39.51 | roxlu | ok |
| 11:39.57 | roxlu | ah.. I see Got 404 Not found on SIP register to service now |
| 11:40.54 | *** join/#asterisk s0lid (n=jlq@210.213.198.98) |
| 11:42.56 | trekroadie | anyone got the Asterisk appliance to work with the demo connected to a phone line (FXO) ports |
| 11:43.17 | *** join/#asterisk ciohsb (n=ciohsb@130.60.242.61) |
| 11:45.09 | ciohsb | Hi, anybody there? |
| 11:46.50 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 11:49.51 | roxlu | yes.. but really quite |
| 11:52.45 | *** join/#asterisk jozu (i=torrent@84.79.51.163) |
| 11:52.51 | jozu | hi to all |
| 11:53.41 | roxlu | hi |
| 11:53.47 | jozu | i have a dude, its possible to put an IVR for inconming and outcoming call's on a SIP? |
| 11:56.05 | jozu | the SIP is a GSM GATEWAY, and have number 6XXXXXX |
| 11:56.20 | tzafrir_home | jozu, sure |
| 11:56.49 | tzafrir_home | do some basic reading on the Asterisk dialplan |
| 11:56.49 | roxlu | how can I use asterisk to use my voip-subscription? (when I do sip show peers, I see that i'm connected) |
| 11:57.10 | tzafrir_home | unlike the IVR in the sample config, you'll probably not want to use the "s" extension there |
| 11:58.55 | beeew | roxlu the s stands for 'start', as most calls start with the s extension.. |
| 12:00.27 | roxlu | okay thanks, I've added the register in the [general] of my sip.conf. Now, I want to test a outgoing/incoming call... but I'm not sure what I need to add to my configurations? |
| 12:14.25 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 12:35.22 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 12:36.14 | roxlu | I'm receiving hundreds of messages like this in the CLI: handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 |
| 12:36.57 | *** join/#asterisk mcquaid (n=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 12:37.46 | mcquaid | I tried asterisk a couple of years back but in the end gave up because of nat issues. I've read that it has gotten better in this regard. |
| 12:38.06 | mcquaid | My question is, can asterisk work even if I don't have access to the firewall? |
| 12:38.39 | mcquaid | What basically happened back then was, outgoing calls (using sip through my voip provider) were fine, but incoming calls had no audio |
| 12:39.24 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 12:39.28 | mcquaid | Some other voip software (ekiga) gave me the same issue, but software like linphone, and my hardware spa 2100 work without any configuration of the firewall |
| 12:40.03 | mcquaid | I always wondered what was my ata and linphone doing to traverse lan that asterisk, ekiga were not. |
| 12:40.24 | mcquaid | And wanted to confirm if progress had been made dealing with nat |
| 12:41.33 | mcquaid | btw, i don't using stun or a proxy server with linphone, or my ata, I just set up the account for voip provider and it works |
| 12:42.24 | *** join/#asterisk muh-die-kuh (n=hco@217.79.190.99) |
| 12:43.21 | muh-die-kuh | hi |
| 12:47.52 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:50.00 | beeew | hi. i have a home asterisk server and i'd simply like to transfer all the config files over to my new server, sip registry, sip trunk config, etc.. |
| 12:53.01 | beeew | would all files in /etc/asterisk suffice? |
| 12:53.10 | beeew | i'm just gonna tar em all up here.. |
| 12:58.54 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:02.17 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:2) |
| 13:04.48 | bminish | any idea how do debug the following incoming SIP or IAX calls wont ring my zaptel extensions but ring to the sip Extensions. incoming zaptel calls ring all extensions problem arose in asterisk update from ver 1.4.1 to 1.4.11 (& later versions) |
| 13:05.57 | bminish | Dial("IAX2/blueface2-out-8", "SIP/brendan&Zap/2&Zap/1&SIP/bmwifi|30|two") in new stack # this does not ring on zaptel extentsions |
| 13:06.06 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 13:06.17 | bminish | <PROTECTED> |
| 13:07.26 | *** join/#asterisk snuff-home (n=snuffy@C-59-101-243-89.bur.connect.net.au) |
| 13:07.34 | *** part/#asterisk snuff-home (n=snuffy@C-59-101-243-89.bur.connect.net.au) |
| 13:11.30 | bminish | for incoming IAX calls it is incorrectly detecting zaptel lines as busy |
| 13:11.49 | bminish | why would it be doing that? |
| 13:12.25 | bminish | it's a TDM400P |
| 13:12.36 | bminish | 2x fxo 2x fxs |
| 13:13.06 | bminish | latest release zaptel drivers |
| 13:18.03 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.89) |
| 13:34.10 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:34.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:39.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:52.04 | *** join/#asterisk anthm_mobile (n=anthm@h46088958.area4.spcsdns.net) |
| 14:06.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:06.41 | *** join/#asterisk bantu (n=Miranda@vpnwww01.rz.uni-karlsruhe.de) |
| 14:16.45 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:22.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:32.39 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:34.21 | *** join/#asterisk bantu (n=Miranda@vpnwww01.rz.uni-karlsruhe.de) |
| 14:38.30 | *** join/#asterisk bintut (n=bintut@cm1.gamma179.maxonline.com.sg) |
| 14:39.47 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 14:40.06 | katsuodo | Hello Everyone |
| 14:41.17 | katsuodo | by default the dial() application will always be priority n+101? |
| 14:41.34 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 14:44.24 | katsuodo | jbot are you in? |
| 14:47.16 | katsuodo | can anyone answer this question? |
| 14:49.18 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.89) |
| 14:56.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:56.43 | *** join/#asterisk cens0r (n=you@p54BE5957.dip.t-dialin.net) |
| 14:56.47 | cens0r | hi all |
| 14:57.56 | cens0r | i set the timeout for a Dial() in my extensions.conf to 30 seconds, but it always timeouts much earlier - even if i set it to something much higher - what could be wrong? |
| 14:58.44 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 14:59.59 | katsuodo | interesting I am having the same issue set at 20 seconds |
| 15:00.23 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 15:00.26 | cens0r | even if i remove the timeout, it goes to s-NOANSWER after +/- 8 seconds |
| 15:03.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:05.13 | *** part/#asterisk ussrback (n=MAX@80.92.183.84) |
| 15:10.48 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:17.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:18.05 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 15:22.50 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:22.50 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:32.50 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 15:36.10 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 15:39.07 | cens0r | hmm, looks like it was a problem with MOH? threw it out, now it works |
| 15:47.08 | *** join/#asterisk lukketto (n=lukketto@host60-36-dynamic.6-87-r.retail.telecomitalia.it) |
| 15:49.45 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 15:50.11 | ussrback | what command can i use to get my callerid? |
| 15:50.56 | blitzrage | eh? |
| 15:51.02 | blitzrage | show function CALLERID |
| 15:51.57 | ussrback | i mean from AGI |
| 15:52.09 | blitzrage | you didn't say that |
| 15:53.30 | ussrback | yeah that my fault :P |
| 15:54.15 | blitzrage | try: agi show get variable |
| 15:55.02 | ussrback | ok |
| 15:55.15 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 15:55.26 | ussrback | is it possible to set such option for meetme, |
| 15:55.44 | blitzrage | I don't understand the question |
| 15:55.49 | ussrback | that when in empty room enterrs |
| 15:55.55 | ussrback | any user |
| 15:55.59 | ussrback | first user |
| 15:56.11 | ussrback | to give him admin rights? |
| 15:56.48 | [TK]D-Fender | ussrback, thats what PIN's are for |
| 15:57.03 | blitzrage | sure, check to see if there are any users in the conference before you execute MeetMe() with the appropriate options |
| 15:57.13 | ussrback | no, without using PIN |
| 15:57.50 | blitzrage | MeetMeCount(1234|CONF_COUNT) |
| 15:59.34 | ussrback | let say there are 3 conference rooms, 1-2-3, if conference room 1 is ampty and you enter this conference, you will became Admin. |
| 15:59.46 | ussrback | any first attendant becames an Admin |
| 15:59.46 | blitzrage | MeetMe(1234,${IF($[${CONF_COUNT} > 1]?a)}) |
| 15:59.59 | blitzrage | ussrback: see above |
| 16:00.10 | ussrback | ohhh i seee nowww |
| 16:00.14 | blitzrage | that is wrong actually |
| 16:00.22 | blitzrage | MeetMe(1234,${IF($[${CONF_COUNT} = 1]?a)}) |
| 16:00.25 | blitzrage | but you get the idea |
| 16:00.33 | blitzrage | s/1/0/ |
| 16:00.40 | blitzrage | (I'm a bit hung over this morning...) |
| 16:00.59 | blitzrage | MeetMe(1234,${IF($[${CONF_COUNT} = 0]?a)}) |
| 16:01.03 | blitzrage | should work |
| 16:01.37 | ussrback | ok thanks |
| 16:02.07 | ussrback | is there any way to call from asterisk to Skype users? |
| 16:02.15 | blitzrage | yes, but no |
| 16:02.32 | blitzrage | there is a hack to make it work... but don't bother |
| 16:02.34 | ussrback | :) between yes and no |
| 16:02.41 | ussrback | why? |
| 16:02.43 | blitzrage | and only 1 channel |
| 16:02.48 | blitzrage | there is no chan_skype |
| 16:03.10 | blitzrage | codefreeze: if I started getting a huge dialplan... what would be more efficient? |
| 16:03.13 | *** join/#asterisk Thus0 (n=nThus0@86.69.158.114) |
| 16:03.25 | blitzrage | MeetMeCount(1234,CONF_COUNT) |
| 16:04.31 | blitzrage | ExecIf($[${CONF_COUNT} = 0]|Set|CONF_OPTIONS=a) |
| 16:04.41 | blitzrage | MeetMe(1234,${CONF_OPTIONS}) |
| 16:04.42 | blitzrage | or |
| 16:05.00 | blitzrage | are you better off doing as much as possible in a single line with multiple dialplan functions? |
| 16:05.10 | blitzrage | (like I described above) |
| 16:05.23 | [TK]D-Fender | ussrback, ... |
| 16:05.25 | [TK]D-Fender | ~skype |
| 16:05.25 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 16:06.45 | ussrback | [TK]D-Fender: what about gtalk and msn? |
| 16:07.14 | blitzrage | there is a chan_gtalk |
| 16:07.29 | blitzrage | there is nothing for MSN |
| 16:07.51 | *** join/#asterisk stunsch (n=stunsch@18.Red-88-27-153.staticIP.rima-tde.net) |
| 16:08.09 | ussrback | ok |
| 16:09.12 | stunsch | I set an asterisk box with 2 BRI PCI-cards. I'm unable to place calls to the outside |
| 16:09.41 | stunsch | I have created a context in extensions.conf to handle outgoing calls like this |
| 16:09.54 | *** part/#asterisk Thus0 (n=nThus0@86.69.158.114) |
| 16:10.00 | stunsch | exten=> _9X.,1,Dial(isdn/g:isdn/${EXTEN},,r) |
| 16:10.10 | stunsch | exten=> _9X.,n,Hangup() |
| 16:10.50 | stunsch | But softphone says "no route to destination". |
| 16:11.35 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:13.08 | stunsch | isdn is the name of the misdn group in the misdn.conf file |
| 16:15.31 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 16:22.47 | *** join/#asterisk blq (n=Bl@dslb-088-064-135-156.pools.arcor-ip.net) |
| 16:22.47 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 16:23.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:23.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:27.36 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 16:29.39 | [TK]D-Fender | stunsch, show us the CLI output of a failed call attempt |
| 16:31.13 | *** join/#asterisk bantu (n=Miranda@dslb-088-064-177-073.pools.arcor-ip.net) |
| 16:32.29 | [TK]D-Fender | ~pb |
| 16:32.29 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:32.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 16:39.29 | *** join/#asterisk saftsack (n=saftsack@pD9E07316.dip.t-dialin.net) |
| 16:43.18 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 16:44.21 | kombi | I'm having trouble compiling zaptel, it complains about missing usb support in the kernel. Can I disable module xxp somehow? |
| 16:45.44 | *** join/#asterisk matsk (n=mk@host-217-213-149-243.mobileonline.telia.com) |
| 16:50.38 | ussrback | http://pastebin.ca/743631 ---- it automaticaly executes hangup |
| 16:50.45 | ussrback | whats a problem ? |
| 16:53.04 | ussrback | ok i found |
| 16:53.22 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 16:58.04 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 17:05.34 | *** join/#asterisk hi365_m (n=hi365@213.151.59.7) |
| 17:06.42 | kombi | never seen it this quiet in here.. |
| 17:07.33 | hi365_m | asterisk seems to be sending my packets to the local ip (instead of the remote ip)! |
| 17:07.48 | hi365_m | bit my rtp settings and the exterip seems to be correct |
| 17:07.59 | [TK]D-Fender | ~pb |
| 17:07.59 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:08.39 | kombi | Fender! long time no see.. |
| 17:08.49 | hi365_m | [TK]D-Fender:me? |
| 17:08.54 | [TK]D-Fender | Yes |
| 17:10.02 | hi365_m | im not sure what exzctly you want to see... http://pastebin.ca/743640 |
| 17:10.20 | hi365_m | would you like to see the rtp debug as well? |
| 17:10.58 | [TK]D-Fender | hi365_m, You whole [general] section + any peers involved |
| 17:11.27 | [TK]D-Fender | hi365_m, And SIP debug of a call would be nice |
| 17:11.33 | hi365_m | sure |
| 17:12.01 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:13.19 | hi365_m | [TK]D-Fender: can i pm the links? |
| 17:13.59 | [TK]D-Fender | sure |
| 17:15.48 | *** join/#asterisk i3inary (i=i3inary@ip72-207-113-253.sd.sd.cox.net) |
| 17:18.57 | hi365_m | [TK]D-Fender:sorry for the delay |
| 17:20.13 | *** join/#asterisk stratadiddy (n=paulkoch@ip72-192-170-113.sd.sd.cox.net) |
| 17:21.36 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 17:27.22 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 17:31.00 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:32.11 | *** join/#asterisk KE6IGZ (i=dave@cpe-66-75-70-232.socal.res.rr.com) |
| 17:33.57 | knarfly | RypPn, so your name is Long Duck Dong....! |
| 17:34.45 | RypPn | put down the crackpipe and step away from the keyboard... |
| 17:36.43 | stunsch | [tk]O-Fender: CLI does not say anything. I've managed to make inbound calls to work. But with outbound calls I've had no success. |
| 17:37.03 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:39.26 | [TK]D-Fender | stunsch, enable SIP debug try again, and pastebin it. |
| 17:40.46 | stunsch | [TK]O-Fender:SIP debug? Sorry, but my asterisk knowlege is rather small... |
| 17:41.08 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 17:41.22 | [TK]D-Fender | stunsch, "sip debug" in * CLI |
| 17:42.26 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 17:43.25 | [TK]D-Fender | ~pb |
| 17:43.25 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:43.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 17:48.10 | stunsch | [TK]O-Fender:Here's it. http://pastebin.com/d144d9673 |
| 17:48.49 | stunsch | The call has been done from extension 300 on a softphone to 972672547 telephone number |
| 17:50.42 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 17:51.00 | luke-jr | yay, so Asterisk 1.4 decides to crash when my baby is born |
| 17:51.10 | luke-jr | and hang if I try to start it |
| 17:51.11 | luke-jr | -.- |
| 17:51.35 | [TK]D-Fender | luke-jr, Did you name your child "Segfault" in memory of the occasion? |
| 17:51.40 | luke-jr | no |
| 17:51.59 | luke-jr | he was named some months ago |
| 17:52.18 | [TK]D-Fender | stunsch, Looking for 972672547 in default (domain 192.168.0.243) SIP/2.0 404 Not Found <--- dialplan error |
| 17:52.57 | luke-jr | so anyhow |
| 17:53.12 | luke-jr | how do I figure out why Asterisk has decided to randomly break? |
| 17:54.57 | [TK]D-Fender | ~8ball will luke-jr find what he is looking for? |
| 17:54.57 | jbot | No. |
| 17:55.04 | luke-jr | … |
| 17:55.06 | [TK]D-Fender | luke-jr, give up while you're ahead :) |
| 17:55.12 | luke-jr | um |
| 17:55.15 | [TK]D-Fender | pwned |
| 17:55.16 | luke-jr | so stop using Asterisk? |
| 17:55.30 | stunsch | dialplan error? |
| 17:55.36 | *** join/#asterisk saftsack (n=saftsack@pD9E07316.dip.t-dialin.net) |
| 17:55.38 | [TK]D-Fender | luke-jr, no, just don't waste time looking for the answer and wait for it to fix itself :) |
| 17:55.48 | luke-jr | [TK]D-Fender: :/ |
| 17:55.57 | luke-jr | stunsch: absolutely nothing has changed |
| 17:56.08 | luke-jr | wtf |
| 17:56.11 | luke-jr | it works in strace |
| 17:56.13 | luke-jr | but not outside |
| 17:56.15 | [TK]D-Fender | stunsch, Correct. You don't have an extension in [default] that can match the 972672547 that ws dialed |
| 17:56.26 | UnixDog | luke there are only 9 numbers in the phone number |
| 17:56.35 | UnixDog | your missing a digit |
| 17:56.44 | luke-jr | … |
| 17:56.54 | stunsch | it's not an extension. That's and outbound telephone number |
| 17:57.02 | UnixDog | your missing a digit |
| 17:57.06 | stunsch | This should place a call to the outside |
| 17:57.09 | UnixDog | look atthe fricking number |
| 17:57.18 | stunsch | through misdn |
| 17:57.18 | [TK]D-Fender | UnixDog, Wake and realize who's conversation you're trying to track and please don't assume what you think it takes for a number to be VALID. |
| 17:57.19 | *** part/#asterisk hi365_m (n=hi365@213.151.59.7) |
| 17:57.49 | [TK]D-Fender | UnixDog, I can make ANY extension pattern I feel like and it will be RIGHT. Just don't think that your rules apply to me or anyone else. |
| 17:58.30 | [TK]D-Fender | stunsch, so go look at the extens in [default[ and see what you're missing for that # you dialed to match the way you'd like it to |
| 18:01.49 | UnixDog | ok tk I was going by the fact 972 is a texas arecode |
| 18:02.15 | UnixDog | and there for the number he dialed would eb missing a nuber |
| 18:03.11 | UnixDog | he also said it was a outbound sip call |
| 18:03.23 | UnixDog | so I just put 2 and 2 together |
| 18:04.23 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 18:05.50 | [TK]D-Fender | UnixDog, and ASSUMED he was even calling to North America. |
| 18:06.08 | [TK]D-Fender | ~assume |
| 18:06.09 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 18:08.51 | [TK]D-Fender | UnixDog, Who dials to North America using **MISDN** with a 10 digit number? |
| 18:09.10 | [TK]D-Fender | stunsch, So... did you find your error yet? |
| 18:14.17 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 18:14.55 | luke-jr | hey, strace is hanging now |
| 18:15.01 | luke-jr | on a futex |
| 18:18.49 | kombi | on a b410p, how do I stop the leds from flashing on unused ports? |
| 18:21.10 | [TK]D-Fender | kombi, electrical tape |
| 18:21.22 | kombi | lol.. |
| 18:22.27 | kombi | just slightly annoying.. I though maybe there was some option in the driver |
| 18:23.26 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:23.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:25.01 | *** join/#asterisk ManxPower (n=manxpowe@160.sub-70-222-47.myvzw.com) |
| 18:28.22 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 18:29.50 | roxlu | hi everyone! |
| 18:30.10 | roxlu | does someone knows a good softphone for the mac, which let you add multipl SIPsA |
| 18:30.13 | roxlu | ? |
| 18:33.40 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-37.dynamic.qsc.de) |
| 18:34.55 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:36.52 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:37.05 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 18:40.45 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:47.16 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:47.18 | *** join/#asterisk Lisa696 (i=julian@200.58.204.164) |
| 18:48.31 | nestAr | Strangly, phone company claimed my config was the problem.. I go home, I come back this morning.. and it works. |
| 18:48.41 | nestAr | love the phone company. |
| 18:49.20 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 18:50.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:54.30 | *** join/#asterisk gardo (n=gardo@121.97.177.221) |
| 18:57.26 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:58.31 | *** join/#asterisk saftsack (n=saftsack@pD9E07316.dip.t-dialin.net) |
| 18:59.23 | *** join/#asterisk saftsack (n=saftsack@pD9E07316.dip.t-dialin.net) |
| 19:00.14 | *** join/#asterisk saftsack (n=saftsack@pD9E07316.dip.t-dialin.net) |
| 19:03.47 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 19:04.01 | codefreeze | blitzrage: still around/conscious? |
| 19:09.33 | *** join/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 19:09.55 | *** part/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 19:10.15 | wwalker | anyone else have trouble with 1.4.13 building and then dying with "illegal instruction" immediately upon execution? |
| 19:12.44 | tzafrir_home | wwalker, sounds like a build system problem |
| 19:13.02 | tzafrir_home | Please give more details on your system |
| 19:13.12 | tzafrir_home | does it pass ./configure ? |
| 19:15.09 | wwalker | appears to be a known problem with via chipset. GOing to try something now. |
| 19:16.36 | *** join/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 19:16.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:17.02 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 19:17.08 | citats | i've seen certain chipsets require compilation with PROC=i586 rather than the detected PROC=i686 |
| 19:17.31 | Qwell | broken friggen i586 kernels.. |
| 19:18.19 | citats | i use a via 800mhz at home that will only work properly if it gets changed to i586 |
| 19:18.51 | wwalker | Qwell: broken hardware from via. it's almost 686 |
| 19:19.27 | DaneM | Hello, all. I'm having an odd problem. I have a Tiger3xx modem (according to lspci), with 2 fxo modules (connected to the phone company), and 2 fxs modules (connected to handsets). Everything seems to be working fine, except that one of the fxs lines doesn't generate a dialtone, and won't take incoming calls. When I pick it up immediately after Asterisk starts, I get some screaching, then silence with a slight echo. If I pick i |
| 19:21.11 | tzafrir_home | wwalker, which CPU exatly? What is the model name in /proc/xpuinfo ? |
| 19:22.09 | wwalker | tzafrir_home: VIA C3 Ezra |
| 19:22.24 | citats | wwalker: edit your makeopts and change PROC=i686 to PROC=i586 |
| 19:23.09 | tzafrir_home | hmmm... I think Ezra is indeed not a "VIA C3-2", and I think its i686 compatibility is indeed poor. |
| 19:23.47 | citats | its the same chip i have on one system and thats the solution to it |
| 19:24.10 | wwalker | tzafrir_home: thx, I was running "make PROC=i586 " |
| 19:24.27 | tzafrir_home | BTW: VIA chipset does not imply "VIA CPU". It does imply, however, "crappy on-board display adapter that performs poorly", FWIW |
| 19:24.27 | wwalker | citats: thx |
| 19:24.41 | wwalker | what video? |
| 19:24.42 | wwalker | <PROTECTED> |
| 19:26.03 | tzafrir_home | that was my desktop at work (crappy amd64 ocmputer with a via chipset). And yeah: asterisk can run on it |
| 19:27.53 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 19:28.10 | blitzrage | codefreeze: back |
| 19:32.28 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:34.31 | DaneM | Can anybody help me with my no dialtone on fxs line problem? |
| 19:34.41 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 19:34.48 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e9bffcae0cae9db3) |
| 19:34.48 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:35.57 | DaneM | Here is the link with my problem's description, and some of my config files: http://www.asteriskguru.org/board/here-vp6848.html#6848 |
| 19:36.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-213-14.dsl.hstntx.swbell.net) |
| 19:38.39 | codefreeze | blitzrage: speed: more in fewer instructions... tip: if you are going to spend time to optimize, optimize the number of extensions, before you optimize the number of priorities. |
| 19:39.29 | blitzrage | codefreeze: so as a call is following through the dialplan, the total number of unique extensions matters, even if the call isn't hitting each of them? |
| 19:41.31 | Cresl1n | blitzrage: !!!!! |
| 19:43.29 | blitzrage | Cresl1n: !!!!!!! |
| 19:43.55 | Cresl1n | what are you up to this saturday? |
| 19:44.21 | blitzrage | cleaning mostly, listening to music... getting ready for the hockey game tonight |
| 19:44.22 | blitzrage | you? |
| 19:44.39 | Cresl1n | nice |
| 19:44.43 | Cresl1n | just doing some working |
| 19:44.47 | blitzrage | hot |
| 19:44.59 | blitzrage | I got my server reimaged... so it's nice and clean again |
| 19:45.10 | Cresl1n | like a spring cleaning |
| 19:45.13 | blitzrage | got a laptop to still do at some point, but probably not today |
| 19:45.20 | blitzrage | ya -- and its sunny and spring like outside :) |
| 19:45.39 | blitzrage | I've never seen it so warm on Oct 20th here |
| 19:46.14 | blitzrage | at some point might work on the asterisk cookbook outline again and get that finished up for tomorrow so I can write some recipes |
| 19:47.48 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 19:49.21 | De_Mon | I would like one for autocallback queues plz |
| 19:49.39 | CCFL_Man2 | anyone get binary data working on a terminal server? |
| 19:51.23 | blitzrage | De_Mon: I plan on doing stuff on queues, yes |
| 19:51.59 | blitzrage | I basically plan on documenting the process of creating a PBX for several sizes of business |
| 19:53.38 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:53.42 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:55.05 | De_Mon | are you going to make extensive use of macros or stick with straight dialplans? |
| 19:55.28 | De_Mon | when I redo this system, I wana macro some of the featuers to make them more modular than they are right now |
| 19:58.45 | *** join/#asterisk enalert (i=trelane@2001:4830:150c:0:20d:61ff:fe31:a58) |
| 19:58.54 | enalert | how do I generate 3 rising tones on an incoming call with Asterisk? |
| 19:58.56 | enalert | (sip) |
| 19:59.52 | [TK]D-Fender | ? |
| 20:00.15 | enalert | [TK]D-Fender, the "info" rising tones. playtones doesn't work in SIP |
| 20:00.40 | [TK]D-Fender | enalert, go record it and play it back |
| 20:01.17 | enalert | I'm sure I can find a wav file online if I'm going that route :) |
| 20:01.18 | enalert | thanks |
| 20:01.26 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:02.03 | DaneM | D-Fender, your suggestions about my queue problem yesterday fixed it! Thanks |
| 20:03.09 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:03.23 | *** join/#asterisk disa-help (n=phobosd@shell.intarwebnetorg.com) |
| 20:03.44 | [TK]D-Fender | DaneM, You're welcome |
| 20:03.56 | disa-help | afternoon TK :) |
| 20:06.05 | ManxPower | enalert: run Answer, then playtones |
| 20:06.49 | ManxPower | also make SURE you have a /etc/asterisk/indications.conf |
| 20:07.07 | roxlu | ManxPower: |
| 20:07.10 | roxlu | oh sorry |
| 20:07.34 | enalert | ManxPower, I do, my default indication is set to us and playtones(info) stopplaytones() doesnt do the trick |
| 20:08.00 | DaneM | Would somebody mind taking a look at this? http://www.asteriskguru.org/board/here-vp6848.html#6848 I'm really stumped. |
| 20:08.06 | roxlu | I'm testing my voip account, which does work when I use it directly in x-lite softphone.. but not when I use it through asterisk. Can someone help me a bit with debuggin? |
| 20:08.29 | ManxPower | enalert: do you run Answer before the Playtones? |
| 20:08.39 | enalert | ManxPower, of course |
| 20:08.44 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:09.17 | ManxPower | enalert: odd. IT should work just fine, after an Answer() then all tones are INBAND (before Answer they are out of band) and Playtones, etc use indications.conf to know what tones to play. |
| 20:09.35 | enalert | ManxPower, I even tried specifying the tones by frequency |
| 20:09.46 | enalert | <PROTECTED> |
| 20:09.50 | enalert | vs Playtones(info) |
| 20:10.05 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 20:10.08 | ManxPower | enalert: then I don't know what might be causing that issue. |
| 20:10.18 | ManxPower | oh! try progressinband=yes in sip.conf [general] |
| 20:10.45 | ManxPower | it is an ugly workaround that I had to use yesterday to get ringback working with in a specific situation |
| 20:13.42 | roxlu | ManxPower: .. when I want to make an outgoing call, do I need to use: Dial(SIP/some_number@provider) ? |
| 20:13.52 | roxlu | or w/o the SIP after Dial() ? |
| 20:17.01 | [TK]D-Fender | roxlu, "show application dial" |
| 20:17.19 | enalert | ManxPower, let me try that real quick |
| 20:18.23 | wwalker | I've got a spa-3102 setup. I see it in sip peers, I think I've set it up right, but I get no sip messages when I call it. |
| 20:18.50 | roxlu | [TK]D-Fender: can you help me a bit? I see info line, with "Executing [..] Dialing..." |
| 20:19.14 | [TK]D-Fender | roxlu, read the INSTRUCTIONS for dial. |
| 20:19.14 | enalert | ManxPower, doesn't help. I can see that it's trying to play it -- Executing PlayTones("SIP/3174361024-08198af0", "!950/330|!1400/330|!1800/330|0") in new stack |
| 20:20.00 | [TK]D-Fender | wwalker, pastebin your peer and your attempt with sip debug enabled. also reconfirm its activation in your pastebin |
| 20:20.01 | [TK]D-Fender | ~pb |
| 20:20.02 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:20.15 | [TK]D-Fender | wwalker, and tell us where it is located relative to your * server |
| 20:20.25 | roxlu | [TK]D-Fender: I think my dial settings are correct: exten => _X.,n,Dial(SIP/${EXTEN}@provider.com) |
| 20:20.53 | [TK]D-Fender | roxlu, good for you then. |
| 20:20.57 | wwalker | asterisk server and spa-3102 are both on the private network inside the firewall at my house. |
| 20:22.12 | enalert | ManxPower, http://www.pastebin.ca/743838 has the script being used (the call is Answer()'d before being directed there) |
| 20:23.18 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-e9bffcae0cae9db3) |
| 20:23.27 | *** join/#asterisk jtoy (n=jtoy@c-76-24-253-215.hsd1.ma.comcast.net) |
| 20:23.34 | enalert | <PROTECTED> |
| 20:23.52 | jtoy | could someone recommend a service where I can get a free US incoming number that I can hook up to asterisk? |
| 20:24.32 | TrentCreek | didx.com |
| 20:24.40 | wwalker | [TK]D-Fender: http://rafb.net/p/TuDUVV46.html |
| 20:25.10 | roxlu | [TK]D-Fender: Can you maye have a look at this: http://paste-it.net/4036, I'm wondering how I can change the @phones:2? or what does this mean> |
| 20:25.25 | wwalker | [TK]D-Fender: with sip debug enabled, there is NO sip message when I call the house (which the SPA is plugged into. |
| 20:25.59 | [TK]D-Fender | roxlu, thats just telling you what part of your dialplan you are executing |
| 20:26.05 | [TK]D-Fender | wwalker, PASTEBIN |
| 20:26.33 | wwalker | sorry, have a button for rafb. what's the difference? |
| 20:26.47 | [TK]D-Fender | wwalker, AH.... well this shows me NOTHING about how you set your SPA to handle calls. |
| 20:27.04 | roxlu | [TK]D-Fender: ahh... thats my exten => outgoing_calls line in [phones] |
| 20:27.15 | *** join/#asterisk axisys (n=axisys@outbound.silenceisdefeat.org) |
| 20:27.53 | roxlu | [TK]D-Fender: do you see a trivial error on that line, which could explain why I can't use my voip account? |
| 20:27.54 | wwalker | the point is the SPA is not sending anything to asterisk. once I have messages from the spa to asterisk, I can probably figure that out. |
| 20:27.56 | blitzrage | De_Mon: I'll be making extensive use of GoSub's instead of Macro's |
| 20:28.28 | blitzrage | I try to employ modular programming practices into my dialplan and reutilize code |
| 20:28.30 | [TK]D-Fender | wwalker, Ok.... I'm only psychic on tuesdays. see you then! |
| 20:28.45 | wwalker | [TK]D-Fender: the real question is "what config on the SPA caused it to forward a call to asterisk when the call comes in?" I've appartenly missed or misconfigured that setting |
| 20:29.19 | [TK]D-Fender | wwalker, you have to set it up to answer. and set its dialplan to dial an exten on * |
| 20:29.39 | [TK]D-Fender | wwalker, www.voxilla.com <- their forums have some good guide |
| 20:29.40 | wwalker | [TK]D-Fender: since there are about 20 pages of configs, I don't even know which one to look on. Right, |
| 20:30.04 | wwalker | I was asking, what sets it to answer and dial *? OK, I'll look there. |
| 20:30.07 | [TK]D-Fender | wwalker, And actually there's only really 4 pages to go for this.... |
| 20:30.35 | [TK]D-Fender | wwalker, its in its dialplan. looks something like <S0:123456789> |
| 20:31.37 | wwalker | ah, that reads:(S0<:8179422517@192.168.0.55:5060>) |
| 20:34.54 | *** join/#asterisk jingles (n=dfbarth@12.146.41.2) |
| 20:35.37 | stratadiddy | anybody know of a voip provider that has an api for registering phone numbers? |
| 20:38.55 | wwalker | [TK]D-Fender: does (S0<:8179422517@192.168.0.55:5060>) |
| 20:39.24 | wwalker | look right? |
| 20:40.24 | [TK]D-Fender | wwalker, go read the guides an loot at the other flags you have to set |
| 20:41.11 | DaneM | In zapata.conf, what is the difference between group, callgroup, and pickupgroup? |
| 20:42.08 | [TK]D-Fender | DaneM, ... |
| 20:42.09 | [TK]D-Fender | ~wikis |
| 20:42.10 | jbot | it has been said that wikis is http://www.voip-info.org |
| 20:42.11 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 20:42.16 | DaneM | aah...sorry |
| 20:44.53 | tzafrir_home | DaneM, group is the number NNN you use for Dial(Zap/gNNN) |
| 20:45.30 | tzafrir_home | dialing to a group of phones, and letting chan_zap get "an available one" from them |
| 20:46.05 | tzafrir_home | callgroup and pickupgroup are for something different |
| 20:46.20 | *** join/#asterisk kanderson (n=kanderso@164.149.188.72.cfl.res.rr.com) |
| 20:47.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:51.59 | DaneM | thanks, tzafrir_home. Was reading wiki :-) |
| 20:52.28 | kanderson | Hey all, my production system recently started to delay the answering of channels by roughly five seconds. When you answer a call or take it off hold you can hear the other party but they can not hear you for five seconds. Any ideas, your help would be appreciated. |
| 20:54.08 | DaneM | When I pick up the handset connected to one of my fxs channels immediately after starting *, I get a screeching noise, then silence, with a slight echo. When I pick up that handset later, after having hung it up, I just get silence and the echo. Any idea what could be causing this? |
| 20:54.21 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-36-20.pskn.east.verizon.net) |
| 20:55.46 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:56.10 | codefreeze | blitzrage: yes, the total number of extensions in your CURRENT context (the one the next priority will come from), matters a great deal, because for every priority, it will search thru every extension to find the best match. |
| 20:56.16 | wwalker | [TK]D-Fender: read the setup guide. all the same settings the first one told me to do. Still no SIP messages to asterisk :( Does show the call received in Voice->Info :( |
| 20:56.42 | [TK]D-Fender | wwalker, keep reading |
| 20:56.44 | blitzrage | codefreeze: aha! the current context matters |
| 20:57.14 | DaneM | does the above problem sound like it has to do with hardware, software, or configuration? |
| 20:57.31 | codefreeze | of the 3, context names, priority numbers, and extension patterns, the extension patterns is the most expensive. |
| 20:57.49 | codefreeze | blitzrage, that is. |
| 20:58.03 | blitzrage | I see... does the number of contexts make a difference? |
| 20:58.44 | codefreeze | yes, because it will search the list of contexts, untils it finds the match. linear search. probability is that it will find it halfway thru the list. |
| 20:59.36 | codefreeze | blitzrage: extensions are expensive because it ALWAYS searches thru ALL of them (in the matching context). |
| 21:00.21 | wwalker | [TK]D-Fender: been reading for 3 hours, need to trouble shoot now, not read conflicting docs |
| 21:00.25 | *** join/#asterisk codec (n=codec@iglu.paranoid-penguin.de) |
| 21:01.16 | kanderson | DaneM: can make/place calls on the fxs? |
| 21:02.28 | blitzrage | codefreeze: ok, so say I'm executing a priority in [context_foo], and that context contains 2 extension patterns, and the currently executing extension has 10 priorities. That means Asterisk is going to parse both extension patterns 10 times during each of the priorities of the matching extension. This means I would get better performance if I did not have the 2nd extension pattern in that priority? |
| 21:02.29 | DaneM | kanderson: not on the one that gets no dialtone, but on my other fxs line, it works perfectly. |
| 21:03.14 | blitzrage | codefreeze: and does the number of priorities for the non-executing extension matter? (seems to be it would not) |
| 21:03.28 | kanderson | DaneM: it may be hardware, but it sounds like config. Has it ever worked or did this just develop? |
| 21:03.33 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 21:03.57 | codefreeze | blitrage: no, the number of prior's in the 'other' exten don't matter. |
| 21:04.52 | DaneM | this just developed yesterday, when I made some changes to extensions.conf and plugged in some splitters (for multiple phones). It worked the day before. I've tried without the splitters, and I *think* my extensions.conf is still OK... |
| 21:05.59 | codefreeze | blitzrage: as to the speed, if you didn't have the other exten, yes, it'd shuffle thru the priorities faster. It doesn't 'hit the wall', in practice, until you get above 20-40 extens in the current context. |
| 21:06.21 | *** part/#asterisk jtoy (n=jtoy@c-76-24-253-215.hsd1.ma.comcast.net) |
| 21:06.21 | kanderson | DaneM: extension.conf would not have caused this, but using splitters may have used to much current and damaged the unit. |
| 21:06.24 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 21:06.36 | DaneM | hmmm...bummer |
| 21:06.41 | blitzrage | codefreeze: what about when you're trying to get more simultaneous calls on a box? |
| 21:07.05 | DaneM | is it not possible/recommended to use multiple phones on my modem? |
| 21:07.11 | codefreeze | Well, 'to be the most you can be', you'll keep your dp simple. |
| 21:07.20 | DaneM | (kanderson) |
| 21:07.37 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 21:07.43 | blitzrage | codefreeze: because I'm curious what the effects would be to do a GoSub() to a separate context instead of doing it to an extension in the current context |
| 21:08.10 | kanderson | DaneM: Specialy when they are ringing it can draw suprising amounts of power, that is a question best answered by the manufacture. |
| 21:08.17 | syle | http://rafb.net/p/VLke6F63.html |
| 21:08.18 | blitzrage | thereby segregating the 2nd extension in another context |
| 21:08.32 | blitzrage | codefreeze: that'd be an interesting comparison |
| 21:08.39 | DaneM | kanderson: ok. Thanks |
| 21:08.44 | kanderson | np |
| 21:08.46 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) [NETSPLIT VICTIM] |
| 21:08.55 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 21:09.02 | codefreeze | blitzrage: you could also include less frequently occurring extensions... |
| 21:09.07 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 21:09.52 | codefreeze | blitzrage: when I switch to hashtabs, it will not matter. They will all execute same speed no matter the size. |
| 21:09.52 | blitzrage | I don't understand... |
| 21:10.11 | blitzrage | the first statement |
| 21:11.21 | *** join/#asterisk anthm (n=anthm@h4608a110.area4.spcsdns.net) |
| 21:11.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:11.25 | codefreeze | If you take all the normal extensions, like i, t, etc, and put them in a diff context, and include that other context, you can cut search times on the average. If they don't happen that much. |
| 21:12.10 | blitzrage | so those extensions are not "searched" unless the current context returns no match? |
| 21:12.18 | codefreeze | right. |
| 21:12.40 | blitzrage | makes sense... but if say the 'h' extension was being called at the end of every call, then it would be cheaper to not include from a separate context? |
| 21:13.02 | codefreeze | yes, if it happens freq. DON'T include it. Have it right there. |
| 21:13.06 | blitzrage | i.e. cheaper to exist in the executing context, instead of being included => |
| 21:13.19 | blitzrage | gotcha |
| 21:13.54 | codefreeze | Well, you add the overhead of dropping and searching the other included contexts. If it doesn't happen much, it won't cost much in terms of time. |
| 21:14.05 | blitzrage | ya, makes sense |
| 21:14.33 | blitzrage | this sounds like a good recipe for the "dialplans: best practices" section I wanna write |
| 21:14.54 | codefreeze | But, like I said, ... when I move to hashtabs, size won't matter. At all. |
| 21:15.31 | codefreeze | Except for realtime. Until it's re-engineered, I cannot apply hashtabs there. |
| 21:15.32 | blitzrage | asterisk will care more about how it's used, and not how big it is |
| 21:16.26 | codefreeze | Well, load/reload times begin to lift their ugly heads at some size point. There's tricks there, too, but... ugh. |
| 21:17.02 | blitzrage | ya, I'm not too concerned about those because I use func_odbc to populate dynamic data from the DB instead of building the DP from a script and reloading constantly |
| 21:17.09 | blitzrage | so that is rarely a concern for me |
| 21:17.09 | De_Mon | blitzrage gosub? |
| 21:17.17 | blitzrage | De_Mon: yep |
| 21:18.07 | codefreeze | dynamic data vs. dynamic instructions (dp is NOT in the db)? |
| 21:18.16 | De_Mon | saving the return address? so it goes to the context and when that ends it comes back to wherever it sub'ed out with that value? |
| 21:18.50 | codefreeze | De_Mon: u gotta say Return() somewhere... the biggest drawback. |
| 21:19.06 | codefreeze | most folks can handle that, tho |
| 21:19.09 | blitzrage | De_Mon: GoSubs do not have the stack size requirement that Macros do. Thus, you are better able to nest modular dialplan. For example, because of the huge stacksize of Macro(), you can't do Macro( Macro( func_odbc ) ) without crashing the stack (and thus asterisk) |
| 21:19.31 | blitzrage | codefreeze: correct -- the DP exists in a static file on the system (extensions.conf loaded from the filesystem) |
| 21:19.46 | De_Mon | hoom, intersting |
| 21:19.53 | blitzrage | ya... need to say Return(), but I've never found that to be an issue |
| 21:20.38 | codefreeze | yes, I try to convince people that the dialplan should be static, and simple, and if you can, use patterns and a DB instead so you don't mult. the extensions. Even with hashtabs, this is a better approach. |
| 21:20.39 | De_Mon | any variables set within the gosub are remembered right? |
| 21:20.51 | blitzrage | yes -- they are just channel variables |
| 21:21.10 | De_Mon | so... why do macros still exist |
| 21:21.15 | codefreeze | but you can do "local" vars in macros/gosubs now, too |
| 21:21.15 | blitzrage | the dialplan executes no differently than if you had just done a Goto() |
| 21:21.33 | blitzrage | De_Mon: because people still use them and they have to have a deprecation phase, and they don't know about them (yet) |
| 21:21.39 | De_Mon | except when you return() it comes back |
| 21:21.47 | blitzrage | I'm on the cutting edge -- people will learn more about them when they look at the func_odbc chapter in TFoT2 |
| 21:21.52 | blitzrage | De_Mon: correct |
| 21:21.59 | De_Mon | wicked cool |
| 21:22.02 | blitzrage | codefreeze: you can? |
| 21:22.09 | codefreeze | oh, and return() works in macros, too. |
| 21:22.09 | blitzrage | codefreeze: what is the format for that? |
| 21:22.22 | blitzrage | LOCAL() function? |
| 21:22.26 | De_Mon | I need a gosub replacement for stdexten |
| 21:22.26 | codefreeze | yep. |
| 21:22.30 | blitzrage | wow... what a guess! |
| 21:22.35 | blitzrage | asterisk is so logical! :D |
| 21:22.53 | blitzrage | Set(LOCAL(variable)=value) I guess is the format |
| 21:22.56 | codefreeze | AEL trunk is now automatically declaring the macro args as local. |
| 21:23.08 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:23.20 | De_Mon | blitzrage are you doing pure dialplan or AEL? |
| 21:23.32 | blitzrage | pure dialplan |
| 21:23.53 | De_Mon | doesn't sound so cutting edge any more O_O |
| 21:23.56 | blitzrage | I recreated the Voicemail() application in pure dialplan :) |
| 21:24.30 | Wonka | AEL rocks |
| 21:24.36 | blitzrage | I use the tool I'm most familiar with... codefreeze doesn't have time for me to learn AEL and start finding things I need right now :) |
| 21:24.40 | De_Mon | codefreeze that doesn't sound as cool |
| 21:25.06 | blitzrage | I've always liked the concept of AEL though -- I just need to learn it |
| 21:25.14 | file | blitzrage: add it to the list |
| 21:25.18 | blitzrage | file: no kidding |
| 21:25.42 | codefreeze | We needed to have NAMED the extensions.conf format something a while back, so we can refer to it more easily!!!!!! |
| 21:25.50 | [TK]D-Fender | blitzrage, However it will not make your dialplan the slightest bit more processor efficient :) |
| 21:26.27 | codefreeze | [TK]D-Fender: Oh, yes it will!!! In certain cases! |
| 21:26.34 | HarryR | uh , I just call it the dialplan.. |
| 21:26.58 | codefreeze | Naw, the dialplan is the sum of AEL + extensions.conf. |
| 21:27.02 | [TK]D-Fender | codefreeze, Since AEL parses back to "extensions.conf" code. how can it do a single thing BETTER than you have in extensions.conf yourself? :) |
| 21:27.18 | HarryR | codec, regardless of AEL, it's just the dialplan |
| 21:27.23 | codefreeze | It all gets mashed up together and ends up in the same structs. |
| 21:27.37 | blitzrage | [TK]D-Fender: slight processor efficiency does make a different when you're trying to max out an asterisk box with simultaneous calls, and when you create complex dialplans, like in the case of a clustered, ITSP/vPBX backend |
| 21:27.43 | Wonka | ael is better readable |
| 21:28.03 | codefreeze | [TK]D-Fender: You will find that using while(), and for() much faster than the apps While(), etc. |
| 21:28.05 | [TK]D-Fender | blitzrage, I understand that. And since all AEL does is get parsed back, all it can do is add BLOAT :) |
| 21:28.20 | HarryR | blitzrage, for anything complex, don't use the dialplan |
| 21:28.27 | [TK]D-Fender | codeits all the same extensions.conf logic at the end of the day. |
| 21:28.27 | blitzrage | no ol' sk00l dialplan can scare me anymore |
| 21:29.24 | blitzrage | HarryR: that is a funny statement to me because I'm the king of writing complex logic into the dialplan |
| 21:29.24 | [TK]D-Fender | codefreeze, So all AEL can do to processing efficiency is add bloat |
| 21:29.24 | HarryR | it just makes it hard to maintain if you do anything complex |
| 21:29.24 | blitzrage | HarryR: maybe for some people... |
| 21:29.25 | [TK]D-Fender | codefreeze, We're talking EXECUTION efficiency, not "code readability" |
| 21:29.26 | HarryR | writing it in a language which was designed to handle more complex things is logically better |
| 21:29.30 | file | the stuff blitzrage has done is crazy O.o |
| 21:29.37 | HarryR | like? |
| 21:29.44 | [TK]D-Fender | codefreeze, I get it perfectly, you've missed the topic :) |
| 21:29.46 | codefreeze | [TK]D-Fender: I've told you before-- checking; better syntax; etc; etc; all you see is the end result. |
| 21:30.27 | HarryR | file, I've built a clustered system from the ground up, basically to do it I had to move just about everything possible away from asterisk |
| 21:30.36 | [TK]D-Fender | codefreeze, blitzrage is SPECIFICALLY looking at the end result, and you keep pimping AEL as a SOLUTION for that when it is clearly not. NOw tell me who has a few screws loose... or perhaps jsut an agenda :P |
| 21:30.50 | blitzrage | HarryR: an example: http://www.pastebin.ca/743892 |
| 21:31.18 | HarryR | holy gods |
| 21:31.35 | blitzrage | something I created after seeing a presentation at TAUG just to see if it could be done in dialplan logic |
| 21:31.38 | HarryR | please don't mix dialplan with shell, you make me cry |
| 21:32.10 | HarryR | Java + FastAGI all nicely abstracted away so it's not reliant on Asterisk = much better |
| 21:32.55 | blitzrage | that is not production quality -- it is simply to show how you HAVE to do it at this point (when you're restricted to using ONLY dialplan and not AGI) -- it is to demonstrate things like that and to give a developer an opportunity to create better dialplan applications to solve deficiencies such as the one you pointed out |
| 21:33.06 | blitzrage | java is a four letter word |
| 21:33.18 | wwalker | blitzrage: absolutely true |
| 21:33.20 | blitzrage | HarryR: much better to java developers I guess |
| 21:33.55 | HarryR | blitzrage, exceptions, type safety, reliability, easier to use logic (none of this goto hell you seein the dialplan) etc. |
| 21:34.00 | blitzrage | _all_ asterisk administrators (should) know how to use the dialplan though |
| 21:34.43 | HarryR | ofcourse, but it doesn't help when people get the wrong idea and do stuff like that instead of agi/fastagi just because they can |
| 21:34.58 | HarryR | it's like writing a logo interpreter in postscript |
| 21:35.01 | codefreeze | [TK]D-Fender: obviously you made your choice, but I'll keep harping at you just for fun. I've been doing large scale coding projects for decades, and I avoid assembler for a reason. |
| 21:35.07 | HarryR | just cos you can doesn't mean you should |
| 21:35.41 | codefreeze | [TK]D-Fender: even if C or C++ adds nothing to assembler to the end. |
| 21:36.04 | codefreeze | uh, /to the end/in the end/ |
| 21:36.13 | [TK]D-Fender | codefreeze, Again, I never disagreed with all of its BENIFITS. Only that it does, in no way, help blitzrage's dialplan execute FASTER than if he coded it himself directly in extensions.conf. |
| 21:36.16 | HarryR | maintainability :D |
| 21:36.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:36.54 | [TK]D-Fender | codefreeze, Now you can go ahead and ask me if I feel his goal is WORTH sacrificing said benifits, and that'll be another matter :) |
| 21:36.55 | blitzrage | [TK]D-Fender: it does if you are not aware of some of the ways to create your dialplan efficently |
| 21:37.07 | codefreeze | [TK]D-Fender: it would not be wise to code for speed at the moment. The reasons for coding that way will soon disappear. |
| 21:37.11 | [TK]D-Fender | blitzrage, Ignorance is NOT an excuse :) |
| 21:37.36 | [TK]D-Fender | codefreeze, I may well CELEBRATE when that day comes :) |
| 21:38.09 | [TK]D-Fender | blitzrage, If you "suck", then it'll only be as an expression of whatever medium you're working with. |
| 21:38.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 21:38.31 | blitzrage | [TK]D-Fender: I've used asterisk everyday for 5+ years and just learned something about dialplan efficiency -- what hope do non-full time asterisk administrators have? |
| 21:38.49 | blitzrage | if there is a tool like AEL to help with some of that -- then all the better |
| 21:39.40 | blitzrage | but I tend to disagree with much of your attitude towards users, so no reason we wouldn't disagree here either |
| 21:39.46 | [TK]D-Fender | blitzrage, well if AEL somehow miraculously coincides with that little nugget you've only just learned, then more power to you. Then again I sort of fear that if such tiny basic dialplan issues are enough to run you so close to the wire, you're already just about baked... |
| 21:40.10 | [TK]D-Fender | blitzrage, Jsut a question of odds.... |
| 21:40.28 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:40.49 | [TK]D-Fender | and timing :) |
| 21:42.08 | blitzrage | the reason I like AEL is the significantly less number of characters I have to type on the keyboard, thus saving me money in savings of having to purchase new keyboards |
| 21:42.36 | [TK]D-Fender | blitzrage, NOW you're finding its true value! :p |
| 21:43.51 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 21:43.58 | kanderson | Does anyone know what would cause a five second delay on answer or return from hold? Also sometimes hangs indiffently on cmd Playback or other audio files. |
| 21:44.57 | kanderson | Audio files are in a ramdisk, and phones are on different networks/ISPs but all have a five second one way audio on answer |
| 21:46.30 | [TK]D-Fender | codefreeze, Seriously, I don't have a "real" beef against AEL. This was more a "what was really asked for" question. Yes, having a pre-parser/validation = GOOD, code can often be more compact = good. Parse time from AEL -> Normal = irrelevant to me so = FINE. |
| 21:47.00 | [TK]D-Fender | codefreeze, was just looking at this from that one specific viewpoint. I should probably start using it myself. |
| 21:47.38 | [TK]D-Fender | codefreeze, But I do acknowledge that it will likely not technically do a single thing more than I can do without it. |
| 21:47.56 | [TK]D-Fender | (minus validation) |
| 21:49.39 | [TK]D-Fender | codefreeze, And of course in summary : extensions.conf is the issue. It is the "broken" lowest common denominator that SHOULD be changed. |
| 21:50.25 | [TK]D-Fender | codefreeze, But if thats so far off I guess AEL will make some people's lives a little nicer, so I guess I can say "thanks" on their behalf :) |
| 21:50.34 | codefreeze | [TK]D-Fender: so "technically do a single thing" can be replaced with at LEAST validation! |
| 21:50.35 | [TK]D-Fender | codefreeze, I'm really not all that bad you know :p |
| 21:50.48 | [TK]D-Fender | codefreeze, yes :) |
| 21:51.07 | codefreeze | Yay! |
| 21:51.17 | De_Mon | yeah, hes only evil encarnate on friday the 13th |
| 21:51.39 | [TK]D-Fender | De_Mon, But at least on tuesday I'll see ti coming ;) |
| 21:51.48 | codefreeze | [TK]D-Fender: I know you are not bad! I look at you helping out with bits of wisdom day to day, and admire you greatly! |
| 21:52.10 | kanderson | Does anyone have any idea about my two questions above? I would appreciate it as I have tried everything I can think of to get ride of it. |
| 21:52.38 | [TK]D-Fender | codefreeze, We all try in our way.... So... who's going to kick some nuts to get one of those chan_sip replacements in? :D |
| 21:53.36 | [TK]D-Fender | We need REAL SIP-B, CPID, proper queues, dear God fix CDRs..... |
| 21:54.16 | TrentCreek | yes my son |
| 21:54.20 | kanderson | CDRs in 1.4.13 are much improved |
| 21:55.40 | [TK]D-Fender | blitzrage, Maybe I can give your new dialplan a work-over in a little bit :) |
| 21:56.34 | [TK]D-Fender | HELP... I'm addicted to another stupid flash game ... http://www.gamedesign.jp/flash/chatnoir/chatnoir.html |
| 22:00.11 | codefreeze | kanderson: really? sigh... you should see what I do in the next week or so |
| 22:01.48 | kanderson | Why, what do you have in mind? |
| 22:03.18 | kanderson | and are the notices that 'cdr was not posted' necessary? |
| 22:09.20 | codefreeze | kanderson: uh, is it 10927? Going to straighten out the CDRs for xfers. And, because its so much work, I'm going to fold it into trunk when I'm done. |
| 22:10.21 | codefreeze | And, I'm writing a letter on Mon, to asterisk-dev, to see if anyone objects to putting NoLossCDR into trunk, in place of the batch stuff. |
| 22:11.17 | codefreeze | kanderson: and yes, 'cdr not posted' will probably go away. |
| 22:12.04 | codefreeze | It's better not to say anything in "normal" operation. |
| 22:13.48 | kanderson | Agreed, it is filling my logs. Have xfers that make sense will be nice. Thank you. My cdrs where unsable untill now. |
| 22:15.01 | TheCops | Someone can tell me a good voip provider for Quebec province ? |
| 22:17.17 | [TK]D-Fender | Unlimitel |
| 22:17.34 | [TK]D-Fender | TheCops, Gota few happy clients with them |
| 22:18.05 | TheCops | Thank |
| 22:26.26 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:26.26 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:32.24 | *** join/#asterisk kn0x (n=pinochle@75.127.83.142) |
| 22:36.14 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 22:37.20 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 22:39.12 | knarfly | 8-) |
| 22:39.25 | knarfly | Can someone help me with this http://www.pastebin.ca/743933 |
| 22:39.55 | knarfly | I want to have this for serveral numbers but don't want to have to repeat the block of code for each one |
| 22:43.12 | [TK]D-Fender | knarfly, http://www.pastebin.ca/743935 |
| 22:46.29 | knarfly | [TK]D-Fender: need a little help here...what exactly does "process" do? Is it a variable or do I just enter the word process and if so how does it know when my special friends are calling? |
| 22:47.01 | [TK]D-Fender | knarfly, its just an exten. it'll get exectude like any other. I could jsut as easily have called ti FRED. |
| 22:47.07 | knarfly | sorry, I think I follow the logic now |
| 22:47.28 | knarfly | thanks, I'll run it up the flagpole |
| 22:47.31 | [TK]D-Fender | knarfly, Thing is that I'm presuming the call is coming in on "s", and you can already see a BUNCH of samples where I dumpt them INTO that exten so they get treated IDENTICALLY |
| 22:47.45 | [TK]D-Fender | dumpted* |
| 22:48.54 | [TK]D-Fender | dumped. |
| 22:48.54 | [TK]D-Fender | ARG |
| 22:49.43 | luke-jr | so any ideas on futex hanging? |
| 22:49.54 | knarfly | Yes, I see...thanks. That's exactly what I needed |
| 22:50.42 | [TK]D-Fender | knarfly, np |
| 22:51.39 | *** join/#asterisk TrN (n=jeff@adsl-074-173-088-193.sip.bct.bellsouth.net) |
| 22:52.10 | knarfly | Another thing I've noticed with 1.4.11 is that this message shows up when I start * |
| 22:52.11 | knarfly | [Oct 20 18:51:16] WARNING[655]: chan_zap.c:11159 process_zap: Ignoring signalling |
| 22:52.12 | knarfly | [Oct 20 18:51:16] WARNING[655]: chan_zap.c:11159 process_zap: Ignoring disconnectsupervision |
| 22:52.38 | knarfly | These are in my zapata.conf file...are they now deprecated |
| 22:53.16 | [TK]D-Fender | Signalling can be ignored... the other I've NEVER seen. |
| 22:53.53 | blitzrage | ignoring disconnect supervision? weird... |
| 22:54.13 | knarfly | if signalling can be ignored, should I remove it from zapata.conf? |
| 22:54.45 | knarfly | that one I found in some example somewhere...I don't know what it means either, I just followed the example. |
| 22:56.48 | [TK]D-Fender | knarfly, Sice they're ignored, do whatever you feel like :) |
| 23:00.18 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:01.19 | lesouvage | Is there any way to record outbound call with the ringing before they pick up included? I tried Monitor() but it skips the ringing. |
| 23:01.59 | *** part/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 23:02.19 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 23:02.29 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 23:03.01 | [TK]D-Fender | lesouvage, dial to chan local, then answer, then dial the real destination |
| 23:03.52 | lesouvage | [TK]D-Fender: I try it out, thanks |
| 23:07.16 | crudpuppy | hey I was noticing that some of you guys recommend against iax2 to voicepulse in favor of sip connections...why is that? |
| 23:07.54 | *** part/#asterisk DaneM (n=DaneM@ppp-71-130-236-78.dsl.chi2ca.pacbell.net) |
| 23:08.42 | [TK]D-Fender | IAX2 when trunked risks losing all calls if the truck fails, and seems to have noticably poorer quality with them |
| 23:09.21 | crudpuppy | k |
| 23:09.59 | crudpuppy | so the default setup with voicepulse is to setup both...so I could just remove the register statements from iax.conf and leave the ones in sip.conf and be set right? |
| 23:10.19 | [TK]D-Fender | crudpuppy, What default? |
| 23:10.30 | crudpuppy | oh, the sample files they provide |
| 23:10.31 | crudpuppy | lol |
| 23:10.58 | crudpuppy | it seems to be that they have register statements in both the sip.conf and iax.conf |
| 23:11.24 | [TK]D-Fender | crudpuppy, they have been editing your dialplan? Sounds like you've been hacked... |
| 23:11.58 | crudpuppy | [TK]D-Fender, no voicepulse gives you a set of configs for a real basic setup to get ya started with |
| 23:12.37 | [TK]D-Fender | crudpuppy, Good.. now take some resposnibility and start calling it YOUR config(which you blindly cut&paste) :p |
| 23:12.44 | crudpuppy | lol |
| 23:12.53 | crudpuppy | ok ok |
| 23:13.00 | [TK]D-Fender | crudpuppy, So yes, go exorcise (or at least comment out) the IAX2 stuff |
| 23:13.30 | crudpuppy | man [TK]D-Fender, you sure do take the long way around to awsner a question |
| 23:13.31 | crudpuppy | lol |
| 23:14.43 | [TK]D-Fender | crudpuppy, Oh, that was quite deliberate, and by no means long. I just like hearing people say exactly what configs they're talking about so I don't waste time on assumptions about what miracle "defaults' you feel like dredging up :) |
| 23:17.22 | *** join/#asterisk KE6IGZ (n=dave@cpe-66-75-70-232.socal.res.rr.com) |
| 23:17.35 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 23:29.32 | *** join/#asterisk jameswf-home (n=jameswf@ip72-204-222-2.ph.ph.cox.net) |
| 23:32.26 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:32.26 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:35.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:37.28 | De_Mon | [TK]D-Fender this reminds me of GO |
| 23:39.12 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:40.47 | lesouvage | [TK]D-Fender: I made the changes and the phone is ringing and the files are made but no recording before answering. Is there anything wrong with what I did. (see: http://pastebin.be/5966) |
| 23:40.56 | De_Mon | bah half the time its impossible to win |
| 23:41.20 | De_Mon | I thought ringing was on the phone, how can you "record" a ring? |
| 23:44.35 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 23:46.35 | De_Mon | [TK]D-Fender your flash game is sooo not adicting |
| 23:51.49 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:54.00 | duki | hello everyone, |
| 23:54.15 | duki | I have a question about DID. |
| 23:54.20 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 23:56.03 | duki | If am in Belgium and want people in spain to call me, I need to DID number from Spain, isn't it? |
| 23:56.54 | duki | I would like to be sure just before takes one DID number. |