| 00:01.14 | cuban | Sweet |
| 00:01.27 | cuban | My smartnet contract for our AS5600 and 7600 routers let met get the SIP image. |
| 00:05.05 | nhuisman | I wonder when the new cisco wireless phones will have sip support |
| 00:06.05 | cuban | I wonder if skinny is any good |
| 00:06.27 | cuban | and if Cisco really things a closed source protocol will become dominent |
| 00:06.40 | tessier | I don't see anything on the wiki about how to send a fax with asterisk |
| 00:06.46 | tessier | All about receiving though. |
| 00:07.53 | PlainWhiteTrash | cuban: skinny seems to work well, and has support for types of services that aren't well achieved by any other VOIP signalling protocols at this time... with the possible exception of some extensions to MGCP. |
| 00:08.02 | nhuisman | does the G extension on a phone name mean wireless? |
| 00:08.02 | *** join/#asterisk webmind (~cme2@217-195-236-172.dsl.esined.net) |
| 00:08.26 | tessier | no |
| 00:08.36 | PlainWhiteTrash | As for the closed protocol issue... it just lets them have a monopoly on a well integrated complete pstn to desktop voip solution. |
| 00:08.36 | nhuisman | whats it mean? |
| 00:08.39 | PlainWhiteTrash | Global |
| 00:08.44 | nhuisman | global means? |
| 00:08.47 | PlainWhiteTrash | text scrubbed off, icons replaced. |
| 00:08.47 | tessier | nhuisman: What phone are we talking about? |
| 00:08.54 | PlainWhiteTrash | Also implies a much later hardware revision. |
| 00:09.02 | nhuisman | like 7912G |
| 00:09.02 | brc_ | ~seen manxpowre |
| 00:09.03 | jbot | brc_: i haven't seen 'manxpowre' |
| 00:09.05 | brc_ | ~seen manxpower |
| 00:09.06 | jbot | manxpower <~eric@dsl-208-164-150-160.datasync.com> was last seen on IRC in channel #asterisk, 2h 4m 41s ago, saying: 'Anyway, I've answer the one or two questions. Ta Ta y'all'. |
| 00:09.09 | PlainWhiteTrash | The LCD screens are better in the G series... Their refrest rate isn't visible. |
| 00:09.15 | nhuisman | ah |
| 00:09.18 | *** join/#asterisk BoRiS (boris@S01060050da67299b.wp.shawcable.net) |
| 00:09.27 | nhuisman | i went back 7.1 and the problem reoccured |
| 00:09.31 | nhuisman | with the phone going nuts |
| 00:09.34 | tessier | I tried getting some 7960's (not G's) to work with Asterisk with no success. The 7960G's worked perfectly. |
| 00:09.47 | nhuisman | what problems did you have tessier? |
| 00:10.01 | nhuisman | I think most of the phones our building are 40's |
| 00:10.02 | tessier | nhuisman: They would not tear down the call properly. |
| 00:10.06 | PlainWhiteTrash | nhuisman - something is either broken with 7.1 generally - or with respect to asterisk - or with respect to your network configuration. |
| 00:10.11 | tessier | nhuisman: It would hang the POTS line open forever. |
| 00:10.13 | nhuisman | interesting |
| 00:10.21 | PlainWhiteTrash | Go all the way back to 5.0, make it work right, and roll forward to 6.3 |
| 00:10.28 | PlainWhiteTrash | see if 6.3 straight from 5 will work right |
| 00:10.28 | tessier | Asterisk Fax Manager only deals with incoming faxes. :( |
| 00:10.29 | nhuisman | yea i'm going to sit here with 5.0 for ab it |
| 00:10.39 | nhuisman | it seemed to happen after a bit of time passed |
| 00:10.47 | tessier | If I have a PRI card in my box how can I send faxes out with it? |
| 00:10.49 | nhuisman | have other people had that issue tessier? |
| 00:11.01 | tessier | nhuisman: Not sure. |
| 00:11.03 | PlainWhiteTrash | The 7960s and 7960Gs run the same firmware and look the same to the network generally speaking.. I've never had trouble getting either to work with * |
| 00:11.24 | tessier | PlainWhiteTrash: I flashed 7960's and 7960G's with the same firmware but the 7960's behaved very differently. |
| 00:11.43 | tessier | They had some different menu navigation, you lock/password them differently, etc. |
| 00:12.04 | PlainWhiteTrash | I suspect you had phones fail to take the flash then.... The code is dead on the same. |
| 00:13.00 | PlainWhiteTrash | This building has a mix of 7960s and 7960Gs and they've always been perfectly interchangeable in my environment. |
| 00:13.07 | tessier | They did take the flash properly. If they hadn't I don't think the phones would have said SIP (as opposed to skinny) and would not have talked to * at all |
| 00:13.37 | tessier | Maybe the client had bogus 7960's then. One of them refused to take a flash because the filename it tried to tftp had a carriage return in the middle of it. |
| 00:13.51 | PlainWhiteTrash | yup - that was an issue... |
| 00:13.54 | PlainWhiteTrash | To get those to update... |
| 00:14.04 | PlainWhiteTrash | You had to run through a special sequence of exact builds in order... |
| 00:14.08 | PlainWhiteTrash | ending with 5.x... |
| 00:14.13 | PlainWhiteTrash | then, they would work normally. |
| 00:14.23 | tessier | Ah. I bet that was the problem then. That is very weak. |
| 00:14.32 | *** join/#asterisk svanlund (~david@as2-5-8.ml.g.bonet.se) |
| 00:14.41 | *** join/#asterisk _ZoR (~anthony@adsl-065-005-181-237.sip.asm.bellsouth.net) |
| 00:14.43 | PlainWhiteTrash | you also had to manually hex edit the OS79XX.TXT to remove line termination in some cases... there are some nuances to it. |
| 00:14.47 | *** join/#asterisk Command (~WampiReLL@219.95.225.25) |
| 00:14.58 | tessier | At this point Cisco's are my least favorite phone. |
| 00:15.18 | tessier | Everyone raves about them having a cool display and everything but I need a phone that works, not a headache. |
| 00:15.18 | *** part/#asterisk PCadach (squid@www.east.telecom.kz) |
| 00:15.21 | PlainWhiteTrash | Are you kidding? I've never found a better SIP desktop phone (once you've figured out all the tricks and automated the process) |
| 00:15.23 | tessier | Even flashing them is a pain. |
| 00:15.47 | tessier | You have to set up a tftp server with just the right special files in it. And to legally get the SIP image you have to go through a bunch of hassle with Cisco. |
| 00:16.00 | *** join/#asterisk shmooz (shmooz@H76.C233.tor.velocet.net) |
| 00:16.01 | tessier | Once you've figured out all the tricks an automated the process? That's a big problem right there. |
| 00:16.07 | tessier | I want a phone not a frickin' computer. |
| 00:16.33 | PlainWhiteTrash | exactly - so write software that makes it act like a phone by handing all that for you intelligently. it's not that hard. |
| 00:16.37 | tessier | The Snom is a lot easier to set up, has no such tricks or nuances, and makes and receives calls and is multiline. |
| 00:16.48 | tessier | And I don't have to write automation software. |
| 00:16.51 | tessier | Programmer time is expensive. |
| 00:17.28 | PlainWhiteTrash | And has no browser... which locks you out of providing lots of nice dynamic information... I have a number of XML apps that I've written for the ciscos and tied back to *... Like extension status... I can punch up a screen on my 79XX and see who's on the phone and with whom. |
| 00:17.43 | Moc__ | programmer is free when you know how to program ;) |
| 00:18.05 | tessier | I don't need dynamic info and I have never had a client ask for dynamic info on their phone. That is what their computer is for. |
| 00:19.09 | *** join/#asterisk xeaded (~xeaded@69.88.201.41) |
| 00:20.52 | PlainWhiteTrash | You'd be amazed how the control freak will come out in a customer when he/she sees you tap a couple buttons on the phone and know who's talking to who... Lot easier than opening a browser or a desktop app... |
| 00:21.08 | nhuisman | um |
| 00:21.27 | nhuisman | how come a person should be able to see who is talking? seems sorta like an invasion privacy |
| 00:21.41 | Moc__ | Anyone have feature request for MeetMe É |
| 00:21.43 | tessier | If it is a company there is no privacy |
| 00:22.04 | PlainWhiteTrash | Naturally, you should provide a configuration option for that sort of thing.. at the end of the day... if you're on company time.. using a company phone... it's their call. |
| 00:22.30 | nhuisman | well i mean management should be able to see it, but not jsut anyone |
| 00:22.34 | PlainWhiteTrash | Same difference with like gastman... those things have to be configurable. |
| 00:22.51 | PlainWhiteTrash | nhuisman - exactly... |
| 00:23.45 | PlainWhiteTrash | Alternatively.. in small businesses.. that's sometimes not true.. my coworkers and i all have it set up on our phones.. we're a small, friendly group and none of us cares. |
| 00:24.04 | nhuisman | yea |
| 00:24.41 | _ZoR | also the info provided probably just says whether their extension is active or not right? |
| 00:26.13 | PlainWhiteTrash | I've written both. |
| 00:26.32 | PlainWhiteTrash | One to show on phone or not, one to show bridged party's caller id or dialed number. |
| 00:26.33 | nhuisman | yea PlainWhiteTrash it appears that just the 7.x images are doing that wierd stuff |
| 00:26.51 | nhuisman | oh nm just did it on 6.3 |
| 00:26.54 | nhuisman | sigh |
| 00:27.05 | PlainWhiteTrash | nhuisman... Either that or something is flawed with your load process of the 7.x's... I suggest running at 6.3, has lots of useful bug fixes and has served me reliably. |
| 00:27.06 | nhuisman | rebooted the phone |
| 00:27.11 | PlainWhiteTrash | uh oh |
| 00:27.17 | PlainWhiteTrash | so it's happening to you on 6.3 |
| 00:27.27 | nhuisman | entered voice mail asked for password, started entering password and died |
| 00:28.29 | nhuisman | yea its just rebooting in the voicemail |
| 00:28.45 | PlainWhiteTrash | What ver. of * are you running? |
| 00:29.20 | Moc__ | Version getdate() |
| 00:29.31 | nhuisman | 5/26 |
| 00:29.33 | cuban | So is it a bitch to get SIP on a 7960G? |
| 00:29.37 | cuban | I'm about to order 4 of them. |
| 00:29.39 | cuban | For testing. |
| 00:29.41 | brc_ | exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) trying to figure out what the X's refer to... |
| 00:29.51 | tessier | cuban: Not as easy as it should be but it is possible IF you can find someone willing to pirate you the SIP image. |
| 00:29.51 | PlainWhiteTrash | cuban: no.. not really... just requires a little patience the first few times... |
| 00:30.02 | cuban | tessier: I have the sip image already. |
| 00:30.14 | cuban | tessier: I got it from one of my other smartnet accounts... |
| 00:30.15 | Moc__ | cuban nah, OLD 7960 with old code it not easy, but once you KNOW you need to upgrade to a older version before installing 6.x or latter.. it easy |
| 00:30.19 | *** join/#asterisk mizzie_ (~mizzie@host217-43-209-44.range217-43.btcentralplus.com) |
| 00:30.29 | tessier | cuban: Then it shouldn't be too bad. Just have to set up your own tftp server with all the right stuff in it |
| 00:30.36 | PlainWhiteTrash | nhuisman.. I can't imagine what could be doing that from * .... in vm.. but .... the fact that the thing is doing it reliably.. makes me wonder... |
| 00:30.38 | cuban | Anyone wrote down any info anywhere on the steps? |
| 00:30.50 | cuban | I'm gonna put 7.1 one on the phone... |
| 00:30.51 | PlainWhiteTrash | nhuisman.. send me your sipdefault and mac specific configs... |
| 00:30.54 | *** join/#asterisk the_grugerR (~iZmir@dialup-107.nas02.azerin.com) |
| 00:30.55 | nhuisman | ok |
| 00:31.10 | Moc__ | also you need to setup your Software version within the SIPDefault or SIPYOURPHONEMAC.cnf so the phone accept the upgrade |
| 00:31.11 | cuban | So it's not just setting up a TFTP server and loading the image to the phone? |
| 00:31.32 | tessier | cuban: You need to create a config file on the phone named by the phones mac address and some other stuff. |
| 00:31.34 | Moc__ | those are the only 2 thing .. (also you need to update OS79xx.txt when updating |
| 00:31.35 | cuban | There should be steps somewhere, nothing on the WIKI? |
| 00:32.06 | cuban | Moc__ Uh, heh you say update but with what |
| 00:32.10 | Moc__ | cuban I have you the 3 major pain I had. You just need the code + sample config. google has some |
| 00:32.12 | cuban | What is OS79XX.txt |
| 00:32.26 | PlainWhiteTrash | cuban: usually it *is* just setting up tftp and loading the files... but it's when things aren't quite right that you really have to get into it. |
| 00:32.26 | cuban | I have you the what? |
| 00:32.28 | Moc__ | it contain the software version |
| 00:32.35 | cuban | oi |
| 00:32.45 | cuban | Someone who knows it should fill in the info on the wiki. |
| 00:33.15 | Moc__ | well get the phone, we will help you set it up.. and add it to the wiki... ;) |
| 00:33.16 | cuban | Are there more than one file you need to load... |
| 00:33.22 | xeaded | Is there any way to get rid of the universal application loader on the 7960G phones with sip 7.1? It takes the phone a lot longer to boot up. |
| 00:33.27 | cuban | Okay, I'll happily add it to the wiki. |
| 00:33.37 | _ZoR | I haven't made any changes to iax.conf and now * all of sudden won't connect when making IAX2 calls. Keeps saying the circuit is busy... what might be causing this? |
| 00:33.55 | Moc__ | OS79XX.TXT SIPDefault.cnf SIPYOURSIPPHONEMAC.cnf and your binary file are the important one |
| 00:33.55 | PlainWhiteTrash | No - the app loader has to be there.. it's part of the execute in flash architecture they built. |
| 00:34.59 | cuban | So I need to put the flash on a tftp server, put a file on the tftp server named $MACADDRESS.cnf and edit OS7960.txt and put 7.1 in there. |
| 00:35.01 | cuban | Correct? |
| 00:35.15 | collinr__ | SIP$MACADDRESS.cnf |
| 00:35.20 | cuban | Ah. |
| 00:35.27 | collinr__ | your OS79XX.TXT |
| 00:35.28 | cuban | And in that file goes what? Some config info? |
| 00:35.28 | collinr__ | will contain |
| 00:35.48 | Moc__ | cuban I'll putup a simple Tar file of my directory |
| 00:35.49 | collinr__ | P0S-07-01-00 |
| 00:35.58 | cuban | Moc__ That would be great |
| 00:36.06 | cuban | Oh and obviously program the phone to point to the TFTP server |
| 00:38.40 | *** join/#asterisk ZoR_ (~anthony@adsl-065-005-181-237.sip.asm.bellsouth.net) |
| 00:39.10 | tessier | hmm...apparently spandsp has a txfax command now |
| 00:39.23 | *** join/#asterisk snewpy_ (~markl@203-217-44-63.dyn.iinet.net.au) |
| 00:43.42 | Moc__ | http://www.moctel.com/asterisk/cisco-79xx-sample-tftpdir.tar.gz |
| 00:43.48 | *** join/#asterisk ToyMan (~stuq@user-0cevfhv.cable.mindspring.com) |
| 00:43.50 | Moc__ | tessier being a while |
| 00:44.29 | Moc__ | cuban via DHCP or set manually on phone for DHCP |
| 00:44.32 | cuban | thanks |
| 00:44.47 | Moc__ | that is the exact thing I use on my side |
| 00:44.48 | *** part/#asterisk mitchel_ (trilluser@216.132.54.140) |
| 00:44.54 | Moc__ | except for the registry hehe |
| 00:45.01 | cuban | How do you set TFTP via dhcp? |
| 00:45.05 | *** join/#asterisk xantus (~xantus@00a0c5e29af5.click-network.com) |
| 00:45.21 | collinr__ | option 150 I believe ? |
| 00:45.42 | Moc__ | I use server-name "10.10.10.254"; |
| 00:45.46 | Moc__ | in dhcpd.conf |
| 00:46.06 | collinr__ | what's your dhcp server? |
| 00:46.14 | cuban | Damn, what if you don't have DHCP? |
| 00:46.19 | cuban | Like a single user? |
| 00:46.28 | Moc__ | cuban you need to setup manually the tftp on the phone |
| 00:46.48 | PlainWhiteTrash | The cisco's almost always need DHCP to be manageable.. It's a pain doing it any other way |
| 00:46.52 | Moc__ | that what I did at the begining |
| 00:46.52 | cuban | So you need TFTP at each damn location? |
| 00:47.00 | cuban | I guess I could put one publically. |
| 00:47.03 | PlainWhiteTrash | No... But DHCP... |
| 00:47.15 | cuban | Damn.... |
| 00:47.33 | PlainWhiteTrash | There are cheap and easy ways to do this... :-) |
| 00:48.18 | collinr__ | you can statically assign IPs in the phones |
| 00:48.43 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
| 00:48.45 | PlainWhiteTrash | Yes... It just becomes a hassle if you ever have to remote admin them.. |
| 00:48.48 | collinr__ | true |
| 00:48.51 | Bobby_Ewing | JerJer: are you still here |
| 00:48.55 | cuban | hmmmm |
| 00:48.59 | Moc__ | cuban also watch out for the dialplan.xml I have in my example... it setup for my need... might cause you pain.. just remove it for the moment and configure it latter once you know how it work |
| 00:49.01 | cuban | We'll I'll look at options once I get one |
| 00:49.15 | cuban | Moc__ Any good docs on it? |
| 00:49.26 | Moc__ | cuban, cisco is the best source for it |
| 00:49.35 | nitram | ciscos sip admninstration guide |
| 00:49.47 | Moc__ | to reset the phone in SIP mode, use * 6 and Settings button.. |
| 00:49.56 | cuban | Okay thanks |
| 00:50.08 | Moc__ | I used to unplug the phone until I found that hehe |
| 00:50.13 | Moc__ | you could also telnet to the phone |
| 00:50.27 | Bobby_Ewing | since updating asterisk today, phones keep dropping off registration from *: after a call, sometimes asterisk removes them from astdb but the phone actually still things it's registered and can connect if you dial a number, just you can't call THAT phone from *, it just says busy. The host is (unspecified) when i do a sip show peers from the CLI |
| 00:50.50 | Bobby_Ewing | i haven't changed any configs, only updated * |
| 00:51.14 | Bobby_Ewing | i tried going down to stable, but it didn't like it and wouldn't run, so i had to recompile cvs head again |
| 00:51.37 | Moc__ | cuban telnet is nice for doing the command : show config |
| 00:51.41 | PlainWhiteTrash | I think the general consensus is to stay away from cvs stable at this point. |
| 00:51.45 | Bobby_Ewing | It was registered, but now is : 2001/2001 (Unspecified) D |
| 00:51.51 | Moc__ | so you can see how it setup and maybe put stuff you want in your .cnf file |
| 00:52.05 | PlainWhiteTrash | Bobby: what kind of phones? |
| 00:52.14 | Moc__ | cuban also you dont need tftp all the time. Unless you have ringtone you want to use |
| 00:52.17 | Bobby_Ewing | PlainWhiteTrash: they are gs, but they have been working fine uptill today |
| 00:52.21 | Moc__ | you could manually setup your extensions |
| 00:52.37 | Bobby_Ewing | what about remote phones that are behind nat |
| 00:52.50 | Bobby_Ewing | they keep loosing registration in * too |
| 00:52.51 | PlainWhiteTrash | Bobby: Hmm... I can't help ya.. I have two gs phones around here in a box.. I gave up on those.. Had high hopes... but when it couldn't even do call waiting right... i gave up on it. |
| 00:53.14 | Moc__ | cuban, my config is made for NAT, check the config and make NAT 0 if you are using it locally |
| 00:53.32 | Bobby_Ewing | PlainWhiteTrash: they have and do work fine on the whole, just after * update today i've been stumped |
| 00:53.43 | PlainWhiteTrash | When had you last updated? |
| 00:53.49 | Bobby_Ewing | i have to keep rebooting the phones manually to get them back in the astdb |
| 00:53.59 | Bobby_Ewing | 05/15 was my last update before today |
| 00:54.51 | *** join/#asterisk bAnU03` (~seWda@client-40-p-1-lns.glfd.dial.virgin.net) |
| 00:54.59 | *** join/#asterisk gbdrbob (~drbob@alltalk.demon.co.uk) |
| 00:55.27 | Bobby_Ewing | PlainWhiteTrash: the only other thing i've done is enable voicemail checking on the phones recently - i'm not sure if that is related or not |
| 00:55.40 | xantus | gus are you here? |
| 00:55.47 | Bobby_Ewing | i.e. so that they flash if there is voicemail waiting |
| 00:56.41 | *** join/#asterisk zotzz (~zotzz@24.231.36.159) |
| 00:56.42 | PlainWhiteTrash | Bobby... I seem to recall kram talking about some scenarios where they changed to code to expire out calls and such if the phone doesn't continue hitting the server correctly... |
| 00:57.18 | Bobby_Ewing | hmm wonder if there is a work around to keep a phone registered in the db |
| 00:57.36 | Moc__ | Bobby_Ewing your being NAT ? |
| 00:57.38 | Bobby_Ewing | so, could it be that voicemail is the cause of the problem perhaps? |
| 00:57.52 | PlainWhiteTrash | Bobby_Ewing - I doubt it... It sounds like a nasty gs bug... they must not be implementing something right. |
| 00:58.13 | Bobby_Ewing | Moc__: i have two phones behind nat at a remote location and the rest are on the same subnet, all experience this same sip loss of registration on * |
| 00:58.26 | *** join/#asterisk magma (~tetsuo@194.250.101.226) |
| 00:58.35 | Moc__ | well NAT should cause that problem if you dont redirect the port correctly.. |
| 00:58.48 | magma | Hi, someone here is using H323 with asterisk ? |
| 00:58.52 | Moc__ | maybe you got firewall on the Linux that dont allow the box to connect to the other phones ? |
| 00:58.52 | Bobby_Ewing | i dont do any redirect at all, never had to |
| 00:58.59 | Bobby_Ewing | it connects to them fine |
| 00:59.01 | magma | when making call with gnugk and 2 endpoints got sound |
| 00:59.18 | Moc__ | it connect, but lose the registration |
| 00:59.30 | Bobby_Ewing | it's that after a while, * removes the ip for the phone from it's db, but the phone still things it's registered and connects fine, you just can't call the extension ... |
| 00:59.34 | magma | but when I use asterisk for a gateway pstn no sound |
| 00:59.40 | magma | any idea ? |
| 01:00.20 | Bobby_Ewing | i've had it working fine for nearly a month, but today / yesterday i started voicemail checking on the phones and updated asterisk at the same time |
| 01:00.28 | Moc__ | Bobby_Ewing, Like I said, I have problem when * can't see the phone that * ask the phone if he still alive, if no answer, then lose the registration |
| 01:00.29 | magma | the signaling seems to work well, it s ringing it catch the hang but no sound |
| 01:00.47 | Bobby_Ewing | i put it down to asterisk update though |
| 01:01.07 | Bobby_Ewing | Moc__: I get the same problem with phones on the same subnet only started today though |
| 01:01.16 | Bobby_Ewing | they're all doing it, whether nat'd or local |
| 01:01.20 | Moc__ | check if you block anything on your linux box |
| 01:01.26 | Bobby_Ewing | nothing blocked |
| 01:01.33 | Bobby_Ewing | nothing has changed in that respect |
| 01:01.35 | Moc__ | check if your SIP setting have NAT setup, or on your phone if stun or nat is activated |
| 01:02.09 | Bobby_Ewing | the phones are fine setup wise for NAT, i'm sure.. they connect fine and used to, but something * side keeps removing their registration |
| 01:02.13 | PlainWhiteTrash | He has phones in two different environs - local & nat - that had been working... i'm starting to believe there may be something in * that killed it's functioning... |
| 01:02.33 | Bobby_Ewing | PlainWhiteTrash: yes, that's the scenario |
| 01:02.54 | Bobby_Ewing | strangely enough, one of the phones in the remote location is actually still registered at the moment |
| 01:03.32 | *** join/#asterisk bing (~bing@216.16.232.250) |
| 01:03.34 | Bobby_Ewing | and a couple here locally are still reg'd too, but i had to reboot a couple of others due to loss of registration. i think sometimes it's after calling the extensions they loose registration |
| 01:04.28 | PlainWhiteTrash | Everything is pointing to a SIP signalling issue... |
| 01:04.54 | PlainWhiteTrash | While a change to * may or may not have broken something, the question will remain who is implementing the spec. correctly. |
| 01:05.24 | Bobby_Ewing | actually i just remembered seeing an error |
| 01:05.39 | Bobby_Ewing | did something change in how i should be setting the local ip of the box in sip.conf |
| 01:05.45 | Bobby_Ewing | It's currently |
| 01:05.45 | Bobby_Ewing | localnet = 192.168.1.13 ; Internal NETWORK address |
| 01:05.45 | Bobby_Ewing | localmask = 255.255.255.0 |
| 01:05.49 | *** join/#asterisk YuzNumaralIAdam (BruceLee@client-82-3-66-64.mant.adsl.virgin.net) |
| 01:05.52 | Bobby_Ewing | is that syntax wrong |
| 01:06.26 | Bobby_Ewing | localnet = 192.168.1.13 ; Internal NETWORK address |
| 01:06.26 | Bobby_Ewing | localmask = 255.255.255.0 |
| 01:06.32 | Bobby_Ewing | sorry |
| 01:06.34 | Bobby_Ewing | Jun 5 02:06:42 WARNING[-1137402960]: chan_sip.c:7645 reload_config: Use of localmask is no long supported -- use localnet with mask syntax |
| 01:07.05 | PlainWhiteTrash | should be locatnet=192.168.1.13/255.255.255.0 |
| 01:08.37 | Bobby_Ewing | cheers, could that have been a cause at all? |
| 01:08.47 | Bobby_Ewing | the old method was fine on the old build |
| 01:08.53 | PlainWhiteTrash | not entirely certain.. i dunno if the gs relies on that stuff or not. |
| 01:08.58 | PlainWhiteTrash | it's a possibility |
| 01:09.01 | Bobby_Ewing | okay |
| 01:09.12 | Bobby_Ewing | maxexpirey=3600 |
| 01:09.17 | Bobby_Ewing | should i try increasing that at all |
| 01:09.58 | Bobby_Ewing | aghh another phone lost registration |
| 01:10.10 | Bobby_Ewing | it seems to be after x seconds of me calling it, it dies off |
| 01:10.30 | Bobby_Ewing | e.g. if i call it now, after so long it looses registration |
| 01:12.17 | PlainWhiteTrash | no... that changes the length of time that * will ask the phone to refresh registration.. if anything lower it... I run mine at 180... |
| 01:12.41 | PlainWhiteTrash | The phone will only loose registration after you've called it once? |
| 01:13.06 | Bobby_Ewing | seems so.. i am testing with another, but it seems to take affect after 5-10 mins or so |
| 01:13.39 | Bobby_Ewing | if you dont call it, it stays up fine |
| 01:13.59 | Bobby_Ewing | actually, saying that, i'm not so sure, as one of the NAT'd phones i tried calling earlier and it's still registered |
| 01:14.20 | h3x | jgfdjhjfd |
| 01:14.28 | h3x | ljk.kjlj |
| 01:16.48 | *** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net) |
| 01:17.03 | h3x | ZX vdfsad |
| 01:17.31 | *** join/#asterisk paulc (~web-irc@S010600062586a0b4.vc.shawcable.net) |
| 01:18.28 | essobi__ | Hey guys.. I got two * boxes I want to interconnect.. where do I start? |
| 01:18.49 | PlainWhiteTrash | bobby - do some more testing - try to identify the scenario |
| 01:18.58 | Bobby_Ewing | PlainWhiteTrash: I'm running a sip debug and i think we may have got it with the subnet mask issue, as it was using *'s external ip before to send udp packets to but now it's using the internal 192.168.1.13 |
| 01:19.12 | Bobby_Ewing | but i'm doing some testing and will try and catch something |
| 01:19.38 | PlainWhiteTrash | aaah, that could have been it. |
| 01:19.54 | PlainWhiteTrash | did you restart asterisk after making the change? |
| 01:20.04 | PlainWhiteTrash | I'm not sure whether that setting will take from a reload or not.. |
| 01:20.20 | Bobby_Ewing | PlainWhiteTrash: I actually see one of the phones behind the NAT connecting to * when i run a sip debug, the other isn't of yet. I just did a reload but it doesn't hurt to do a restart |
| 01:20.32 | Bobby_Ewing | (he says) |
| 01:20.55 | *** join/#asterisk BukLeMun (~dj_didem@adsluser-2605.adsl.ttnet.net.tr) |
| 01:21.26 | Bobby_Ewing | PlainWhiteTrash: shouldn't "restart now" normally restart * |
| 01:21.35 | PlainWhiteTrash | yes... |
| 01:21.50 | PlainWhiteTrash | what happened when you did that? |
| 01:21.51 | Bobby_Ewing | it's not doing anything, strange |
| 01:21.56 | Bobby_Ewing | nothing, just goes to a new line |
| 01:22.01 | PlainWhiteTrash | it's stuck with no command line? |
| 01:22.10 | essobi__ | hey guys.. I got two * boxes .. I want to dialout the zap of one from the other.. what do I use to interconnect them? IAX2? |
| 01:22.16 | Bobby_Ewing | it gives me a new line on the CLI |
| 01:22.20 | *** join/#asterisk alan (~alan@mozzer.routingloop.com) |
| 01:22.29 | PlainWhiteTrash | a new line to enter a command or a blank new line? |
| 01:23.00 | Bobby_Ewing | to enter command |
| 01:23.07 | Bobby_Ewing | actually no commands seemed to be responding |
| 01:23.13 | Bobby_Ewing | i've killed the process and manually restarted |
| 01:23.19 | PlainWhiteTrash | hehe... sounds like a deadlock :-) |
| 01:23.21 | PlainWhiteTrash | That can't be good. |
| 01:23.30 | alan408 | can someone point me at keywords or documentation to figure out how to amke asterisk automatically launch calls, as called by a cronjob or script, please? |
| 01:23.33 | essobi__ | busted spinlock |
| 01:23.35 | essobi__ | ctrl-c |
| 01:23.38 | Bobby_Ewing | OoO have not had that happen before |
| 01:23.52 | essobi__ | alan408 /var/spool/asterisk/outgoing |
| 01:24.10 | alan408 | essobi - thanks - what keywords do i look up to learn what to put in taht dir? |
| 01:24.12 | *** join/#asterisk Cript (~K@ip68-224-74-19.lv.lv.cox.net) |
| 01:24.27 | Cript | Does Asterisk Support two B-Channel Transfer? |
| 01:25.09 | essobi__ | http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out |
| 01:25.24 | PlainWhiteTrash | Cript on PRI? I don't think so, actually. |
| 01:25.36 | alan408 | thanks essobi |
| 01:25.40 | alan408 | yo're my new best friend |
| 01:25.45 | essobi__ | or you can use the manager interface. |
| 01:26.55 | Cript | what about transfering a call once its already connected to asterisk? |
| 01:27.12 | Cript | thx ess |
| 01:27.42 | PlainWhiteTrash | transfering the call back out over the pri? |
| 01:27.47 | essobi__ | Cript that link was for alan408 |
| 01:27.49 | Cript | PlainWhiteTrash yes |
| 01:27.53 | Cript | yea just realized =) |
| 01:27.58 | *** part/#asterisk collinr__ (~trillian@ip2.collinr.net) |
| 01:28.06 | PlainWhiteTrash | you can bridge (obviously), but I don't think you can do a divert... |
| 01:28.14 | essobi__ | Cript Umm.. yea it can be done.. but you're eating two lines. |
| 01:28.15 | *** join/#asterisk menger (~menger@dsl-244.84.240.220.rns02-dryb-mel.dsl.comindico.com.au) |
| 01:28.16 | Cript | id like to not have to use 2 lines to connect the call during the whole conversation |
| 01:28.20 | Cript | yea dont want to bridge =/ |
| 01:28.27 | PlainWhiteTrash | It could be implemented (ostensibly), theere are providers in the US who support that... |
| 01:28.39 | *** join/#asterisk venix (~venix@CPE0050bf77ff8d-CM000a7363ecf6.cpe.net.cable.rogers.com) |
| 01:28.53 | PlainWhiteTrash | However.. The messages for doing that aren't currently implemented in libpri as of yet.. unless it's relatively new. |
| 01:28.54 | Cript | Yea looking at Mpower right now they have that |
| 01:28.54 | Cript | Two B-channel transfer - This allows PRI to connect two calls, transfer the calls together and then release the parties from the PRI. |
| 01:28.55 | essobi__ | PWT: Inband transferance to a dual call leg? |
| 01:29.07 | Cript | just if asterisk doesnt support it, its no good =) |
| 01:29.10 | essobi__ | PWT: Soemthing is bridging the line reguardless.. you or someone else. ;) |
| 01:29.45 | Cript | as long as it frees my channels =) |
| 01:29.53 | essobi__ | Oh.. they're doing it OOB with two B channels. |
| 01:30.03 | essobi__ | Nope. * doesn't support OOB transfering. |
| 01:30.07 | PlainWhiteTrash | Nope... AT&T supports it on their PRIs... A PRI message goes out bridging up the two calls (potentially at your expense) and releases them, freeing your PRI channels. You loose call control, the two parties are bridged together.. your PRI line becomes 2 B channels freer :-) |
| 01:30.48 | alan408 | how do I turn on debugging from asterisk -r ? |
| 01:30.50 | essobi__ | I've just wrote DTMF collecting bridge. ;) |
| 01:31.02 | essobi__ | alan408 asterisk -rvvvv |
| 01:31.09 | essobi__ | or how many you want.. |
| 01:31.15 | essobi__ | that's not debug..that's verbose. |
| 01:31.16 | Cript | Crap! nevermind |
| 01:31.33 | Cript | oh read that wrong |
| 01:31.37 | essobi__ | you have to start your server with -dddd |
| 01:31.43 | Bobby_Ewing | PlainWhiteTrash: actually, my phone just lost registration again - this time it didn't call anything, i actually was using it to call |
| 01:31.44 | Cript | thought they wanted $45 per channel for the bchannel transfer lol its per pri |
| 01:32.06 | alan408 | I am trying to launh a call useing directory outgoing - it is failing but I get no debugging |
| 01:32.12 | PlainWhiteTrash | Bobby, but you did place a call through it.. I think that may be the key... if a call gets handled by it. |
| 01:32.13 | alan408 | I launched asterisk with -rvvvvv but nothing there |
| 01:32.18 | Bobby_Ewing | sorry, it didn't receive calls, i placed the call thru it |
| 01:32.18 | alan408 | where can I find debugging? or how? |
| 01:32.46 | Bobby_Ewing | i was calling an extension group that it is part of if that helps at all |
| 01:33.15 | Bobby_Ewing | i.e., i dialed 1000 which then dials 2000,2001,2002,2003,2004,2005 etc.. and the phone in question is 2000 |
| 01:33.16 | essobi__ | alan408 just use asterisk -vvvvr |
| 01:33.37 | alan408 | okay |
| 01:33.40 | alan408 | it didn't give me info |
| 01:33.42 | alan408 | but iax2 debug did |
| 01:33.54 | PlainWhiteTrash | bobby hmmm.. it definately sounds like a sip issue... see what gs says |
| 01:34.13 | alan408 | I copy /tmp/callfile into ...outgoing |
| 01:34.16 | alan408 | soon file goes away |
| 01:34.17 | alan408 | very quick! |
| 01:34.20 | alan408 | but no call placed |
| 01:34.22 | alan408 | hard to figure out why |
| 01:34.35 | Bobby_Ewing | plain: actually, the gs just crashed too |
| 01:34.42 | Bobby_Ewing | screen all scrambled |
| 01:34.45 | alan408 | is the timestamp on the file relevant? |
| 01:35.11 | PlainWhiteTrash | Bobby: definately try to take it up with their support... |
| 01:35.30 | essobi__ | alan408 If the file diapappeared.. it got processed. |
| 01:35.36 | essobi__ | You've got something wrong in the file. |
| 01:35.47 | essobi__ | asterisk -vvvvvr bub |
| 01:35.49 | alan408 | right, i think i have something wrong - how to get debugging to figure out what? |
| 01:35.59 | essobi__ | what'cha it before you start the copy. |
| 01:36.06 | essobi__ | oh.. and you shouldn't be copying. |
| 01:36.34 | alan408 | why not copy? |
| 01:36.43 | essobi__ | cp /tmp/whatever /var/spool/asterisk/outgoing/whatever.tmp ; mv tmp to .call |
| 01:36.47 | h3x | PlainWhiteTrash: XO supports 2 B channel transfer too but asterisk dosen't |
| 01:36.50 | essobi__ | cause copy isn't atomic. |
| 01:37.11 | essobi__ | h3x Is it OOB controlled on the D-channel or what? |
| 01:37.30 | h3x | Cript: Your only workaround is ordering like E&M Wink trunks and get centrex transfer feature |
| 01:37.54 | h3x | which is really tricky because you have to make sure the wink timing and flash timing are set up correctly to match the telco or you will be hanging up calls |
| 01:38.03 | essobi__ | alan408 move is atomic only when it's in the same directory. |
| 01:38.08 | PlainWhiteTrash | If B channel transfer is that important, there are probbaly people who'd write it for a price... |
| 01:38.23 | alan408 | I will use move |
| 01:38.25 | *** join/#asterisk XARiUS (~xarius@wsip-68-224-170-22.sd.sd.cox.net) |
| 01:38.26 | PlainWhiteTrash | I'm certainly no PRI guru.. but there must be some out there who run *.... |
| 01:38.27 | essobi__ | Well.. Yea. |
| 01:38.35 | essobi__ | I know a few. |
| 01:38.44 | essobi__ | Who are pretty sharp on the PRIs. |
| 01:39.03 | essobi__ | Not around thou. |
| 01:39.06 | h3x | PlainWhiteTrash: Well the first thing libpri needs is NFAS support |
| 01:39.21 | h3x | because its useless unless you happen to have one channel free on the same T1 your original call was made on |
| 01:39.26 | essobi__ | You'd have to talk to kram about that.. or get someone to write the patch. |
| 01:39.34 | h3x | I asked kram to do it |
| 01:39.41 | essobi__ | Hehehe. |
| 01:39.42 | h3x | he said something along the lines of it being a waste of time |
| 01:39.42 | h3x | heh |
| 01:39.44 | XARiUS | Anyone know if there are any pre-config'd adsi scripts for the 7960's? I saw a very basic one included in the config samples, but I thought there were some for browsing VM on screen and whatnot... |
| 01:39.55 | h3x | that and he dosent have any way to test it |
| 01:40.00 | PlainWhiteTrash | hehehehehehehhehehe |
| 01:40.12 | h3x | I have aculab cards that support 2B transfer if you want to buy them |
| 01:40.13 | essobi__ | so a waste of his time in other words. ;) |
| 01:40.39 | PlainWhiteTrash | XARiUS.. The 7960 is a SIP phone... it's not an adsi phone... |
| 01:40.40 | alan408 | <PROTECTED> |
| 01:40.43 | alan408 | I run that |
| 01:40.44 | h3x | if your application is really simple i could write you the software to handle the calls |
| 01:40.51 | PlainWhiteTrash | The 7960 does have a rather useful XML web browser though.. which can do nice things... |
| 01:40.51 | alan408 | after I place the call file [using move] I get no debugging telling me where it hosed up |
| 01:40.58 | XARiUS | pwt: I thought you could still load config's on them though? |
| 01:41.07 | h3x | er. where did cript go anyway |
| 01:41.07 | alan408 | any suggestions how to get debugging on why callfile doesn't do 'what I want' ? |
| 01:41.25 | PlainWhiteTrash | sure you can -- but not adsi... it's very different concept.. The 7960 is more like a cellphone with a WAP browser. |
| 01:41.31 | XARiUS | I see |
| 01:41.35 | h3x | but anyway, NFAS is far more important than 2B transfer coz all you gotta do is just get trunks and flash em |
| 01:41.35 | XARiUS | so it's all done with xml then |
| 01:41.37 | h3x | or just get more T1s |
| 01:41.48 | h3x | I think its in digium's best interest for you to order more T1s so they sell more boards :PO |
| 01:41.52 | h3x | thats the real reason i think |
| 01:41.56 | XARiUS | I was reading up on the wiki about the directory and some other xml stuff, but there weren't really any examples to go by |
| 01:41.58 | PlainWhiteTrash | The ADSI stuff is more like a voice patch with a scriptable tty interface inband... |
| 01:42.07 | XARiUS | yeah I saw it was only 1200b/s |
| 01:42.23 | PlainWhiteTrash | And while it's drawing screens you loose voice... |
| 01:42.28 | timecop | what does "C" mean during cvs checkout |
| 01:42.28 | XARiUS | ewie. |
| 01:42.30 | h3x | adsi rules but no one has written decent scripts for it |
| 01:42.34 | XARiUS | ok adsi sucks then. xml it is. |
| 01:42.36 | PlainWhiteTrash | and it has lots of other horrible options... |
| 01:42.44 | h3x | printer! |
| 01:42.44 | PlainWhiteTrash | h3x.. i actually wrote a decent one for it... |
| 01:42.47 | PlainWhiteTrash | for *... |
| 01:42.49 | Bobby_Ewing | PlainWhiteTrash: what's the max time you should ever be dialing someone on an extension in extensions.conf ? If i have a dial string and it's anything over 60 seconds, once it reaches 60 seconds anyway, all the phones stop ringing plus when i disconnect from the dialing phone, that crashes - is that a normal feature or something wrong with the gs |
| 01:42.49 | h3x | whats it do |
| 01:42.52 | PlainWhiteTrash | and fixed the adsi compiler to load it... |
| 01:42.59 | XARiUS | I'll post to the list, I've heard chatter in here about people browsing their vm on the 7960's, I'm sure someone has config samples they'd share |
| 01:43.02 | PlainWhiteTrash | the adsi compiler changes were contributed back and merged... |
| 01:43.06 | PlainWhiteTrash | the script so far has not been. |
| 01:43.15 | h3x | so whats the script do ? |
| 01:43.27 | XARiUS | it just sets up like vm buttons and junk'n'stuff |
| 01:43.33 | XARiUS | speed dials, basically. |
| 01:43.45 | XARiUS | the included script does, anyway. |
| 01:43.50 | PlainWhiteTrash | Provides nice soft keys and maintains all the right state engies to show who's one the phone with you, switch between call waiting calls, conference up calls, etc. |
| 01:44.02 | h3x | oh thats cool |
| 01:44.08 | PlainWhiteTrash | It's not perfect.. but much better than what's out there... |
| 01:44.15 | h3x | so can you call waiting more than 2 calls then or ? |
| 01:44.27 | h3x | my workaround was just to use 2 line phones |
| 01:44.30 | PlainWhiteTrash | Nope... That's theoretically possible but would require changes in zaptel... |
| 01:44.33 | h3x | and have groups for every extension |
| 01:44.33 | Cript | sorry back |
| 01:44.36 | h3x | so that calls roll to the 2nd line |
| 01:44.37 | Cript | reading past messages now |
| 01:44.47 | h3x | fuck call waiting... |
| 01:45.14 | XARiUS | thats why I wanted 7940's or 7960's. call appearences, I hate call waiting. |
| 01:45.19 | PlainWhiteTrash | if someone took the time to implement "call waiting deluxe" in zaptel, it would open up some major improvement to *'s ADSI support. |
| 01:45.21 | alan408 | anyone available for some minor consulting, pls message me - trying to automate outbound calls |
| 01:45.34 | PlainWhiteTrash | XARiUS... that works well... |
| 01:45.36 | *** join/#asterisk beedauchon (~fractal@3ffe:bc0:8000:0:0:0:0:1d7) |
| 01:45.41 | PlainWhiteTrash | The 7960s are quite good at that.. |
| 01:45.44 | XARiUS | good, I hope so. |
| 01:45.47 | XARiUS | thats why i bought'em hehe |
| 01:45.58 | XARiUS | hell if xlite can do it, surely cisco can. =) |
| 01:45.58 | timecop | HELLO what hte FUCK does "C" mean during CVS checkout i'm trying to udpate to latest asterisk and i'm getting "C" status for chan=sip.c |
| 01:46.10 | PlainWhiteTrash | timecop: conflicts of merge! |
| 01:46.21 | timecop | PlainWhiteTrash: it didnt show any errors |
| 01:46.37 | h3x | 7960s are stupid |
| 01:46.41 | h3x | you cant drop one part of a conf call |
| 01:46.42 | PlainWhiteTrash | timecop... so blame cvs... if it's status is C, then you have a corrupt sip.c :-) |
| 01:46.44 | h3x | thats fucking RETARDED |
| 01:47.09 | timecop | no, I have my own code in it that kram wont accept |
| 01:47.14 | timecop | the thing is it usually doesnt mind |
| 01:47.15 | PlainWhiteTrash | h3x - i agree - the only thing i don't like about the 7960s call management is the internal conf. bridge... |
| 01:47.22 | timecop | and hwen it does it says conflicts during merge |
| 01:47.23 | timecop | that I can fix |
| 01:47.26 | timecop | but now it jsut says "C" |
| 01:47.31 | timecop | without any explanation |
| 01:47.40 | Cript | All these companys that offer Biz lines with Interlata / InterState Wide Calling for free All say now allowed to connect it to a PBX or VOIP etc. But no real way for them to find out as long as your only using the pbx to dial out right? |
| 01:47.50 | PlainWhiteTrash | timecop - just the same... C is a conflict in merge... go check the file.. you should find it marked up with the conflict sections... |
| 01:47.57 | Cript | now = not |
| 01:48.14 | timecop | oh, right |
| 01:48.20 | timecop | oh i see what htey did |
| 01:48.25 | timecop | they finally implemented user-agent changing |
| 01:48.30 | Merlin | timecop sup |
| 01:48.31 | timecop | that was one of my mods |
| 01:48.32 | PlainWhiteTrash | <<<<<<<< >>>>>>> ====== :-) |
| 01:48.33 | timecop | merlin hi whore |
| 01:48.36 | Merlin | hahah |
| 01:48.39 | timecop | merlin what are YOU ding here |
| 01:48.41 | timecop | er doing |
| 01:48.45 | Merlin | asterisk: dude, i love this project |
| 01:48.48 | Merlin | err |
| 01:48.52 | Merlin | you know what i mean |
| 01:49.00 | Merlin | timecop: are you using it yet? |
| 01:49.06 | XARiUS | I'm stoked, I finally get to put * in production next week |
| 01:49.07 | PlainWhiteTrash | Cript - of course they'll know ;-) You'll probably be using high volume... |
| 01:49.10 | Bobby_Ewing | thanks for the help Moc__, PlainWhiteTrash etc.. much appreciated. I will try and get a response out of gs about the crashing phone |
| 01:49.11 | *** join/#asterisk r0d3nt-600m (~RatMan@wsip-24-234-241-186.lv.lv.cox.net) |
| 01:49.12 | XARiUS | been waiting just over a month now! |
| 01:49.14 | Bobby_Ewing | ciao3 |
| 01:49.16 | timecop | um i've been, for about a year at least |
| 01:49.21 | Merlin | timecop: very cool |
| 01:49.44 | timecop | PlainWhiteTrash: yep, fixed. ok cool thanks |
| 01:49.53 | PlainWhiteTrash | np |
| 01:49.53 | Merlin | i want to extend the voicemail to link to LDAP |
| 01:49.59 | timecop | ew |
| 01:50.03 | timecop | Merlin: good luck |
| 01:50.08 | XARiUS | yeah seriously. |
| 01:50.11 | Merlin | why? |
| 01:50.15 | alan408 | anyone have a minute to help me figure out why my callfiles in outgoing aren't causing calls to launch? |
| 01:50.34 | XARiUS | actually, it shouldn't be too bad with the right permissions |
| 01:50.40 | XARiUS | port 389 I think it is to the ldap server |
| 01:50.47 | XARiUS | should be able to do your queries there. |
| 01:50.52 | Merlin | it seems like a no brainer to me... you only have to maintain your employee directory in one place |
| 01:52.35 | *** join/#asterisk xeaded (~xeaded@69.88.201.41) |
| 01:53.17 | Cript | PlainWhiteTrash: But if I wont be using high volume. And even if I am. I could just say i make a lot of calls =) |
| 01:54.35 | Merlin | timecop: when are you moving back to the US? |
| 01:54.35 | Merlin | haha |
| 01:54.36 | PlainWhiteTrash | Cript... they'll eventually choose not to do business with you if they catch it.. read the fine print... In general, if a business partner doesn't want to do business with you... find someone else :-) |
| 01:55.55 | timecop | Merlin: huhu |
| 01:57.48 | xeaded | Could someone help with my Cisco 7960G phone? I always get the error "Registration from 'sip:101@192.168.0.3' failed for '192.168.0.51' I have both the username and password set in both the sip.conf file and on the phone. |
| 01:58.13 | *** join/#asterisk michael1234 (mick1234@202.43.239.10) |
| 01:58.35 | michael1234 | How can I get rid of old versions of openh323 so I can install the lastest ones |
| 01:59.42 | h3x | Cript: typically the 2B channel transfer and centrex transfer features are limited to only letting you place twice the number of calls of your span |
| 01:59.51 | h3x | so for instance you cant have more than 46 calls up on a single PRI |
| 02:00.18 | h3x | and... considering that some other CLEC besides AT&T would cost less than half as much per PRI |
| 02:00.25 | h3x | you're better off just doing normal call bridging |
| 02:04.58 | Cript | how much per pri h3x? |
| 02:05.07 | Cript | about $500 is standard isnt it? |
| 02:05.34 | Connor | ~seen jerjer |
| 02:05.35 | jbot | jerjer is currently on #asterisk (1d 1h 5m 12s). Has said a total of 169 messages. Is idling for 2h 23m 57s |
| 02:06.04 | Cript | No idea what AT&T charges |
| 02:06.19 | *** part/#asterisk essobi__ (kstone@75.137.26.216.host.teledvance.com) |
| 02:09.47 | timecop | um what hte fuck |
| 02:10.08 | timecop | Jun 5 11:08:39 NOTICE[131081]: chan_iax2.c:4998 socket_read: Rejected connect attempt from 65.39.205.121 |
| 02:10.14 | timecop | i get this after doing cvs udpate |
| 02:10.30 | JerJer | you need a type=user |
| 02:10.54 | timecop | instead of type=friend? |
| 02:11.11 | venix | is type=friend being steadily deprecated? |
| 02:11.30 | venix | its use is obviated most of the time |
| 02:11.38 | Cript | Digim have more than 1 Analog rj11s on a single Card yet? or still only the 1 port? |
| 02:11.56 | timecop | Cript: um? |
| 02:11.58 | h3x | at&t charges upwards of a grand |
| 02:12.03 | timecop | Cript: tdm400 is 4 ports |
| 02:12.07 | h3x | they charge for the loop seperately from the pri itself |
| 02:12.11 | timecop | can be either fxo or fxs |
| 02:12.13 | h3x | same thing with xo |
| 02:12.31 | h3x | sometimes at&t charges for local outbound even in non-metered markets |
| 02:12.38 | h3x | what city are you in |
| 02:12.54 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 02:13.47 | h3x | oh you are here in vegas? |
| 02:14.05 | h3x | i have a xspedius PRI at home |
| 02:14.10 | h3x | its the cheapest thing there is here for sure |
| 02:14.20 | timecop | ok what the fuck |
| 02:14.28 | h3x | and im getting a ds3 from them at my data center |
| 02:14.44 | timecop | god fucking damn why do fucking cvs update break shit that worked for months |
| 02:14.51 | timecop | what hte fuck I cant get any incoming calls after todays update |
| 02:15.05 | timecop | Jun 5 11:14:14 NOTICE[131081]: chan_iax2.c:4998 socket_read: Rejected connect attempt from 65.39.205.121 |
| 02:15.22 | venix | if you want stable, get stable. |
| 02:15.36 | JerJer | stable is NOT stable |
| 02:15.41 | JerJer | it will become stable |
| 02:15.43 | venix | yeah, well. |
| 02:15.45 | JerJer | get a clue |
| 02:15.47 | venix | just sayin |
| 02:16.05 | timecop | anyway |
| 02:16.08 | timecop | so what hte hell is wrong |
| 02:16.18 | timecop | i cant get any incoming SIP calls from FWD |
| 02:16.33 | snewpy_ | timecop: insecure=very for each peer thats affected |
| 02:16.41 | timecop | uh |
| 02:16.43 | timecop | "peer" |
| 02:16.48 | timecop | those are incoming calls from whatever |
| 02:17.01 | timecop | where is this shit documented? |
| 02:17.18 | Merlin | timecop: your mom |
| 02:17.21 | snewpy_ | head tries to authenticate all incoming calls, by the looks of it, and it would appear that no one actually does this |
| 02:17.45 | Cript | damnYea vegas h3x |
| 02:18.00 | timecop | i hope i can jsut put that shit into globals |
| 02:18.24 | XARiUS | it's on the mailing list all over the place |
| 02:18.40 | timecop | so how the fuck do I disable it |
| 02:18.50 | XARiUS | insecure=very |
| 02:18.50 | timecop | i'm wasting time wiht this shit not working |
| 02:18.55 | timecop | insecure=very in global context doesnt work |
| 02:19.08 | XARiUS | thats because it goes next to each peer you have defined. |
| 02:19.13 | h3x | cool |
| 02:19.14 | *** join/#asterisk NuMLock (~DeviL_OfG@dialup-179.nas01.azerin.com) |
| 02:19.15 | Cript | no not a tdm400p |
| 02:19.16 | h3x | so whats your application then? |
| 02:19.16 | Darwin35 | gui.c:294: error: conflicting types for `gtk_widget_modify_bg' |
| 02:19.17 | Darwin35 | /usr/X11R6/include/gtk-2.0/gtk/gtkwidget.h:656: error: previous declaration of `gtk_widget_modify_bg' |
| 02:19.17 | Darwin35 | gmake: *** [gui.o] Error 1 |
| 02:19.30 | Cript | i need connected to phone lines not analog phones =) |
| 02:19.37 | Darwin35 | and I just updated gtk |
| 02:19.41 | timecop | Cript: as I said |
| 02:19.44 | timecop | Cript: tdm400 |
| 02:19.52 | timecop | wiht 4 fxo modules |
| 02:19.57 | timecop | connects you to 4 phone lines. |
| 02:20.26 | Cript | hmm |
| 02:20.27 | *** part/#asterisk lightn (lightn@lightn.org) |
| 02:20.40 | timecop | XARiUS: what hte fuck, if i have like 200 peers in my sip conf |
| 02:20.50 | Cript | Digium makes it? dont see it on their site got a link for me? |
| 02:20.52 | timecop | who came up wiht the idea it couldnt be global? |
| 02:21.04 | *** join/#asterisk paulc (~web-irc@S010600062586a0b4.vc.shawcable.net) |
| 02:21.05 | XARiUS | no clue, just know it's a new "feature" |
| 02:21.11 | XARiUS | and thats the resolution for it. |
| 02:21.14 | XARiUS | conf wise, anyway. |
| 02:21.33 | timecop | http://store.yahoo.com/asteriskpbx/newitd4pofxo.html |
| 02:21.36 | timecop | Cript: ^^ |
| 02:22.30 | timecop | http://store.yahoo.com/asteriskpbx/newitd2pofxo.html |
| 02:22.39 | timecop | green modules are fxs, red are fxo |
| 02:22.46 | Connor | who else offers 1800 #'s ? |
| 02:22.48 | timecop | you can mix and match them in whatever way |
| 02:23.40 | *** part/#asterisk paulc (~web-irc@S010600062586a0b4.vc.shawcable.net) |
| 02:24.17 | Darwin35 | over priced |
| 02:24.19 | *** join/#asterisk paulc (~web-irc@S010600062586a0b4.vc.shawcable.net) |
| 02:24.21 | Cript | Thanks |
| 02:24.28 | Darwin35 | digium need to lower its prices |
| 02:24.46 | Darwin35 | yes the money helps support the dev |
| 02:25.02 | Darwin35 | but still to expensive for the average person |
| 02:25.18 | JerJer | um no |
| 02:25.22 | ReniRmx_ | a developers kit isn't too bad |
| 02:25.28 | timecop | Darwin35: wtf are you talking about |
| 02:25.31 | JerJer | Dialogic is expensive |
| 02:25.35 | timecop | equivalent equipment from elsewhere is 2x that price |
| 02:25.39 | timecop | at least |
| 02:26.03 | timecop | JerJer: does h323 havea built in gatekeeper yet? |
| 02:26.09 | XARiUS | buying half price knock offs from china isn't exactly supporting * hehe |
| 02:26.19 | Darwin35 | I knw |
| 02:26.26 | *** join/#asterisk Inca (~Inca@c207.134.5-98.clta.globetrotter.net) |
| 02:26.33 | JerJer | timecop: it will if you are willing to fund the development |
| 02:27.11 | Darwin35 | but I just wish there was a program to help those who are working to port * to other os to make payments not giving it all up at once would even help |
| 02:27.32 | Darwin35 | 2 to 3 paysmenst foor a x100p would make it easier |
| 02:27.46 | XARiUS | isn't an x100p like, 95 bucks? |
| 02:27.47 | timecop | darkthrn: like hat OS are you porting asterisk to? |
| 02:27.55 | timecop | theres a couple grand bounty last I checked for getting it to work on freebsd |
| 02:28.03 | timecop | there's a bug on that on bugs.digium |
| 02:28.17 | PlainWhiteTrash | Later guys.... |
| 02:28.45 | JerJer | and Digium did have a program to acquire free Digium hardware for asterisk development |
| 02:28.51 | timecop | huhu |
| 02:28.51 | JerJer | not many took them up on the offer |
| 02:29.02 | timecop | Darwin35: what OS are you porting asterisk to |
| 02:29.08 | Darwin35 | <PROTECTED> |
| 02:29.24 | Darwin35 | we are still workign on the fbsd port fixing bugs |
| 02:29.25 | JerJer | generate more revenue then |
| 02:29.33 | michael1234 | does anyone have trouble installing oh323 |
| 02:29.34 | XARiUS | Darwin: not to knock your fixed income or anything, but anyone dabbling in voip must understand there are costs involved. |
| 02:29.36 | Darwin35 | if I could I would |
| 02:29.38 | Inca | Hi. Trying to compile Asterisk and getting all kinds of errors. I am simply type "make install" inside the directories that were created with cvs. Looks like some headers might be missing. Am I missing somethin obvious? |
| 02:29.52 | timecop | Darwin35: you know that whoever gets it working will get a couple grand according to that pounty I mentioned right? |
| 02:29.55 | JerJer | Inca: kernel source |
| 02:30.09 | XARiUS | 95 isn't too much for a quality card, but if you had any idea what it's like running a billing department that handles "payments", it can be a nightmare. |
| 02:30.09 | timecop | Darwin35: i duno the bug number but search bugs.digium.com |
| 02:30.12 | XARiUS | Losses are horrendous. |
| 02:30.39 | Darwin35 | timecop thats for drivers |
| 02:30.46 | XARiUS | it's not so bad in software, but in hardware, you're out product. |
| 02:30.48 | Darwin35 | we have the x100p working |
| 02:30.48 | Inca | JerJer: might you have a quick reference URL? |
| 02:31.07 | Darwin35 | but they are still porting the other cards |
| 02:31.07 | JerJer | so start sending Digium money and when you've sent them enough, they will ship your hardware |
| 02:31.09 | JerJer | simple |
| 02:31.13 | Darwin35 | it wiol take time |
| 02:31.33 | timecop | Darwin35: cool, the sooner the better |
| 02:31.38 | Darwin35 | I do not see where they do layaway |
| 02:31.46 | timecop | Darwin35: how did you resolve gethostbyname_r |
| 02:31.49 | Darwin35 | not stated on thier sight |
| 02:31.55 | JerJer | call Malcolm or Greg |
| 02:32.09 | timecop | doesnt freebsnd not have that |
| 02:32.21 | Darwin35 | we did it a diff way |
| 02:32.26 | JerJer | i think they #defined it |
| 02:32.28 | XARiUS | someone's been working on bsd stuff, I saw some updates for zaptel I think it was go in the other day |
| 02:32.30 | JerJer | for bsd |
| 02:32.33 | Darwin35 | yes |
| 02:32.36 | *** join/#asterisk ncjp (~switch@61.206.115.4.user.ad.il24.net) |
| 02:32.47 | michael1234 | get this error message when compiling oh323 |
| 02:32.48 | michael1234 | chan_oh323.c:1855: too few arguments to function `ast_dsp_process' |
| 02:32.48 | michael1234 | make[1]: *** [chan_oh323.o] Error 1 |
| 02:32.59 | Darwin35 | gui.c:294: error: conflicting types for `gtk_widget_modify_bg' |
| 02:32.59 | Darwin35 | /usr/X11R6/include/gtk-2.0/gtk/gtkwidget.h:656: error: previous declaration of `gtk_widget_modify_bg' |
| 02:32.59 | Darwin35 | gmake: *** [gui.o] Error 1 |
| 02:33.06 | Darwin35 | now on gastman I getthis |
| 02:33.10 | Darwin35 | and its killing me |
| 02:33.12 | JerJer | michael1234: find the developer of that code... |
| 02:33.34 | michael1234 | I have oh323-astersik |
| 02:33.51 | JerJer | and i only support chan_h323 |
| 02:34.08 | JerJer | nobody else here is dumb enough to try |
| 02:34.34 | file | Nacho Cheese Doritos For All1 |
| 02:34.49 | timecop | does anyone still use gastman? |
| 02:35.01 | timecop | that crap didnt work worth a shit last time I tried it |
| 02:35.35 | Darwin35 | I had iton 4.9 and the cli worked fine |
| 02:35.48 | timecop | iton? |
| 02:36.04 | Darwin35 | iton it on |
| 02:36.22 | timecop | i could never get it to match channels to configured extensions |
| 02:36.30 | Darwin35 | dont correct small spelling like that its anoying |
| 02:37.00 | timecop | um |
| 02:37.05 | michael1234 | JerJer: Well how can i get more debug on chan_h323 |
| 02:37.16 | timecop | i was assuming "iton" was some asterisk manager cli |
| 02:37.22 | timecop | that I didnt know about |
| 02:37.31 | timecop | since all the GUIs I know of blow ass |
| 02:37.36 | Darwin35 | no if you read the sentance you can tell what it ment |
| 02:37.38 | JerJer | h.323 debug |
| 02:37.40 | timecop | and i'm still looking for a usable asterisk maanger |
| 02:37.42 | *** join/#asterisk hwstar (~hwstar@wsip-68-15-21-168.sd.sd.cox.net) |
| 02:37.48 | Darwin35 | but lets not get a war going here |
| 02:37.48 | JerJer | or h.323 trace 4 |
| 02:37.54 | XARiUS | so's the rest of the * community, timecop heh |
| 02:38.01 | JerJer | timecop: write one... I did |
| 02:38.01 | Darwin35 | back to trying to get it all working |
| 02:38.23 | timecop | JerJer: is it opensores? |
| 02:38.24 | *** join/#asterisk ptblank (~MURDER1@68-233-230-124.anhmca.adelphia.net) |
| 02:39.18 | *** join/#asterisk hwstar_ (~hwstar@wsip-68-15-21-168.sd.sd.cox.net) |
| 02:39.43 | JerJer | do you see me offering it up? |
| 02:40.01 | *** part/#asterisk Inca (~Inca@c207.134.5-98.clta.globetrotter.net) |
| 02:40.09 | XARiUS | haha |
| 02:40.32 | timecop | JerJer: didnt think so |
| 02:40.46 | Cript | Oh ok ic They just made FXO mods for the tdm400p |
| 02:41.11 | Cript | Anyway to order jsut the Four (4) FXO Modules (red)? |
| 02:41.18 | Cript | If i already have a 400p? |
| 02:41.21 | timecop | Cript: if yu ohave the card, yes |
| 02:41.24 | timecop | what revision is it? |
| 02:41.42 | JerJer | better yet, what color is it ? |
| 02:41.43 | Cript | had the Power Connector /shrug dont know off hand |
| 02:41.50 | JerJer | the pcb |
| 02:42.15 | *** join/#asterisk |^Angel^| (~angel@fw.aub.dk) |
| 02:42.44 | Cript | got the packing slip |
| 02:42.48 | Cript | doesnt say rev though |
| 02:42.53 | Cript | just TDM40B |
| 02:43.42 | Connor | what voip providers can provide 1-800 #'s both SIP and PSTN ? |
| 02:44.20 | paulc | Connor: I was looking for 800 SIP/IAX yesterday.. www.kall8.com do PSTN + VoIP |
| 02:44.35 | Darwin35 | where do I get g723 ? |
| 02:44.44 | Darwin35 | or is it in the src tree |
| 02:44.58 | JerJer | G.723.1 is encumbered by patents |
| 02:44.59 | Connor | paulc: What kind of rates? |
| 02:45.04 | Alric | g723 is only passthrough. |
| 02:45.21 | Alric | But if you have a few six-digit figure bank accts, they'll talk to you :) |
| 02:46.09 | Darwin35 | then we should pull it out of * |
| 02:46.44 | JerJer | there is no G.723.1 in asterisk |
| 02:46.55 | Darwin35 | codec_g723_1.c:46:28: g723.1/typedef.h: No such file or directory |
| 02:46.56 | Darwin35 | codec_g723_1.c:47:28: g723.1/cst_lbc.h: No such file or directory |
| 02:46.59 | JerJer | but it can support passing thru if its already encoded |
| 02:47.09 | JerJer | read the Makefile |
| 02:47.12 | Cript | timecop: How much are the mods each for the fxos? |
| 02:47.30 | timecop | Cript: the store has them. I think $75 each. yo uare better off getting the 4 port bundle at $377 |
| 02:47.33 | JerJer | mark isn't stopping anyone from getting their own Patent Indemifcation |
| 02:47.39 | timecop | or maybe not |
| 02:47.46 | timecop | 75*4 is 300 |
| 02:47.58 | Darwin35 | then it needs to be fixed this is from a fresh tree pull |
| 02:48.26 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 02:48.44 | JerJer | Darwin35: those are dependcy checks |
| 02:48.53 | JerJer | if those two flles exists the code compiles |
| 02:48.57 | JerJer | otherwise it moves on |
| 02:49.04 | paulc | Connor: Delayed reply, sorry.. umm.. I think 6.5c/minute.. not the cheapest but not bad.. |
| 02:49.08 | *** join/#asterisk cuban (~cuban@link.cuban.cc) |
| 02:49.28 | paulc | Connor: Alternative is to get a DID with free incoming that terminates VoIP then get a traditional 800 pointed at that number. That's what I've done for now, interim measure while I'm testing |
| 02:49.28 | JerJer | 6.5c is highway robbery |
| 02:49.29 | michael1234 | paulc: to where that is expensive |
| 02:49.34 | chap- | 6.5... wow. |
| 02:49.45 | *** join/#asterisk justinnnnnn (~justin@c211-28-201-105.eburwd1.vic.optusnet.com.au) |
| 02:49.47 | justinnnnnn | hey guys |
| 02:49.51 | michael1234 | I can terminate into the us for 1 cent per minute |
| 02:49.57 | michael1234 | from australia |
| 02:50.03 | justinnnnnn | im from aus 2 |
| 02:50.03 | paulc | JerJer: I know.. nufone = 2.9.. but I wanted Canadian origination and your Hayzell wasn't too up on it conversing via email yesterday |
| 02:50.18 | paulc | I get 1p/min from the UK to USA via voiptalk |
| 02:50.26 | JerJer | she seemed to converse with you pretty well |
| 02:50.28 | michael1234 | justinnnnnn: who do you use for voice |
| 02:50.29 | justinnnnnn | who does it for 1c/min from aus to usa ? |
| 02:50.46 | justinnnnnn | atm were using a small iax provider in melbourne |
| 02:50.48 | justinnnnnn | and our own pri's |
| 02:50.54 | JerJer | if you want canadian toll-free numbers its 10 cents per minute for any call...no strings attached |
| 02:50.57 | justinnnnnn | pri's in testing atm |
| 02:51.09 | file | JerJer: can I grab a second toll-free number? |
| 02:51.15 | paulc | JerJer: I was happy with speed of response, but would have liked a yes or no on the canadian origination question rather than being referred to phone sales.. just my £0.02 of grumpiness yesterday - I'm cool with it today |
| 02:51.20 | justinnnnnn | michael what about u ? |
| 02:51.42 | JerJer | paulc: huh? i specifically read her reply to you |
| 02:51.46 | JerJer | its called RT |
| 02:51.49 | justinnnnnn | bloody i hate those little square rock bolt thingies |
| 02:51.57 | JerJer | and i'm an AdminCC on every email that goes thru it |
| 02:52.01 | justinnnnnn | i cut my hands up crying to put them in /take them out yesterday |
| 02:52.56 | JerJer | yeah RT rules |
| 02:53.05 | JerJer | that's why i get so pissed off when ppl say we don't respond to them |
| 02:53.14 | JerJer | i have proof either way |
| 02:53.29 | JonR800 | getting pissed might not be the right response lol |
| 02:53.44 | paulc | so.. question is.. can I get a Canadian accessible toll free number, and for what rate? I'm totally happy paying above 2.9c a min, hell - I'd almost be happy paying double that.. but obviously it's got to work for you guys as well.. |
| 02:53.56 | JerJer | sure it is...espcially when they take their matter public |
| 02:54.27 | JonR800 | nah jerjer |
| 02:54.27 | JerJer | paulc: just as hayzell said if you want a Canadian accessable toll=free number the cost is currenlty 10 cents per minute for any call |
| 02:54.34 | JonR800 | that was their experience.. |
| 02:55.10 | JonR800 | getting mad/defensive about anything hurts the cause |
| 02:55.19 | JerJer | file: you can have as many toll-free numbers as youwant, but you have to email sales@nufone.net |
| 02:55.34 | michael1234 | justinnnnnn: Will just have to check with MD that is ok and then I will tell you |
| 02:55.36 | JerJer | JonR800: i disagree... i have no problems with firing customers |
| 02:55.46 | Darwin35 | I need a job where I can work from home. |
| 02:55.47 | E|ny | WOW. who charges 10c/ min? |
| 02:56.00 | JonR800 | JerJer: but when that firing is public it drives other new potential customers away. |
| 02:56.07 | justinnnnnn | michael1234: u mean to tell me who u use for voice ? |
| 02:56.11 | E|ny | I could do US/Canada 800#s for 7c |
| 02:56.14 | JerJer | JonR800: i can prove that statement wrong |
| 02:56.17 | justinnnnnn | where do u work michael ? |
| 02:56.25 | JonR800 | JerJer: i can prove it right. |
| 02:56.34 | paulc | JerJer: Ok thanks.. although I never got a quote for 10c/min from Hayzell.. For now I think I'll stick with what I'm doing for a laugh, it's just for testing, but I'll be talking more in the next few months |
| 02:56.49 | Alric | There are probably situations and effects that support both viewpoints. |
| 02:56.57 | JerJer | everytime someone has complained on the asterisk mailing list about us we get flooded with new customers |
| 02:57.02 | alan408 | can asterisk play mp3 files? |
| 02:57.07 | JonR800 | lol |
| 02:57.27 | JonR800 | how does that work? |
| 02:57.32 | JonR800 | i read them and it steered me away |
| 02:57.37 | Darwin35 | mpg123 |
| 02:57.42 | JerJer | and lots of them act suprized when we actually answer our fone, unlike what they were expecting |
| 02:57.43 | Darwin35 | if you installed it |
| 02:57.55 | Darwin35 | its needed for moh |
| 02:58.22 | Darwin35 | get the jisum out of your eyes |
| 02:58.24 | JerJer | ...or respond to email ... works the same |
| 02:58.29 | JonR800 | hehe |
| 02:58.37 | alan408 | anone know a good text to speech convertor, ie put text in, get GSM or AU/WAV out? |
| 02:59.07 | paulc | JerJer: I'm not up for bashing nufone at all, they seem to be well liked, well respected.. and apologies if this is a FAQ for the 10 millionth time.. but have you got plans for online signup with a credit card, prepay account top up etc? |
| 02:59.08 | justinnnnnn | that festival program does that ? |
| 02:59.14 | paulc | alan408: Festival |
| 02:59.15 | JonR800 | well whatever works for you. long as your making a profit i guess it doesn't matter much does it? :) |
| 02:59.25 | alan408 | thanks! |
| 02:59.29 | JerJer | paulc: when its ready |
| 02:59.47 | alan408 | is asterisk working on freebsd these days, out of the box/ |
| 02:59.49 | alan408 | ? |
| 03:00.21 | Darwin35 | not 100 % |
| 03:00.27 | Darwin35 | but we are close |
| 03:00.37 | Darwin35 | we now have the x100p supported |
| 03:00.45 | Darwin35 | and a driver in ports |
| 03:00.47 | alan408 | I give money to help if it would help. I just need SIP/VOIP stuff, no hardware on box |
| 03:01.06 | Darwin35 | it would |
| 03:01.06 | JerJer | run linux, yo |
| 03:01.09 | JerJer | keep your money |
| 03:01.20 | alan408 | linux/vmware/windows2k baby |
| 03:01.22 | alan408 | it's all that |
| 03:02.11 | *** join/#asterisk {-LeYLaaa-} (Paste@c211-28-62-206.sunsh2.vic.optusnet.com.au) |
| 03:02.33 | paulc | JerJer: Diplomatic answer :) s'all good |
| 03:03.09 | JerJer | put it this way: people keep throwing money at us to solve their problems, instead of us solving ours |
| 03:03.46 | JerJer | solving our own |
| 03:03.47 | alan408 | so you're a whore, you're saying? |
| 03:03.55 | paulc | Yeah - I know how that goes.. stuff that pays the bills generally gets done ahead of the "wouldn't it be nice" stuff.. |
| 03:04.02 | paulc | rude! |
| 03:04.13 | alan408 | oh your virgin ears |
| 03:04.58 | paulc | hang on.. wasn't it me talking about phone sex and my previous employer yesterday? ;- |
| 03:04.59 | alan408 | my kingdom for p orts |
| 03:05.23 | paulc | alan408: so a solid state type box, BSD, no hardware? |
| 03:05.49 | alan408 | yeah, using voip <-> voicepulse; asterisk serves as voip pbx for me |
| 03:07.15 | *** part/#asterisk njustinnnnn (~justinm@solid.mpa.net.au) |
| 03:07.32 | bkw_ | never fear bkw is back |
| 03:07.35 | bkw_ | back again |
| 03:07.57 | *** join/#asterisk switch (~switch@61.206.115.4.user.ad.il24.net) |
| 03:08.24 | JonR800 | guess who's back? |
| 03:09.13 | paulc | uh.. bkw_? at a guess.. |
| 03:09.36 | JonR800 | tell a friend |
| 03:09.56 | chap- | bkw: I have been gone all day. Get power back yet? |
| 03:10.37 | derrick | ~seen swk |
| 03:10.38 | jbot | swk is currently on #asterisk. Has said a total of 20 messages. Is idling for 1d 19h 49m 25s |
| 03:10.42 | paulc | LOL.. yeah, what was the story there? You wanted to get the power disconnected to the people next door or something, but you got cut off instead? |
| 03:11.05 | file | so I finished my computer science course today, 100%... teacher was speechless over my final project |
| 03:11.05 | chap- | paulc: Major storms. |
| 03:11.16 | chap- | file: What was your project? |
| 03:11.28 | file | an AOL Instant Messenger client |
| 03:11.37 | derrick | neato file :) |
| 03:11.49 | alan408 | dear teacher |
| 03:11.51 | alan408 | those who can do |
| 03:11.53 | alan408 | those who can't teach |
| 03:11.54 | alan408 | bye! |
| 03:12.32 | derrick | file, where ya going to school? |
| 03:12.34 | JonR800 | anything special about the aim client? |
| 03:12.36 | chap- | file: Thats cool. I guess your does not have all the bullshit ads and media that come up- even like on ICQ. |
| 03:12.47 | file | derrick: my local high school lol |
| 03:12.54 | derrick | badass file! |
| 03:13.03 | chap- | kids today. |
| 03:13.07 | file | JonR800: not particularly |
| 03:13.13 | file | it was cool to be allowed to chat in class though |
| 03:13.19 | JonR800 | ahh.. lol |
| 03:13.23 | JonR800 | good excuse.. i see |
| 03:13.30 | file | I've been done it awhile |
| 03:13.31 | JonR800 | im wise to your trickery |
| 03:13.54 | file | few days I worked on asterisk-win32 (which is on my workstation at school) and a messaging platform |
| 03:13.57 | derrick | file, my biggest HS code accomplisment was writing a ti calc 'memory eraser' emulater... obviously simple..but over 10yrs later..it's still traded at my old HS ;) |
| 03:14.13 | derrick | file, where do you want to go for college? |
| 03:14.14 | file | derrick: haha |
| 03:14.37 | file | derrick: I have a seat at the community college, and they've got a decent computer science course plus cisco networking |
| 03:14.47 | derrick | cmu? umich? ut? mit? |
| 03:15.08 | JerJer | hehe i dropped out of CMU :) |
| 03:15.18 | derrick | i dropped out since they didn't use to teach netwroking..and that's all i ever wanted to do |
| 03:15.25 | alan408 | canadia is the state up north by alaska, right? |
| 03:15.26 | derrick | i dropped out of PSU |
| 03:15.32 | file | well I'm trying to get an independent study for cisco networking |
| 03:15.47 | file | which would basically give me 2 semesters of it |
| 03:15.51 | _E_ | derrick is not to smart tho |
| 03:16.00 | JonR800 | PSU actually has a very good program now |
| 03:16.00 | derrick | not at all |
| 03:16.02 | _E_ | ; ) |
| 03:16.11 | derrick | they did then too |
| 03:16.12 | JonR800 | though most of them are geared toward the business end. |
| 03:16.38 | paulc | yeah - and we all got pet moose |
| 03:17.11 | JerJer | ~moose penis |
| 03:17.12 | jbot | MOOOOSE PENIS!!! |
| 03:17.15 | alan408 | text2wave myfile.txt -o myfile.wav |
| 03:17.17 | alan408 | yeah baby |
| 03:17.25 | derrick | file, why the focus on cisco? |
| 03:17.32 | alan408 | cisco networking is dead |
| 03:17.38 | alan408 | you will make less than an auto mechanic |
| 03:17.41 | alan408 | do something else |
| 03:17.53 | file | only thing really even offered or attempted to be offered |
| 03:17.55 | derrick | wow |
| 03:17.56 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 03:17.57 | paulc | alan408: you seriously think so? |
| 03:17.59 | file | and it opens up more possibilities |
| 03:18.01 | *** join/#asterisk epitron (~epitaph@Toronto-HSE-ppp3702728.sympatico.ca) |
| 03:18.01 | derrick | file, understood :\ |
| 03:18.02 | alan408 | i know so |
| 03:18.08 | alan408 | i used to not be able to spell network engineer |
| 03:18.10 | alan408 | now I is one |
| 03:18.11 | file | I've already been offered a position at a company in Mississippi |
| 03:18.22 | _E_ | ooof Mississippi ? |
| 03:18.25 | derrick | file, in my opinion, take it |
| 03:18.36 | derrick | work experience > school exp. |
| 03:18.36 | file | the trouble is moving, getting a visa, etc |
| 03:18.47 | paulc | file: where you at? |
| 03:18.48 | _E_ | derrick: agreed wholeheartedly |
| 03:18.53 | derrick | dude, us want's all of canadia it can hve |
| 03:18.56 | file | paulc: New Brunswick |
| 03:19.00 | JonR800 | work will also make you want to go back to school even more .. you'll have more motivation :-P |
| 03:19.28 | derrick | file, come share you magic number w/ the us |
| 03:19.30 | paulc | file: oot east eh? |
| 03:19.42 | file | paulc: yeah |
| 03:19.45 | alan408 | damn festival takse forever to compile |
| 03:19.48 | alan408 | poor me whine whine |
| 03:19.50 | alan408 | indian food time |
| 03:19.52 | paulc | <-- BC, Canadian born but grew up in the UK.. so I'm the Brit guy in Vancouver as far as the locals are concerned.. |
| 03:20.01 | file | paulc: haha |
| 03:20.06 | file | BC... where it rains alot |
| 03:20.13 | alan408 | For example in Latex files we do not want to here "left brace, backslash |
| 03:20.19 | derrick | <--wishes he was from ca |
| 03:20.20 | alan408 | I hate it when tech docs have mis-spellings |
| 03:20.24 | paulc | file: www.katkam.ca - been nice today, gonna rain all weekend |
| 03:20.38 | file | paulc: been raining here the last week, was finally nice today and should be nice tomorrow |
| 03:21.08 | alan408 | I wish make had a progress meter |
| 03:21.34 | file | I wish stupid people ... I'm gonna stop there |
| 03:22.40 | andrewg | alan408: distcc it up ;) |
| 03:23.03 | file | know what would be cool? to use the collective computing power of the school I work at during the summer |
| 03:23.22 | andrewg | someone already tried that and had to pay fines I think in the .us |
| 03:23.29 | alan408 | what would you do with it? |
| 03:23.31 | alan408 | serve porn? |
| 03:23.37 | paulc | what else?! |
| 03:23.38 | file | compile asterisk in seconds! |
| 03:24.06 | derrick | it already compiles in seconds |
| 03:24.18 | andrewg | nah |
| 03:24.21 | andrewg | convert asterisk to cpp |
| 03:24.25 | derrick | why? |
| 03:24.27 | JonR800 | unless you're on a p3 500 |
| 03:24.27 | andrewg | then use it to compile it in seconds :) |
| 03:24.30 | file | ugh, cpp |
| 03:24.35 | andrewg | derrick: because cpp is slowass :) |
| 03:24.38 | file | g++ is slow |
| 03:24.40 | derrick | heh |
| 03:24.47 | andrewg | having a cluster in search of a problem :) |
| 03:24.48 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 03:24.48 | andrewg | hahahah |
| 03:24.52 | derrick | that's the goal..make it slwer :) |
| 03:25.00 | andrewg | yep |
| 03:25.00 | alan408 | The system is too big. It takes a long time to compile even on quite large machines, and its foot print is still in the 10s of megabytes as is the run-time requirement. |
| 03:25.25 | paulc | are we gonna get in to a C versus C++ religious flame war? |
| 03:25.28 | alan408 | on a scale of 1 to 10 ; how good is the quality of festival's digitization of text to speech? |
| 03:25.36 | derrick | paul, so what sorta arch was/is it that ran your old project? |
| 03:25.37 | file | paulc: no |
| 03:25.37 | *** mode/#asterisk [+o denon] by denon_ |
| 03:25.45 | andrewg | alan408: no. people here agree c++ sucks. |
| 03:25.46 | paulc | derrick: the phone sex stuff? |
| 03:25.48 | derrick | yeah |
| 03:25.49 | andrewg | => |
| 03:25.56 | derrick | but i thought it wasn't phone sex :p |
| 03:26.10 | paulc | We had dual processor pentiums using Dialogic hardware with all call control stuff written in VOS (which is now Intel CT-ADE). |
| 03:26.18 | paulc | And no, it's not phone sex, it's "people meeting people", right? ;-) |
| 03:26.24 | derrick | and it could handle that much volume? |
| 03:26.26 | alan408 | I like C-- |
| 03:26.29 | file | DROP THE BOMB! |
| 03:26.30 | alan408 | like my GPA |
| 03:27.03 | paulc | derrick: Yeah.. we had 20 T1s in most systems and the CPU chugged along at around 40% loading |
| 03:27.22 | paulc | the new boxes they're putting in are like 28 T1s.. using the new(ish) Quadspan cards.. DMV/9600 etc |
| 03:28.05 | paulc | now the question is.. how to do a similar thing with Asterisk? I can't write C for shit.. but I'm alright(ish) with Perl.. go with AGI? or someone pointed me in the direction of anthm's perl module type thing yesterday |
| 03:28.12 | derrick | paul, yeessh..so the dialogics offload all the processing? that's insane |
| 03:28.44 | paulc | Scott Stingel talks about 4 T1s per box.. 8 would be nice.. but if it's 4, then it's 4.. I can live with that.. lots of pizza boxes, linked together with IAX when we need to bridge callers on separate boxes etc |
| 03:28.53 | *** join/#asterisk xeaded (~xeaded@69.88.201.41) |
| 03:29.14 | paulc | derrick: yeah - the Dialogic cards have 486 cpus on them I think.. plus there's not a lot of processing involved really in playing/recording speech |
| 03:29.22 | derrick | have ya thought of how to load between the pizza boxes? |
| 03:29.25 | derrick | is this all in-bound? |
| 03:29.33 | paulc | yeah.. vast majority is.. |
| 03:29.46 | *** join/#asterisk PrintScReeN (~BoTMeN@pop-ultel-1-156.azeronline.com) |
| 03:29.47 | paulc | so.. T1s with cyclic hunting.. or "super groups" as they're called on this side of the pond I think.. |
| 03:29.51 | derrick | my whole prob is xcoding tons of streams..and no hardware other than cpu to do it |
| 03:30.09 | SwK | thats what I'm concerned about on 1 project |
| 03:30.19 | xeaded | How can I get rid of the echo in calls over a X100P from my Cisco 7960? I have echocancel set to yes. |
| 03:30.19 | derrick | y0 ken |
| 03:30.21 | SwK | not more then prolly 2 T spans going at once |
| 03:30.24 | h3x | paulc: Its cheaper to get MAX TNTs with 0day firmware that does SIP |
| 03:30.25 | derrick | ken, did the sales droid call you yet? |
| 03:30.27 | h3x | and hook that up to asterisk |
| 03:30.28 | SwK | but howmuch CPU to throw at it |
| 03:30.30 | paulc | Yeah.. see most of our stuff would be just Zap channels on T1.. no transcoding.. a few VoIP calls for mucking around maybe.. |
| 03:30.35 | h3x | fuck t1 cards |
| 03:30.35 | SwK | derrick: nope |
| 03:30.37 | JerJer | xeaded: echotraining=yes |
| 03:30.39 | derrick | herm |
| 03:30.53 | h3x | compress a whole DS3 for about 10 grand |
| 03:30.57 | paulc | h3x: how's that work then? MAX TNT = T1 concentrators? like DS-3 splitters? |
| 03:31.02 | xeaded | thanks, i'll try that |
| 03:31.04 | h3x | DS3 in, ethernet out |
| 03:31.15 | *** join/#asterisk michael1234 (mick1234@202.43.239.10) |
| 03:31.19 | paulc | hmm.. now that's stirred things up.. |
| 03:31.22 | h3x | haha |
| 03:31.24 | paulc | JerJer: Why giggling? |
| 03:31.28 | h3x | these ex-ISP admins dont like lucent shit |
| 03:31.28 | michael1234 | JerJer: What do you do |
| 03:31.31 | paulc | derrick: why choking/coughing? |
| 03:31.33 | bkw_ | ok time to find these people and smack them around |
| 03:31.36 | bkw_ | who's next |
| 03:31.43 | h3x | cisco whores |
| 03:31.46 | h3x | well guess what |
| 03:31.51 | Moc__ | compress a DS3 ? |
| 03:31.54 | h3x | cisco AS* series cost more than digium cards |
| 03:32.00 | derrick | do tnt's scale now? |
| 03:32.09 | *** join/#asterisk Bay-X (~BackDown@62.217.137.156) |
| 03:32.12 | JerJer | hehe |
| 03:32.12 | h3x | scale? its a 1024 DS0 backplane |
| 03:32.14 | h3x | and DSPs |
| 03:32.23 | michael1234 | h3x: Yeah but lucent provider shit support and dropped callers |
| 03:32.27 | JerJer | lmao |
| 03:32.32 | paulc | I got to admit to being keen on a Digium/Asterisk solution.. compared to Dialogic, it saves 10s of thousands.. and that's before you figure in any VOS software licensing.. |
| 03:32.33 | h3x | lucent dosent do support anymore |
| 03:32.38 | h3x | they handed it off to some other company |
| 03:32.43 | paulc | so.. lots of pizza boxes with 4 T1s in each = scaleable, no? |
| 03:32.54 | h3x | even if the t1 cards were free it would cost more |
| 03:32.57 | h3x | cpu $$$ |
| 03:33.00 | h3x | M13 muxes $$$ |
| 03:33.05 | derrick | h3x, that's good..my 3 lucent guys were all borderline homicidal :) |
| 03:33.08 | michael1234 | h3x: Probably sick of the complaints. I dealt with lucent in australia and I will never deal with lucent again |
| 03:33.10 | h3x | G.729 Licenses if you need them $$$ |
| 03:33.26 | h3x | I've never really called tech support for anything before |
| 03:33.26 | bkw_ | all da pbx guys fear us |
| 03:33.29 | h3x | so i wouldnt know |
| 03:33.35 | bkw_ | you know that don't ya |
| 03:33.39 | bkw_ | they shake in their boots |
| 03:33.43 | h3x | all im like is "how much does it cost for a firmware upgrade". "$3500"... Hhmmmmm thats not so bad |
| 03:34.00 | michael1234 | h3x are you talking about cisco |
| 03:34.07 | JerJer | hehe funny shit |
| 03:34.09 | tessier_ | I've got a few TNT's and they are ok so far |
| 03:34.09 | h3x | no, im talking about using a shitload of asterisk boxes |
| 03:34.21 | h3x | tessier: with voip? |
| 03:34.23 | derrick | regardelss, a large room full of tnts couldn't do what a sonus gsx can |
| 03:34.30 | h3x | sonus shit sucks ass |
| 03:34.35 | derrick | elaborate |
| 03:34.40 | SwK | see... I gotta project now that needs voip off a legacy system ~48 channels (at once)... sure put a * at both end to connect the legacy gear, but seriously, what kinda CPU power is that gonna require when looking to use G729 |
| 03:34.41 | h3x | that crap has more cross talk problems and echo and clicking and shit |
| 03:34.44 | h3x | than anything ive ever seen |
| 03:35.00 | derrick | is that why 400 mil mins a month don't hae issues for us? |
| 03:35.07 | h3x | heh |
| 03:35.09 | Moc__ | JerJer, how car can the TDMoE can be used ? local only or can be used over a extended Wan ? |
| 03:35.10 | tessier_ | h3x: Unfortunately no, need to upgrade the shelf controller. Got them coming. |
| 03:35.11 | derrick | must be |
| 03:35.11 | michael1234 | JerJer: What do you do for a living |
| 03:35.20 | JerJer | Moc__: just sue IAX2 |
| 03:35.23 | *** join/#asterisk mizzie (~mizzie@host217-43-209-44.range217-43.btcentralplus.com) |
| 03:35.26 | JerJer | michael1234: code |
| 03:35.32 | paulc | so.. once we've all finished slagging off all the other companies.. is there anything wrong with taking a bunch of pizza boxes with 4 T1s in each? or for a system with more than a handful of T1s am I better looking at something else? |
| 03:35.32 | h3x | tessier: do you have ADI or 96DSP cards? |
| 03:35.40 | paulc | JerJer = god of nufone ? |
| 03:35.41 | h3x | (e.g. not 48MOD) |
| 03:35.57 | michael1234 | JerJer: is that why you like h323 so much. I see |
| 03:36.19 | michael1234 | sorry chan_h323 |
| 03:36.22 | JerJer | H.323 is a joke |
| 03:36.57 | JonR800 | SwK: i wouldn't use g729 for a project like that... why g729 anyways? |
| 03:37.01 | h3x | s/joke/joke on asterisk/ |
| 03:37.13 | JerJer | h.323 is a joke, period |
| 03:37.24 | SwK | JonR800: ever tried to get bandwidth outta central america to the states? |
| 03:37.26 | Moc__ | JerJer, can * keep a B channel intact from a T1 ? |
| 03:37.42 | Moc__ | and route it via IAX2 to send it to another B channel on remote side |
| 03:37.47 | JonR800 | SwK: lol.. no... i see. |
| 03:38.18 | derrick | swk, i'll follow up on monday about the sales idiots |
| 03:38.22 | michael1234 | JerJer: If I send you come debug can you tell me what is wrong so I can tear shreads off my provider |
| 03:38.25 | SwK | aight |
| 03:38.28 | JonR800 | SwK: http://www.digium.com/index.php?menu=asterisk_g729 |
| 03:38.42 | JonR800 | <PROTECTED> |
| 03:38.57 | h3x | fuck xeon |
| 03:39.01 | SwK | exactly |
| 03:39.01 | derrick | ? |
| 03:39.02 | h3x | P4 Extreme baby |
| 03:39.10 | SwK | Xeon's are so over priced |
| 03:39.12 | h3x | Server Works 7200 chipset |
| 03:39.25 | JonR800 | well.. im sure it doesn't have to be a xeon |
| 03:39.40 | JonR800 | but that gives you a base comparison at least. |
| 03:39.43 | h3x | you can do 96 calls with gsm and iax2 trunking on a P4 |
| 03:39.45 | michael1234 | but if you dont use a xeon the card doesnt fit |
| 03:39.48 | h3x | 3.0ghz or something |
| 03:39.52 | SwK | I'm thinking of tossing some AthlonMPs at it |
| 03:40.07 | derrick | i still haven't tested that load :\ |
| 03:40.08 | MysticOne | anyone have any advice for getting volume to an acceptable level on an x101p as well as making echo cancellation ont suck? :) |
| 03:40.11 | JonR800 | what card? |
| 03:40.19 | michael1234 | the digium card |
| 03:40.45 | JonR800 | umm heh |
| 03:41.35 | *** join/#asterisk SteveWrightNZ (~steve@pop11-port79.jetstart.maxnet.co.nz) |
| 03:41.39 | JonR800 | SwK: athlon mp's would probably do the trick |
| 03:41.58 | JerJer | MysticOne: are you running a Digium X100P or a knock off? |
| 03:42.11 | h3x | he said 101P |
| 03:42.15 | MysticOne | JerJer: digium |
| 03:42.27 | michael1234 | How much does it cost to manufactor an e1 card |
| 03:42.28 | JerJer | h3x: i'm always verifiying... i will not suport lamers |
| 03:42.43 | SwK | I really just wanna get a couple of 405s and see what I can get it to do |
| 03:42.45 | h3x | actually the Digium X100P is a IE-92 knock off if you wanna get your shit straight |
| 03:43.00 | MysticOne | seriously, the card came from digium a few months ago :) |
| 03:43.08 | file | You say yes... I say no... you say start... and I say go go go |
| 03:43.09 | SteveWrightNZ | gurus, [FAQ] where/how can I test linphone ? |
| 03:43.15 | JerJer | no Mark chose a specific card for specific reasons |
| 03:43.19 | file | oh no! you say goodbye, and I say hello |
| 03:43.23 | MysticOne | file: hehehe |
| 03:43.34 | JerJer | it just so happens other cards use a simular and driver compatable chipset |
| 03:43.37 | JerJer | hence inferor |
| 03:44.14 | MysticOne | so annnyway ... |
| 03:44.16 | *** join/#asterisk PlainWhiteTrash (~matt@user-12hcv1u.cable.mindspring.com) |
| 03:44.19 | JonR800 | is the x101p a lot better than the x100p? |
| 03:44.24 | JerJer | not really |
| 03:44.31 | Moc__ | let all get TDM400 FXO module instead |
| 03:44.46 | JerJer | motorolla stopped producing the chip used on the X100P card |
| 03:44.53 | JerJer | so mark had to punt |
| 03:44.54 | MysticOne | JerJer: so, any advice? :) |
| 03:45.01 | JerJer | do what Moc__ said |
| 03:45.06 | JonR800 | ahh so no difference really |
| 03:45.19 | JonR800 | lol.. TDM400 might be overkill for some folks |
| 03:45.31 | JerJer | its called expandaiblity |
| 03:45.38 | MysticOne | which was what? TDM400? :) |
| 03:45.46 | JerJer | you may not need the extra capacity today, but what about in 6 months? |
| 03:45.47 | Moc__ | I love my TDM400 |
| 03:45.51 | JonR800 | which some people don't need. |
| 03:45.56 | JonR800 | :) |
| 03:46.02 | Moc__ | JerJer, I find the quality better than x100p |
| 03:46.03 | MysticOne | I'm trying to get problems with this analog card ironed out, though |
| 03:46.06 | JerJer | then only get one module |
| 03:46.10 | Moc__ | answer faster,dial faster |
| 03:46.12 | JerJer | Moc__: most certianly |
| 03:46.39 | JerJer | i shitcanned my X100Ps the instant mark sold me an FXO module |
| 03:46.42 | JonR800 | it's still more money jerjer |
| 03:47.07 | Moc__ | hehe about the same here.. |
| 03:47.07 | MysticOne | so ... |
| 03:47.34 | MysticOne | the solution is get a TDM400 and PRI? |
| 03:47.54 | PlainWhiteTrash | You'll go far with a TDM400 & PRI :-) |
| 03:48.11 | SteveWrightNZ | which #forum for linphone or other SIP newbies / testers ? |
| 03:48.18 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 03:48.24 | JonR800 | mystic; i would check out the zapata.conf info on voip-info.org |
| 03:48.28 | MysticOne | well the thing is, right now I have the x101p ... (or x100p or wahtever the heck it is) and I'm trying to get it to work to the best of its abilities now, because it's what I have |
| 03:48.29 | JonR800 | namely settin rxgain and txgain |
| 03:48.30 | *** join/#asterisk epi (~epitaph@HSE-MTL-ppp72408.qc.sympatico.ca) |
| 03:48.39 | MysticOne | JonR800: so far what I've noticed is that rxgain and txgain seem to do absolutely nothing |
| 03:48.52 | JonR800 | did you stop/start asterisk after changing it? |
| 03:49.05 | MysticOne | yes |
| 03:49.05 | JonR800 | you have to restart.. a reload won't do it |
| 03:49.11 | MysticOne | this has been on-going for a few months |
| 03:49.19 | MysticOne | been trying to improve it, but, nothing has seemed to work so far |
| 03:49.26 | MysticOne | txgain/rxgain is in dB, right? |
| 03:49.29 | JonR800 | seems to do nothing .. you've tested with calls? |
| 03:49.52 | JerJer | have you followed bkw's instrucitons on how to tune tx and rx gain values ? |
| 03:50.06 | MysticOne | JerJer: where might those be? |
| 03:50.08 | *** part/#asterisk SteveWrightNZ (~steve@pop11-port79.jetstart.maxnet.co.nz) |
| 03:51.01 | MysticOne | the other thing is I'm not quite sure what to expect |
| 03:51.07 | MysticOne | should I reasonably be able to eliminate echo? |
| 03:51.12 | JonR800 | http://www.voip-info.org/tiki-index.php?page=Asterisk%20x100p%20echotraining |
| 03:51.22 | JonR800 | yes you should |
| 03:51.29 | MysticOne | JonR800: just loaded that page up, actually :) |
| 03:51.30 | JonR800 | it takes a few seconds for the card to get the echo down |
| 03:51.50 | JonR800 | but on mine.. generally ten seconds into the call and echo is gone. |
| 03:52.15 | MysticOne | I'd also heard that txgain/rxgain disabled echo cancellation |
| 03:52.17 | MysticOne | is that untrue? |
| 03:52.34 | JonR800 | i believe that's untrue. |
| 03:52.41 | JerJer | false |
| 03:52.50 | JerJer | where did you hear this horseshit? |
| 03:52.55 | JonR800 | lol |
| 03:52.59 | JerJer | (MysticOne) |
| 03:53.09 | MysticOne | so ... I should be able to get a normal volume level with the x100p and echo cancellation should work pretty well? |
| 03:53.14 | MysticOne | either in here or on voip-info |
| 03:53.20 | MysticOne | I can't recall where exactly |
| 03:53.29 | JonR800 | yes you should |
| 03:53.36 | derrick | trust no one |
| 03:53.36 | JonR800 | make sure to read the comment on that page |
| 03:53.39 | JonR800 | that i linked you |
| 03:54.22 | *** join/#asterisk DanGeR (~PrEnSeS__@219.95.194.81) |
| 03:54.23 | JonR800 | is echo better on the tdm400? |
| 03:54.23 | *** join/#asterisk `skygirl_ (~Aqua_Baby@219.95.194.81) |
| 03:54.47 | *** join/#asterisk epi (~epitaph@HSE-MTL-ppp72580.qc.sympatico.ca) |
| 03:55.17 | MysticOne | so I should have echotraining=yes and echocancel=yes? |
| 03:55.25 | JonR800 | yes |
| 03:55.33 | *** join/#asterisk jeffpc_ (~jeffpc@ool-44c218a8.dyn.optonline.net) |
| 03:56.05 | brc_ | ~seen manxpower |
| 03:56.06 | jbot | manxpower <~eric@dsl-208-164-150-160.datasync.com> was last seen on IRC in channel #asterisk, 5h 51m 41s ago, saying: 'Anyway, I've answer the one or two questions. Ta Ta y'all'. |
| 03:58.01 | *** join/#asterisk _beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 03:58.26 | *** join/#asterisk snewpy__ (~markl@203-217-35-65.dyn.iinet.net.au) |
| 04:01.50 | JonR800 | woot.. my GS phone should be here monday |
| 04:02.05 | JonR800 | it's no fancy pants 7960g but should be fun to play with |
| 04:02.57 | JerJer | hehe ok sure, if you say so |
| 04:03.20 | JerJer | who's giving odds on JonR800's loosing hair starting tuesday? |
| 04:03.21 | paulc | hehe.. I like my Cisco 7940 that I got off eBay |
| 04:03.40 | paulc | aren't GS phones those ones with the annoying ring tone that no one likes (especially for call waiting?) |
| 04:03.43 | paulc | or.. is that budgettone |
| 04:03.50 | JonR800 | hahah |
| 04:03.51 | JerJer | barbietone |
| 04:03.54 | michael1234 | JerJer: My providers notes say The call should be initialised as Gateway-Gateway not using a RAS/ARQ |
| 04:03.57 | JonR800 | i've been losing hair since birth |
| 04:04.00 | JonR800 | should be no different |
| 04:04.03 | michael1234 | is that how asterisks does things |
| 04:04.24 | JerJer | michael1234: are you trying to register to thier GK? |
| 04:04.29 | JerJer | that is ras |
| 04:04.43 | michael1234 | JerJer: No register |
| 04:04.47 | *** join/#asterisk adkr (adker@216.130.231.23) |
| 04:04.52 | JonR800 | barbietone lol |
| 04:04.55 | PlainWhiteTrash | DON'T DO IT! |
| 04:04.55 | JerJer | then you are talking directly to their gateway |
| 04:04.59 | derrick | mm barbie |
| 04:05.06 | PlainWhiteTrash | I JUST READ SOMEONE WAS LOOKING FORWARD TO GETTING A GS! |
| 04:05.13 | JonR800 | they're adding IAX so they say |
| 04:05.19 | JerJer | hehe yeah ok |
| 04:05.27 | PlainWhiteTrash | Doesn't matter.. The phone will still suck :-) |
| 04:05.30 | JonR800 | i know probably not for 10 years |
| 04:05.33 | JerJer | I volunteered about 6 months ago to do it for free |
| 04:05.38 | JerJer | they wouldn't even talk to me |
| 04:06.01 | JonR800 | eh it was $65 :) it's just going in my basement anyways.. if it sucks it sucks |
| 04:06.21 | JerJer | oh it sucks alright |
| 04:06.30 | PlainWhiteTrash | :-) I'm just really prejudiced against that phone because I invested way too much time and sweat on it... |
| 04:06.42 | JonR800 | what's so tough about it? |
| 04:06.48 | PlainWhiteTrash | I had this vision that maybe the phone was right and I was wrong... but no... That phone is just evil. |
| 04:06.55 | JonR800 | shit just doesn't work? |
| 04:07.08 | PlainWhiteTrash | Granted.. It's been a while since I played with it... |
| 04:07.09 | JerJer | i think its a Microsoft developer writing the firmware for the barbietone |
| 04:07.12 | JerJer | bloat |
| 04:07.28 | PlainWhiteTrash | But last time I checked.... Hold would not allow you to put the handset back in the cradle... |
| 04:07.28 | JonR800 | ahh too much crammed in there.. |
| 04:07.46 | PlainWhiteTrash | Call waiting didn't actually work at all, though the phone would ring for the second call... |
| 04:07.46 | JonR800 | lol.. i have a few analog phones that do that. |
| 04:07.51 | PlainWhiteTrash | All kinds of disasters... |
| 04:07.55 | JonR800 | ahh |
| 04:08.24 | JerJer | and the damn ringtone |
| 04:08.38 | PlainWhiteTrash | GrandStream killed my inner child :-( |
| 04:08.42 | JonR800 | HAHAHA |
| 04:08.45 | PlainWhiteTrash | ugh... especially the ringtone. |
| 04:08.46 | JerJer | and the fact that it looks like a toy i would give to my 5 year old kid or something |
| 04:08.59 | JonR800 | i might end up doing that too jerjer |
| 04:09.31 | PlainWhiteTrash | what I loved most was the cheap almost transparent white plastics.. the keypad backlighting leds shone right through the plastics and just looked ridiculous... In the dark it looked like this nasty off-pink eyesore. |
| 04:09.32 | JonR800 | give it to one of my cousins or something |
| 04:09.44 | MysticOne | JerJer: hey, after submitting money to you guys for nufone, how long until I get a number? :) |
| 04:10.10 | JonR800 | plain: it's a $65 phone.. you can't even get a decent analog for that |
| 04:10.32 | JerJer | MysticOne: depends on the number... vanity numbers take 7-10 business days |
| 04:10.57 | MysticOne | JerJer: no, just normal number ... not vanity |
| 04:10.58 | PlainWhiteTrash | Very true... It has a niche... If they can ever make it work.... And invest about $2/unit more on plastics :-) |
| 04:11.27 | JonR800 | haha yeah it does looke cheesy |
| 04:11.47 | JonR800 | currently though there aren't many mid range voip phones |
| 04:11.58 | JonR800 | the virbiage one looks awesome.. but as everyone has said.. vapor |
| 04:12.01 | PlainWhiteTrash | They could even do wonders for it... just by removing the brand label... Who wants to be reminded that they're talking on a "BudgeTone?!?" |
| 04:12.33 | JonR800 | lol the label won't bother me.. |
| 04:12.49 | JonR800 | i can't imagine putting them in a business setting hahahaha |
| 04:13.02 | JerJer | MysticOne: a few minutes then |
| 04:13.04 | paulc | LOL.. yeah, bit of a bad choice of name really innit |
| 04:13.05 | SwK | fuck |
| 04:13.09 | SwK | down to 1 cig |
| 04:13.14 | SwK | that sucks ass |
| 04:13.15 | JonR800 | look real professional on your budgetone |
| 04:14.23 | *** join/#asterisk Shado (shado@maxx.mc.net) |
| 04:14.33 | PlainWhiteTrash | Me either... but still... I have this vision of some asian techo-whorehouse churning these things out while the managers laugh all the way to the bank... "Stupid Americans! Even the name says cheap! When will they learn?" |
| 04:14.45 | file | JerJer: how much are vanity numbers? |
| 04:15.09 | JonR800 | heh cheap is what it's all about these days |
| 04:15.10 | Shado | Anyone good at TFTP updates to Cisco phones? |
| 04:15.14 | JonR800 | gotta cut those costs :) |
| 04:15.28 | doughecka | file: free I think |
| 04:15.33 | MysticOne | the BTs wouldn't be so bad if they didn't just *feel* so cheap |
| 04:15.36 | doughecka | just takes 2 weeks to get it |
| 04:16.01 | michael1234 | JerJer: With h323 how can I reduce this to 20 |
| 04:16.02 | michael1234 | <PROTECTED> |
| 04:16.03 | file | interesting |
| 04:16.04 | JonR800 | lol.. but they are cheap! |
| 04:16.15 | Shado | lddefault.cfg isn't making my phone download the SIP image... |
| 04:16.19 | MysticOne | JonR800: I know :) |
| 04:16.32 | JonR800 | i agree though.. if BT could make another phone with the same features (that worked).. for $30 more with better plastics |
| 04:16.36 | JonR800 | they'd sell a lot |
| 04:16.38 | PlainWhiteTrash | I agree... But there's a minimum of quality people will tollerate.. The idea of a home voip handset (other than cordless) is pretty much just silly... Almost no one wants a single handset versus an FXS port... And that phone has no place in typical office settings... So where do they see it going? |
| 04:17.19 | JonR800 | lol |
| 04:17.21 | *** join/#asterisk M-A-R-I-A (~Berfu_eLi@dialup-42.nas02.azerin.com) |
| 04:17.29 | JonR800 | well.. in my home setting it's probably getting tucked in my basement |
| 04:17.38 | hez | PlainWhiteTrash: not so true, some people will buy them for home use =) just not many ... heh |
| 04:17.54 | PlainWhiteTrash | Which is cool.. It has some limited applications. |
| 04:18.09 | JonR800 | other than that.. yes.. not much purpose |
| 04:18.24 | PlainWhiteTrash | But... really... I hope they weren't banking on volume :-) |
| 04:18.33 | JonR800 | i still think the killer would be a decent voip cordless system |
| 04:18.48 | hez | mmmm wifi iax2 phone ;) |
| 04:18.56 | doughecka | ilbc too |
| 04:19.01 | JonR800 | right now a decent SIP/IAX ATA is $100.. add in the cost of the phone |
| 04:19.05 | doughecka | with high wep too |
| 04:19.09 | JonR800 | i'd like to see a cordless voip system |
| 04:19.09 | *** join/#asterisk ardor (~ardorgof@ip68-224-74-19.lv.lv.cox.net) |
| 04:19.18 | JonR800 | that includes the AP in the charger |
| 04:19.25 | JonR800 | just drop and go |
| 04:19.33 | hez | doughecka: plah, you can get a cheap 900mhz here for 35 cdn |
| 04:19.34 | PlainWhiteTrash | No... I think that's the mistake everyone's been making... Why do wifi on the thing when you could do the voip part back in the base, and have multiple cheaper digital handsets with features communicated out of band... |
| 04:19.50 | JonR800 | that too plain |
| 04:19.50 | hez | PlainWhiteTrash: one word... roaming |
| 04:20.01 | PlainWhiteTrash | Cordless voip phone - yes... but I think wifi to the handset for general household use is a waste... |
| 04:20.04 | PlainWhiteTrash | The roaming this is cool... |
| 04:20.04 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 04:20.06 | JonR800 | that's also true.. but a home user won't roam |
| 04:20.07 | postel | JonR800: just get an ATA and connect a 900mhz on it and off you go |
| 04:20.20 | PlainWhiteTrash | Most home users will not be roaming... |
| 04:20.27 | JonR800 | postel: i know hehe.. i have a few.. just i think for a home user an ATA is a bit much to config. |
| 04:20.31 | JerJer | even sipuras are better than barbietones |
| 04:20.34 | hez | JonR800: oh yeah... sitting in starbucks... pull it out... and make calls =) |
| 04:20.35 | PlainWhiteTrash | The notion that a roamable wifi phone will replace a cell is almost laughable. |
| 04:20.51 | JerJer | PlainWhiteTrash: motorolla is building one |
| 04:20.52 | JonR800 | hez: who.. other than tech geeks would do that? |
| 04:20.57 | JerJer | cell fone+wifi sip |
| 04:21.02 | JonR800 | who other than geeks goes to starbucks? haha |
| 04:21.02 | hez | JonR800: true =) |
| 04:21.03 | paulc | how about a phone that's GSM (or whatever) when you're out and about, and a WiFi phone when you're home? |
| 04:21.12 | PlainWhiteTrash | JerJer - yea they're doing it on the iDEN stuff too, which I track pretty closely... |
| 04:21.14 | PlainWhiteTrash | But... |
| 04:21.16 | doughecka | JerJer: never |
| 04:21.18 | michael1234 | JerJer: Can you tell me how to force my end of h323 to be 20ms |
| 04:21.21 | michael1234 | <PROTECTED> |
| 04:21.21 | PlainWhiteTrash | They have the right base to do that. |
| 04:21.26 | doughecka | cell fone+wifi iax with ilbc! |
| 04:21.26 | doughecka | :P |
| 04:21.37 | JonR800 | paulc: that's a hack |
| 04:21.44 | hez | doughecka: sign me up! |
| 04:21.47 | JonR800 | i'd like to see 802.16 deployed |
| 04:21.51 | doughecka | lol |
| 04:21.54 | JonR800 | 802.11 nationwide is silly |
| 04:21.55 | SwK | hah |
| 04:21.58 | doughecka | JonR800: yea! |
| 04:22.03 | PlainWhiteTrash | The motorola iDEN phones (Nextel units) are common in corporate environments... Carrier installed wifi minicells inside the building.. Fix huge build inside coverage problems in free spectrum. |
| 04:22.41 | JonR800 | that's true.. |
| 04:22.50 | *** join/#asterisk miopea (~miopia@dhcp024-166-121-187.neo.rr.com) |
| 04:22.59 | JonR800 | lol.. low orbit sat's.. there we go |
| 04:23.01 | PlainWhiteTrash | I suspect we'll find that those phones will not just connect to any wifi system and signal calls on anyone's termination... It'll be an enhancement coordinated with the associated cellular carrier to relieve downtown type congestion one large corporate customer at a time... |
| 04:23.12 | PlainWhiteTrash | And... to provide superior in building coverage. |
| 04:23.25 | file | JerJer: RT is b0rken |
| 04:23.35 | miopea | Is there a dummy line for zapata.conf to allow for meetme/conference calls? I'm dealing in a 100% SIP/IAX environment |
| 04:23.37 | paulc | hmm.. don't see it working/happening though.. massive MASSIVE work innit |
| 04:23.49 | paulc | miopea: ztdummy? |
| 04:23.52 | JerJer | file: ? |
| 04:23.59 | miopea | paulc: what is that? |
| 04:24.02 | JonR800 | PlainWhiteTrash: they already do that for a few $$'s |
| 04:24.04 | JonR800 | hehe |
| 04:24.22 | file | pipe to |/opt/rt3/bin/rt-mailgate --queue sales --action correspond --url http://rt.nufone.net/ |
| 04:24.23 | file | <PROTECTED> |
| 04:24.23 | file | <PROTECTED> |
| 04:24.27 | JonR800 | i read what the cost of a private nextel tower was.. wasn't cheap |
| 04:24.39 | PlainWhiteTrash | Yea, I'm aware... They usually put in a cheap BDA and way overcharge for it... |
| 04:24.45 | PlainWhiteTrash | But that has serious disadvantages... |
| 04:24.46 | miopea | shame nuphones LD rates are soo bad |
| 04:24.53 | miopea | it was cheaper to call international with MCI |
| 04:25.14 | PlainWhiteTrash | wifi will let them do it cheaper and relieve congestion in their corporate owned/leased spectrum. |
| 04:25.35 | paulc | miopea: ztdummy is a dummy timing device needed for MOH, conferencing etc in boxes that you don't have zaptel hardware installed in |
| 04:25.44 | paulc | file: I fucking hate it when RT goes wrong like that |
| 04:25.44 | JonR800 | yeah.. i read they were going to put up 802.11 APs |
| 04:25.50 | JonR800 | err 802.11a APs |
| 04:26.00 | PlainWhiteTrash | Previously the only advantage to inbuilding coverage was for the tenants... Adding well implemented wifi would let the manufacturer tilt the benefits to the carrier. |
| 04:26.06 | miopea | Just at a line to my zapata.conf file? and make that a device? its on the wiki right? |
| 04:27.46 | paulc | miopea: yeah - you need to check the makefile and remove the # before ztdummy, recompile/install.. wiki should cover it :) |
| 04:28.09 | miopea | thanks a bundle |
| 04:28.36 | E|ny | are the asterisk variables, DNID, CALLERIDNUM read only variables? |
| 04:28.42 | E|ny | I'm unable to set them using setvar or setglobalvar |
| 04:28.43 | derrick | paul, i got distracted... |
| 04:28.50 | derrick | wha's yer goal for * ? :) |
| 04:29.12 | derrick | (from earlier convo on fux0r line) |
| 04:29.17 | Shado | are there 7905G firmware versions that don't like to update? |
| 04:29.55 | paulc | derrick: uh.. *tries to remember* we were talking about the chat line thing I think? |
| 04:30.03 | derrick | yeah |
| 04:30.21 | derrick | paul, but i'm curious what yer current job wants/needs |
| 04:30.34 | derrick | ie, where you need/want to go /w it |
| 04:30.53 | derrick | ie, what to contrib |
| 04:31.00 | paulc | current issues are size/scaleability of the boxes, how to get the traffic in, and whether I can technically do what I want to do |
| 04:31.42 | derrick | yeaaaaahhh |
| 04:31.49 | derrick | that's what it was..all in-bound? |
| 04:31.57 | derrick | what do ya need to do? |
| 04:32.23 | derrick | i'm in teh scalability box too |
| 04:32.35 | derrick | time allowing will correct that |
| 04:33.01 | *** join/#asterisk juLiet (~TuRKisH@218.111.10.197) |
| 04:33.06 | derrick | at least for my implementaion...but if you have one too...i'llmake it work for both |
| 04:33.15 | paulc | yeah.. vast majority of calls would be inbound. Most of the traffic is IVR type stuff.. message passing relay style.. database work.. but there comes a time I need to connect the caller to either a customer service rep, or another caller. In the first case, once the CSR hangs up, the caller continues in IVR.. in the second case, if either caller hangs up we need to signal the other one and they'll continue with IVR.. or if either caller |
| 04:33.56 | SwK | a date line |
| 04:34.07 | paulc | seems that in the TDM world (which I know and love) anything above 4 T1s in a box is pushing it.. fine, ok, lets have lots of pizza boxes.. I'm fine with that.. and am sure with some tweaking I can get callers talking to each other across boxes.. so then we're on to "how do we get traffic in to the box(es) from all over the place?" |
| 04:34.10 | paulc | swk: yup |
| 04:34.25 | SwK | paulc: what about TDNoE? |
| 04:34.29 | SwK | err TDMoE |
| 04:34.48 | *** part/#asterisk hwstar (~hwstar@wsip-68-15-21-168.sd.sd.cox.net) |
| 04:35.01 | paulc | not sure about that.. heard it's dodgy if nodes come up and go down.. plus you're then fixing the number of timeslots between boxes.. which you wouldn't be doing with IAX? |
| 04:35.15 | Shado | Anyone know the syntax for OpFlags in lddefault.txt? need to set 0x40 |
| 04:35.18 | SwK | I dunno |
| 04:35.36 | hez | what advantage would tdmoe have over iax2? |
| 04:35.48 | *** join/#asterisk uN{IQ}uE`18f (~oPtions@adsl-81-7-119-160.takas.lt) |
| 04:36.27 | file | JerJer: did I break it good? |
| 04:36.34 | paulc | yeah - I'd like to know that to.. is there any? |
| 04:36.58 | hez | from what ive been seeing iax2 is more flexable and better supported |
| 04:37.55 | paulc | yeah.. and it does clever things.. like detecting tromboned paths and dropping out of the equation |
| 04:38.11 | derrick | i'm confused...what do you need specifically? |
| 04:38.12 | hez | =) |
| 04:38.16 | Shado | yay... No thanks to you guys I got it done... |
| 04:39.16 | SwK | derrick: he's trying to figure out how to setup one of those telephone dating services you call all the time |
| 04:39.47 | paulc | LOL.. I know how they work, I've written them already.. using Dialogic/VOS.. I'm trying to suss out how it works in the Asterisk world, with multiple nodes forming a larger virtual system. |
| 04:40.09 | paulc | with the added north american headache of how to get traffic in to the systems for a fixed rate. |
| 04:40.15 | *** join/#asterisk menger (~menger@dsl-244.84.240.220.rns02-dryb-mel.dsl.comindico.com.au) |
| 04:40.51 | h3x | personally if i was doing that particular application id use a couple performance technologies dual ds3 cards and a aculab prosody or two on the backplane |
| 04:41.00 | h3x | (just because i wrote software for it of course) |
| 04:41.03 | *** part/#asterisk miopea (~miopia@dhcp024-166-121-187.neo.rr.com) |
| 04:41.06 | h3x | but you hardly ever need DSP resources |
| 04:41.08 | paulc | LOL.. slightly biased.. |
| 04:41.25 | h3x | not really because asterisk is theroetically cheaper, but on the other hand it isnt |
| 04:41.26 | h3x | in that scale |
| 04:41.31 | paulc | nah, a lot of what goes on is relay messaging.. I message you, you message me, back and forth, a bunch of times, before we finally decide to connect live with each other |
| 04:41.38 | h3x | Oh i see |
| 04:41.40 | h3x | so lots of play and record. |
| 04:42.05 | h3x | yeah i guess asterisk would be good for that |
| 04:42.06 | paulc | yeah.. don't have ratios between that and live chat.. but I'd say that the majority is relay messaging and mailbox work.. |
| 04:42.28 | paulc | live chat happens.. but it's not all live chat.. like those 1-900 saucy actress type lines |
| 04:42.32 | h3x | maybe the voip did idea i had isnt such a bad idea |
| 04:42.51 | h3x | if they can do it |
| 04:42.55 | h3x | what kind of coverage do you need |
| 04:43.02 | SwK | those 1-900 saucy actress type lines would work more like on a bridging basis anyway |
| 04:43.10 | paulc | Yeah - doing away with TDM T1s and having someone terminate DIDs over VoIP to me.. |
| 04:43.13 | SwK | which is prett much how they are all set up now |
| 04:43.13 | *** join/#asterisk file (~file@mctn1-7091.nb.aliant.net) |
| 04:43.15 | paulc | swk: yeah, exactly |
| 04:43.25 | h3x | how many cities / proviences ? |
| 04:43.31 | SwK | paulc: i know a couple of "operators" heh |
| 04:43.33 | paulc | h3x: north american wide eventually.. coast to coast.. |
| 04:43.40 | h3x | oh ok |
| 04:43.53 | paulc | swk: LOL.. Tracy, who talks dirty whilst breast feeding the baby and ironing a shirt or two at the same time? |
| 04:43.53 | h3x | well i should be able to fix your US problem |
| 04:44.09 | h3x | canadian DIDs are gonna be tricky but should be doable |
| 04:44.32 | *** join/#asterisk SaGopa_KaJmer (~KesHmen@170.cust20.vic.dsl.ozemail.com.au) |
| 04:44.34 | SwK | paulc: yeah something like that |
| 04:44.45 | paulc | s'funny.. cos it's easier for us to launch up here then move south.. but technically not so much :-s |
| 04:47.29 | *** join/#asterisk twisted (~twisted@user-69-73-55-199.knology.net) |
| 04:47.29 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:48.04 | michael1234 | what does this mean Read error: PER decode failure in Q.931 User-User Information Element |
| 04:49.54 | MysticOne | oh, that happens in Q.931 when the User-User Information Element has a PER decode failure |
| 04:50.07 | PlainWhiteTrash | Someone who knows for sure: does the out of band dtmf signalling in IAX convey duration and spacing of dtmf digits dialed? |
| 04:51.20 | MysticOne | :D |
| 04:51.26 | MysticOne | (obviously that was a joke, as I have no idea what that means) |
| 04:51.47 | PlainWhiteTrash | I think it's essentially a debug message from a recent patch to the PRI stuff... |
| 04:53.38 | JonR800 | the newish sipura firmware has a Hook Flash Tx Method would this possibly allow me to hook/flash a zap FXO ? |
| 04:53.50 | PlainWhiteTrash | Anybody reading know the answer to my out-of-band DTMF timing & spacing issue? |
| 04:54.33 | paulc | PWT: Nah.. But my gut feel is that it wouldn't know about duration and timing.. they're just events.. and I'm guessing DTMF detection is done on trailing edge.. |
| 04:54.58 | PlainWhiteTrash | That's kind of what I figured. |
| 04:57.25 | *** join/#asterisk WorLDCup (~Lesbiaaan@210.186.115.1) |
| 04:57.26 | *** join/#asterisk _kiz^ (~asaLak@dsl81-215-4892.adsl.ttnet.net.tr) |
| 04:57.54 | JonR800 | hmmm |
| 04:58.03 | JonR800 | Jun 5 00:56:25 WARNING[229391]: chan_sip.c:5434 receive_info: Unable to parse INFO message from 2f51ece23ffa07eb7eab4c28301f6df5@192.168.1.200. Content |
| 04:58.23 | JonR800 | so asterisk doesn't know how to parse that i flashed |
| 04:58.52 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 05:03.12 | michael1234 | MysticOne: What does the pdu do |