| 00:00.04 | *** join/#asterisk annonimous (n=annonimo@189.136.58.51) |
| 00:00.11 | annonimous | hiya! |
| 00:00.52 | JT | vlt: make sure you've stopped all other isdn drivers from running |
| 00:01.00 | JT | and check they're not running now |
| 00:01.07 | JT | vlt: hisax etc |
| 00:01.20 | *** part/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 00:01.38 | vlt | JT: I only did rmmod hfc8... How do I unload hisax? The same way? |
| 00:02.10 | JT | yep, you must make sure they do not load at boot either |
| 00:02.15 | JT | or ever |
| 00:03.35 | vlt | JT: I unloaded and blacklisted hfc8... and hisax. Trying to unload "isdn" tells me it was busy ... |
| 00:04.00 | JT | pm or pastebin the output of lsmod |
| 00:07.17 | vlt | JT: http://paste.ubuntu-nl.org/5848/ |
| 00:07.33 | foobar778 | <PROTECTED> |
| 00:08.29 | JT | ffs, stop repeating foobar778 |
| 00:08.33 | JT | you've been warned already |
| 00:09.10 | annonimous | can anyone help me with a little trouble with a fxs 4 port gateway? cause partialy registers and i cant get the incoming calls but it can dials for outgoing calls ='( |
| 00:09.43 | JT | vlt: try rmmoding slhc and isdn |
| 00:09.56 | JT | vlt: i can't believe there's so many modules loaded |
| 00:10.40 | JT | horrible ubuntu ;) |
| 00:11.21 | vlt | JT: I'm not very familiar with kernel stuff. What's wrong abou many modules? What would be better? |
| 00:11.52 | JT | only the module you need... |
| 00:12.29 | diclophis-work | how would i reset a queue to its orignal settings from queues.conf (including the members defined there?) |
| 00:12.50 | JT | on my systems, i'd call 20 the highside of loaded modules |
| 00:13.00 | JT | yours has 84 |
| 00:14.07 | *** join/#asterisk thor_n (n=thor@rrcs-24-153-226-142.sw.biz.rr.com) |
| 00:15.01 | annonimous | filosofical doubt if i update my asterisk box ill can get the gateway to register? cause it didnt register since i got a power fail (electrical) |
| 00:16.20 | vlt | JT: (OT) So which of my 84 loaded modules would you unload? |
| 00:19.25 | JT | it's pointless unless you stop them loading in the first place |
| 00:19.44 | JT | looks like heaps of useless shit like "battery" is it a laptop, i think that's for a laptop |
| 00:19.55 | JT | and some of that stuff should be compiled into a kernel not as modules |
| 00:21.39 | tzafrir_laptop | JT, what' the problem? |
| 00:22.25 | tzafrir_laptop | oh, it's vlt |
| 00:22.33 | tzafrir_laptop | kernel module or asterisk module? |
| 00:23.10 | JT | talking about kernel atm |
| 00:23.15 | vlt | tzafrir_laptop: kernel module -- off topic ;-) |
| 00:23.17 | vlt | JT: Maybe we should talk about the advatages of compiling stuff into the kernel later ;-) |
| 00:23.23 | JT | just looking at the horror that is ubuntu :P |
| 00:23.30 | JT | for a server at least |
| 00:24.18 | tzafrir_laptop | vlt, grep for the card's pci id in /lib/modules/`uname -r`/modules.pcimap |
| 00:25.05 | tzafrir_laptop | If a module gets loaded automatically by bus ID (and not from being pulled as a dependency), it would probably appear there |
| 00:25.17 | tzafrir_laptop | (for a PCI card) |
| 00:25.56 | tzafrir_laptop | You can blacklist a module in modprobe.d , but I don't recall the syntx |
| 00:27.42 | vlt | tzafrir_laptop: I already blacklisted the hfc4s8s... and hisax module. |
| 00:28.12 | JT | blacklisted from next time you boot? |
| 00:28.27 | tzafrir_laptop | vlt, one possibility: maybe they are actually loaded in the initrd? (the horros) |
| 00:30.06 | tzafrir_laptop | If you really want to check that, boot the system with 'init=/bin/bash' |
| 00:30.19 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 00:30.34 | vlt | tzafrir_laptop: Yes, blacklisted from next boot. And addidionally I did `rmmod hfc...` and `rmmod hisax` successfully. But I can't remove isdn (don't know if that would help). |
| 00:31.41 | vlt | tzafrir_laptop: "[23:55] <vlt> JT: Sorry, the card is about 80 miles away from me ..." |
| 00:32.03 | vlt | tzafrir_laptop: So rebooting to /bin/bash is not the way to go now ;-) |
| 00:32.52 | JT | vlt: did you remove the other module i suggested to? |
| 00:33.15 | vlt | JT: "battery"? not yet. |
| 00:33.22 | JT | no |
| 00:33.31 | JT | that won't directly help |
| 00:33.36 | vlt | JT: sorry, You mean idsn and |
| 00:33.42 | JT | < JT> vlt: try rmmoding slhc and isdn |
| 00:33.44 | vlt | slhc |
| 00:34.11 | vlt | JT: "ERROR: Module slhc is in use by isdn" |
| 00:34.25 | vlt | JT: "ERROR: Module isdn is in use" |
| 00:34.31 | vlt | :( |
| 00:35.05 | JT | you might need to reboot, i dunno |
| 00:35.07 | JT | anyway |
| 00:35.16 | JT | even more likely is you'll need to patch qozap |
| 00:35.55 | annonimous | can anyone could tell me about the trouble of an extensions that partially register? http://www.pastebin.ca/356301 |
| 00:36.14 | vlt | JT: Ok, I'll reboot. Should I blacklist isdn and slhc before, too? |
| 00:36.21 | JT | yes |
| 00:40.48 | [TK]D-Fender | annonimous : What do you mean "partiall registered"? And which of those? |
| 00:41.25 | annonimous | [TK]D-Fender, ok i mean partial that if i call to the extensions 200 to 203 i got a 503 sip error |
| 00:42.00 | annonimous | but if i pickup my phones into the 200 to 203 extensions and call to another extension "example the 101" i can call |
| 00:42.46 | annonimous | [TK]D-Fender, and i have this trouble about 4 days ago when the box receives a electrical down fail (we got out of electrical power) |
| 00:43.13 | annonimous | [TK]D-Fender, so the ups entered and we poweroff all the system and when we power up later the system we got tihs trouble |
| 00:44.50 | *** join/#asterisk netsurfer (n=bbjunkie@user-514d724a.l2.c2.dsl.pol.co.uk) |
| 00:49.29 | The_DoC^ | damn non paying customers = disabled account |
| 00:49.56 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:54.21 | [TK]D-Fender | annonimous : How is it you have 3 extension on the same ip? |
| 00:54.28 | [TK]D-Fender | annonimous : AND port? |
| 00:54.44 | annonimous | yep its a gateway audiocode 4 port fxs |
| 00:55.07 | [TK]D-Fender | annonimous : Should be on seperate port#'s at least... |
| 00:55.19 | JT | err why |
| 00:55.23 | annonimous | but the audiocodes dont have that funtion =S |
| 00:56.05 | The_DoC^ | I have a pap2 with the same ip for each ext and same port as well |
| 00:56.20 | JT | The_DoC^: does it work fine? |
| 00:56.27 | The_DoC^ | perfect |
| 00:56.29 | [TK]D-Fender | The_DoC^ : normally you'll see 5060 for one, and 5061 for the other... |
| 00:56.34 | JT | yeah i'd think so if setup right |
| 00:56.44 | JT | umm that makes no sense, unless to get around an asterisk bug |
| 00:56.56 | JT | should be able to go on destination sip address |
| 00:57.29 | The_DoC^ | it registers with asterisk and I am able to dial to and from each ext |
| 00:57.47 | JT | :) |
| 00:58.31 | The_DoC^ | now the rt31p2 I have that thing was a little bitch to set up |
| 00:58.52 | JT | what is it? |
| 00:59.28 | The_DoC^ | its a router with 2 fxs's |
| 00:59.39 | annonimous | like an utstarcom? |
| 00:59.40 | JT | ah ok, who makes it |
| 00:59.50 | The_DoC^ | linksys/cisco |
| 01:00.03 | JT | the rt31p2? |
| 01:00.15 | The_DoC^ | yea |
| 01:00.28 | annonimous | its a pap wit fxs fxo and router isnt it? |
| 01:00.30 | JT | i see |
| 01:01.24 | The_DoC^ | problem with it is that the fxs's are only accessable on the wan side, do you have to do port forwarding like you would with a normal router connected to the internet |
| 01:01.43 | The_DoC^ | just 2 fxs's no fxo |
| 01:01.48 | annonimous | oh ok |
| 01:01.49 | JT | ahh that's dumb |
| 01:02.03 | annonimous | [TK]D-Fender, curious case isnt it? |
| 01:02.05 | The_DoC^ | no kidding, but hell it was free |
| 01:02.38 | vlt | JT: Ok, I blacklisted and rebooted. The card is not activated during boot. But when I modprobe qozap I get the same timeout error messages. Will now try to patch qozap |
| 01:03.15 | The_DoC^ | is there a way to disable a linux user's login for a short time |
| 01:03.17 | JT | vlt: ok, probably easiest to patch, only other thing would be to load qozap with debug on see if it tells you anything |
| 01:03.21 | JT | but probably unnecessary |
| 01:03.49 | vlt | JT: debugging how? |
| 01:03.50 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-80-108.vic.bigpond.net.au) |
| 01:04.25 | JT | there are options you can pass to the driver when loading that will activate more debugging output |
| 01:04.42 | JT | are you loading qozap correctly? |
| 01:04.50 | JT | you must load qozap with the ports= option |
| 01:05.02 | JT | with an octal encoded span definition |
| 01:06.17 | vlt | JT: My fault ... modinfo tells me "parm: debug:i" How to tell modprobe those parms? |
| 01:07.10 | JT | modprobe qozap options............. |
| 01:10.36 | vlt | I ran `modprobe qozap debug=1`. Now it looks like this: http://paste.ubuntu-nl.org/5856/ |
| 01:10.45 | vlt | What does "octal encoded span definition" mean? |
| 01:11.15 | JT | did you read the readme/installation documents at all in the bristuff tarball? |
| 01:11.24 | JT | it tells you what you need to do for it to work |
| 01:11.31 | vlt | oops .. brb |
| 01:11.45 | JT | if you follow all that and it doesn't work, then you may need to patch qozap |
| 01:12.03 | The_DoC^ | I think I am finally going to order a TDM this week |
| 01:12.20 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:12.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:13.13 | JT | and is anything plugged into the card? |
| 01:13.41 | JT | the driver seems to be assuming all TE as far as i can tell, but better put a port definition in to be safe |
| 01:28.24 | vlt | JT: Ok, I read the install guide and told the driver the octal port definition. When using "ports=8", for ex., it loads [TE TE TE NT]. But the problem doesn't disappear. -- Patching. Can I use the bristuffed ubuntu-packaged asterisk and only compile qozap myself or do I have to start all over? |
| 01:32.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:35.32 | JT | vlt: possibly, if your source is the right version |
| 01:35.50 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 01:36.12 | joe | if you are playing an mp3 is there a way to make it fastforward or rewind like a voicemessage? background an playback don't show that functionality |
| 01:37.28 | *** join/#asterisk fholmes (n=fholmes@cpe-72-177-234-192.houston.res.rr.com) |
| 01:37.47 | JT | ControlPlayback does, but it's not for mp3s |
| 01:38.56 | Qwell | where is 626? |
| 01:39.34 | joe | JT: what does it support? |
| 01:40.48 | *** join/#asterisk yanek (i=yanek@atlantis.mitranet.cz) |
| 01:40.57 | JT | wav |
| 01:41.02 | JT | maybe other prompt formats? |
| 01:41.54 | joe | I'll look into it. thanks JT |
| 01:44.59 | *** join/#asterisk ManxPower (n=manxpowe@169.sub-70-216-230.myvzw.com) |
| 01:46.26 | mpruett | Hello Everyone |
| 01:46.56 | mpruett | I have an issue I am hoping someone can help me with |
| 01:47.08 | JT | interesting |
| 01:47.20 | mpruett | I am getting a 480 "Temporarily Unavailable" when I try to bridge a call |
| 01:48.01 | mpruett | I have explained the entire scnario and documented the output and sip debug at http://pastebin.ca/354465 |
| 01:48.21 | mpruett | Any help would be greatly GREATLY appreciated |
| 01:49.35 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 01:52.09 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 02:01.10 | JT | mpruett: do you run 204.16.49.121? |
| 02:01.19 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 02:03.11 | *** join/#asterisk xpot (n=jim@71-213-33-19.slkc.qwest.net) |
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| 02:14.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:17.33 | vlt | Does anyone know how to read my BRI card's vendor id that is checked in the original qozap.c:113: "if ((qoztmp->type == 0xb520)"? |
| 02:17.55 | JT | did you find the instructions for patching it? |
| 02:18.41 | vlt | I found a patch for beronet cards that jsut adds "|| (qoztmp->type == 0xb560))" but I don't know if this will match my card ... |
| 02:18.49 | vlt | just* |
| 02:20.24 | JT | vlt: lspci -vv tells you your vendorid |
| 02:20.49 | JT | Subsystem: |
| 02:23.03 | vlt | JT: Subsystem: Cologne Chip Designs GmbH: Unknown device 08b4 |
| 02:23.19 | vlt | I thought that was the chip identifier. |
| 02:25.39 | JT | 08b4 is the vendor id i believe |
| 02:25.40 | vlt | JT: hmm, I tried to open a "/msg JT" and paste it there. Did it work? |
| 02:25.53 | JT | the paste did not |
| 02:26.59 | vlt | Can you direct me to a patch that covers _all_ non-Junghanns devices (not a specific BeroNET one)? |
| 02:28.15 | JT | just replace the patch's beronet vendor id with yours |
| 02:40.18 | vlt | JT: Maybe it's a little late (here) but I'm afraid i can't find what exactly to replace. There are several different numbers added by the (beronet) patch: http://paste.ubuntu-nl.org/5868 -- look: 0xb552, 0xb562, 0xb520, 0xb560, 0xb566 ... |
| 02:41.49 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
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| 02:44.45 | JT | vlt: i guess you'd replace all instances of 0xb560 with 0x08b4 |
| 02:45.15 | JT | and replace the message for that case to say something else, not beronet |
| 02:45.21 | *** join/#asterisk Lurchtoke (i=professi@adsl-75-37-75-140.dsl.frs2ca.sbcglobal.net) |
| 02:45.22 | vlt | ;-) |
| 02:45.31 | JT | 0xb560 is the beronet 4 port card i believe |
| 02:45.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:45.43 | JT | that patch adds support for the beronet 4 and 8 port card |
| 02:45.48 | JT | and 2 port it seems |
| 02:45.50 | Lurchtoke | evening folks |
| 02:46.04 | Lurchtoke | has anyone here set up sips behind a linksys router? |
| 02:46.20 | JT | sips? |
| 02:46.23 | vlt | JT: Aaah, I forgot that this is one driver for several cards ... Thank you. |
| 02:46.29 | *** join/#asterisk ocnarfid800 (n=user@cpe-68-175-111-8.nyc.res.rr.com) |
| 02:46.30 | JT | yeah |
| 02:46.35 | JT | different multiport hfc cards |
| 02:46.41 | JT | single port hfc card is a different driver |
| 02:46.52 | vlt | zaphfc |
| 02:46.54 | vlt | ? |
| 02:46.55 | JT | yes |
| 02:47.03 | Lurchtoke | I got adsl....so I take it I set it up as bridge mode...and do the pppoe in the linksys router...then forward my 5060 and 5061 ports (along with the 16384 or whaterver ports) |
| 02:47.35 | JT | Lurchtoke: where are the sip clients, where is asterisk? |
| 02:47.45 | Lurchtoke | sips are spa-2000's |
| 02:47.51 | Lurchtoke | and the server is at another location |
| 02:47.57 | JT | ATAs you mean |
| 02:47.58 | JT | ok |
| 02:48.19 | Lurchtoke | well...its halfway done |
| 02:48.36 | Lurchtoke | we grabbed the sips from a store we closed.....so they know where the server is allready |
| 02:48.52 | Lurchtoke | I just tried to set them to dhcp in the sip menu...but they didnt work |
| 02:48.53 | *** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au) |
| 02:49.00 | Lurchtoke | I have the router set to dhcp |
| 02:49.03 | orlock | Does anybody know when SIP_HEADER replaced SPIGetHeader? |
| 02:49.12 | Lurchtoke | but I could not get the damn things to grab an ip |
| 02:49.21 | JT | ATAs |
| 02:49.30 | orlock | or.. how would i tell if a version of asterisk has that function available? |
| 02:49.46 | Lurchtoke | I set them as static ips inside my router network...and configured the appropriate settings gateway..etc |
| 02:49.52 | Lurchtoke | but still couldnt get them to work |
| 02:49.56 | Lurchtoke | couldnt even ping them |
| 02:50.05 | JT | they are not "sips" |
| 02:50.11 | Lurchtoke | ok |
| 02:50.18 | Lurchtoke | point taken :P |
| 02:50.55 | JT | umm, i don't even know if they need all that portforwarding |
| 02:51.04 | Lurchtoke | I dunno |
| 02:51.06 | Lurchtoke | (shrugs) |
| 02:51.09 | JT | i can usually make sip work fine without if the sip server is on a public ip |
| 02:51.16 | Lurchtoke | yeah |
| 02:51.18 | Lurchtoke | me too |
| 02:51.33 | JT | wait, you can't even ping these devices? |
| 02:51.35 | Lurchtoke | I setup the ones at another store no problem...we had cable modem set to bridge mode |
| 02:51.56 | Lurchtoke | maybe I should set the speedstream 5100 adsl modem to bridge mode...add a couple ips and try it them |
| 02:52.05 | Lurchtoke | err then |
| 02:52.25 | Lurchtoke | but dont you need to do the pppoe handshake to establish connection to the sbl dsl?? |
| 02:52.42 | Lurchtoke | dont you have to do that in thee router? |
| 02:52.55 | Lurchtoke | err sbc dsl |
| 02:53.08 | JT | i assume you would bneed to authenticate, and ppoe is encapsulation, not just a handshake |
| 02:53.15 | JT | it needs to be done before the ATAs |
| 02:53.23 | Lurchtoke | ya |
| 02:54.14 | Lurchtoke | so...setting the dsl modem to bride mode...would I just need a hub/switch on the backside.....going directly to the atas? and also use one of the block of ips for my network... |
| 02:54.48 | Lurchtoke | in essense....two of the ips going to ATA's....and the other for my puters (via a router) |
| 02:55.09 | Lurchtoke | or should I hook the spa-2000 ata's into the backside of the router |
| 02:55.17 | Lurchtoke | then router to dsl modem |
| 02:55.28 | Lurchtoke | out to the copper pair... |
| 02:55.49 | Lurchtoke | danm...been so long...cant rememeber... |
| 02:56.34 | JT | you must do ppoe before the ATAs, either in the modem, or a router behind it |
| 02:56.57 | Lurchtoke | ok... |
| 02:57.03 | Lurchtoke | ya |
| 02:58.01 | Lurchtoke | if I do it in the modem.....can i hook up everything to the backside (LAN) and router...and the ips will be available to the atas..and the router |
| 02:58.13 | Lurchtoke | I got my block of 5 ips |
| 02:58.36 | JT | you will need to run a static route on the router if you do it that way |
| 02:58.46 | JT | in theory they shouldn't need public IPs anyway |
| 02:59.14 | *** part/#asterisk thor_n (n=thor@rrcs-24-153-226-142.sw.biz.rr.com) |
| 02:59.15 | vlt | JT: You're right: qozap.c: switch (tmp->subsystem_device) { case 0x08b4: printk(KERN_INFO "qozap: CologneChip HFC-4S evaluation board configured ..." case 0xb520: printk(KERN_INFO "qozap: Junghanns.NET quadBRI card ..."; |
| 02:59.32 | Lurchtoke | cool |
| 02:59.43 | vlt | and 0x08b4 is _my_ vendor id, so I saw the first msq in dmesg |
| 02:59.56 | Lurchtoke | ok....lemme run off the configuration plan I got and tell me if I fogot anything |
| 02:59.59 | Lurchtoke | err forgot |
| 03:00.32 | JT | vlt: was that already in the source? |
| 03:00.49 | JT | vlt: i didn't see your vendorid in the patch, but i didn't look at the original source |
| 03:00.54 | vlt | Yes. Source from junghanns.net |
| 03:01.08 | JT | maybe it's set to ignore it, i dunno |
| 03:01.15 | Lurchtoke | I set the speedstream 5100 ADSL modem to bridge mode.....then on the lan side of that modem I connect a hub.....then I hook up two ATA's and a router |
| 03:01.35 | vlt | my vendor id was always 2 or three lines before the diff part ;-) |
| 03:01.56 | Lurchtoke | the 2 ATA's I configure with the ip/gateway/dns info given by SBC...to get my phones..... |
| 03:02.13 | Lurchtoke | then I use an additional IP for my data network |
| 03:02.21 | Lurchtoke | right? |
| 03:02.26 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 03:03.22 | JT | no |
| 03:03.28 | foobar778 | Hey guys sorry for being a pest asking same question but I solved it have two way audio no more critical packet dropped call can someone call me very excited |
| 03:03.29 | JT | you must deal with ppoe first |
| 03:03.42 | JT | either the modem or router must deal with it |
| 03:03.50 | Lurchtoke | <PROTECTED> |
| 03:03.57 | foobar778 | Jyt hi sorry for earlier |
| 03:03.58 | JT | foobar778: what fixed it? |
| 03:04.09 | foobar778 | Jt |
| 03:04.13 | JT | Lurchtoke: read the speedstream 5100 documentation, i am unfamiliar with it |
| 03:04.15 | foobar778 | <PROTECTED> |
| 03:04.23 | foobar778 | no two way audio |
| 03:04.27 | foobar778 | <PROTECTED> |
| 03:04.57 | foobar778 | an error crtical packet frop lose call in 10 secomnds both resolved nat issue |
| 03:05.01 | vlt | During compiling (from junghanns's .sh) I got "Link /usr/src/linux-2.6 to your kernel sources first!" Do I need sources for compiling? Not only headers? |
| 03:05.05 | foobar778 | drop not frop |
| 03:05.31 | JT | vlt: yes |
| 03:05.36 | foobar778 | Jt can u call my pbxc am very excited |
| 03:05.46 | JT | i guess |
| 03:05.53 | foobar778 | great |
| 03:05.57 | foobar778 | Im pm u |
| 03:07.51 | JT | qozap is a kernel module, it needs the kernel source to compile against |
| 03:08.14 | *** join/#asterisk dj-fu (n=deejay@smtp.sigmaco.co.nz) |
| 03:09.53 | vlt | JT: (off topic): Yes, but why? Why aren't headers enough? |
| 03:10.17 | JT | ask a kernel hacker :) |
| 03:13.00 | vlt | JT: maybe next year. |
| 03:21.55 | Bobthehunter | \trying to format a drive for windows but in unix |
| 03:21.57 | Bobthehunter | any idea ? |
| 03:22.21 | Bobthehunter | l |
| 03:22.46 | vlt | Bobthehunter: #unix ? |
| 03:22.56 | Bobthehunter | tried |
| 03:23.10 | vlt | Bobthehunter: mfks.vfat |
| 03:23.23 | Bobthehunter | hmm im still in fdisk |
| 03:23.30 | JT | err |
| 03:23.36 | vlt | Bobthehunter: fdisk is for partitoning |
| 03:23.41 | JT | fdisk is for partitioning NOT making a filesystem/formatting |
| 03:23.56 | Bobthehunter | yeah well i had to make a part; ) |
| 03:24.31 | Bobthehunter | <PROTECTED> |
| 03:24.31 | Bobthehunter | mkfs.vfat 2.8 (28 Feb 2001) |
| 03:24.31 | Bobthehunter | mkfs.vfat: Will not try to make filesystem on '/dev/sda' |
| 03:24.33 | Bobthehunter | bad bad |
| 03:24.53 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 03:25.21 | vlt | Bobthehunter: sda1 or similar |
| 03:25.55 | vlt | Bobthehunter: did you create a new partition? |
| 03:26.05 | Bobthehunter | not uyet |
| 03:27.29 | Bobthehunter | workign now |
| 03:27.34 | Bobthehunter | firs 4 tiems it didnt stick |
| 03:27.57 | Bobthehunter | so i used type 6 for fat16 then i used that on sda1 |
| 03:28.00 | Bobthehunter | its going |
| 03:28.16 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 03:28.35 | JT | well it sounds like you don't know what you're doing, Bobthehunter :P |
| 03:28.43 | JT | you must use fdisk to make partitions |
| 03:28.48 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 03:28.57 | JT | then use mkfs to format a specific partition |
| 03:29.09 | JT | you do not mkfs the MBA (/dev/sda) |
| 03:29.22 | vlt | bad bad |
| 03:29.27 | JT | MBR even |
| 03:29.48 | JT | vlt: damn |
| 03:30.02 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 03:31.23 | *** join/#asterisk BOLIVIAN (n=klkl@201.222.93.229) |
| 03:32.07 | *** part/#asterisk BOLIVIAN (n=klkl@201.222.93.229) |
| 03:34.01 | Bobthehunter | lol |
| 03:34.02 | Bobthehunter | oups |
| 03:34.24 | Bobthehunter | can unix mount that vfat ? |
| 03:35.54 | Bobthehunter | also says wond to a full disk.. hmm |
| 03:36.53 | Qwell | dseeb_: should hcid be running with chan_cellphone? |
| 03:38.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:38.40 | JT | vfat is fat32 |
| 03:39.57 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:45.38 | Bobthehunter | yeah |
| 03:46.44 | Bobthehunter | thing is my main has 2 sata slots.. and one is the mba and os on #2.. lol bad thing i know.. so i got a sata to usb sapteor.. but since that drive waas ext3 windwos cant see/format/or partition.. so i pushed it on my linux box.. repartiion fat and mkfs right now.. then ill see if windows sees it.. |
| 03:46.53 | Bobthehunter | <PROTECTED> |
| 03:46.58 | Bobthehunter | i wahs i just used |
| 03:47.01 | Bobthehunter | and still going |
| 03:47.29 | Bobthehunter | omg.. lots of typos.. |
| 03:47.41 | Bobthehunter | sapteor is a neat word but i meant adaptor |
| 03:48.07 | Bobthehunter | so mount /dev/sda1 /usbdrive -t vfat |
| 03:48.11 | Bobthehunter | shoudl work after ? |
| 03:49.27 | *** join/#asterisk rafee (n=rafee@201.19.252.12) |
| 03:50.15 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 03:50.29 | rafee | I'm having trouble with one configuration. This channel is the best place to ask for help? |
| 03:51.49 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 03:52.05 | *** join/#asterisk Growly (n=growly@ftp.hibs.school.nz) |
| 03:52.36 | Lurchtoke | shoot |
| 03:52.36 | [TK]D-Fender | rafee : Depends. Just ask. |
| 03:52.44 | Lurchtoke | another stupid question |
| 03:53.18 | Lurchtoke | If I set an adsl modem to bridge mode....does that mean it doesnt do the pppoe stuff....and the ata's must do it? |
| 03:53.24 | Lurchtoke | (if supported) |
| 03:54.09 | rafee | I have this line o extensions.conf exten => 888,21,System(/usr/bin/lame --silent -h /tmp/${message_id}.wav /tmp/${message_id}.mp3 |
| 03:54.16 | rafee | But lame is not being executed |
| 03:54.59 | [TK]D-Fender | rafee : And you see it being called from * CLI? |
| 03:55.30 | Lurchtoke | fender....can you scroll up and read the stuff I posted? I almost have a grasp of what I need to do...but need answer to this |
| 03:55.51 | [TK]D-Fender | Lurchtoke : I have no clue on how to set up an ISP provided subnet, sorry |
| 03:56.15 | rafee | [TK]D-Fender: Nothing appears on CLI |
| 03:56.22 | Lurchtoke | hmmm |
| 03:56.53 | [TK]D-Fender | rafee : Make sure you're on high verbose (10 is my norm). If you don't see it, then you aren't even attempting to execute that line |
| 03:56.59 | Lurchtoke | in theory...if the modem is set to bridge...does it do the pppoe and then rout ip and traffic?? |
| 03:57.16 | rafee | [TK]D-Fender: Thanks I will change verbose |
| 03:57.32 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:57.39 | *** join/#asterisk ledoktre (n=x@209.152.67.6) |
| 03:57.41 | [TK]D-Fender | Lurchtoke : No clue |
| 03:57.42 | ledoktre | hello |
| 03:57.48 | k-man | hello |
| 03:58.41 | Qwell | okay, so, this chan_cellphone rocks |
| 03:58.42 | rafee | [TK]D-Fender: Do you run asterisk with -vvvvvvvvvv (10 Vs)? |
| 03:59.02 | JT | Lurchtoke: whatever does the PPPoE, after that you need the IPs to be routed |
| 03:59.22 | JT | and probably setup as a virtual interface on the device if possible |
| 03:59.36 | JT | unless you don't need it on your lan subnet |
| 03:59.45 | JT | in which case it can be on the primary interface |
| 04:01.24 | ledoktre | anyone here know, if programming some softkeys, how do you program in "flash"? |
| 04:01.32 | rafee | [TK]D-Fender: I could see two errors |
| 04:01.34 | rafee | /bin/sh: -c: line 0: unexpected EOF while looking for matching `"' |
| 04:01.34 | rafee | /bin/sh: -c: line 1: syntax error: unexpected end of file |
| 04:03.31 | *** join/#asterisk dorphalsig (i=dorphals@pcsp163-73.supercabletv.net.co) |
| 04:03.45 | dorphalsig | Hey, anybody has used Plantronics CT-11 with asterisk? |
| 04:04.23 | JT | ledoktre: from memory, you can't |
| 04:04.34 | Qwell | dorphalsig: what is it? |
| 04:05.07 | dorphalsig | Qwell --> Wireless headset phones |
| 04:05.15 | Qwell | is it an analog phone? |
| 04:05.18 | dorphalsig | yes |
| 04:05.21 | Qwell | then sure |
| 04:05.34 | ledoktre | JT: there isnt some key combination so you could hit a button while on the phone and <flash>,100# ? |
| 04:05.52 | JT | ledoktre: on a sip phone? |
| 04:06.02 | ledoktre | aastra 480i |
| 04:06.07 | dorphalsig | because I read somewhere that CT12 had problems with digital PBX systems |
| 04:06.10 | JT | that is a sip phone |
| 04:06.17 | Qwell | somebody should send me a bluetooth headset, so I can play with chan_cellphone some more :P |
| 04:06.23 | Qwell | dorphalsig: asterisk isn't a digital PBX |
| 04:06.25 | JT | ledoktre: i don't think you can, it's not a very pbxish thing to do |
| 04:06.29 | flenders | hey, can I have distinctive rings on SPA-941s? |
| 04:06.29 | Qwell | at least, not in the same sense |
| 04:06.55 | Qwell | dorphalsig: when somebody says "digital PBX" they mean "proprietary piece of garbage, with proprietary phones" |
| 04:06.55 | dorphalsig | Qwell ... could you elaborate on that? (feeling blatantly ignorant) |
| 04:06.57 | ledoktre | JT: ok, there isnt really anything you can program in to make a 1 button transfer then? |
| 04:07.10 | *** join/#asterisk xpotx (n=jim@71-213-33-19.slkc.qwest.net) |
| 04:07.10 | dorphalsig | Ooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooh |
| 04:07.14 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 04:07.37 | JT | ledoktre: err you can do sip transfers, it's not a hook flash though |
| 04:08.23 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 04:09.13 | ledoktre | JT: Back when I setup * at my office a couple years ago, and using older ata/analog phones, flash/extension was the way I was told to do it. Apparently there is a cooler way to do it .... how? |
| 04:09.41 | dorphalsig | Qwell... thnx! |
| 04:10.03 | JT | ledoktre: yeah, via sip signalling, since you can't do a real hook flash on a sip phone, not to mention it would be more cumberson even if you could signal it |
| 04:10.12 | JT | search voip info for sip transfers? |
| 04:10.15 | dorphalsig | I have Another question... I think its pretty much self answered, but I just want to double check |
| 04:10.25 | dorphalsig | I have 10 analog phone lines |
| 04:10.32 | dorphalsig | can I dump them into an FXS channelbank |
| 04:10.53 | Qwell | dorphalsig: yeah, if you have a T1 card, or something to get the channelbank connected to asterisk |
| 04:11.00 | Qwell | (usually it's via T1) |
| 04:11.17 | ledoktre | JT: will search. Im already using the flash button & transfer on the 480i's, but I am setting up the softbuttons and trying to find a one button solution for my 4 users. Something simple. |
| 04:11.29 | dorphalsig | Thnx :) |
| 04:11.30 | dorphalsig | Byw |
| 04:11.33 | dorphalsig | Bye |
| 04:11.53 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 04:12.05 | JT | ledoktre: there's a button called flash on your sip phone? |
| 04:15.51 | ledoktre | JT: a soft button yes. |
| 04:16.20 | ledoktre | was just reading about transfers, and they mention using #, or *2. trying them both but no luck so far :-( |
| 04:16.52 | [TK]D-Fender | ledoktre : You need to be dialing your phone with certain options for DTMF transfers to work.. |
| 04:17.03 | Bobthehunter | http://members.calbar.ca.gov/search/member_detail.aspx?x=169888 |
| 04:17.06 | Bobthehunter | lol |
| 04:17.24 | *** join/#asterisk AJaymn (n=Me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 04:18.49 | ledoktre | TKD-Fender: I dont suppose you would mind elaborating a bit? ;-) |
| 04:19.25 | vlt | JT: Absolutely no clue what to check now: What am I doing wrong here when compiling qozap.ko? http://paste.ubuntu-nl.org/5870/ |
| 04:19.34 | JT | [TK]D-Fender: he has sip phones, why does he need dtmf transfers at all? |
| 04:19.56 | flenders | Hi, I'm trying to have distinctive ring on SPA-921 and SPA-941... I tried "exten => 35,1,Set(__ALERT_INFO=n=Simple-4;w=4;c=1)" but that doesn't work. any ideas? |
| 04:20.25 | [TK]D-Fender | JT : He doesn't. He's jsut trying to kiss ass to a few users and give them "1-button transfer direct to extensions" |
| 04:20.54 | [TK]D-Fender | flenders : "show application sipaddheader" |
| 04:21.25 | [TK]D-Fender | ledoktre : "show application dial" |
| 04:22.02 | *** join/#asterisk infosys24 (n=infosys2@63.243.179.210) |
| 04:22.08 | JT | vlt: something seems majorly wrong |
| 04:22.20 | JT | vlt: your symlinking to kernel stuff is probably incorrect |
| 04:22.25 | JT | or wrong kernel stuff |
| 04:22.32 | infosys24 | gm chan |
| 04:22.52 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 04:23.01 | *** join/#asterisk xpot (n=jim@71-213-33-19.slkc.qwest.net) |
| 04:23.07 | infosys24 | i am in afghanistan and i dont have time to research something so if someone could maybe give me a quick answer |
| 04:23.22 | Bobthehunter | ? |
| 04:23.38 | ledoktre | TKD-Fender: Not really trying to kiss anyone anywhere, just trying to set it up to be a KISS system, for the 4 users that will be using it, and avoid calls to me, how do I do this or that |
| 04:23.39 | Bobthehunter | you a soldier ? |
| 04:23.45 | Bobthehunter | lol |
| 04:23.46 | infosys24 | i am looking for a virtual pri maybe with flat rate or lowrate |
| 04:23.52 | infosys24 | yes i am a soldier |
| 04:23.54 | infosys24 | army |
| 04:23.59 | *** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 04:23.59 | vlt | JT: I created a symlink /usr/src/linux-2.6 -> /usr/src/linux-source-2.6.15/ where the source is. Btw, asterisk itself compiled successfully. |
| 04:24.04 | murdmath | Howdy all. |
| 04:24.11 | Bobthehunter | ok to where |
| 04:24.18 | Bobthehunter | vpri you mean dedicated channels ? |
| 04:24.25 | JT | asterisk doesn't have kernel modules itself |
| 04:24.31 | Bobthehunter | and where to. ;) not afgan i hope |
| 04:24.38 | infosys24 | lol no |
| 04:24.45 | infosys24 | i need some channels |
| 04:24.58 | infosys24 | i have a pri at home on oc-198 |
| 04:25.05 | murdmath | Is it normal for a call that has been parked and then picked up to not have it's caller id information? |
| 04:25.08 | JT | vlt: /usr/src/linux should symlink to the source too |
| 04:25.09 | Bobthehunter | home base ;) |
| 04:25.11 | JT | oc-192 don't you mean? |
| 04:25.15 | infosys24 | and i am using it for incomming |
| 04:25.31 | vlt | jt |
| 04:25.33 | infosys24 | but i need some virtual channels prob about 46 for outgoing |
| 04:25.35 | Bobthehunter | so waht you nead.. say it ;) |
| 04:25.38 | vlt | thx, trying |
| 04:25.41 | JT | vlt: looks like problems with the kernel headers though |
| 04:25.51 | Bobthehunter | for a guys whos in a hurry.. long to get the words out |
| 04:25.56 | vlt | do i need a symlink to the headers too? |
| 04:26.18 | Bobthehunter | EX: i need 2 channels and a did in the NXX area code ;) |
| 04:26.19 | Bobthehunter | that would be a start |
| 04:26.36 | JT | vlt: look at all those errors, they seem to be header errors |
| 04:26.44 | JT | "include" is usually headers |
| 04:26.49 | infosys24 | bob pm me? |
| 04:30.15 | Bobthehunter | mount: wrong fs type, bad option, bad superblock on /dev/sda1, |
| 04:30.22 | Bobthehunter | ok that didnt work |
| 04:30.45 | flenders | [TK]D-Fender: thanks mate! |
| 04:31.07 | flenders | [TK]D-Fender: can we also control the volume it rings on all phones? |
| 04:32.21 | Bobthehunter | dev/sda1 1 29356 235801910+ 6 FAT16 |
| 04:32.24 | Bobthehunter | hmm |
| 04:35.37 | vlt | Both linux-source and linux-headers are installed in /usr/src/ but I get 3663 lines of compiler errors. Any idea? |
| 04:36.01 | vlt | (for qozap.ko) |
| 04:37.35 | k-man | [TK]D-Fender, any idea where i might find doco on doing remote provisioning with the SPA942? ie, what file does the phone look for on the tftp server? |
| 04:40.58 | [TK]D-Fender | k-man : http://spc.pifiu.com |
| 04:41.28 | [TK]D-Fender | flenders : Not a clue. |
| 04:41.37 | JT | vlt: what about /usr/include? |
| 04:41.58 | vlt | JT: That exists. |
| 04:42.10 | [TK]D-Fender | flenders : I doubt it. I mean if you want your ringer low, do you really want some ass "paging" you really loud int he middle of a meeting with a potentially exta-obnoxious ring-tone? |
| 04:42.18 | k-man | [TK]D-Fender, ah, thanks |
| 04:42.20 | JT | does it have kernel headers in it? |
| 04:42.29 | k-man | thats nice of linksys to make the info hard to come by |
| 04:42.31 | vlt | JT: oh, no, it doesn't ... |
| 04:43.00 | vlt | JT: nm, it actually does. |
| 04:43.18 | JT | the correct ones? |
| 04:43.29 | JT | does it match /usr/src/linux's version? |
| 04:43.34 | vlt | JT: How do I know? |
| 04:43.49 | *** part/#asterisk infosys24 (n=infosys2@63.243.179.210) |
| 04:43.50 | JT | umm, i dunno, work it out :) |
| 04:43.52 | JT | sorry |
| 04:44.07 | JT | i hate using default kernels |
| 04:46.15 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 04:48.23 | *** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca) |
| 04:51.21 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 04:51.40 | flenders | [TK]D-Fender |
| 04:51.50 | flenders | [TK]D-Fender: yeah, makes sense |
| 04:53.09 | b1shop | looking for some beginner help. i read the wiki and other misc articles. i have an office with 4-5 lines and a 7+ yr old nortel pbx. looking to switch to a voip solution and get a number for a branch office in another area code. |
| 04:53.14 | k-man | [TK]D-Fender, what is the spc utility for? any idea? |
| 04:53.22 | *** join/#asterisk KaiHanari (n=kaihanar@stjhnf0122w-142162043055.pppoe-dynamic.nl.aliant.net) |
| 04:53.39 | b1shop | someone point to to hardware guides... do i need new phones? etc... |
| 04:53.53 | JT | most likely |
| 04:53.59 | JT | look at getting sip phones, b1shop |
| 04:54.51 | k-man | b1shop, do you do more calling out or receiving calls? |
| 04:55.04 | b1shop | hmm. ok. any idea how much they run? trying to figure out how much to budget for the whole upgrade |
| 04:55.27 | JT | check out voipsupply.com but a few hundred each for good ones |
| 04:55.40 | [TK]D-Fender | b1shop : How many phones do you have? |
| 04:55.58 | J4k3 | can anyone make a call out on vitelity right now? |
| 04:56.00 | k-man | b1shop, technicaly, you could connect voip adaptors to your existing phone system... but if you can afford to spend a bit more, its probably worth ditching your existing phone system and go totaly with asterisk |
| 04:56.13 | b1shop | k-man, about equal actually. we have 7-8 phones that came with the old system |
| 04:56.31 | k-man | b1shop, are you in australia? |
| 04:56.44 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:56.49 | [TK]D-Fender | b1shop : that in mind, time to ditch the entire setup. |
| 04:57.11 | b1shop | sucks. cause i bet we paid almost 10K for that old system... problem is that i have NO IDEA how to program it. so any time we need to change greeting, add/change extensions. i have to call someone in |
| 04:57.22 | b1shop | k-man, us based |
| 04:57.28 | k-man | b1shop, i'm in the same boat as you |
| 04:57.46 | k-man | b1shop, i'm learning asterisk so i can ditch our phone system and support it myself |
| 04:58.07 | k-man | b1shop, i hate paying a contractor to come and make a different extension ring or change a minor setting |
| 04:58.09 | b1shop | i called out ISP which offers voip service. but it not cheap (setup + hardware + serivce costs, etc. |
| 04:58.53 | k-man | b1shop, in fact, I even managed to find the programming guide for our phone system on the internet but its no use.. its so bloody complex to program you really need to have attended the manufaturers course to understand it |
| 04:58.54 | [TK]D-Fender | b1shop : Ok, I'm betting rewiring the place isn't worth it, so : 7 x http://www.telephonydepot.com/product_p/105-058-501.htm |
| 04:59.14 | [TK]D-Fender | b1shop : 1 x http://www.telephonydepot.com/product_p/105-058-601.htm (receptionist) |
| 04:59.50 | b1shop | [TK]D-Fender, well. we're actually moving into larger office space in 2-3mo. so rewire is not that big a deal |
| 05:00.07 | [TK]D-Fender | b1shop : And for your lines : http://www.telephonydepot.com/product_p/105-052-20002d-a.htm |
| 05:01.22 | JT | k-man: hey, what area are you in again? |
| 05:01.23 | [TK]D-Fender | b1shop : then if you wish you would be advised to wire 2 sets of Cat5E to each station,a nd use the other for your phones. You'd want a PoE switch, at which point my common user phone suggestion would be : http://www.telephonydepot.com/product_p/105-058-430.htm |
| 05:01.24 | b1shop | which providers are good? and about how much a month do you pay? |
| 05:01.30 | k-man | jt, sydney |
| 05:02.00 | [TK]D-Fender | b1shop : No providers, no monthly fees. this is for normal analog lines |
| 05:02.00 | vlt | Maybe I really should try to compile a kernel myself ... tomorrow. Thanks for your help (JT). |
| 05:02.19 | JT | k-man: ah right |
| 05:02.40 | JT | k-man: was i talking with you about parking, or flenders? |
| 05:02.42 | b1shop | hmm. i'm looking to have a couple numbers in a fllorida office roll over to chicago if possib;e |
| 05:02.43 | JT | vlt: np, good luck |
| 05:03.02 | JT | b1shop: sounds possible |
| 05:04.00 | k-man | jt, must have been flenders |
| 05:04.11 | J4k3 | wow, bummer |
| 05:04.13 | k-man | jt, what about parking though? |
| 05:04.14 | J4k3 | my DID is dead again too |
| 05:04.17 | b1shop | what kind of hardware do you run? typical x68 ststem? |
| 05:04.36 | [TK]D-Fender | b1shop : Basic P4 class would do jsut fine |
| 05:05.09 | k-man | jt, are you in sydney? |
| 05:05.32 | J4k3 | bleh... f this local DID. I'm getting an 800#. |
| 05:05.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:05.56 | b1shop | <-- not a phone guru. any professional servfices available to help me scope this oout? or are they not worth it? |
| 05:06.19 | JT | k-man: yes, flenders must be too then |
| 05:06.41 | k-man | jt, are you going to webjam? |
| 05:07.01 | b1shop | i have bandwith questions to take into concideration as well |
| 05:07.06 | b1shop | im sure |
| 05:07.10 | *** part/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 05:07.48 | [TK]D-Fender | b1shop : You looking to do away with analog lines and go VoIP for PSTN termination? |
| 05:08.06 | b1shop | PSTN? |
| 05:09.05 | ShadowHntr | Public Switched Telephone Network |
| 05:09.14 | b1shop | i have not really weight the pros and cons of analog vs voip |
| 05:09.17 | ShadowHntr | normal analog telephone lines |
| 05:09.56 | [TK]D-Fender | b1shop : I would normally advise against VoIP as your path to the normal phone network for business purposes |
| 05:10.09 | b1shop | i need to put together a couple different senarios to weigh costs. |
| 05:10.21 | JT | ShadowHntr: no, you're talking about POTS |
| 05:10.29 | b1shop | if voip means more $ monthly for extra bandwidth.. then things could change |
| 05:10.34 | JT | POTS allows you to connect to the PSTN, just like a SIP ITSP can |
| 05:10.55 | ShadowHntr | JT: my bad - got my correlations off slightly. |
| 05:11.04 | JT | ISDN also allows you to connect to the PSTN |
| 05:11.23 | ledoktre | anyone here know of an ADSI shared phonebook? |
| 05:11.32 | JT | the majority of first world telephone networks are all digital/ISDNish in the backend |
| 05:11.52 | *** join/#asterisk bmg505 (n=leon@c1-33-16.rndf.isadsl.co.za) |
| 05:12.20 | JT | k-man: what is webjam? |
| 05:12.47 | murdmath | Is it normal for a call that has been parked and then picked up to not have it's caller id information? |
| 05:12.57 | k-man | jt, webjam.com.au |
| 05:14.36 | JT | k-man: wow, that looks so devoid of info on what actually is |
| 05:14.40 | JT | is it just a web thing |
| 05:14.47 | k-man | jt, yeah |
| 05:14.57 | JT | doesn't sound very interesting ;) |
| 05:15.06 | k-man | just a bunch of people showing off their interesting web sites in under 3 minutes |
| 05:15.12 | k-man | it was quite interesting last time i went |
| 05:15.25 | JT | what made their sites interesting? ;) |
| 05:16.38 | k-man | mostly they were sites that used ajax or some other new tech thing |
| 05:16.45 | JT | ah ok |
| 05:17.27 | k-man | some talks were dead boring |
| 05:17.31 | k-man | and some were very interesting |
| 05:17.44 | JT | i'm more interested in a voip/telephony group |
| 05:17.47 | k-man | and as they were only 3 minutes long it didn't matter too much if it was boring |
| 05:17.54 | k-man | jt, fair enough |
| 05:18.02 | k-man | jt, let me know when you find one ;) |
| 05:18.14 | JT | there was talk of one being started in sydney |
| 05:18.19 | JT | haven't heard anything more now |
| 05:19.18 | k-man | thats a shame |
| 05:22.08 | *** join/#asterisk jaxxan (n=jhelmich@202.70.119.84) |
| 05:22.15 | jaxxan | hey guys |
| 05:22.31 | jaxxan | qwell are ya here ? |
| 05:22.36 | Qwell | no |
| 05:22.41 | jaxxan | damn |
| 05:23.00 | jaxxan | dude, does 1.2.15 have the snmp capabilities that 1.4 has ? |
| 05:23.05 | Qwell | no |
| 05:23.20 | jaxxan | will 1.2 be ditched for 1.4 soon ? |
| 05:23.51 | jaxxan | bah, i'm at the airport, gotta run. bbl |
| 05:28.20 | k-man | does asterisk get many patches from the community? |
| 05:28.47 | Qwell | yes |
| 05:29.00 | JT | k-man: bugs.digium.com |
| 05:34.01 | k-man | jt, hmm.. interesting |
| 05:41.29 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:44.54 | *** join/#asterisk tg (i=tg@2001:618:1a23:0:0:0:0:1) |
| 05:46.59 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:46.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:53.35 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:53.38 | zeeesh | hi |
| 05:54.32 | *** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 05:55.14 | k-man | hi |
| 05:57.28 | JT | k-man: what area of sydney if you mind me asking? |
| 05:57.43 | k-man | jt, alexandria |
| 05:57.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:00.24 | k-man | jt, you? |
| 06:06.24 | *** join/#asterisk AJaymn (n=mypocket@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 06:16.02 | murdmath | Is it normal that after a call is parked and then picked up it is missing the caller id information? |
| 06:19.43 | *** join/#asterisk BOLIVIAN (n=ewrtwert@201.222.93.229) |
| 06:25.57 | BOLIVIAN | i downloaded asterisknow is an iso file, how will burn it to a cd |
| 06:26.22 | BOLIVIAN | wich burn option, burn and image? |
| 06:27.10 | JT | you open it up in the burning software |
| 06:27.19 | JT | and it should be read to make a cd of it |
| 06:27.34 | JT | do not burn it as a file on the cd |
| 06:28.02 | BOLIVIAN | i have an option in nero wich says burn image to a cd |
| 06:28.04 | BOLIVIAN | is that the one? |
| 06:28.11 | BOLIVIAN | i pick that and then add the file? |
| 06:29.50 | JT | sounds like the right option |
| 06:29.55 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 06:30.05 | JT | mind you this isn't really an asterisk question |
| 06:30.12 | BOLIVIAN | but i have in my C installed win xp |
| 06:30.18 | BOLIVIAN | and 2 other partitions |
| 06:30.29 | BOLIVIAN | will asterisknow will erase my winxp? |
| 06:30.36 | JT | probably |
| 06:31.01 | JT | asterisk runs on linux, just get an old machine to use for testing |
| 06:34.29 | k-man__ | BOLIVIAN, you could install it in vmware... but i have had sound issues with doing that |
| 06:34.53 | k-man__ | BOLIVIAN, i beleive there are some vmware iso's with it all set up |
| 06:34.59 | k-man__ | images i should say |
| 06:35.27 | aptura | man im tired. JT how are you doing tonight? |
| 06:39.19 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 06:39.21 | *** join/#asterisk k-man___ (n=jason@unaffiliated/k-man) |
| 06:39.22 | JT | does no-one keep old pcs anymore? ;) |
| 06:39.28 | anthony] | no. |
| 06:39.33 | k-man___ | jt, i do |
| 06:39.52 | BOLIVIAN | i have but its an ibm 486 |
| 06:40.06 | anthony] | okay, I lied ;) |
| 06:40.12 | BOLIVIAN | with win 3.1 |
| 06:40.19 | J4k3 | I have my original IBM PC |
| 06:40.24 | anthony] | I wanted to feel cool. |
| 06:40.24 | J4k3 | two 360k floppies |
| 06:40.28 | J4k3 | techmar captains card |
| 06:40.29 | J4k3 | etc. |
| 06:40.43 | BOLIVIAN | oh wait i have one celeron integrated with 90 MB of RAM |
| 06:40.50 | BOLIVIAN | do u think will it work on that |
| 06:40.56 | anthony] | The oldest machine I have in my office right now, is |
| 06:41.03 | anthony] | a Pentium 4 2.8 Ghz |
| 06:41.13 | J4k3 | Kraft dual-port analog game port adapter card. which replaced my original single port (with only one sticks worth of ADCs) |
| 06:41.13 | anthony] | and that is a server.. heh |
| 06:41.14 | J4k3 | ;) |
| 06:42.05 | *** join/#asterisk apardo (n=apardo@87.217.144.179) |
| 06:42.29 | BOLIVIAN | j4k3 u mean u r chating from it now? |
| 06:42.52 | J4k3 | no, but I could. |
| 06:43.00 | J4k3 | I'd have to IRC via telnet, but I could. |
| 06:43.01 | BOLIVIAN | how much RAM has |
| 06:43.06 | J4k3 | 640K |
| 06:43.18 | BOLIVIAN | mine has 4MB |
| 06:43.28 | BOLIVIAN | do u think i can browse with that |
| 06:43.46 | BOLIVIAN | or at least use msn |
| 06:43.47 | J4k3 | with mosaic and windows 3.1, sure. |
| 06:43.59 | BOLIVIAN | is there msn for windows 3.1? |
| 06:44.06 | J4k3 | your best bet is using it as a vnc client |
| 06:44.35 | osiris | anyone here use trixbox ? |
| 06:44.36 | BOLIVIAN | vnc? u mean the remote program |
| 06:44.57 | J4k3 | yes. |
| 06:49.00 | Qwell | osiris: see topic |
| 06:50.07 | osiris | Qwell, see question. |
| 06:50.10 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 06:50.19 | osiris | im not asking for support |
| 06:51.30 | *** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 06:52.42 | Qwell | I haven't seen a question |
| 06:52.49 | Qwell | only a survey |
| 06:53.17 | osiris | a survey is a question |
| 06:53.35 | osiris | there cant be one without the other |
| 06:53.55 | JT | a celeron could probably run asterisk |
| 06:53.59 | JT | depending on what it does |
| 06:54.02 | osiris | i take that back, you can have a question without a survey |
| 06:54.05 | Qwell | then no, nobody uses trixbox here |
| 06:55.50 | J4k3 | nobody is willing to admit it |
| 07:01.58 | BOLIVIAN | i installed vmware but it is asking me a congfiguration file .vmx |
| 07:02.13 | BOLIVIAN | where do u get that |
| 07:04.24 | *** join/#asterisk wannabeaastguru (n=wannabea@brenibba.ifi.uio.no) |
| 07:21.12 | fetcher | JT: I just set up an old Celeron 533 * box ... works ok |
| 07:21.28 | JT | cool |
| 07:21.42 | fetcher | transcoding 4 iLBC channels to ulaw uses about 33% of the CPU |
| 07:22.15 | fetcher | (IAX to SIP gateway) |
| 07:22.38 | JT | iLBC lol |
| 07:22.45 | J4k3 | I ran a 4-concurrent-calls-at-most * box on an athlon 1.2 @ 900 and PC133 ram, and can tell no difference between it and an athlonxp 2400+ on PC2700. |
| 07:22.49 | JT | what do you have, a satellite link? |
| 07:22.55 | fetcher | a little embedded / appliance box to do that would be nice, though |
| 07:22.57 | JT | J4k3: 4 calls :P |
| 07:23.16 | J4k3 | jt: exactly. but I doubt a ppc 405GP@266 could handle it? :) |
| 07:23.21 | J4k3 | I should try. |
| 07:23.32 | JT | heh, depends on transcoding more than anything |
| 07:23.33 | fetcher | JT: 802.11 1-mile radio link on one end |
| 07:23.46 | JT | only 802.11 not even 802.11b? |
| 07:23.50 | J4k3 | fetcher: in the USA? |
| 07:23.51 | fetcher | JT: you don't care for ilbc? |
| 07:23.58 | JT | i'm in australia, i don't need miles |
| 07:24.11 | fetcher | J4k3: yup. 802.11b, actully |
| 07:24.16 | JT | oh it's good for low bandwidth and otherwise poor connections, it's just not the highest quality |
| 07:24.25 | J4k3 | fetcher: is the link not stable or something? |
| 07:24.44 | JT | fetcher: weren't you in australia? |
| 07:24.59 | fetcher | JT: what would you use, if G.729's not an option? |
| 07:25.13 | *** join/#asterisk marc7 (n=marc@S0106000f66461bdb.gv.shawcable.net) |
| 07:25.19 | JT | i dunno, compare it to gsm, see which wins |
| 07:25.26 | JT | don't you have the bandwidth to run g.711? |
| 07:25.43 | fetcher | J4k3: occasional interference troubles, after a rain (some trees in the way) |
| 07:26.05 | J4k3 | fetcher: if its NLOS, try the Ubiquiti SR9 radio |
| 07:26.14 | J4k3 | it cares not about trees |
| 07:26.52 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:27.37 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:28.35 | JT | fetcher: are you in australia? |
| 07:28.54 | fetcher | JT: no, US. |
| 07:29.21 | fetcher | have never been to .au as yet |
| 07:29.30 | JT | i swear you were in australia, maybe it was fletcher? |
| 07:29.47 | fetcher | yeah, must me someone else |
| 07:30.23 | fetcher | JT: iLBC sounds better than gsm to me, even on good-quality links. G.726 ADPCM, about the same.. |
| 07:30.44 | JT | ilbc just kills bandwidth i guess |
| 07:30.46 | JT | err |
| 07:30.51 | JT | ilbc just kills CPU i guess |
| 07:31.03 | JT | speex is another one to try |
| 07:31.10 | JT | g.729 even sounds better than gsm |
| 07:31.16 | fetcher | JT: yeah, it's implemented in floating point, which would rule out a lot of embedded boards |
| 07:31.38 | fetcher | JT: speex is extremely CPU intensive, isn't it? Moreso than ilbc. |
| 07:31.48 | JT | i thought they were similar |
| 07:32.08 | *** join/#asterisk shinux__ (n=shinux@196.220.25.12) |
| 07:32.22 | fetcher | hmm, may be worth a try then. |
| 07:32.39 | fetcher | can any other codecs do 30ms packet size? |
| 07:33.06 | JT | dunno |
| 07:33.37 | k-man___ | jt, when you do a new asterisk install, do you do the make sample? or just start with a blank /etc/asterisk? |
| 07:33.51 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 07:34.01 | fetcher | That helps a lot on wireless links (especially 802.11g and a), where per-packet overhead is extremely high |
| 07:34.03 | tzafrir | k-man___, 'make samples' is generally safer. |
| 07:34.07 | endre | blank && scp oldbox:/etc/asterisk* /etc/asterisk |
| 07:34.22 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:34.22 | fetcher | (dropping from 50pps/channel to 33.3_) |
| 07:34.32 | k-man___ | tzafrir, but i found it filled up /etc/asterisk with hundreds of files |
| 07:34.43 | tzafrir | Alternatively, try starting with an empty dir (just astertisk.conf and maybe modules.conf), it would be an interesting learning experince |
| 07:34.48 | JT | k-man___: mm, depends i guess |
| 07:34.55 | JT | makes samples is good for reference |
| 07:35.01 | k-man___ | ok |
| 07:35.06 | JT | but the actual files i use are usually from scratch |
| 07:35.25 | k-man___ | but you and [tk] told me to turf half of the configs |
| 07:35.49 | k-man___ | ok |
| 07:35.59 | k-man___ | from scratch sounds good to me |
| 07:36.01 | fetcher | Are the Asterisk prompt recordings (/var/lib/asterisk/sounds/*) available in a format other than .gsm ? Maybe uncompressed wav? |
| 07:36.04 | JT | i don't actually use the samples |
| 07:36.11 | JT | they are all blah.conf.sample in my setup |
| 07:36.31 | JT | fetcher: yes, you can download them in almost all supported formats |
| 07:36.39 | JT | ulaw alaw slin g729 etc |
| 07:36.46 | k-man___ | jt, are there any australian prompts? |
| 07:36.50 | fetcher | JT: from digium's site? |
| 07:36.53 | JT | slin is the internal asterisk format |
| 07:36.56 | JT | fetcher: yes |
| 07:37.08 | fetcher | ah, cool |
| 07:37.13 | JT | k-man___: yes, but the free ones aren't too my liking |
| 07:37.16 | JT | s/too/to/ |
| 07:37.39 | JT | there's one set of free ones with a fairly strong male accent |
| 07:37.43 | JT | sounds too bogan too me |
| 07:38.05 | JT | he's a radio announcer, so sound more like radio announcements than phone prompts, too |
| 07:39.12 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:39.33 | k-man___ | jt, oh |
| 07:39.37 | k-man___ | interesting |
| 07:39.53 | k-man___ | jt, which actor would you get to do it if you could? mel gibson? ;) |
| 07:39.58 | k-man___ | georgie parker maybe? |
| 07:39.59 | k-man___ | shes nice |
| 07:40.24 | k-man___ | cate blanchette |
| 07:40.31 | JT | i think i prefer female usually |
| 07:40.53 | JT | but the sbs voiceover guy or the cityrail announcement guy would be funny as hell |
| 07:40.59 | k-man___ | i wonder who the woman is who does all the telstra prompts |
| 07:41.12 | JT | yeah, she's got an extremely neutral voice |
| 07:41.14 | k-man___ | oh yeah, hes cool |
| 07:41.21 | JT | maybe it's a computer generated voice ;) |
| 07:41.26 | k-man___ | maybe |
| 07:41.44 | k-man___ | there are a few pretty good computer generated voices nowadays too |
| 07:42.03 | JT | The train on platform 16 has been delayed due to... fatality. Cityrail appologises for any inconvenience caused. |
| 07:42.35 | k-man___ | *shudder* |
| 07:42.40 | JT | i've actually heard that one |
| 07:42.46 | JT | i might've got the platform number wrong |
| 07:42.48 | k-man___ | oh god... thats terrible |
| 07:42.58 | JT | but yeah, the prerecorded system has "fatality" |
| 07:43.21 | JT | i think the average is 1 a week actually |
| 07:43.27 | JT | deaths by cityrail trains |
| 07:43.30 | JT | from jumpers |
| 07:43.59 | k-man___ | in japan, if someone commits suiside by train, the family has to pay for the cleanup and the cost of the delays |
| 07:44.08 | k-man___ | they did that in an efford to deter the jumpers |
| 07:44.32 | JT | all media agencies have an unofficial agreement not to report most suicides, as it's been statistically shown to increase the incidence of suicide |
| 07:45.00 | fetcher | "hey, I can be on TV too!" |
| 07:45.15 | k-man___ | oh geez |
| 07:45.39 | JT | but in sydney alone there's probably at least a dozen per week |
| 07:45.44 | k-man___ | hmm... i can;t satisfy the build requirements for 1.4 under debian stable easily |
| 07:45.51 | JT | lol |
| 07:45.56 | JT | debian stable... yeah |
| 07:46.01 | k-man___ | well... |
| 07:46.11 | k-man___ | this machine was installed a long time ago |
| 07:46.11 | JT | can be pretty old packages |
| 07:46.16 | k-man___ | yeah |
| 07:46.35 | k-man___ | i need a newer version of libtonezone1 |
| 07:46.43 | k-man___ | but that would require libc6 to be upgraded |
| 07:46.45 | JT | they once had the cityrail voiceover guy on big brother up late, but i reckon he hardly sounded the same |
| 07:46.55 | k-man___ | oh |
| 07:46.58 | k-man___ | interesting |
| 07:47.16 | JT | probably need the right environment to get the voice just right |
| 07:47.23 | k-man___ | yeah |
| 07:47.33 | k-man___ | and a $10,000 mic |
| 07:47.38 | k-man___ | in a sound proof room |
| 07:47.38 | JT | sbs voiceover guy would be good of you ran some sort of erotica business |
| 07:47.45 | k-man___ | supported on an air cushion |
| 07:47.47 | JT | yeah and scripted lines |
| 07:47.55 | k-man___ | yeah |
| 07:47.58 | creativx | all that to play it out at 8khz ? |
| 07:48.07 | JT | the sbs dude sounds like he's always trying to seduce you |
| 07:48.15 | JT | creativx: nah, PA system :P |
| 07:48.20 | k-man___ | jt, yeah, he's got a great voice |
| 07:48.29 | JT | i think foxtel use him too |
| 07:48.45 | k-man___ | he used to do a bit for TripleJ too |
| 07:48.50 | creativx | oic |
| 07:49.06 | JT | if i had somewhere to record, i could try doing voiceovers, be curious to see how good they sounded |
| 07:49.19 | k-man___ | jt, yeah.... |
| 07:49.21 | JT | i have a very telephone friendly voice i'm led to believe |
| 07:49.32 | k-man___ | i wonder how many takes they need to do |
| 07:49.40 | k-man___ | can you say "1" again please? |
| 07:49.45 | JT | heh no idea |
| 07:50.03 | k-man___ | jt, you know the guy who plays the cop on blue heelers |
| 07:50.10 | k-man___ | John Wood i think |
| 07:50.16 | JT | know of him, yes |
| 07:50.18 | k-man___ | he has a great voice for voice overs |
| 07:50.31 | k-man___ | he does the voice over on this claymation an aussie guy does |
| 07:50.33 | k-man___ | very good |
| 07:50.46 | JT | hmm maybe, can't quite picture his voice |
| 07:51.11 | JT | not a big fan of the vodafone voiceover chick |
| 07:52.06 | JT | the only advantage of her "Hello" being an octave higher than everything else on the "mobile unavailable..." message is it's easy to recognise and hit disconnect very fast |
| 07:53.14 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 07:53.16 | k-man___ | http://www.atomfilms.com/film/brother.jsp |
| 07:54.33 | k-man___ | jt, that cartoon has john woods voice i think |
| 07:55.49 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:58.10 | *** part/#asterisk BOLIVIAN (n=ewrtwert@201.222.93.229) |
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| 08:01.03 | kuto | hi all can anyone tell me what ids stand for? |
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| 08:25.35 | marc7 | any idea why this wouldn't be working? http://www.voip-info.org/wiki/view/Asterisk+tips+fileexistance |
| 08:26.02 | marc7 | by "not working", i mean when the script is invoked, it immediately goes to the next step in the context (eg, n+1) instead of n+100 |
| 08:33.10 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
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| 08:36.44 | zapp-branigan | mac7 are you there ? |
| 08:36.47 | wannabeaastguru | Hey guys : is it possible to use g729 with sjphone |
| 08:37.11 | wannabeaastguru | im very satisified with SJphone, works well in Linux |
| 08:37.21 | zapp-branigan | use fopen in mode read if returm NULL is not present |
| 08:38.01 | wannabeaastguru | i know its closed source code, but any hacks to make it work with the free ver of g.729 |
| 08:38.08 | *** join/#asterisk vlt (n=dm@p54B33559.dip0.t-ipconnect.de) |
| 08:38.41 | marc7 | zapp: not sure if you meant me or nott |
| 08:39.09 | marc7 | but i solved my problem, the "priorityjumping" command was set to "no" |
| 08:39.32 | zapp-branigan | wanachuru http://kvin.lv/pub/Linux/Asterisk |
| 08:40.08 | *** part/#asterisk EyeCue (n=eyecue@miranda/admin/eyecue) |
| 08:41.18 | wannabeaastguru | zapp-branigan: thanx man, cool, g729 codec for ast was there |
| 08:41.40 | zapp-branigan | but do not work very very good |
| 08:41.53 | zapp-branigan | some times lose packets |
| 08:42.10 | wannabeaastguru | i see |
| 08:42.18 | zapp-branigan | avery 5 secons o more |
| 08:42.20 | wannabeaastguru | did u test g729? |
| 08:42.26 | zapp-branigan | yes |
| 08:42.28 | wannabeaastguru | oh i see |
| 08:42.56 | wannabeaastguru | u mean it drops packets on a average of 5 sec? |
| 08:44.07 | zapp-branigan | the sound is bad this codec is only for text |
| 08:44.18 | wannabeaastguru | shit, i thought i found the goldmine |
| 08:44.20 | wannabeaastguru | :) |
| 08:44.29 | Strom_C | wannabeaastguru: what are you trying to accomplish? |
| 08:45.19 | zapp-branigan | intel have a g729 who work fine |
| 08:45.21 | wannabeaastguru | zapp-branigan: free? |
| 08:45.28 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:45.31 | zapp-branigan | you can download the icc compiler the codec |
| 08:45.41 | zapp-branigan | apli a patch and run |
| 08:45.41 | wannabeaastguru | Strom_C: test g729 without paying |
| 08:45.51 | Strom_C | you realize that you're committing patent infringement by doing that, right? |
| 08:46.00 | zapp-branigan | but only in fedora 4 and uguntu and some else |
| 08:46.15 | zapp-branigan | fedora 5 and 6 is not soported |
| 08:46.31 | Strom_C | anyway, g729 sounds like gsm but uses 5 kbps less bandwidth |
| 08:46.32 | wannabeaastguru | zapp-branigan: ok. i see, i run CentOS 4.4(RHEL 4.4) |
| 08:46.39 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 08:46.48 | zapp-branigan | i'm bot sure |
| 08:46.59 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 08:47.08 | wannabeaastguru | Strom_C: i see, so it may be not worth all the stress |
| 08:47.23 | zapp-branigan | and another codec is freebsb53 |
| 08:47.28 | Strom_C | wannabeaastguru: if you're DYING to try it out, just spend the $10 |
| 08:47.37 | wannabeaastguru | Strom_C: im very very satisified with the gsm |
| 08:47.37 | Strom_C | but really, it sounds like gsm |
| 08:47.44 | zapp-branigan | http://www.ingenjörsfirman.se/share/text/school/telecom/voip/softswitch/asterisk/codec/ |
| 08:48.00 | wannabeaastguru | Strom_C: ok, i will think about it :) |
| 08:48.40 | wannabeaastguru | zapp-branigan: Er du svensk? |
| 08:48.50 | zapp-branigan | no, hablo español |
| 08:49.07 | zapp-branigan | spanish |
| 08:49.13 | wannabeaastguru | zapp-branigan: si si |
| 08:49.17 | wannabeaastguru | zapp-branigan: :) |
| 08:49.25 | zapp-branigan | :P |
| 08:50.33 | zapp-branigan | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 08:50.45 | zapp-branigan | this works fine in from intel |
| 08:50.50 | wannabeaastguru | zapp-branigan: This one i know |
| 08:50.56 | wannabeaastguru | i see |
| 08:51.12 | wannabeaastguru | what about this swedish codec? goof? |
| 08:51.16 | wannabeaastguru | good* ? |
| 08:51.27 | zapp-branigan | work free 1 month |
| 08:51.52 | wannabeaastguru | swedish codec u mean? |
| 08:52.20 | wannabeaastguru | Strom_C: ok, u've tested g729 & sounds like gsm? |
| 08:52.40 | Strom_C | yes |
| 08:52.55 | Strom_C | there are small differences, but overall they sound about the same |
| 08:53.07 | Strom_C | i'd stick with gsm unless you're starved for bandwidth |
| 08:53.27 | *** join/#asterisk lorinc (n=ang@caracas-1502.adsl.interware.hu) |
| 08:53.41 | wannabeaastguru | Strom_C: ok, isee, then its for voip servers with many many clients really |
| 08:54.07 | Strom_C | how many concurrent calls are you trying to run? |
| 08:55.19 | wannabeaastguru | not many |
| 08:55.36 | Strom_C | "not many" is not a number |
| 08:55.43 | Strom_C | can you be more specific please? :) |
| 08:56.06 | wannabeaastguru | sorry |
| 08:56.22 | wannabeaastguru | im testing ast+openser and installed recently |
| 08:56.42 | wannabeaastguru | im only testing with my friends |
| 08:56.53 | wannabeaastguru | 2-3 concurrently |
| 08:57.12 | zapp-branigan | use the first codec who i send you |
| 08:57.12 | Strom_C | and how much bandwidth do you have? |
| 08:57.14 | zapp-branigan | to text |
| 08:57.25 | zapp-branigan | if you like buy from digium |
| 08:57.57 | wannabeaastguru | ok |
| 08:58.26 | wannabeaastguru | my servers is at my friends house, with only 1 Mb upload i think |
| 08:58.39 | zapp-branigan | the codec digium work very very better than the others |
| 08:59.02 | Strom_C | wannabeaastguru: if you have that much upstream bandwidth available, you should try using g.711 |
| 08:59.08 | Strom_C | it sounds much better |
| 09:00.05 | wannabeaastguru | Strom_c :i see |
| 09:00.25 | zapp-branigan | if you have very low conecction use lpc10 :P |
| 09:00.37 | wannabeaastguru | oki |
| 09:00.40 | zapp-branigan | ha ha |
| 09:00.45 | wannabeaastguru | :) |
| 09:01.01 | zapp-branigan | this codec you hear a robot |
| 09:01.13 | wannabeaastguru | zapp-branigan: joke :) |
| 09:01.16 | wannabeaastguru | zapp-branigan: i see |
| 09:01.54 | zapp-branigan | http://www.voip-info.org/wiki-Codecs |
| 09:02.12 | wannabeaastguru | Strom_c : i wonder what codec a pro VOIP provider uses to its mass customers? |
| 09:02.21 | zapp-branigan | downloas speex and install |
| 09:02.27 | Strom_C | wannabeaastguru: likely g.711 |
| 09:02.33 | Strom_C | best quality, lowest latency |
| 09:02.37 | drako | Ok im having serious problem with a TDM on Debian, it hangsup the system , freezing the whole OS when i try to make a phone call trought the TDM card. |
| 09:02.38 | wannabeaastguru | Strom_c : really? |
| 09:02.41 | Strom_C | yes |
| 09:02.56 | zapp-branigan | # ITU G.711 - 64 Kbps, sample-based Also known as alaw/ulaw |
| 09:02.58 | Strom_C | drako: which version of debian? |
| 09:03.04 | drako | Strom_C, Etch |
| 09:03.10 | wannabeaastguru | Strom_c : but it will consume a lot bandwith with 1000 calls simultaneously?? |
| 09:03.12 | zapp-branigan | # ITU G.729 - 8 Kbps, 10ms frame size |
| 09:03.13 | zapp-branigan | # Speex - 2.15 to 44.2 Kbps |
| 09:03.37 | Strom_C | wannabeaastguru: yes, but any ITSP that actually cares will have enough bandwidth available |
| 09:03.45 | zapp-branigan | ok |
| 09:04.10 | wannabeaastguru | Strom_c : of course, that shouldn't be a problem for them |
| 09:04.28 | wannabeaastguru | zapp-branigan: i will do that |
| 09:04.49 | wannabeaastguru | zapp-branigan: i'll read about speex |
| 09:05.20 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:05.52 | zapp-branigan | download install an re make asterisk |
| 09:06.09 | zapp-branigan | and the speex.so apear |
| 09:06.41 | wannabeaastguru | zapp-branigan: u mean i've to install ast again? |
| 09:07.13 | zapp-branigan | donload firt speex if you want |
| 09:07.20 | wannabeaastguru | zapp-branigan: i've been to this site on voip-info about codecs, but didn't spend time on every codecs .) |
| 09:07.32 | zapp-branigan | what version of asterisk you have ? |
| 09:07.35 | wannabeaastguru | 1.4 |
| 09:07.48 | zapp-branigan | then you must to edit some file |
| 09:08.01 | zapp-branigan | make something |
| 09:08.01 | wannabeaastguru | Strom_c : R u scandinavian? |
| 09:08.07 | Strom_C | no |
| 09:08.09 | zapp-branigan | to compile speex |
| 09:08.19 | wannabeaastguru | Strom_c : ok, ur name sounds like |
| 09:08.29 | zapp-branigan | 1'4 not compile speex alone |
| 09:08.50 | wannabeaastguru | zapp-branigan: oh i see, u have to 'make' speex only |
| 09:08.56 | zapp-branigan | no |
| 09:09.06 | zapp-branigan | download speex |
| 09:09.12 | wannabeaastguru | zapp-branigan: yesm then |
| 09:09.18 | zapp-branigan | ./configure make make install |
| 09:09.23 | wannabeaastguru | yes ok |
| 09:09.34 | wannabeaastguru | then there should be a speex.o ? |
| 09:09.41 | zapp-branigan | then in the asterisk go to the directory of the source |
| 09:09.48 | wannabeaastguru | ok |
| 09:09.58 | zapp-branigan | and vi makesomthing |
| 09:10.05 | zapp-branigan | remove a line |
| 09:10.15 | wannabeaastguru | ok |
| 09:10.22 | zapp-branigan | and ./configure asterisk make make insstall |
| 09:10.33 | wannabeaastguru | oki i see |
| 09:10.48 | wannabeaastguru | so first u make speex then make ast |
| 09:10.48 | zapp-branigan | and the .so apear in the codec directory |
| 09:10.53 | wannabeaastguru | i see |
| 09:11.21 | zapp-branigan | i not remenber the file |
| 09:11.26 | wannabeaastguru | zapp-branigan: no prob |
| 09:11.36 | wannabeaastguru | zapp-branigan: but what about the qulity? |
| 09:11.47 | wannabeaastguru | zapp-branigan: voicequality? |
| 09:12.25 | zapp-branigan | use gsm :P |
| 09:12.52 | zapp-branigan | or g729 from digium |
| 09:13.08 | drako | Ok im having serious problem with a TDM on Debian, it hangsup the system , freezing the whole OS when i try to make a phone call trought the TDM card... |
| 09:13.43 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:13.57 | wannabeaastguru | zapp-branigan: ok, thanx for now |
| 09:14.17 | wannabeaastguru | zapp-branigan & Strom_c: Thank u very much for help |
| 09:14.41 | wannabeaastguru | zapp-branigan & Strom_c: ttyl |
| 09:14.57 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:16.24 | *** join/#asterisk lately (n=doug@60-242-152-212.tpgi.com.au) |
| 09:16.58 | *** join/#asterisk Modcuts (n=Moducts@88-111-26-64.dynamic.dsl.as9105.com) |
| 09:17.07 | yansolo90 | Does the option "buggymwi" works in SIP.conf with Cisco phones and * 1.4 ? |
| 09:17.29 | Strom_C | drako: FWIW, my TDM cards work just fine under sarge |
| 09:18.01 | Strom_C | drako: which version of zaptel are you running? |
| 09:18.37 | drako | Strom_C, 1.2.13 |
| 09:18.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:18.54 | drako | Strom_C, where I can say debian to stop load zaptel modules |
| 09:19.09 | drako | Strom_C, i commented everything in /etc/modutils/zaptel |
| 09:19.09 | Strom_C | i'm sorry, that sentence makes no sense. |
| 09:19.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:19.45 | drako | Strom_C, its loading the zaptel module on startup |
| 09:19.54 | *** join/#asterisk s1gny|wrk (n=s1gny@p549174E5.dip.t-dialin.net) |
| 09:19.58 | drako | Strom_C, but i don't know from where its taking that orden. |
| 09:20.22 | Strom_C | from what I recall, it sees the card on boot and loads the appropriate driver |
| 09:21.10 | lately | Can someone please help me out here. Upgraded to 1.4 the other day now lost ${CALLERIDNUM} |
| 09:21.11 | lately | exten => 8500,1,Macro(myvoicemail,CALLERID(num)) |
| 09:21.24 | lately | Can someone explain what's wrong with above line? |
| 09:22.09 | Strom_C | lately: CALLERID is a function |
| 09:22.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:22.23 | lately | Strom_C: Aren't I calling it as such? |
| 09:22.25 | Strom_C | lately: you treat it just like a variable |
| 09:22.25 | Strom_C | no |
| 09:22.36 | Strom_C | you need to treat it like a variable |
| 09:22.37 | lately | ${CALLERID(num)} ? |
| 09:22.40 | Strom_C | yes |
| 09:22.45 | lately | ah... bugger me |
| 09:22.48 | lately | :-) |
| 09:22.53 | Strom_C | you should have known that CALLERIDNUM was deprecated in 1.2 |
| 09:23.00 | Strom_C | didn't you read the documentation? :) |
| 09:24.13 | lately | I saw in the changelog that CALLERIDNUM was deprecated so I found that the function CALLERID replaced it. Was trying to figure out why the function call wasn't working. |
| 09:24.37 | lately | Now it works :-) Thanks heaps |
| 09:26.44 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:27.15 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 09:35.46 | *** part/#asterisk s1gny|wrk (n=s1gny@p549174E5.dip.t-dialin.net) |
| 09:35.46 | tzafrir | drako, you use kernel 2.4 or 2.6? if not 2.6, consider using 2.6 (e.g 2.6.8 of Sarge) |
| 09:36.38 | tzafrir | load zaptel modules from /etc/modules if you want to guarantee their load order. Otherwise they'll be loaded by hotplug and/or discover |
| 09:37.21 | drako | tzafrir, 2.6.18 |
| 09:37.47 | tzafrir | then you should know that /etc/modutils is relevant only to the modutils of kernel 2.4 |
| 09:38.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 09:38.34 | tzafrir | Take a look at the init.d script of the zaptel deb (of Etch) |
| 09:39.47 | *** join/#asterisk ghenry (n=ghenry@82-69-53-89.dsl.in-addr.zen.co.uk) |
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| 09:42.34 | drako | tzafrir, im thinking about compile zaptel from source, since im getting all this freez problems |
| 09:42.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:42.42 | drako | tzafrir, the OS crash so bad |
| 09:43.16 | tzafrir | go ahead. However the init.d script of Zaptel is not the finest. The one of the deb is much better |
| 09:44.06 | tzafrir | drako, could you be more specific? I'd appreciate a detailed bug report (to either bugs.digium.com or bugs.debian.org ...) |
| 09:46.59 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
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| 09:50.36 | drako | tzafrir, my english is not good enought to make a detailed bug report :/ |
| 09:50.48 | drako | tzafrir, i can tell you whats the problem in my way :/ |
| 09:50.55 | tzafrir | good enough |
| 09:51.31 | drako | the problem is, when I use echocancel=yes and i try to make a phone call throught the pstn it hangs |
| 09:51.32 | drako | freez |
| 09:51.35 | drako | crash |
| 09:51.47 | drako | wont even response to the keyboard press |
| 09:53.08 | *** join/#asterisk pasqu (n=pasqu@151.38.43.46) |
| 09:53.28 | pasqu | hi all |
| 09:53.51 | pasqu | can anyone help me for make a trunk between cisco cme & asterisk |
| 09:53.53 | pasqu | ? |
| 09:59.35 | drako | tzafrir, any idea? |
| 10:00.16 | *** join/#asterisk shinux_ (n=shinux@196.220.30.42) |
| 10:02.12 | tzafrir | drako, what hardware? |
| 10:02.31 | tzafrir | And what type of "crash"? kernel panic? 100% CPU time? |
| 10:04.40 | drako | tzafrir, tdm04b |
| 10:04.54 | drako | tzafrir, im remote |
| 10:05.02 | drako | tzafrir, it just stop work |
| 10:05.06 | drako | wont response to ping |
| 10:05.26 | drako | nor the webserver |
| 10:05.45 | tzafrir | has it worked with any version of zaptel before? |
| 10:06.03 | drako | on a trixbox yes |
| 10:06.07 | drako | it had randoms hangs too |
| 10:06.10 | tzafrir | ping should be answered from the kernel directly. Other the kernel is very busy or it has panicked |
| 10:06.22 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 10:06.27 | drako | but this is everytime i try to call |
| 10:07.49 | tzafrir | BTW: to somewhat reduce the echo, have you set opermode properly? Also use fxotune (get fxotune.c from zaptel 1.4) |
| 10:08.37 | Zefk | Hi all! Anyone succeeded with TE412P ? |
| 10:08.52 | drako | tzafrir, well thats a problem, to get the right opermode i have to rmmode the modules |
| 10:09.10 | drako | and do it by hands because i can't find out where to put it to load at startup |
| 10:09.26 | tzafrir | stop asterisk, then rmmod wctdm |
| 10:09.47 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 10:10.15 | tzafrir | No, you don't need to rmmod it in your init.d script |
| 10:10.58 | tzafrir | If you have any problems with passing module options, remove the silly /etc/modprobe.d.zaptel |
| 10:11.18 | tzafrir | The default contents of it only gets in your way |
| 10:11.23 | drako | tzafrir, i also have this weirds WARNINGS |
| 10:11.28 | drako | tzafrir, http://pastie.caboo.se/40474 |
| 10:12.23 | vlt | Hello (again). I'm trying to compile qozap module on Ubuntu Dapper running 2.6.15. I installed linux-source and -headers, unzipped the source and created some symlinks that point to it. When running make I get 3,663 lines of stderr output: http://paste.ubuntu-nl.org/5870 --- What am I doing wrong here? Maybe something wrong with /usr/include/linux? Anyone here ever faced a similar problem? |
| 10:12.48 | *** join/#asterisk nas_lslsa (n=chatzill@athedsl-239577.otenet.gr) |
| 10:13.27 | tzafrir | drako, you can ignore those warnings: some of the changes in zaptel.conf require a full restart to get applied |
| 10:14.05 | tzafrir | vlt, don't bother with linux-source |
| 10:14.13 | tzafrir | apt-get install zaptel-source |
| 10:14.26 | tzafrir | apt-get install zaptel-source linux-headers-`uname -r` |
| 10:14.31 | tzafrir | m-a a-i zaptel |
| 10:14.59 | drako | tzafrir, where i set up the opermode options? in init.d/zaptel ? |
| 10:15.17 | yansolo90 | Does the option "buggymwi" works in SIP.conf with Cisco phones and * 1.4 ? |
| 10:15.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:15.50 | zamba | can i define what MoH to be used when using the M-option to meetme? |
| 10:16.59 | tzafrir | drako, it is a kernel module parameter |
| 10:17.13 | drako | tzafrir, ok but where i put it to be loaded? |
| 10:17.43 | tzafrir | options wctdm opermode=WHATEVER in /etc/modprobe.d/somefile |
| 10:18.17 | drako | ok |
| 10:19.28 | drako | ill try that |
| 10:19.50 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 10:23.14 | marc7 | if there's a few seconds being clipped off the front of my welcome greeting, should I put a Wait(1) after the Answer statement? |
| 10:23.21 | marc7 | before my welcome greeting plays |
| 10:27.04 | *** join/#asterisk FunnyManVA (n=coriley@p57A36832.dip.t-dialin.net) |
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| 10:30.06 | drako | tzafrir, stuff are working now but without echo cancelation which is a terrible thing |
| 10:30.27 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:30.35 | tzafrir | what version of zaptel is it? |
| 10:30.54 | tzafrir | (btw: have you contacted support@digium.com ?) |
| 10:31.23 | tzafrir | Do you see any kernel traces in /var/log/kern ? |
| 10:31.35 | tzafrir | right before the new startup |
| 10:32.40 | *** part/#asterisk marc7 (n=marc@S0106000f66461bdb.gv.shawcable.net) |
| 10:33.50 | drako | tzafrir, 1.2.13 |
| 10:34.49 | drako | tzafrir, no i havent contact support yet |
| 10:37.38 | drako | just tried again, crashed. |
| 10:38.53 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:39.56 | LyosNorezel | just wondering... can Asterisk be used entirely Voice-only? |
| 10:40.12 | rafee | Which charcaters should I ident on System() syntax ? |
| 10:40.43 | LyosNorezel | ie., say a name and Asterisk connects you to that person's phone? |
| 10:42.09 | giasai68 | hello |
| 10:42.11 | qdk | LyosNorezel: that isnt up to Asterisk, but your client. |
| 10:42.25 | giasai68 | any idea how I can insert codecs preference in asterisk? |
| 10:42.33 | Ahrimanes | speech recognition |
| 10:42.33 | Ahrimanes | hm |
| 10:42.34 | LyosNorezel | qdk: my client? |
| 10:42.49 | RoyK | giasai68: insert codecs?? |
| 10:42.51 | qdk | giasai68: list them in the order you prefer. |
| 10:43.01 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 10:43.11 | Ahrimanes | LyosNorezel: http://www.sineapps.com/news.php?rssid=1505 |
| 10:43.46 | giasai68 | I mean to menage 'codecs preference' |
| 10:43.51 | LyosNorezel | ah |
| 10:43.58 | qdk | LyosNorezel: client = the thingy that sends and receive voice where ever you are located. |
| 10:44.05 | LyosNorezel | ah |
| 10:44.12 | Ahrimanes | qdk: http://www.sineapps.com/news.php?rssid=1505 <- serverside |
| 10:45.05 | qdk | Ahrimanes: not what he is asking for. |
| 10:45.17 | LyosNorezel | I think I need to study the concepts of VoIP a bit more... |
| 10:45.31 | qdk | Ahrimanes: he wanna shout to his phone "call momma". |
| 10:45.46 | giasai68 | qdk: in which .conf file I can do it? |
| 10:46.16 | qdk | giasai68: where ever you need it... eg. in sip.conf if you are using sip accounts. |
| 10:46.48 | giasai68 | qdk: I'm using ooh323.conf |
| 10:46.55 | Ahrimanes | qdk: yes, could be done server side, speech recognition ivr |
| 10:47.01 | giasai68 | so I can insert in that file? |
| 10:47.17 | giasai68 | qdk: any example have you? |
| 10:47.32 | LyosNorezel | qdk: actually... I was thinking of setting up a system that would link a person's name to his/her phone number. So that anyone connected to my system... can say the person's name (and maybe city/state) and be connected to that person |
| 10:47.35 | qdk | Ahrimanes: and the phone magically transmit voice to the server? |
| 10:47.43 | Ahrimanes | qdk: off-hook autodial |
| 10:47.46 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:47.57 | qdk | giasai68: no i dont use h323. |
| 10:48.36 | LyosNorezel | qdk: biggest issue will be the sheer size of a telephone database |
| 10:48.57 | giasai68 | qdk: is it possible emulate h323 in sip ? |
| 10:49.02 | Ahrimanes | or call 877-977-0707 and say "Demo." <- hehe |
| 10:49.10 | drako | tzafrir, there is other way to echocancel ? |
| 10:49.27 | qdk | LyosNorezel: well you should make the logic and database first and the put voice-recon on top of it. |
| 10:50.08 | tzafrir | drako, can you try 1.2.12? 1.2.13 had some changes in the echo canceller area |
| 10:50.46 | qdk | giasai68: afaik h323 is its own channel with its own properties, and i would be surprised if you disallow/allow doesnt work as in SIP. |
| 10:50.48 | tzafrir | (again, if you have anything in your kernel logs it will be of great help to debug this issue) |
| 10:51.32 | *** join/#asterisk shinux__ (n=shinux@196.220.30.42) |
| 10:52.00 | drako | tzafrir, i don't see any info on kernel logs |
| 10:52.20 | drako | tzafrir, ok so compiling from source? im using a deb packages from the site you gave me. |
| 10:53.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 10:55.17 | giasai68 | qdk: please, can u explain? |
| 10:56.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:57.23 | qdk | giasai68: http://www.voip-info.org/wiki-Asterisk+config+sip.conf <- search for disallow and see how its done in SIP. |
| 10:58.18 | giasai68 | qdk: any idea regard h323 and g729r8 codec? |
| 10:59.13 | qdk | giasai68: http://www.mail-archive.com/ooh323c-devel@lists.sourceforge.net/msg00233.html |
| 10:59.30 | giasai68 | qdk: thanks |
| 11:04.23 | vlt | tzafrir: In the makefile I found "Link /usr/src/linux-2.6 to your kernel sources first!". That's what I did and then got 3000+ error lines. Because I couldn't understand why compiling needs not headers but source I changed the symlink to linux-headers instead. Now I get only a few error lines. One of them is '*** Warning: "zt_register" [/home/dm/bristuff-0.3.0-PRE-1x/qozap/qozap.ko] undefined!' When trying to load the now compiled qozap.ko I |
| 11:04.23 | vlt | ol zt_alarm_notify_no_master_change". |
| 11:05.06 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 11:05.35 | tzafrir | vlt, that is no longer necessary for quite a while |
| 11:05.55 | vlt | tzafrir: What's not necessary? |
| 11:06.05 | tzafrir | vlt, look for KSRC in the makefile |
| 11:06.49 | tzafrir | vlt, I don't really like bristuff's build scripts |
| 11:07.10 | tzafrir | If you're on Debian, do use recent debs, as hey are already bristuffed |
| 11:07.18 | vlt | tzafrir: There's no match for KSRC in the Makefile |
| 11:07.49 | tzafrir | vlt: zaptel/Makefile |
| 11:08.09 | vlt | tzafrir: I need to compile it myself because later, when compiling finally works, I want to add some cosmetic corrections to qozap.c |
| 11:08.12 | *** join/#asterisk nuny (n=nuny@200.102.160.1) |
| 11:08.31 | tzafrir | BTW: the error you quote is because zaptel.ko has failed to load |
| 11:08.55 | vlt | tzafrir: Yes, I got the same errors on compiling zaptel |
| 11:09.14 | vlt | tzafrir: It's done in one compile.sh from Junghanns |
| 11:10.45 | nuny | hi, anybody already used Planet VIP gateways with asterisk ? |
| 11:12.01 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 11:13.23 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:14.43 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 11:21.44 | tzafrir | vlt, I'd appreciate it if you communicate that information to JJunghanns |
| 11:24.42 | JT | tzafrir: he needed to modify qozap because it does not appear he has a junghanns card |
| 11:28.32 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.in.comcast.net) |
| 11:28.46 | tzafrir | what card is it? |
| 11:28.57 | JT | that's the weirded bit |
| 11:28.59 | JT | unkown |
| 11:29.07 | JT | neither beronet or junghanns |
| 11:29.22 | JT | qozap has code that identifies it as a reference design |
| 11:29.35 | JT | and he's 60miles from the server with the card in |
| 11:29.44 | JT | i suspect he must've inherited it or something |
| 11:30.43 | coppice | a refererence design, eh? must be goodm then :-) |
| 11:31.24 | tzafrir | which one? from pbxhardware.com? |
| 11:31.52 | JT | dunno, you try extracting that info out of him :) |
| 11:31.53 | JT | i tried |
| 11:32.11 | JT | he just kept saying the server was 60miles away and he had no idea where the card was from |
| 11:35.24 | coppice | has anyone else noticed that octasic have started supplying their echo canceller has host software for asterisk? |
| 11:41.14 | coppice | s/has/as |
| 11:48.51 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 11:50.23 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:55.24 | vlt | JT: Hello. I really don't know the manufacturer of the card, but I'm at the server now and will have a look ... |
| 11:56.12 | *** join/#asterisk dj-fu (n=deejay@202-89-157-131.dyn-dsl.ubs.qsi.net.nz) |
| 12:07.08 | tzafrir | coppice, on their own or through Digium? |
| 12:07.34 | phearless | hello everybody |
| 12:07.41 | coppice | on their own. digium's one is apparently licenced from adaptive digital |
| 12:07.44 | phearless | when I make a phone call via PRI I got this kind of errors : |
| 12:07.45 | phearless | http://paste.lisp.org/display/36893 |
| 12:07.55 | phearless | does it makes sense for anybody ? |
| 12:08.04 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 12:11.45 | phearless | anybody? |
| 12:12.05 | phearless | Feb 15 12:05:40 NOTICE[6350]: chan_zap.c:6500 handle_init_event: Alarm cleared on channel 6 |
| 12:12.10 | phearless | Feb 15 12:05:52 WARNING[6350]: chan_zap.c:6507 handle_init_event: Detected alarm on channel 2: Yellow Alarm |
| 12:12.10 | phearless | Feb 15 12:05:52 WARNING[6350]: chan_zap.c:1584 zt_disable_ec: Unable to disable echo cancellation on channel 2 |
| 12:12.14 | phearless | it is quite strange. |
| 12:15.52 | phearless | is it an IRQ conflict? |
| 12:16.33 | tzafrir | irq conflict? |
| 12:16.40 | tzafrir | what card is it? |
| 12:17.22 | phearless | Digium Wildcard TE110P E1 |
| 12:17.33 | phearless | cat /proc/interrupts|grep te11 |
| 12:17.33 | phearless | 169: 66621515 0 IO-APIC-level wcte11xp, eth0 |
| 12:17.49 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:18.22 | phearless | on the web some people say that "chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1" is caused by IRQ conflicts |
| 12:18.35 | phearless | I have never heard about IRQ conflicts since years! |
| 12:18.51 | vlt | JT, tzafrir: I tried all I could. The card is in the lowest pci slot near to the case bottom, so I can only see its back. I even took our receptionist's pocket mirror sliding it between the card and the case ...... but I can only see that it's the HFC-4S chip on the card. Maybe I can shutdown the server this evening (CET) ... tzafrir: I looked for KSRC in zaptel/makefile but think that all symlinks needed exist. I even created "/usr/src/linux -> |
| 12:18.51 | vlt | .. hmmmm ... |
| 12:21.08 | phearless | tzafrir? |
| 12:21.24 | phearless | i will try to move the card to another slot |
| 12:21.27 | phearless | no? |
| 12:21.32 | phearless | is it a stupid idea? :p |
| 12:21.52 | tzafrir | phearless, no idea. But on zttool do you see irq misses from that span? |
| 12:22.03 | tzafrir | (it should also be in pri show span, IIRC) |
| 12:23.46 | phearless | http://paste.lisp.org/display/36893#1 |
| 12:23.55 | phearless | please have a look at this |
| 12:24.04 | phearless | and about zttool, I can not compile it |
| 12:24.38 | phearless | zttool.c:579: warning: implicit declaration of function ‘newtDrawRootText’ |
| 12:24.40 | phearless | etc |
| 12:25.00 | *** join/#asterisk robsdesk (n=bab610c5@adsl.ntsols.com) |
| 12:25.26 | phearless | ok i compiled it |
| 12:25.39 | robsdesk | hi, would anyone be able to help with getting a B410 going on Business Edition? |
| 12:26.03 | phearless | zttool says: |
| 12:26.06 | phearless | IRQ Misses: 49 |
| 12:27.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:30.44 | drako | tzafrir, btw , I used m-a a-i to install zaptel modules |
| 12:30.50 | drako | tzafrir, is that ok? |
| 12:31.09 | tzafrir | <PROTECTED> |
| 12:31.16 | tzafrir | <PROTECTED> |
| 12:31.17 | tzafrir | yes |
| 12:31.41 | tzafrir | it should build and install the built package |
| 12:31.59 | drako | but the built package is 1.2.13 |
| 12:32.41 | *** part/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 12:34.27 | phearless | tzafrir, any ideas for me? |
| 12:34.32 | tzafrir | no |
| 12:34.45 | tzafrir | drako, hmmm... where did you get that zaptel-source deb from? |
| 12:37.27 | drako | tzafrir, deb http://pkg-voip.buildserver.net/debian etch main |
| 12:39.14 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 12:39.19 | ThoMe | hiho |
| 12:39.25 | ThoMe | said. how i can play a wave-file? |
| 12:39.36 | ThoMe | with Playback ? |
| 12:39.41 | phearless | <PROTECTED> |
| 12:39.48 | phearless | so waht does this thing means? |
| 12:42.00 | ThoMe | how i can play wave-files? |
| 12:43.29 | tzafrir | play file.wav |
| 12:43.35 | tzafrir | (if you have sox installed) |
| 12:43.41 | tzafrir | play where? |
| 12:44.16 | ThoMe | tzafrir: aha play (file) ? |
| 12:44.31 | ThoMe | exten => s,n,Play(Ansage_Arbeistzeit_1) |
| 12:44.32 | ThoMe | tzafrir: ? |
| 12:44.36 | tzafrir | Playback |
| 12:45.34 | tzafrir | Playback(file); not Playback(file.wav) |
| 12:45.59 | tzafrir | drako, get the package from unstable, for the moment. http://packages.debian.org/zaptel-source |
| 12:46.14 | tzafrir | I suspect that you should be able to just manually install it with dpkg |
| 12:46.26 | tzafrir | drako, it is probably still 1.2.12 |
| 12:46.30 | ThoMe | tzafrir: |
| 12:46.30 | ThoMe | Feb 15 13:46:06 WARNING[11357]: file.c:436 ast_filehelper: Unable to open file on /usr/share/asterisk/sounds/Ansage_Arbeistzeit_1.wav |
| 12:46.34 | ThoMe | Feb 15 13:46:06 WARNING[11357]: file.c:824 ast_streamfile: Unable to open Ansage_Arbeistzeit_1 (format alaw): No such file or directory |
| 12:46.55 | ThoMe | ls /usr/share/asterisk/sounds/Ansage_Arbeistzeit_1.wav |
| 12:46.55 | ThoMe | /usr/share/asterisk/sounds/Ansage_Arbeistzeit_1.wav |
| 12:47.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:49.53 | ThoMe | tzafrir: |
| 12:49.53 | ThoMe | servetux*CLI> show modules like format_wav.so |
| 12:49.53 | ThoMe | Module Description Use Count |
| 12:49.57 | ThoMe | format_wav.so Microsoft WAV format (8000hz Signed Line 0 |
| 12:49.59 | ThoMe | 1 modules loaded |
| 12:51.02 | tzafrir | ~pb |
| 12:51.07 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 12:51.22 | cpm | ~areyouadog |
| 12:51.25 | jbot | bark bark! |
| 12:51.34 | tzafrir | ~bot abuse |
| 12:51.35 | jbot | ACTION huddles in the corner, whimpering 'please, please stop' |
| 12:51.35 | cpm | silly jbot |
| 12:51.52 | drako | tzafrir, thats only the source, |
| 12:52.04 | tzafrir | yeah |
| 12:52.18 | drako | m-a a-i will make also the package? |
| 12:52.26 | drako | no right? just modules |
| 12:52.51 | tzafrir | 'm-a build' just builds the package. 'm-a a-i' build and installs |
| 12:53.55 | *** join/#asterisk netsurfer (n=bbjunkie@user-514d724a.l2.c2.dsl.pol.co.uk) |
| 12:56.31 | vlt | I don't believe it. I ran `make clean all` in zaptel/ manually again. And now it worked. I can now load the qozap module ... |
| 12:57.05 | *** join/#asterisk rbw_ (n=rowi@slave.rixport80.se) |
| 12:57.06 | *** join/#asterisk enots (i=dimka@freelsd.net) |
| 12:57.20 | *** part/#asterisk enots (i=dimka@freelsd.net) |
| 12:57.36 | tzafrir | drako, zaptel-modules-<KERNEL_VERSION> is a package |
| 12:57.44 | vlt | The card is detected as "qozap: CologneChip HFC-4S evaluation board" as defined in qozap.c:1538 |
| 12:58.02 | drako | tzafrir, but i need also the zaptel packages |
| 12:58.05 | drako | not only the modules |
| 12:58.09 | vlt | (didn't apply any further patches yet) |
| 12:59.41 | tzafrir | apt-get install zaptel |
| 13:00.28 | drako | it will install zaptel 1.2.13 then |
| 13:01.03 | ThoMe | how i can wav encode to wave: 8KHZ 16 bits MONO |
| 13:01.05 | ThoMe | under debian? |
| 13:01.40 | creativx | ThoMe: sox |
| 13:01.54 | ThoMe | creativx: ah, ok, and how? |
| 13:01.59 | creativx | ThoMe: http://astrecipes.net/?n=152 |
| 13:03.45 | ThoMe | creativx: is it correct? sox Ansage_Arbeistzeit_1.wav -r 22050 /usr/share/asterisk/sounds/Ansage_Arbeistzeit_1.wav |
| 13:05.04 | vlt | ThoMe: maybe a -c 1 is missing |
| 13:05.11 | vlt | ThoMe: for the mono part |
| 13:05.20 | creativx | please read the url i gave you |
| 13:05.28 | creativx | it shows you how to make 8khz mono any bits |
| 13:06.13 | ThoMe | vlt: ah, yes. thx! |
| 13:06.52 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 13:07.00 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
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| 13:24.48 | tcseke | Hello |
| 13:24.50 | tcseke | can anybody help me? i would like to pass sip calls from a sip proxy (openser) to asterisk, but it seems it doesn't hit the dialplan, i tried sip debug but i don't see nothing, with ngrep a see |
| 13:24.51 | tcseke | invites like this http://pastebin.com/881385, is there any error in this request, why asterisk doesn't handle it? |
| 13:24.56 | *** join/#asterisk faab (i=fabio@red.digitized.nl) |
| 13:25.06 | faab | anyone here using a cisco 7961G with chan_sccp? |
| 13:25.19 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 13:25.38 | faab | yansolo90: you can set the username/passwd in the cnf.xml? |
| 13:26.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:31.51 | *** join/#asterisk ManxPower (n=manxpowe@140.sub-70-196-120.myvzw.com) |
| 13:35.42 | orlock | ssh? |
| 13:35.57 | orlock | mine only had telnet and console and the LCD |
| 13:35.57 | Nugget | telnet is eeeeeeevil! |
| 13:36.16 | orlock | yes, telnet is evil, as are all other unencrpted comms that may carry evil data |
| 13:36.41 | orlock | now begone, all of you, as i make the channel dissapear in a puff of logic! |
| 13:36.58 | orlock | and get killed on the next zebra crossing after proving black is white and white is black |
| 13:37.09 | orlock | thanks everyone, i will be here all night.. |
| 13:37.14 | `Sauron | orlock: Bad Pratchett reference |
| 13:37.23 | orlock | `Sauron: Bah Bow |
| 13:37.40 | orlock | a bad pratchett reference would involve ambulatory luggage |
| 13:37.57 | `Sauron | the puff of logic one |
| 13:38.02 | orlock | that was DOuglas Adams |
| 13:38.05 | ManxPower | All references to Pratchett are bad. |
| 13:38.10 | orlock | 42! |
| 13:38.13 | `Sauron | hum |
| 13:38.13 | `Sauron | err |
| 13:38.15 | `Sauron | right |
| 13:39.25 | *** join/#asterisk whocares123 (n=asdas@wbs-41-208-220-134.wbs.co.za) |
| 13:39.26 | orlock | `Sauron: http://www.positiveatheism.org/hist/quotes/douglas.htm |
| 13:39.35 | whocares123 | hi all |
| 13:39.37 | ManxPower | Apparently today is Feb 15th, not Feb 16th. The guy at the Amtrak ticket counter informed me of this. |
| 13:39.41 | whocares123 | hope someone can help me. |
| 13:40.06 | whocares123 | Is there a way to set the length of the called id number in asterisk |
| 13:40.09 | whocares123 | ? |
| 13:40.09 | ManxPower | whocares123: I doubt it, but the world will never know unless you ask your question. |
| 13:40.15 | orlock | heh |
| 13:40.17 | ManxPower | whocares123: yes. |
| 13:40.18 | whocares123 | lol |
| 13:40.30 | whocares123 | may I get a hint as to how |
| 13:40.36 | ManxPower | the called number is ${EXTEN} |
| 13:41.46 | whocares123 | isn't that your extension you have entered |
| 13:42.24 | whocares123 | I get an international number that has 00 in front in the place of the + and that causes that the last 2 digits gets lost |
| 13:43.09 | ManxPower | whocares123: get your terms correct. That is CALLER ID, not CALLED ID. |
| 13:43.37 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 13:43.37 | whocares123 | oops sorry my bad |
| 13:43.49 | ManxPower | whocares123: Asterisk will accept up to 255 characters of Caller*ID number information. Perhaps you are using some sissy evil GUI that should rot in hell that is messing it up? |
| 13:44.23 | whocares123 | im using astesik-java to process it in java |
| 13:44.37 | whocares123 | might be asterisk-java that causing the problem not asterisk |
| 13:45.01 | ManxPower | you can find out by putting Noop(${CALLERID}) before you call your AGI |
| 13:45.42 | faab | noone here with a cisco 7961G on sccp? :( |
| 13:46.05 | ManxPower | faab: nobody runs SCCP with Asterisk if they have any other option. |
| 13:46.10 | faab | why not? |
| 13:46.22 | faab | chan_sccp supports more stuff than cisco's own callmanager ;) |
| 13:46.27 | ManxPower | because Asterisk has far, far better support for SIP. |
| 13:46.41 | faab | cisco's sip images suck like h*ll :) |
| 13:46.57 | faab | its a really waste to load sip on a phone like the 7961G |
| 13:47.18 | whocares123 | will try to send the calledid to java in another way. Thanks for the assistance |
| 13:47.56 | faab | not even mentioning the sound quality of SIP ;) |
| 13:48.18 | SplasPood | faab: SIP doesn't exactly have anything to do with sound quality... |
| 13:48.31 | faab | let me rephrase |
| 13:48.36 | `Sauron | faab: sip vs. sccp has very little to do with the actual sound quality... |
| 13:48.39 | ManxPower | In fact SIP, MGCP, SCCP, and H323 all use the exact same protocol for audio. |
| 13:48.41 | faab | the SIP firmware for cisco phones |
| 13:48.47 | ManxPower | It is called RTP. |
| 13:48.48 | faab | screws up the sound quality |
| 13:48.49 | orlock | hmm |
| 13:48.50 | orlock | goddamn |
| 13:49.09 | faab | happy now? ;) |
| 13:49.12 | SplasPood | faab: Hrm... well mine sound fine.. but then again, I haven't used any with SCCP |
| 13:49.20 | orlock | this is giving me the shits.. two asterisk setups, same version, yet DID's work on one and break on the other |
| 13:49.21 | faab | try and you will notice the difference |
| 13:49.32 | faab | perhaps its model dependant as well |
| 13:49.42 | ManxPower | faab: I'll stick to Polycom, thankyouverymuch |
| 13:49.53 | SplasPood | i've tried 7905s 7940s and 7960s |
| 13:49.59 | SplasPood | I'm with manx tho |
| 13:50.04 | SplasPood | polycom 4 lyfe! |
| 13:50.12 | orlock | faab: i have changed a pile of cisco phones from skinny to sip/rtp, no issues |
| 13:50.31 | faab | well |
| 13:50.44 | faab | the 7912 has no problems with sip |
| 13:50.45 | faab | but the 7961G.. :( |
| 13:50.45 | orlock | and SIP is just for signalling, nothing to do with audio |
| 13:50.48 | faab | so i'm kinda forced to use sccp |
| 13:50.50 | faab | I know |
| 13:50.53 | faab | its not the protocol |
| 13:50.55 | faab | its the firmware! |
| 13:50.56 | orlock | thats prettu much down to the codec, etc |
| 13:51.10 | faab | the sip firmware for certain cisco's just suck |
| 13:51.11 | faab | s |
| 13:51.14 | orlock | yeah |
| 13:51.20 | orlock | wouldent suprise me |
| 13:51.34 | orlock | its cisco.. sell hardware with a broken OS< then charge for an upgrade |
| 13:51.37 | ManxPower | My main issue with Cisco is licensing |
| 13:51.49 | orlock | and charge to get somebody to tell you theres even a problem with the default IOS |
| 13:51.51 | faab | hehe |
| 13:51.53 | faab | well |
| 13:51.57 | faab | chan_sccp works fine |
| 13:52.01 | faab | except for a few quirks |
| 13:52.02 | dhill | do people typically require registration from sip clients on public static IPs? |
| 13:52.08 | faab | which I can sort out my self, but I'm a bit lazy |
| 13:52.15 | faab | so I wondered if someone already had a perfect setup |
| 13:52.19 | faab | but I guess not ;) |
| 13:52.41 | ManxPower | dhill: The ONLY thing registration does is inform the remote server of the IP address of the client. |
| 13:53.04 | dhill | ManxPower: so it's kind of a waste to do on clients on static public IPs :) |
| 13:53.14 | ManxPower | dhill: correct. |
| 13:53.40 | faab | its also handy to do when one end is behind a firewall |
| 13:53.44 | vlt | Me again, hello. After loading qozap module successfully "zap show status" in *CLI shows me "quadBRI PCI ISDN Card 1 Span 1 [TE] (ca? UNCONFIGUR 0 0 0" and three more lines (Span 2-4). Where (and how) do I have to configure the channels? what's the difference between zapata.conf and zaptel.con? |
| 13:54.10 | vlt | f |
| 13:54.32 | *** join/#asterisk jack404 (i=jack@404.be) |
| 13:54.37 | *** join/#asterisk gr1ncheux_ (n=devine@unaffiliated/gr1ncheux) |
| 13:55.09 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 13:55.30 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 13:55.38 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net) |
| 13:55.47 | kippi | hey, really silly question, how do you set the CID in a queue? |
| 13:56.29 | tcseke | can anybody help me? i would like to pass sip calls from a sip proxy (openser) to asterisk, but it seems it doesn't hit the dialplan, i tried sip debug but i don't see nothing, with ngrep a see |
| 13:56.30 | tcseke | invites like this http://pastebin.com/881385, is there any error in this request, why asterisk doesn't handle it? |
| 13:56.48 | jack404 | i downloaded asterisk 1.4; chap.zap.so is not being build and i can't figure out why |
| 13:56.50 | *** part/#asterisk whocares123 (n=asdas@wbs-41-208-220-134.wbs.co.za) |
| 13:57.17 | ManxPower | jack404: did you also download zaptel 1.4.x? |
| 13:57.34 | ManxPower | kippi: set the callerid before the call hits the queue |
| 13:57.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:00.14 | jack404 | ManxPower: yes |
| 14:00.42 | jack404 | can't find errors in config.log |
| 14:02.42 | kippi | ah cool |
| 14:02.44 | kippi | thanks! |
| 14:04.23 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:09.11 | *** join/#asterisk netsurfer (n=bbjunkie@user-514d724a.l2.c2.dsl.pol.co.uk) |
| 14:10.31 | kippi | anyone got anyideas why music on hold would stop? it starts and then stops staright away |
| 14:10.51 | ManxPower | kippi: VAD would be one reason |
| 14:11.13 | kippi | VAD? |
| 14:12.23 | suma | how can i make zaptel drivers load automatically when my system boots and also how can i make asterisk automatically start on boot ? |
| 14:14.07 | ManxPower | suma: you need to learn something about linux |
| 14:14.28 | suma | ManxPower: there is no sample scripts available ? |
| 14:14.31 | Aurs | suma: rc scripts |
| 14:14.38 | Aurs | suma: what distro are you running |
| 14:14.42 | ManxPower | suma: they are included in asterisk and in zaptel |
| 14:14.50 | *** join/#asterisk apardo (n=apardo@87.217.144.124) |
| 14:15.02 | suma | fc6 |
| 14:15.09 | suma | i c |
| 14:15.22 | Aurs | chkconfig zaptel on |
| 14:15.26 | Aurs | chkconfig asterisk on |
| 14:15.39 | Aurs | if you have the startup scripts installed |
| 14:16.07 | suma | Aurs:I did not have startup scripts installed |
| 14:16.19 | suma | error reading information on service asterisk: No such file or directory |
| 14:16.29 | suma | says like this for chkconfig asterisk on |
| 14:16.31 | kippi | how can I stop asterisk from striping a 0 on incomming calls? |
| 14:16.42 | Aurs | suma: make config in source tree |
| 14:17.13 | Aurs | 'make config' that is |
| 14:18.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:19.14 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 14:20.23 | ManxPower | kippi: Are you using PRI? |
| 14:21.00 | kippi | yea |
| 14:21.19 | suma | Aurs: it worked, thanks |
| 14:21.42 | Aurs | np |
| 14:22.35 | ManxPower | kippi: There are several options in /etc/asterisk/zapata.conf with regards to dialplan stuff. I don't know for sure what exact option to set. |
| 14:26.19 | *** join/#asterisk wunderkin- (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:27.27 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:30.28 | suma | i get an error when i load zaptel |
| 14:30.29 | suma | line 4: Cannot get number of tones for channel 4 |
| 14:30.37 | suma | totally of 8 errors |
| 14:31.11 | suma | http://pastebin.ca/357154 |
| 14:31.19 | suma | some one please help with this error |
| 14:31.24 | suma | i never encountered this before |
| 14:34.44 | *** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122) |
| 14:35.49 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
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| 14:38.33 | *** join/#asterisk IPmonger (n=ipmonger@68-114-72-181.dhcp.mant.nc.charter.com) |
| 14:40.06 | robsdesk | hi can someone help with a B410? |
| 14:42.20 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 14:43.21 | *** part/#asterisk blitz[training] (n=blitzrag@66.135.99.122) |
| 14:44.36 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
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| 14:50.29 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:51.16 | ManxPower | suma: make sure you have /etc/asterisk/indications.conf |
| 14:52.51 | mocker | Can someone point me to some info on what the purpose of the Jabber integration is on 1.4? |
| 14:53.06 | mocker | I read the doc that came w/ the source, but I'm still kinda fuzzy on how it integrates.. |
| 14:53.17 | drako | tzafrir, i tried with zaptel 1.2.12 |
| 14:53.20 | drako | tzafrir, same error |
| 14:53.37 | drako | tzafrir, i'd same crash |
| 14:54.04 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 14:54.14 | sweeper | is there something I need to specify in the dialplan to tell * to listen for dtmf on active calls? |
| 14:57.11 | phearless | i got issues to call OUT with PRI E1 |
| 14:58.26 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:58.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:58.41 | *** join/#asterisk xezz (n=asdas@83.235.189.2) |
| 15:00.29 | ManxPower | sweeper: see features.conf and the t,T,w, and W options to Dial |
| 15:01.13 | ManxPower | sweeper: and READ THE DAMN ASTERISK BOOK |
| 15:01.14 | ManxPower | ~book |
| 15:01.20 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:01.30 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:01.39 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:02.17 | sweeper | ManxPower: I have it open in front of me |
| 15:02.27 | sweeper | but it lacks a proper index :P |
| 15:03.31 | eald | isn't waitexten function in dialplan what you want? |
| 15:03.50 | sweeper | no, T/t is what I want D: |
| 15:04.01 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 15:04.21 | phearless | i not not understand "g1" |
| 15:04.26 | phearless | I use PRI E1 |
| 15:04.30 | phearless | with 8 channels |
| 15:04.35 | phearless | to call out I use : |
| 15:04.41 | phearless | exten => _0XXXXXXX.,1,Dial(Zap/1/${EXTEN}) |
| 15:04.44 | phearless | is it fine? |
| 15:05.22 | sweeper | phearless: no |
| 15:05.32 | sweeper | you need to put your bchannels in a group |
| 15:05.39 | phearless | ok |
| 15:05.49 | sweeper | and then Dial(Zap/g1/${EXTEN}) |
| 15:06.09 | phearless | I got : |
| 15:06.11 | phearless | group=1 |
| 15:06.11 | phearless | channel => 1-8 |
| 15:06.11 | sweeper | that way you use the first available channel in that group |
| 15:06.15 | phearless | in zapata.conf |
| 15:06.52 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 15:07.05 | sweeper | that should do it |
| 15:07.12 | phearless | exten => _0XXXXXXX.,1,Dial(Zap/g1/${EXTEN}) |
| 15:07.15 | suma | ManxPower: i will check it now, Is zaptel.conf associated with indications.conf ? |
| 15:07.17 | phearless | with this, now |
| 15:07.19 | sweeper | yea |
| 15:07.20 | phearless | cool |
| 15:07.23 | phearless | let's try. |
| 15:08.03 | *** join/#asterisk Emrah (n=Me@adslgva0491.worldcom.ch) |
| 15:09.05 | phearless | http://paste.lisp.org/display/36900 |
| 15:09.10 | phearless | ok I got this error message |
| 15:09.30 | suma | ManxPower: i check it is there and also entry for [us] too |
| 15:09.37 | ManxPower | phearless: did the call fail? |
| 15:09.43 | phearless | yes it failed |
| 15:09.46 | ManxPower | suma: I have no more suggestions |
| 15:09.49 | phearless | I pasted the log on http://paste.lisp.org/display/36900 |
| 15:09.54 | ManxPower | phearless: the failure has nothing to do with groups |
| 15:10.02 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 15:10.03 | ManxPower | it has to do with a PCI or motherboard issue |
| 15:10.12 | putzz | morning! |
| 15:10.30 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 15:10.37 | ManxPower | actually the failure is because the destination number was busy and you did not handle the busy condition in your dialpan |
| 15:10.40 | phearless | ManxPower, the incoming calls works perfectly |
| 15:10.46 | pifiu | morning everyone |
| 15:10.52 | phearless | ok |
| 15:10.56 | phearless | ok ManxPower |
| 15:11.03 | phearless | but why is this busy? |
| 15:11.07 | ManxPower | phearless: you have never use VoIP or PRI before. |
| 15:11.14 | ManxPower | phearless: because the person was on the phone. |
| 15:11.21 | sweeper | errrr |
| 15:11.24 | phearless | 1s I check the phone |
| 15:11.55 | sweeper | that error will happen if the call fails, not necesarily because it's busy |
| 15:12.01 | ManxPower | phearless: you need Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after the Dial. Remember on PRI the telco will not send you a busy tone, you need to handle those conditions be looking at hangupcause and deciding what to do |
| 15:12.29 | ManxPower | the noop will tell you the code the telco send you telling you why the call failed. |
| 15:12.54 | phearless | http://paste.lisp.org/display/36900#1 this is the new log when the phone is not busy |
| 15:12.58 | ManxPower | Look here for the meaning of the codes http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 15:13.02 | phearless | ok ManxPower |
| 15:13.26 | ManxPower | phearless: I do not see the Noop |
| 15:13.33 | phearless | I am adding it ;) |
| 15:13.37 | phearless | 1s |
| 15:14.10 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:14.11 | phearless | exten => _0XXXXXXX.,1,Dial(Zap/g1/${EXTEN}) |
| 15:14.12 | phearless | exten => _0XXXXXXX.,2,Noop(HANGUPCAUSE is ${HANGUPCAUSE} |
| 15:14.16 | phearless | ok. |
| 15:14.22 | phearless | now I reload and call.. |
| 15:14.25 | ManxPower | You are missing the closing ) |
| 15:14.52 | *** join/#asterisk doomsday- (n=doomsday@home.cameuh.net) |
| 15:14.53 | phearless | <PROTECTED> |
| 15:15.32 | ManxPower | phearless: what what is cause code 1 listed in the PDF link I sent you? |
| 15:15.33 | phearless | oh yes I forgot the ) |
| 15:16.05 | ManxPower | phearless: the telco is saying that the number you dialed does not exist. |
| 15:16.21 | phearless | right |
| 15:16.24 | phearless | unallocated number |
| 15:16.33 | phearless | let's try with another phone number |
| 15:17.00 | *** part/#asterisk IPmonger (n=ipmonger@68-114-72-181.dhcp.mant.nc.charter.com) |
| 15:17.12 | ManxPower | phearless: you need to look at [macro-std-exten] in extensions.conf.sample |
| 15:17.23 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 15:17.43 | kuku5 | How can I check the reason why the asterisk process takes 100% cpu the moment I start it ? |
| 15:17.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:17.49 | kuku5 | Any debugging I can do ? |
| 15:17.55 | phearless | ok ManxPower |
| 15:18.03 | phearless | BTW the number is right |
| 15:18.16 | phearless | let's have a look at [macro-std-exten] |
| 15:18.45 | ManxPower | phearless: the telco does not think the number is correct. contect your telco and say "I'm getting a Q.931 cause code of 1 when I dial [whatever number you are dialing], but the number is valid. |
| 15:18.52 | sweeper | phearless: but they may not be expecting it in the format you're sending |
| 15:19.26 | *** join/#asterisk dshrout (n=chatzill@c-76-18-159-29.hsd1.tn.comcast.net) |
| 15:20.51 | phearless | ok I have read macro-stdexten |
| 15:21.21 | phearless | it is made to do a different action for each dialstatus |
| 15:21.43 | phearless | sweeper, ManxPower, thanks for the help |
| 15:21.55 | phearless | maybe I need to put a 9 before the phone number or something like this? |
| 15:22.38 | ManxPower | phearless: you must send the number exactly as the telco expects it. |
| 15:23.05 | putzz | doubt it needs a 9 goint to telco maybe they expect a 10 or 11 digit number |
| 15:23.24 | phearless | ok |
| 15:23.31 | phearless | I use british telecom |
| 15:23.37 | phearless | PRI E1 |
| 15:23.57 | putzz | well how ever u dial it there is how they expect it |
| 15:24.10 | putzz | if u have to dial 1 for long distance they might expect a 1 in front of every number |
| 15:24.24 | putzz | phone them and ask how they want the numbers sent to them |
| 15:24.24 | putzz | heh |
| 15:24.28 | ManxPower | perhaps you need the prinationaldialplanand locadialplan options in zapata.conf |
| 15:24.47 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 15:25.54 | kuku5 | How can I check the reason why the asterisk process takes 100% cpu the moment I start it ? |
| 15:26.53 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 15:26.55 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:27.44 | phearless | I switched , in zapata.conf, national to euroisdn for switchtype |
| 15:28.35 | *** join/#asterisk DaveCanoe (n=Dave@strike-dsl.dclg.ca) |
| 15:28.52 | phearless | let's have a look at pridialplan |
| 15:29.18 | puzzled | hi |
| 15:29.57 | phearless | pridialplan=unknown |
| 15:29.58 | phearless | wow |
| 15:30.02 | phearless | it seems taht it works |
| 15:30.05 | phearless | ?! |
| 15:36.24 | phearless | ok |
| 15:36.37 | phearless | I can call out but after something like 30s I got many lines like : |
| 15:36.39 | phearless | Feb 15 15:31:51 WARNING[4629]: chan_zap.c:6507 handle_init_event: Detected alarm on channel 2: Yellow Alarm |
| 15:36.39 | phearless | Feb 15 15:31:51 WARNING[4629]: chan_zap.c:1584 zt_disable_ec: Unable to disable echo cancellation on channel 2 |
| 15:36.45 | phearless | and it cuts the call |
| 15:41.11 | RoyK | pridialplan=whatever |
| 15:41.53 | phearless | pridialplan=unknown |
| 15:41.58 | phearless | prilocaldialplan=unknown |
| 15:42.03 | phearless | signalling=pri_cpe |
| 15:42.05 | *** join/#asterisk cocaleca (n=lecaus@www.isoenlinea.com) |
| 15:42.09 | phearless | switchtype=euroisdn |
| 15:42.14 | phearless | these are the interesting lines |
| 15:42.17 | cocaleca | Hello there |
| 15:42.40 | RoyK | ~pb |
| 15:42.42 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:42.58 | cocaleca | anyone able to give some help for a TE110P configuration |
| 15:43.01 | cocaleca | ? |
| 15:43.03 | *** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122) |
| 15:43.09 | blitz[training] | hey all -- anyone know how to get a SmartNet account? |
| 15:43.31 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:43.41 | cocaleca | I am running AsteriskNow (nice work by the way!) but cannot configure my E1 Card! |
| 15:45.42 | elriah | Would a local SIP proxy (like Sip Express Router) resolve NAT issues with phones to a public Asterisk server? |
| 15:47.56 | *** join/#asterisk ^chewie (n=chewie@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 15:48.04 | cocaleca | elriah!, can you see my messages? |
| 15:48.21 | HarryR | elriah, if you use OpenSER and the MediaProxy plugin, you can effectively use it as a SIP and RTP proxy for your users |
| 15:48.41 | *** join/#asterisk Skaag (n=skaag@212.199.180.157.static.012.net.il) |
| 15:48.41 | HarryR | elriah, or you could use something like YATE as a frontend switch, which can also handle transcoding and media proxying etc. |
| 15:48.53 | Skaag | I've just been reading about the res_perl addon, can I also execute shell scripts? |
| 15:49.03 | Skaag | (within a dialplan) |
| 15:49.41 | blitz[training] | elriah: all I've ever needed for SIP behind NAT is to add nat=yes to the channel config in sip.conf |
| 15:50.45 | blitz[training] | I have 6 phones behind NAT at home without any forwarding or issues |
| 15:51.46 | cocaleca | Anyone have some good information about configuring E1 Cards for Asterisk? |
| 15:51.48 | elriah | blitz[training]: Apparently, you don't have Cisco phones.. lol (Polycoms and Aastra phones appear to work fine in straight NAT situations) |
| 15:52.13 | phearless | <phearless> Feb 15 15:31:51 WARNING[4629]: chan_zap.c:6507 handle_init_event: Detected alarm on channel 2: Yellow Alarm |
| 15:52.13 | phearless | <phearless> Feb 15 15:31:51 WARNING[4629]: chan_zap.c:1584 zt_disable_ec: Unable to disable echo cancellation on channel 2 |
| 15:52.16 | phearless | any ideas for this? |
| 15:52.29 | phearless | I got cut each ~ 1mn30s |
| 15:52.34 | blitz[training] | elriah: I have a 7960, a 7970, a 7912, a Polycom IP501, a Linksys SPA942, and a Mitel 5220 |
| 15:52.57 | blitz[training] | elriah: all work without issue |
| 15:53.01 | *** join/#asterisk s1gny|wrk (n=s1gny@p549158CF.dip.t-dialin.net) |
| 15:53.26 | putzz | yellow alarm = no channels open |
| 15:53.52 | ^chewie | hello, folks |
| 15:53.58 | *** part/#asterisk s1gny|wrk (n=s1gny@p549158CF.dip.t-dialin.net) |
| 15:54.08 | ^chewie | I've got a question regarding the tftp procol on Sipura devices |
| 15:54.28 | ^chewie | we have built a knoppix custom CD with a custom tftpd server |
| 15:54.39 | elriah | blitz[training]: How did you get the Cisco phones to work with NAT and Asterisk? |
| 15:54.40 | ^chewie | the server source code is derived from the inetutils-1.4.2 code base |
| 15:54.54 | elriah | blitz[training]: And which firmware are you using on your phones? |
| 15:54.57 | blitz[training] | elriah: 8.4 |
| 15:55.02 | blitz[training] | on the 7960 |
| 15:55.03 | ^chewie | but includes a config file generation application |
| 15:55.07 | *** part/#asterisk cocaleca (n=lecaus@www.isoenlinea.com) |
| 15:55.23 | ^chewie | the daemon works well with a simple linux tftp client |
| 15:55.23 | elriah | blitz[training]: That's what I have (well, 8.4SR1). Can you PM me a copy of one of your working config xml files and your default xml? It would be tremendously helpful. |
| 15:55.24 | blitz[training] | sip.conf has nat=yes in the [7960] peer |
| 15:55.28 | ^chewie | but the sipura devices are dropping connections |
| 15:55.41 | ^chewie | I'm getting tftpd: read : Connection refused errors |
| 15:55.46 | phearless | <putzz> yellow alarm = no channels open |
| 15:55.46 | phearless | ok |
| 15:55.50 | elriah | blitz[training]: And how do yo handle TFTP from behind NAT? |
| 15:56.32 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 15:56.37 | phearless | putzz, how can I open the channels? |
| 15:56.42 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:56.47 | blitz[training] | elriah: if you're behind NAT, it doesn't matter because you're opening the connection to the server |
| 15:56.47 | elriah | blitz[training]: We have 25 Cisco 7941's. |
| 15:56.52 | putzz | phearless: are you getting random messages ? |
| 15:57.00 | phearless | putzz, not random, just : |
| 15:57.04 | blitz[training] | the NAT server maintains the association |
| 15:57.13 | elriah | blitz[training]: I can't get any of our Cisco phones to register properly. I keep |
| 15:57.15 | phearless | Feb 15 15:56:08 NOTICE[4812]: chan_zap.c:8346 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 15:57.15 | phearless | Feb 15 15:56:08 WARNING[4813]: chan_zap.c:6507 handle_init_event: Detected alarm on channel 1: Yellow Alarm |
| 15:57.15 | phearless | Feb 15 15:56:08 WARNING[4813]: chan_zap.c:1584 zt_disable_ec: Unable to disable echo cancellation on channel 1 |
| 15:57.15 | phearless | F |
| 15:57.23 | phearless | each 1 or 2 minutes |
| 15:57.27 | blitz[training] | phearless: please paste to pastebin |
| 15:57.31 | phearless | sorry |
| 15:57.36 | ^chewie | the strange thing is that I don't get this with a binary built in a Knoppix 3.3 environment works |
| 15:57.45 | phearless | and it kills all the incoming or outcoming calls |
| 15:57.46 | ^chewie | the binary built in a Debian sarge environment appears to fail |
| 15:57.50 | blitz[training] | elriah: msg me and I'll paste my simple config |
| 15:57.59 | blitz[training] | elriah: actually nevermind, I'll just put it in a pastebin |
| 15:58.06 | ^chewie | the device is connected over a x-over cable to the workstation |
| 15:58.09 | ^chewie | I'm really at a loss |
| 15:58.18 | elriah | blitz[training]: You're "The Man". Hopefully this will get us squared away. |
| 15:58.22 | ^chewie | does anyone know of quirks I should be aware of for sipura devices |
| 15:58.43 | elriah | Does anyone know of a linux TFTP server that handles NAT? |
| 15:59.27 | *** join/#asterisk overridex-work (n=Dan@mail.bizfon.com) |
| 15:59.30 | blitz[training] | elriah: http://www.pastebin.ca/357281 |
| 15:59.43 | putzz | phearless: I have pvt you info on yellow alarms you can read up and check your configurations |
| 15:59.54 | overridex-work | odd question, does caller id provide the year in what it sends or is it just day/month? |
| 16:00.05 | *** join/#asterisk dwhitehead (n=dwhitehe@194.127.216.216.transedge.com) |
| 16:00.13 | elriah | blitz[training]: Thanks. Our configs are XML based but essentially the same thing as what you have there... hrm... |
| 16:00.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:00.42 | ^chewie | elriah: there is a nat kernel module for tftp |
| 16:01.02 | dwhitehead | This may be an RTFM moment, but I cannot seem to find in any online resources the answer to the following question: |
| 16:01.06 | elriah | ^chewie: Which distro? (and thanks) |
| 16:01.16 | ^chewie | elriah: vanilla kernel |
| 16:01.24 | ^chewie | elriah: should be available for most |
| 16:01.29 | ^chewie | elriah: not sure which kernel version |
| 16:01.36 | elriah | ^chewie: Hrm... I'll go see if I can find some info on it, thanks! :) |
| 16:01.41 | dwhitehead | What is needed to redirect an incoming zap call to an external long-distance number in terms of configs, if it is possible at all. |
| 16:01.56 | *** join/#asterisk Curi (n=creinero@200.24.227.122) |
| 16:02.33 | ^chewie | elriah: ip_conntrack_tftp.ko (in 2.6.8) |
| 16:02.36 | Curi | hi everyone, does any one knows what's included in the md5 hash that's sended in the digest message? |
| 16:02.45 | elriah | blitz[training]: The 79x1's are java based, I wonder if they are just bugged. |
| 16:02.47 | ^chewie | elriah: and ip_nat_tftp.ko |
| 16:02.49 | elriah | ^chewie: Thanks! |
| 16:03.08 | ^chewie | elriah: NP |
| 16:03.15 | elriah | ^chewie: Will it work with a daemonized atftpd? |
| 16:03.41 | blitz[training] | elriah: what kind of phone are you using? 7961 I guess? My 7970 is a PITA to get working. I got it working on the LAN, but not WAN. |
| 16:04.05 | ^chewie | elriah: the daemon is running on a box behind the nat? |
| 16:04.21 | elriah | We have 25 7941's. They were extremely easy to get working (until we went to configure some at remote locations, i.e., the NAT issue) |
| 16:04.24 | ^chewie | elriah: I would assume it would work with any tftpd server, it's the protocol that is NAT'ed |
| 16:04.24 | Curi | blitz[training]: do you know where i can get 7970 SIP Firmware? |
| 16:04.32 | blitz[training] | Curi: Cisco has a copy |
| 16:04.53 | ^chewie | elriah: hmm... are you going through two NAT's or one? |
| 16:04.55 | elriah | ^chewie: I'm looking up the docs now... |
| 16:05.06 | Curi | blitz[training]: there's no way of getting that without paying cisco? |
| 16:05.12 | blitz[training] | Curi: not legally |
| 16:05.20 | ^chewie | i.e. tftpd ->(nat1)->internet->(nat2)->device? |
| 16:05.27 | elriah | ^chewie: Just one. Nothing special. Asterisk is getting the register request, it never makes it back to the phone. |
| 16:05.35 | elriah | blitz[training]: Getting what from cisco? |
| 16:05.40 | Curi | blitz[training]: i've bought a couple here, but they came with skinny |
| 16:05.53 | elriah | ^chewie: Oops, that was meant for blitz[training]. |
| 16:06.12 | elriah | Oh. If you want the 8.4SR1 I'll be happy to send it to you, PM me your email address. |
| 16:07.11 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 16:08.18 | ^chewie | elriah: I would think it would be sufficient to load the nat and conntrack modules |
| 16:09.06 | elriah | Just for clarification on Cisco and firmware. You don't have to PAY for firmware, you just need a SmartNet contract to download firmware. If you call Cisco, they will give you the firmware. |
| 16:09.15 | elriah | Regardless if you have a contract. |
| 16:09.50 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:10.51 | Makenshi | are there any efforts toward a system of transcribing voicemail messages before the email gets sent out? |
| 16:11.19 | elriah | Makenshi: Speech to text? |
| 16:11.43 | blitz[training] | elriah: where do I get a SmartNet account? |
| 16:11.53 | blitz[training] | elriah: I'm trying to document that, and no one seems to know where to get one :) |
| 16:12.14 | elriah | blitz[training]: It costs like $11 per phone, but if you just call Cisco they will setup a TAC case and give you the firmware. |
| 16:12.34 | blitz[training] | elriah: great! Got a number for Cisco that I would call? That's the info I'm looking for. |
| 16:13.25 | elriah | It's the main support number, on their website. Something that ends with NETS |
| 16:13.39 | elriah | blitz[training]: I can email you the 8.4SR1 if you want it. |
| 16:14.12 | elriah | ^chewie: Any docs that you know of on that module? I can load it but it doesn't seem to do anything for me. Google doesn't yeild much. |
| 16:14.24 | blitz[training] | elriah: nah, I've got firmware, I'm trying to document for the book how other people can get a SmartNet account to legally get the firmware |
| 16:14.56 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 16:17.10 | *** join/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.co.comcast.net) |
| 16:17.32 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:17.32 | *** join/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 16:17.36 | *** join/#asterisk Creathir (n=Creathir@mobile-166-214-111-009.mycingular.net) |
| 16:17.45 | Creathir | hello everyone |
| 16:17.46 | davidcsi | anyone knows what this is? ast_read: exception flag set on sip/blah but no exception handler |
| 16:17.57 | [[blah]asfd | what does this mean? Feb 15 09:15:28 ERROR[7963]: cdr_addon_mysql.c:228 mysql_log: mysql_cdr: Failed to insert into database: (1062) Duplicate entry '' for key 4 -- Hungup 'Zap/2-1' |
| 16:18.05 | [[blah]asfd | i had a power outage and now I am getting this error |
| 16:18.32 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 16:18.34 | Creathir | I am running FPBX, and understand you cannot support this, but I am trying to get an issue resolved with a digium card, and it is giving me a bugger of a time |
| 16:19.08 | mercestes | Creathir: Your support channels are digium and #freepbx. |
| 16:19.22 | Creathir | * answers the call, but does not send a pickup ack to the phone company |
| 16:19.25 | mercestes | We really can't help you. We'd just be guessing. |
| 16:19.46 | ^chewie | elriah: http://sipx-wiki.calivia.com/index.php/HowTo_configure_iptables |
| 16:19.59 | ^chewie | elriah: try that, used "conntrak NAT HOWTO" on google to find that one |
| 16:20.06 | elriah | ^chewie: thanks again! |
| 16:20.29 | Creathir | okay, so, should I call Digium? or do they have an IRC room? |
| 16:20.32 | ^chewie | if anyone is a guru on tftpd stuff, I'd love to talk to you |
| 16:20.54 | mercestes | Creathir: We definately can't help you with that. We're very sorry. You should very much call digium. I don't think they have an IRC room. #freepbx does tho. |
| 16:21.41 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 16:22.36 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-49030bcf8bf94850) |
| 16:22.55 | Qwell[] | ManxPower: Do not taunt the macro-happy-fun-ball |
| 16:23.12 | blitz[training] | [[blah]asfd: it means Asterisk is trying to write a CDR record to your database with a null field that is not allowed to be null in your database |
| 16:23.46 | dlynes_laptop | Creathir: which country are you in? |
| 16:23.49 | [[blah]asfd | found the issue... the power outage corrupted my database. |
| 16:24.02 | dlynes_laptop | Creathir: the US? |
| 16:24.07 | Corydon-w | blitz[training]: actually, it means there's a unique index across a particular column, and he tried to insert a duplicate record |
| 16:24.24 | blitz[training] | Corydon-w: ahhh I see -- I don't use MySQL, so I don't know the exact errors :) |
| 16:24.36 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:24.51 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:24.59 | dlynes_laptop | good morning, tk |
| 16:25.47 | elriah | Let the sarcasm begin! |
| 16:25.47 | [TK]D-Fender | y0 |
| 16:25.47 | [TK]D-Fender | elriah: Sarcastic? MOI?!? *NEVER* |
| 16:25.49 | elriah | lol |
| 16:27.45 | Makenshi | elriah, yes |
| 16:28.20 | Qwell[] | Corydon-w: technically he was right :P |
| 16:28.25 | Qwell[] | just for the wrong reasons, heh |
| 16:29.29 | dwhitehead | Can anyone point me to what is necessary to take an incoming zap call and after some processing inside *, redirect it back out the zap? |
| 16:29.29 | elriah | dial() |
| 16:29.55 | dwhitehead | elriah: will that be able to operate on the same incoming analog, or do I need an additional analog? |
| 16:29.58 | dlynes_laptop | dwhitehead: take a look at dial, answer, and flash |
| 16:30.02 | Curi | blitz[training]: do you have a .cnf file for cisco 7970 SIP, that i can use as an example? |
| 16:30.27 | putzz | dwhitehead: additional analog unless u got multiple channels |
| 16:30.31 | dlynes_laptop | dwhitehead: you'll need two analog lines |
| 16:30.45 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:30.48 | dlynes_laptop | dwhitehead: but if you have flash feature, it'll release both lines after it's flashed |
| 16:30.49 | dwhitehead | crap, that' |
| 16:30.57 | dwhitehead | that's what I thought. Thanks folks! |
| 16:31.23 | dwhitehead | dlynes_laptop: is that an * feature or a POTS feature? |
| 16:31.23 | dlynes_laptop | dwhitehead: flash is also known as 'link' on some phone systems |
| 16:31.26 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 16:31.41 | dlynes_laptop | dwhitehead: it's a feature you can order on your line from your telco |
| 16:31.50 | elriah | dwhitehead: You can flash transfer with some carriers, i.e., bellsouth. There's a faq out there somewhere. |
| 16:31.56 | dlynes_laptop | dwhitehead: You need to use the 'flash' dialplan application to control it |
| 16:32.09 | elriah | oops, yea what they said |
| 16:32.14 | [TK]D-Fender | dwhitehead: Your analog line doesn't support a "transfer" feature, so theres nothing * or any other solution can do with just that. |
| 16:32.53 | dlynes_laptop | [TK]D-Fender: flash for an analog line is almost the same thing as transfer on a sip phone |
| 16:32.58 | [TK]D-Fender | dlynes_laptop: unless you're hoping for a 3-way call though multiple flashes, thats not gonna work... |
| 16:33.23 | [TK]D-Fender | dlynes_laptop: Name me a telco that lets you transfer a call.... not in any telco guide I've every seen |
| 16:33.41 | elriah | BellSouth has a consumer line (1fb/pots) feature that does it. |
| 16:33.44 | dlynes_laptop | [TK]D-Fender: nah...you receive the call, do a flash, dial a number, do a flash, and then it releases both lines |
| 16:33.45 | fall0ut | bellsouth lets you |
| 16:34.13 | [TK]D-Fender | Rare at best.... very it manually first... |
| 16:34.18 | [TK]D-Fender | verify* |
| 16:34.27 | robl^ | [TK]D-Fender: actually even SBC, the new AT&T lets you transfer calls if you subscribe to their centrx service |
| 16:34.29 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 16:34.32 | dlynes_laptop | [TK]D-Fender: we use it on our nortel system regularly |
| 16:34.35 | dlynes_laptop | [TK]D-Fender: it's not rare |
| 16:34.40 | elriah | ahhh! centrx! RUN RUN RUN |
| 16:34.46 | dlynes_laptop | [TK]D-Fender: most of our customers have that feature, too |
| 16:34.46 | fall0ut | Specifies whether the switch is currently configured to support Call Transfer. Values: |
| 16:34.48 | [TK]D-Fender | dlynes_laptop: Never seen in QC/ON |
| 16:34.55 | fall0ut | its a pretty standard feature |
| 16:34.56 | dlynes_laptop | [TK]D-Fender: bell has it, too |
| 16:35.12 | [TK]D-Fender | dlynes_laptop: Must be well hidden and specifically requested. |
| 16:35.16 | dlynes_laptop | [TK]D-Fender: bell might call it 'link' instead of 'flash' |
| 16:35.22 | dlynes_laptop | [TK]D-Fender: but they don't publicize it |
| 16:35.26 | dlynes_laptop | [TK]D-Fender: you have to ask for it |
| 16:35.45 | dlynes_laptop | [TK]D-Fender: your interconnect will usually request it for you |
| 16:35.58 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 16:36.05 | [TK]D-Fender | dlynes_laptop: link/flash is just a term/means fo doing a hook-flash. That doesn't imply that you can use it as a means of effectively doing a 1BCT on a 1FL |
| 16:36.22 | dlynes_laptop | [TK]D-Fender: i have no idea what a 1BCT or a 1FL is |
| 16:36.28 | dlynes_laptop | [TK]D-Fender: all I know is 'link' works |
| 16:36.37 | elriah | dlynes_laptop: Who is your telco? |
| 16:36.38 | dwhitehead | so if I have a current system with two incoming POTS using (I know, I now) Visual Vioce it can't be redirecting the incoming POTS call on the same POTS line, correct? |
| 16:36.38 | [TK]D-Fender | dlynes_laptop: But if indeed a service thats possible to order, NOBODY advertises it, even in business |
| 16:36.52 | dlynes_laptop | [TK]D-Fender: 'link' uses a different length hook flash than the standard hook flash |
| 16:37.02 | *** join/#asterisk mendol (i=mendol@bobas.nowytarg.pl) |
| 16:37.03 | [TK]D-Fender | dlynes_laptop: 1FL is the technical tern for a normal analog line. |
| 16:37.06 | dlynes_laptop | [TK]D-Fender: no, it's never advertised..you're correct |
| 16:37.25 | robl^ | [TK]D-Fender: http://www.att.com/gen/general?pid=3596#ref37 |
| 16:37.26 | [TK]D-Fender | dlynes_laptop: Timing specific? Dear God... |
| 16:37.33 | fall0ut | stkn_: http://cpr.bellsouth.com/pdf/sc/filings/appr/SC2002-063.pdf |
| 16:37.36 | fall0ut | er |
| 16:37.44 | fall0ut | [TK]D-Fender: see URL |
| 16:37.45 | mendol | sorry to interrupt guys, but anybody has experience with wifi voip phones + mesh ap? |
| 16:37.57 | fall0ut | rules for it |
| 16:38.07 | dlynes_laptop | elriah: allstream (AT&T), rogers, telus, Bell, group telecom, ... |
| 16:38.31 | dlynes_laptop | [TK]D-Fender: Have you ever taken a look at the flash application for the dialplan? |
| 16:38.36 | [TK]D-Fender | fall0ut: Thats USA, and Centrex specific, not a global analog service. |
| 16:38.50 | fall0ut | uhm |
| 16:38.51 | fall0ut | thats not centrex |
| 16:38.58 | fall0ut | thats 1fb |
| 16:38.59 | dlynes_laptop | [TK]D-Fender: yes, it is a centrex feature, but you don't need a centrex line to take advantage of it |
| 16:39.13 | [TK]D-Fender | Call Transfer-Deluxe - Allows you to transfer incoming calls to other Centrex lines or any line outside the Centrex system, such as to a cellular phone. Callers are spared the inconvenience of redialing and, once the call is transferred, the line is immediately free to take other calls. |
| 16:39.21 | *** join/#asterisk RoyK (n=roy@81.191.113.238) |
| 16:39.42 | [TK]D-Fender | that very clearly implies that YOU are on a centrex system. |
| 16:39.59 | Creathir | FYI, the problem was progressdetection was turned on |
| 16:40.00 | [TK]D-Fender | "other" was a keyword in there. |
| 16:40.05 | Creathir | and it did not need to be |
| 16:40.30 | *** part/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 16:40.31 | [TK]D-Fender | Creathir: Progress detection on analog is BAD. Can also cause random hangups, etc.... |
| 16:40.50 | Creathir | lol yup yup |
| 16:40.50 | Creathir | it was VERY bad |
| 16:40.58 | Creathir | turned it off in zapata.conf |
| 16:41.04 | Creathir | and all problems went away |
| 16:41.09 | Creathir | I thank you all for your help |
| 16:41.37 | dlynes_laptop | [TK]D-Fender: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash, and check the area about 'link-transfer' capability |
| 16:42.18 | dwhitehead | dlynes_laptop: so if I understand correctly, if the old system was handling this (taking incoming POTS calls and rerouting them using the same POTS line to an external agency, the lines must already have this link/flash enabled and I only need to config * to handle it? |
| 16:42.19 | kippi | how can I block people from making outgoing calls? |
| 16:42.34 | dlynes_laptop | dwhitehead: correct....see the link I gave [TK]D-Fender |
| 16:43.25 | dwhitehead | dlynes_laptop: thank you so very much! |
| 16:43.44 | [TK]D-Fender | dlynes_laptop: I don't deny the possibility of this service's existance. That link does not imply a different length of time like you suggested was needed (or perhaps having this option devalidate certain other options). It also doesn't tell you which telco's will offer it. |
| 16:43.58 | dlynes_laptop | [TK]D-Fender: take a look at the beginning of the page |
| 16:44.04 | [TK]D-Fender | dlynes_laptop: In essence you can be entirely right, but it just doesn't tell me much ;) |
| 16:44.11 | dlynes_laptop | [TK]D-Fender: you'll see that you can vary the length of the hook flash |
| 16:44.33 | dlynes_laptop | [TK]D-Fender: 200 to 1200ms |
| 16:45.24 | fall0ut | I forget the GR that specifies hook-flash lengths for loop-start |
| 16:45.24 | fall0ut | but 200-1500 covers it |
| 16:45.24 | dlynes_laptop | [TK]D-Fender: a centrex line is a digital line (no dial tone, just battery) |
| 16:45.29 | dlynes_laptop | [TK]D-Fender: this link feature is available on lines that have dial tone |
| 16:45.35 | [TK]D-Fender | dlynes_laptop: It doesn't imply that the TELCO uses 2 differnt timing for different things, it only says that differnt SYSTMES will have different timings |
| 16:46.00 | dlynes_laptop | [TK]D-Fender: yes, but I've read another document that specifies different timings for different features |
| 16:46.06 | dlynes_laptop | [TK]D-Fender: most importantly for the link feature |
| 16:46.13 | fall0ut | dlynes_laptop: for loopstart lines, that's usually not true |
| 16:46.28 | dlynes_laptop | fall0ut: referring to what? |
| 16:46.28 | [TK]D-Fender | dlynes_laptop: I had centrex lines at my old office and had perfectly normal tone. On real diff was the PBX-like extras that were possible (and never taken advantage of) |
| 16:46.50 | dlynes_laptop | [TK]D-Fender: all the centrex lines I've seen are basically nortel digital ports |
| 16:46.54 | dlynes_laptop | [TK]D-Fender: so no dial tone |
| 16:47.01 | fall0ut | dlynes_laptop: different hook-flash times for different features |
| 16:47.21 | [TK]D-Fender | dlynes_laptop: Nortel... *shudder* |
| 16:48.05 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 16:48.08 | fall0ut | dlynes_laptop: GR-506-CORE states that a "hookflash" is cycling off-hook -> onhook for 300-1100 msec |
| 16:48.24 | dlynes_laptop | [TK]D-Fender: you're not Canadian? |
| 16:48.33 | fall0ut | usually it gets stretched to 200msec |
| 16:48.43 | fall0ut | so anything that falls in there |
| 16:48.47 | fall0ut | is a hook-flash |
| 16:49.17 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:49.21 | *** join/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 16:49.32 | davidcsi | anyone knows what this means? Exception flag set on 'SIP/1130-081e5d88', but no exception handler |
| 16:49.33 | fall0ut | changing what asterisk does for a hookflash on the card, is only for inter-op between some analog lines that may not be in that 300-1100 range |
| 16:49.54 | [TK]D-Fender | dlynes_laptop: Yes, and I LOATHE Nortel. DMS100 = trouble, Norstar = ICK, Nortel stock dropping like a rock = UGH |
| 16:50.11 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 16:50.19 | mercestes | Nortel stock dropping like a rock = YAY! because it means they are going awa7 |
| 16:50.20 | [TK]D-Fender | <- Montreal, QC , remember? |
| 16:50.22 | dlynes_laptop | [TK]D-Fender: nortel stock just went up |
| 16:50.28 | mercestes | no! |
| 16:50.30 | dlynes_laptop | [TK]D-Fender: they had a reverse stock split :) |
| 16:50.33 | [TK]D-Fender | dlynes_laptop: To what,, TEN cents? |
| 16:50.52 | pifiu | hey fender how are you |
| 16:50.56 | vooduhal | If I do a reload chan_agent.so to reread the agents.conf file, will the location of the agents currently logged in be lost? |
| 16:51.22 | [TK]D-Fender | pifiu: Still breathing, and caught my second probable super-bug within a month... |
| 16:51.22 | mercestes | vooduhal: Shouldn't. |
| 16:51.29 | mercestes | voipman: notice I said *won't* I said *shouldn't*. |
| 16:51.39 | sweeper | sigh |
| 16:51.52 | [TK]D-Fender | mercestes: You're supposed to save comments like that until AFTER they try accusing you ;) |
| 16:51.55 | vooduhal | Is there any other way to reread agents.conf without doing a full reload? |
| 16:51.58 | [TK]D-Fender | mercestes: For full effect... |
| 16:52.03 | mercestes | True |
| 16:52.36 | *** join/#asterisk sourceset (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 16:52.46 | sourceset | Is it possible to have multiple application set in the call file ? |
| 16:52.58 | nas_lslsa | hum , finally I connected 2 SIP phones on my Asterisk :) .. so sip.conf is fine (at the moment) .. but what about my extensions.conf ? how can I call from one SIP phone the other one ? ! |
| 16:53.20 | vooduhal | mercestes, that worked. Thanks. :) |
| 16:53.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:53.40 | davidcsi | anyone knows what this means? Exception flag set on 'SIP/1130-081e5d88', but no exception handler |
| 16:53.42 | davidcsi | ?? |
| 16:55.49 | suma | register => 099001:pass099001pass@chumma.com:5098 <--- using IAX, I want to forward any of the incoming call to this number to a context, how can i do it ? |
| 16:56.13 | blitz[training] | nas_lslsa: exten => 100,1,Dial(SIP/john_smith) |
| 16:56.21 | dlynes_laptop | [TK]D-Fender: here's another link for you guys |
| 16:56.26 | dlynes_laptop | [TK]D-Fender: http://www.tek-tips.com/viewthread.cfm?qid=1073812&page=18 |
| 16:56.41 | blitz[training] | nas_lslsa: don't hurt yourself -- see www.asteriskdocs.org for a copy of the book |
| 16:57.12 | nas_lslsa | giiiii ... thanks .. |
| 16:57.28 | kippi | please can someone check this? http://pastebin.com/881499 it dosn't seem to ring for 6 rings |
| 16:57.47 | kippi | just seems to keep on ringing |
| 16:59.21 | blitz[training] | kippi: why are you using the Ringing application? |
| 16:59.37 | [TK]D-Fender | dlynes_laptop: "The only way to do that with analog lines is by using Centrex. You don’t have to order the whole Centrex package though…you can just get the off premise transfer feature" - confirms the nature of analog Centrex. Again, it confirms the services existance, which I can entirely believe. And finally again, doesn't say you can get it on REALLY normal lines, or who supports/offers |
| 16:59.39 | [TK]D-Fender | it. |
| 17:00.23 | fall0ut | stkn_: you have 3-way calling? |
| 17:00.27 | fall0ut | [TK]D-Fender: rather |
| 17:00.40 | kippi | should use more |n|||12 ? |
| 17:00.40 | dlynes_laptop | [TK]D-Fender: yeah...like i said a long time ago...it's a centrex feature, but you don't need a centrex line to get it |
| 17:00.53 | dlynes_laptop | [TK]D-Fender: but you always choose to read what you want to read, not everything :) |
| 17:01.29 | [TK]D-Fender | dlynes_laptop :Where does that link say you can get it on a non-centrex line? |
| 17:01.59 | [TK]D-Fender | dlynes_laptop: The response's first sentence seemed to completely confirm the opposite. |
| 17:02.28 | [TK]D-Fender | dlynes_laptop: "The only way to do that with analog lines is by using Centrex." <- Whats not clear about this? |
| 17:02.28 | suma | when i run asterisk i get Segmentation Fault ! it is 1.4, i did not add any modules, please help |
| 17:02.48 | dlynes_laptop | [TK]D-Fender: yeah...what's not clear about using centrex on analog lines about that statement? |
| 17:03.02 | sweeper | wtf... |
| 17:03.04 | dlynes_laptop | [TK]D-Fender: 'analog' |
| 17:03.21 | sweeper | musiconhold starts, stops immediately, with no errors :/ |
| 17:03.31 | [TK]D-Fender | dlynes_laptop: That "Centrex analog" != "any analog not specifically ordered as Centrex" |
| 17:03.39 | fall0ut | http://smallbusiness.bellsouth.com/ls/callingfeatures.asp |
| 17:04.39 | dlynes_laptop | [TK]D-Fender: according to you then, callwaiting, call forwarding and all that other stuff are all centrex features that are only available on centrex lines as well |
| 17:04.55 | dlynes_laptop | [TK]D-Fender: those are all centrex features as well, that you can get on analog lines |
| 17:06.15 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 17:06.58 | kippi | should use |n|||12 and not http://pastebin.com/881499 ? |
| 17:07.00 | dlynes_laptop | [TK]D-Fender: and if that's the case then 'pure analog lines' in a purist sense really don't even exist anymore |
| 17:07.06 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 17:07.16 | [TK]D-Fender | dlynes_laptop: By no means. That would mean I assume ALL functionaly was centrex-only. I'm just following that what was stated tells us straight up that THAT feature was centrex-only. |
| 17:07.43 | dlynes_laptop | [TK]D-Fender: it's a centrex feature...I never denied that...I just said you could order it on an analog line |
| 17:07.50 | [TK]D-Fender | dlynes_laptop: Welcome to our world of grey :) I'm just very good at boolean arguments ;) |
| 17:08.18 | [TK]D-Fender | dlynes_laptop: Fine. You tell me that. And then provide a link where they guy "telling it like it is" begs to differ ;) |
| 17:08.34 | dlynes_laptop | [TK]D-Fender: well fine...if you can't read English, that's not my problem |
| 17:09.11 | dlynes_laptop | anyways...I'm tired of arguing over semantics...this is a pointless argument |
| 17:09.43 | [TK]D-Fender | dlynes_laptop: I can. Quite clearly. He said in no uncertain terms -> "The only way to do that with analog lines is by using Centrex.". And you'd cite this as a reference that it is available on NON Centrex lines. Please tell me where I'm wrong. |
| 17:09.57 | dlynes_laptop | 'with analog lines' |
| 17:10.00 | fall0ut | [TK]D-Fender: because you're wrong |
| 17:10.01 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 17:10.02 | dlynes_laptop | what part of that do you not understand? |
| 17:10.47 | [TK]D-Fender | dlynes_laptop: That same quote implies Centrex is a seconday analog service to "normal" analog (whatever you'd like to define the difference as) |
| 17:10.53 | dlynes_laptop | centrex feature on an analog line...I don't think the author could make that any more clear |
| 17:11.26 | dlynes_laptop | [TK]D-Fender: yes and that's been my point all along |
| 17:11.37 | dlynes_laptop | [TK]D-Fender: as opposed to pure centrex |
| 17:11.42 | [TK]D-Fender | dlynes_laptop: It would say that I can't take my home line and just order "transfer" on it as I know by bill sure as heck doesn't say "Centrex" like my old work bill did. |
| 17:12.10 | dlynes_laptop | [TK]D-Fender: you can have an analog line, add $5/mo to your bill and have link transfer capability |
| 17:12.28 | dlynes_laptop | [TK]D-Fender: same as you can add call forwarding, advanced call forwarding, ... |
| 17:12.59 | [TK]D-Fender | dlynes_laptop: Changes some other switching, etc, and may involve other things (often centrex has a "9" prefix which you can specifically order to be suppressed) |
| 17:13.00 | dlynes_laptop | [TK]D-Fender: but you CANNOT call centrex extensions |
| 17:13.40 | [TK]D-Fender | dlynes_laptop: Can't confirm much as as far as what you can/can't do mind you, just following the text and specific points of experience. |
| 17:14.15 | dlynes_laptop | [TK]D-Fender: anyways...instead of arguing about it, order the link transfer functionality on your home 2 line service and see what we're talking about |
| 17:14.18 | [TK]D-Fender | dlynes_laptop: But yes I know you can have them change your line from "home", "business" / "other", to one billed as "centrex" without much issue. |
| 17:14.39 | dlynes_laptop | [TK]D-Fender: it's not billed as centrex...it's billed as an ordinary business or home line |
| 17:14.42 | [TK]D-Fender | dlynes_laptop: I would.... if I even HAD a phone line with tone on it any more ;) |
| 17:15.00 | [TK]D-Fender | <- Dry-loop DSL + SIP |
| 17:15.40 | [TK]D-Fender | And since I run a Sangoma S518, not even a modem most would recognize anymore ;) |
| 17:18.15 | drako | Ok I have this big issue |
| 17:18.21 | drako | Ok im having serious problem with a TDM on Debian, it hangsup the system , freezing the whole OS when i try to make a phone call trought the TDM card. |
| 17:19.47 | *** join/#asterisk nas_lslsa (n=chatzill@athedsl-239577.otenet.gr) |
| 17:20.14 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 17:20.19 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:20.30 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 17:20.59 | *** join/#asterisk AJaymn (n=mypocket@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 17:21.23 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 17:21.41 | overridex-work | odd question, does caller id provide the year in what it sends or is it just day/month? |
| 17:22.10 | CunningPike | overridex-work: CID doesn't contain date at all, afaik |
| 17:22.50 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 17:22.57 | overridex-work | CunningPike, it at least sends the day of the month and the month |
| 17:23.02 | overridex-work | just not sure if it sends the year... |
| 17:23.21 | overridex-work | every phone i've looked at just shows dd/mm and wikipedia doesn't really say :) |
| 17:23.33 | overridex-work | (i'm setting the time and date of my phone system using the cid data) |
| 17:23.46 | overridex-work | old analog system with no net connection so no ntp hehe |
| 17:25.24 | suma | can anyone please help me to clear of the seg fault with asterisk ? http://pastebin.ca/357383 |
| 17:25.34 | suma | this is the first time i'm facing this problem |
| 17:25.43 | suma | it is asterisk 1.4 |
| 17:25.56 | ^chewie | hmm, it appears not |
| 17:26.00 | ^chewie | doh, wrong window |
| 17:26.39 | suma | ^chewie: for me ? |
| 17:27.57 | fall0ut | overridex-work: doesn't contain year |
| 17:27.57 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 17:28.27 | overridex-work | thanks fall0ut |
| 17:28.52 | fall0ut | np |
| 17:28.56 | variable_office | is it possible to just get colo space in a co and get cheap voice-t1 that way instread of trying to buy through a reseller? |
| 17:29.41 | overridex-work | people will just have to set their year on their own... i think they can handle that... maybe |
| 17:29.55 | fall0ut | yea |
| 17:29.58 | nas_lslsa | question ... Asterisk supports commands like forward my call from my phone to another ? or answer a group call or that kind of staff functionalities ? |
| 17:30.29 | fall0ut | variable_office: it is technically possible, but you'd end up spending more money |
| 17:30.29 | *** join/#asterisk JeffLink (n=JeffLink@70.140.194.11) |
| 17:30.42 | variable_office | fall0ut how come rack rental space? |
| 17:33.12 | variable_office | for example, voice t1 here would be $400, in a colo I bet I could get that down to next to nothing, and then just run it off that? |
| 17:33.18 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:33.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:33.27 | variable_office | i am looking for t.38 support, but it is near impossible to find |
| 17:33.37 | fall0ut | in a colo, but not in a CO |
| 17:34.20 | Qwell[] | fall0ut: colos can often be found in the same building as a CO |
| 17:34.26 | fall0ut | and your charges for the T1 in somewhere would only be cheaper because you're not paying the ilec for the loop to your bldg |
| 17:34.27 | fall0ut | etc |
| 17:34.46 | variable_office | Qwell[] ya loop is everything here(at least for data) |
| 17:35.05 | Qwell[] | it could also be cheaper because the colo has a deal with the lec |
| 17:35.11 | ManxPower | nas_lslsa: yes and np. You can set up such features, but it requires you do to so in the dialplan. The Wiki has examples. |
| 17:35.26 | variable_office | how much is a typical voice t1 in a colo ? |
| 17:35.35 | fall0ut | voice T1 is pretty generic |
| 17:36.06 | nas_lslsa | cool , thanks |
| 17:37.01 | variable_office | fall0ut you mean the term is generic or it is readily available? |
| 17:37.12 | fall0ut | well, the term is generic |
| 17:37.15 | fall0ut | depends on what you want |
| 17:37.31 | fall0ut | some places you can get incoming only PRIs for say $125-200/month |
| 17:38.37 | drako | I'm having serious problem with a TDM on Debian, it hangsup the system , freezing the whole OS when i try to make a phone call trought the TDM card. |
| 17:38.39 | drako | any idea? |
| 17:39.03 | variable_office | fall0ut Do i need to get the colo in the same area as I want the numbers? |
| 17:39.11 | sweeper | what the bloody fuck..... |
| 17:39.26 | fall0ut | depends |
| 17:39.35 | variable_office | any way to check? |
| 17:39.41 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 17:40.02 | fall0ut | talk to a local CLEC and get a clueful sales guy and tell him what you're trying to do |
| 17:42.05 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:42.05 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.15 and Zaptel 1.2.13 released! (Feb. 9, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 17:44.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:44.48 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.15 and Zaptel 1.2.13 released! (Feb. 9, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 17:45.07 | *** join/#asterisk Lloydie-t (n=Lloydie-@host81-149-42-129.in-addr.btopenworld.com) |
| 17:45.40 | Lloydie-t | I need some help transfering incoming external sip calls to outgoing external sip calls. At the moment I get oneway conversation. Can you help? |
| 17:48.47 | ManxPower | Lloydie-t: search the mailing list archives for NAT and SIP |
| 17:49.28 | ManxPower | sweeper: # stuff is controlled by feature.conf |
| 17:49.42 | *** part/#asterisk jack404 (i=jack@404.be) |
| 17:50.55 | sweeper | ManxPower: http://pastebin.ca/357415 |
| 17:51.10 | variable_office | fall0ut or is there no single factor in determining how portable numbers are for the colo/ |
| 17:52.25 | *** join/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net) |
| 17:54.50 | [TK]D-Fender | sweeper: Pastebin the entire CLI output of your call including the attempt... |
| 17:56.16 | nosbig | I am still having trouble with IAX trunking... Both sides see the connection, but I cannot dial the other side... |
| 17:56.37 | nosbig | I get the following error: chan_iax2.c:6907 socket_read: Rejected connect attempt from XXX.XXX.XXX.XXX, who was trying to reach '32@people |
| 17:59.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:00.41 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:01.03 | *** join/#asterisk mhammerton (n=mhammert@a24-244-140-15.maxil.com) |
| 18:02.00 | mhammerton | hey i am new to asterisk and want to migrate a current voip system from an NT server to Debian with asterisk, does asterisk support Brookrout cards? |
| 18:04.29 | [TK]D-Fender | mhammerton: Not that I've ever heard of... |
| 18:04.51 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:04.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:05.33 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 18:05.52 | [TK]D-Fender | russellb: Hey, here's a question right up your alley : |
| 18:06.01 | [TK]D-Fender | mhammerton>hey i am new to asterisk and want to migrate a current voip system from an NT server to Debian with asterisk, does asterisk support Brookrout cards? |
| 18:06.18 | Dr-Linux | Feb 15 10:03:41 WARNING[25129]: chan_iax2.c:691 jb_warning_output: Resyncing the jb. last_delay -921, this delay -5456, threshold 2766, new offset 184 |
| 18:06.23 | Dr-Linux | why is this? :S |
| 18:07.02 | russellb | [TK]D-Fender: i have never touched them. |
| 18:07.17 | russellb | but, there is no official support built in ... |
| 18:07.42 | mhammerton | so you guys would advise that i purchase a digium card and use that to replace the brookrout one, cause it is kinda old |
| 18:08.11 | russellb | what kind of card is it? |
| 18:08.36 | mhammerton | its a Brookrout TR114-P4D DID card |
| 18:09.18 | anthony] | any looking for a dedicated server to host their pbxs on? |
| 18:09.24 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:10.14 | Qwell[] | anthony]: please don't advertise in here |
| 18:10.27 | Qwell[] | not unless somebody specifically asks |
| 18:11.36 | *** join/#asterisk evisu (i=hIRC@bzq-88-154-4-53.red.bezeqint.net) |
| 18:12.00 | mhammerton | @russellb: currently what the Brookrout card does is it has 4 analog DID lines, it supports DID inbound calling, call hunting, and it need to support 100 DID numbers for voicemail |
| 18:12.30 | russellb | um, asterisk would handle all of those features ... you just need a digium analog card |
| 18:14.31 | mhammerton | ohh ok digium analog card thanks, i want to read up more about the digium cards so that i get the correct analog card they have all the info on hte site? Also does asterisk support any other brand of analog cards besides digium? |
| 18:15.21 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:15.23 | russellb | not officially, but yeah |
| 18:16.08 | russellb | but you want to support the people that actually make asterisk, don't you? :) |
| 18:16.46 | mhammerton | yea keep the community going |
| 18:18.34 | [TK]D-Fender | sweeper: *crickets* |
| 18:19.25 | russellb | ~lart [TK]D-Fender |
| 18:19.32 | russellb | :-p |
| 18:19.41 | [TK]D-Fender | :O |
| 18:19.50 | russellb | that was not one of my favorite lartings. |
| 18:19.59 | [TK]D-Fender | russellb: So am I your random "lart" victim of the day? :) |
| 18:20.04 | russellb | yup |
| 18:20.20 | russellb | let's try that again. |
| 18:20.23 | russellb | ~thwack [TK]D-Fender |
| 18:20.26 | jbot | ACTION bludgeons [TK]D-Fender on the toe with a Holy Bible |
| 18:20.35 | russellb | that's a bit cleaner |
| 18:20.38 | nas_lslsa | I put thiese lines under [default] in extensions.conf : exten => 117,1,Dial(SIP/snom) exten => 115,1,Dial(SIP/konnosp) , I did CLI> module reload , and I still can't call 115 or 117 from my SIP phones :( I am connected right with asterisk at the moment .. |
| 18:21.01 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 18:21.08 | russellb | nas_lslsa: make sure your sip phones are configured with context=default |
| 18:21.09 | nas_lslsa | also I tried to reconnect to asterisk the sip phones .. nothing :( |
| 18:21.33 | [TK]D-Fender | nas_lslsa: And what proves to us that that context is actually being USED for anything? |
| 18:22.31 | nas_lslsa | [TK]D-Fender: I didn't understand you .. you mean that it proves that SIP phones are connected ? |
| 18:22.40 | sourceset | Is it possible to have multiple application set in the call file ? |
| 18:23.01 | nas_lslsa | WORKED ! ! ! ! ! |
| 18:23.40 | [TK]D-Fender | nas_lslsa: No, what proves that they are supposed to be using that context in tthe first place. |
| 18:24.21 | nas_lslsa | context=default ! ! ! it was context=from-sip .. |
| 18:24.34 | anthony] | Qwell, sorry |
| 18:25.02 | nas_lslsa | don't know .. |
| 18:26.17 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:26.43 | [TK]D-Fender | nas_lslsa: "Morning, gentlemen. This is your wakeup call." - John Sheridan |
| 18:27.57 | Nugget | Trust Ivanova, Trust yourself. Anyone else? Shoot 'em. |
| 18:28.41 | [TK]D-Fender | Nugget: Fitting near-death words :) |
| 18:28.58 | [TK]D-Fender | Nugget: She was expected to ACTUALLY get killed off in that episode |
| 18:30.38 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:31.38 | *** join/#asterisk gork (n=gork@gw.amarillo.energynet.com) |
| 18:32.01 | [TK]D-Fender | sourceset: Instead of calling a specific application, have it dial a Local channel and do whatever you want in there. |
| 18:35.53 | *** join/#asterisk IPmonger (n=ipmonger@68-114-72-181.dhcp.mant.nc.charter.com) |
| 18:36.14 | *** part/#asterisk IPmonger (n=ipmonger@68-114-72-181.dhcp.mant.nc.charter.com) |
| 18:37.03 | Nugget | Totally unrelated to anything, but the BOFH-mobile has been overhauled. http://macnugget.org/carrera/ |
| 18:37.23 | Nugget | I'm not allowed to swap the plates yet, though |
| 18:37.26 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 18:39.09 | edison\ | anyone know if there's something I could have screwed up in my dialplan to make CALLED exts continue to ring after the CALLING ext has hung up? |
| 18:39.35 | thekidrio | oh man oh man |
| 18:39.35 | *** join/#asterisk cantspeakslowly (n=chatzill@24.219.76.135) |
| 18:39.41 | thekidrio | how in the heck did i miss that |
| 18:39.44 | edison\ | I'm trying to determine if this is a bug in 1.4. |
| 18:39.48 | thekidrio | digium cards require pci 2.2 |
| 18:39.54 | thekidrio | and i have pci 2.0 |
| 18:39.57 | Nugget | d'oh! |
| 18:40.01 | thekidrio | almost 3 days of attempted debugging |
| 18:40.23 | endre | thekidrio: what mainboard is that? |
| 18:40.30 | thekidrio | tiny little letters catch my eye as i pull card out to return it |
| 18:40.44 | thekidrio | its a dell poweredge 2650 |
| 18:40.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:41.00 | Qwell[] | 2650 only has pci 2.0? |
| 18:41.08 | thekidrio | this one seems too |
| 18:41.36 | thekidrio | and the error i have been getting points in that direction |
| 18:41.55 | thekidrio | i just did not think it was possible |
| 18:42.04 | thekidrio | 2.0 had such a small window |
| 18:42.24 | *** part/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.co.comcast.net) |
| 18:43.00 | thekidrio | i am pulling it from the rack now to get a closer look |
| 18:43.08 | thekidrio | brrr so cold in server rooms haha |
| 18:43.22 | *** join/#asterisk Sloom (i=Sloom@cpc2-bagu5-0-0-cust981.bagu.cable.ntl.com) |
| 18:43.45 | *** part/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net) |
| 18:46.07 | nas_lslsa | people, goodnight ! ( here is 8:45 PM ) Thanks all for your help! , bye |
| 18:47.15 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 18:47.57 | drako | can't believe it... |
| 18:48.23 | drako | it keeps hanging the whole OS |
| 18:50.19 | *** join/#asterisk flip2 (i=azbmw@azeuros.org) |
| 18:50.20 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:51.13 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:51.45 | *** join/#asterisk qufk (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:53.19 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 18:55.54 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:55.57 | diclophis-work | hello all |
| 18:56.10 | diclophis-work | is it possible to do a "batphone" style setup with SIP phones? |
| 18:56.34 | cpm | wtf is a batphone? |
| 18:56.51 | diclophis-work | where you pickup the line and it immediatly rings another phone |
| 18:56.54 | mercestes | you pick up the phone and it immediately calls someone, just like the red phone in batman |
| 18:56.56 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 18:57.04 | mercestes | diclophis-work: I've heard of it. search asterisk batphone |
| 18:57.26 | diclophis-work | yea i have only found stuff related to zapata.conf |
| 18:57.38 | mercestes | http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
| 18:57.40 | mercestes | there ya go |
| 18:57.42 | cpm | should be doable |
| 18:57.54 | mercestes | diclophis-work: oh. SIP won't know when you pick up the phone. |
| 18:58.09 | diclophis-work | damn |
| 18:58.14 | mercestes | diclophis-work: * just has no clue that you have picked up the handset and are preparing to make a call so it has no way of doing "batphone" mode. |
| 18:58.41 | diclophis-work | i guess maybe if my sip phone had an auto-dial upon lifting of the handset i could do it |
| 18:59.58 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 19:00.57 | mercestes | diclophis-work: That's how it would have to work were it to owk. |
| 19:01.03 | mercestes | s/owk/work/ |
| 19:01.26 | Lloydie-t | ManxPower: I have made some changes to sip.conf based on what I have found, but I still get One way speech. The incoimg call call hear the outgoing but not vice versa |
| 19:01.36 | diclophis-work | damn |
| 19:02.14 | Lloydie-t | If I use the asterisk # feature the call cuts off |
| 19:03.03 | *** join/#asterisk DougSmith (n=dsmith@12.43.48.194) |
| 19:03.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 19:03.17 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 19:03.58 | GaVak | Has anyone run into the problem where ZAP to ZAP bridges somtimes make a terrible noise for about 5-10 seconds during the initial bridging? |
| 19:04.00 | [TK]D-Fender | mercestes: Didn't the Bat-Phone have a single large red button in the middle you had to press? :) |
| 19:04.47 | mercestes | [TK]D-Fender: No, that was a light that blinkied. You just had to pick up the handset to bring in the men in tights. |
| 19:04.50 | mercestes | well.... |
| 19:05.02 | mercestes | batman had tights.....robin on the other hand just had....*shudders* ew |
| 19:05.15 | mercestes | I was happy to have forgotten those days |
| 19:06.45 | Strom_C | we're men / we're men in tights |
| 19:06.51 | [TK]D-Fender | diclophis-work: For your question about "dial on handset pickup", it depends on the phone. |
| 19:07.55 | diclophis-work | yea, my phone doesnt support it |
| 19:08.01 | diclophis-work | that would have been cool though |
| 19:08.05 | *** join/#asterisk qufk (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 19:08.17 | diclophis-work | anyone experience a "click" when dealing with SIP phones... |
| 19:08.22 | diclophis-work | that doesnt make sense to me at all |
| 19:10.08 | drako | how can I drop echo from a TDM400 without use echocancel=yes ? |
| 19:10.12 | drako | any other tricky way> |
| 19:10.13 | drako | ? |
| 19:11.08 | *** join/#asterisk ManxPower (n=manxpowe@140.sub-70-196-120.myvzw.com) |
| 19:11.12 | mercestes | drako: Adjsut tx/rx values. |
| 19:11.38 | drako | mercestes, how ? |
| 19:11.59 | drako | i mean with which values should i try |
| 19:12.09 | mercestes | drako: 97/97 |
| 19:12.22 | drako | mercestes, i can't use echocancel=yes cuz it freez the whole OS |
| 19:12.33 | mercestes | drako: Sounds like you have a different issue then. |
| 19:13.24 | drako | rxgain=97 |
| 19:13.25 | drako | txgain=97 |
| 19:13.26 | drako | like that? |
| 19:13.28 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:13.32 | PakiPenguin | hello everyone |
| 19:13.34 | [TK]D-Fender | drako: that is just PSYCHO |
| 19:13.35 | mercestes | yea, exactly. |
| 19:13.45 | mercestes | >.> |
| 19:13.51 | Cresl1n | drako: use fxotune |
| 19:13.57 | Cresl1n | why? |
| 19:14.03 | mercestes | lol. Damnit. That would have been amusing results. |
| 19:14.14 | drako | i don't understand. |
| 19:14.32 | Cresl1n | use fxotune to reduce echo on your tdm card |
| 19:14.36 | mercestes | drako: You should adjust by +1 at a time, maybe to a value of +3 or -3 on tx. No where near 97. |
| 19:14.39 | Cresl1n | do you have fxo modules on it? |
| 19:14.42 | mercestes | drako: And run fxotune |
| 19:14.47 | Cresl1n | do that first |
| 19:14.51 | Cresl1n | before anything else |
| 19:14.59 | Cresl1n | don't start messing around with gains until you run fxotune |
| 19:15.21 | ManxPower | drako: If your OS freezes, notihng will work until you fix that issue |
| 19:15.23 | Cresl1n | and messing around with gains isn't going to do a whole lot for you unless you are planning on setting echocancel=yes |
| 19:15.28 | mercestes | rxgain +97 will send a pressurized sonic wave into your brain and kill you immediately....and fry your card. |
| 19:15.44 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:15.45 | ManxPower | Cresl1n: drako's machine is hard locking when he runs Asterisk |
| 19:15.58 | Cresl1n | ..... |
| 19:16.00 | diclophis-work | how would i rig up a queue, so that it dials X Y Z in order everytime... but skips over an interface if its in use? |
| 19:16.04 | drako | not when running |
| 19:16.09 | Cresl1n | that's pretty opaque |
| 19:16.10 | drako | when calling throught the pstn |
| 19:16.14 | diclophis-work | i tried roundrobin with penalties, but it never gets to the second interface |
| 19:16.30 | ManxPower | drako: Asterisk should never ever ever hard lock your machine. |
| 19:16.34 | drako | only when echocancel=yes is on zaptel.conf |
| 19:16.40 | drako | well it does. |
| 19:17.02 | mercestes | drako: What os? |
| 19:17.04 | diclophis-work | drako: have you checked your memory using something like memtest86 ? |
| 19:17.09 | Cresl1n | drako: that's weird |
| 19:17.10 | drako | mercestes, Debian/Etch |
| 19:17.27 | Cresl1n | what version of zaptel/asterisk are you using? |
| 19:17.28 | Lloydie-t | can anyone advise me how to use the asterisk transfer featue |
| 19:17.53 | drako | creativx, zaptel 1.2.12 and asterisk 1.2.14 |
| 19:18.46 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.10) |
| 19:18.57 | drako | creativx, fxotune says i don't have fxotune.conf |
| 19:20.45 | cpm | or turn your brain into a virtual clone of betty ford |
| 19:21.04 | Cresl1n | use zaptel-1.4's fxotune |
| 19:21.13 | Cresl1n | it's a lot better than 1.2's |
| 19:21.26 | Qwell[] | Cresl1n: stupid question, but 1.4 fxotune works with 1.2 zaptel, right? |
| 19:21.40 | Cresl1n | yes |
| 19:21.59 | Cresl1n | just do the line calibration with the 1.4 version of fxotune because it's much better |
| 19:22.09 | Cresl1n | (fxotune -i, I think) |
| 19:22.26 | Cresl1n | and then you can use 1.2 fxotune to set the line (fxotune -s) |
| 19:22.35 | Qwell[] | heh |
| 19:22.52 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 19:23.06 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 19:23.27 | Lloydie-t | Is there a problem with blind transfer(#)? I just get cut off after I enter a number on a sip phone |
| 19:23.57 | Qwell[] | You're supposed to get "cut off" after a blind transfer |
| 19:24.01 | drako | pbx:/etc/asterisk# fxotune -s |
| 19:24.03 | drako | Cannot open /etc/fxotune.conf! |
| 19:24.23 | Lloydie-t | But the call is not transferred |
| 19:24.29 | Qwell[] | oh |
| 19:24.40 | Qwell[] | Why are you using a # transfer on a SIP phone? |
| 19:24.46 | Cresl1n | drako that's because you haven't done an fxotune -i yet |
| 19:24.54 | Cresl1n | read README.fxotune |
| 19:25.10 | drako | ayah bit it was for some values |
| 19:25.27 | tzanger | [TK]D-Fender: there's *got* to be a better way to use the ip501/601 forward feature than putting in "SIP/extension#" |
| 19:25.29 | drako | how i'd know which values should I use... |
| 19:25.38 | Lloydie-t | because when I try to transfer to an external sip number I get One way speech |
| 19:25.57 | ManxPower | tzanger: Huh? You press Forward, put in the number, press enable |
| 19:26.00 | Qwell[] | Lloydie-t: that's a NAT issue |
| 19:26.14 | tzanger | ManxPower: for me, I put in "202" and press enable, and it says "try again" |
| 19:26.15 | ManxPower | Qwell: I told him that hours ago, he didn't listen. |
| 19:26.20 | drako | pbx:/etc/asterisk# fxotune -s |
| 19:26.20 | drako | fxotune: successfully set echo coeffecients on FXO modules |
| 19:26.21 | drako | ok |
| 19:26.30 | ManxPower | tzanger: then you have some other issue |
| 19:26.35 | tzanger | ManxPower: heh |
| 19:26.39 | Cresl1n | drako: don't use zaptel-1.2's fxotune |
| 19:26.43 | Cresl1n | use zaptel-1.4's |
| 19:26.48 | Lloydie-t | Yes I did listen an I am still trying to find a resolution |
| 19:27.15 | [TK]D-Fender | tzanger: HUH?! |
| 19:27.28 | ManxPower | Lloydie-t: you will get one-way audio until you fix the NAT issue. |
| 19:27.55 | ManxPower | tzanger: you can dial 202 from the phone? |
| 19:28.00 | tzanger | ManxPower: yep |
| 19:28.17 | ManxPower | tzanger: I've never seen that issue |
| 19:28.29 | drako | Cresl1n, is a separated packages? because this fxotune is not working at all |
| 19:29.05 | ManxPower | drako: yes, zaptel version 1.4 is different from zaptel 1.2 |
| 19:29.19 | Lloydie-t | OK I have tried various solutions on voip-info.org except using a stun server. Will using a stun server make a difference? |
| 19:29.29 | ManxPower | Lloydie-t: I doubt it. |
| 19:30.00 | ManxPower | Lloydie-t: diagram your setup like: SIP Phone <-> Asterisk <-> NAT Router <-> Internet <-> SIP provider |
| 19:30.06 | ManxPower | or whatever your setup is |
| 19:30.14 | [TK]D-Fender | Lloydie-t: pastebin the [general] section of your sip.conf. |
| 19:30.16 | [TK]D-Fender | ~pb |
| 19:30.31 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:30.43 | [TK]D-Fender | Lloydie-t: I'm betting your scenarion looks exactly like ManxPower sampled for you |
| 19:30.55 | Lloydie-t | Yes it is |
| 19:31.19 | tzanger | aha |
| 19:31.25 | tzanger | asterisk segfaults |
| 19:31.28 | tzanger | 3 IP501s |
| 19:31.37 | tzanger | identical config except for extension/reg info |
| 19:31.43 | ManxPower | Lloydie-t: what are your externip= and localnet= set to in sip.conf? |
| 19:31.45 | tzanger | call 201 from 203 -- ok |
| 19:31.48 | Lloydie-t | my sip.conf is at http://pastebin.ca/357537 |
| 19:31.55 | tzanger | forward 203 to 202 on the phone itself |
| 19:31.59 | tzanger | er sorry |
| 19:32.02 | tzanger | forward 201 to 202 on the phone |
| 19:32.29 | tzanger | call 201 from 203... asterisk sees "moved temporarily" and then tries Local/203@internal and segfaults |
| 19:32.33 | Lloydie-t | externip=81.149.42.129 localnet=192.168.7.0/255.255.255.0 |
| 19:32.35 | diclophis-work | ringinuse, is useless |
| 19:32.39 | [TK]D-Fender | Lloydie-t: You need "canreinvite=no" in [general]. Not having it will kill your calls |
| 19:32.44 | ManxPower | Lloydie-t: it is disallow=all not disable=all Also, only allow ONE codec. |
| 19:32.59 | ManxPower | allow=ulaw would be a good one |
| 19:33.08 | ManxPower | diclophis-work: why? |
| 19:33.14 | diclophis-work | it doesnt work |
| 19:33.38 | diclophis-work | i have it set to ringinuse=no, but it still dials the interface |
| 19:33.42 | diclophis-work | and its a SIP interface no less |
| 19:33.43 | ManxPower | tzanger: I did say you had "other issues" |
| 19:33.52 | tzanger | ManxPower: :-) yes, yes you did |
| 19:33.59 | ManxPower | diclophis-work: does your phone have call waiting enabled? If so, then the interface is not in use |
| 19:34.03 | tzanger | why would that crash out though |
| 19:34.13 | diclophis-work | hmm call waiting you say.. |
| 19:35.11 | diclophis-work | is it possible to disable call-wating with sip.conf? |
| 19:35.49 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 19:36.06 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 19:37.08 | ManxPower | tzanger: From what I hear 1.4 crashes all the time. *ducks* |
| 19:37.17 | tzanger | ManxPower: hehe |
| 19:40.24 | ManxPower | tzanger: you know my stance on 1.4, right? |
| 19:40.24 | diclophis-work | so.. how would i setup a queue so that it dials X Y Z in that order everytime ? |
| 19:40.24 | tzanger | ManxPower: yes |
| 19:40.35 | mercestes | ManxPower: no..No! tell us... |
| 19:40.41 | mercestes | well, tell me..I wanna know. |
| 19:41.22 | Kritter | anyone know how big the AsteriskNOW initrd is? |
| 19:41.30 | ManxPower | mercestes: Until Digium is confident enough in 1.4.x (not CVS/SVN) to run it on IAXTel AND the Digium corporate PBX, I won't run 1.4 |
| 19:41.40 | Qwell[] | ManxPower: iaxtel runs trunk |
| 19:41.46 | Qwell[] | just fyi |
| 19:42.02 | ManxPower | Qwell: exactly. that tells me that digium thinks 1.4 won't work for that. |
| 19:42.19 | Qwell[] | we use iaxtel for testing trunk |
| 19:42.31 | ManxPower | Qwell: and their corporate PBX. |
| 19:42.47 | ManxPower | Qwell: Fair enough, IAXTel isn't really a production system. |
| 19:43.20 | Lloydie-t | still no joy transfer calls. Get following warning. handle_response: Notify answer on an owned channel? |
| 19:43.43 | ManxPower | Lloydie-t: you are not doing portforwards on your NAT router, correct? |
| 19:44.52 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:45.18 | ManxPower | Every single time Asterisk fails on a production server the other department heads try to convince management that IT is incompetent and should be put under their department. |
| 19:45.24 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 19:46.19 | Lloydie-t | I have set min and max ports in rtp.conf to 8000-8050 and have forwarded those ports on the firewall. I will check again |
| 19:46.55 | ManxPower | Lloydie-t: Your SIP device should not have any NAT stuff enabled. |
| 19:47.25 | cpm | ManxPower, yeah, what is that about anyway? I mean who would *want* IT ? |
| 19:47.58 | ManxPower | cpm: it has the largest budget in the company |
| 19:48.13 | *** join/#asterisk weazahl (n=weazahl@adsl-69-152-175-185.dsl.ksc2mo.swbell.net) |
| 19:48.15 | cpm | it does? |
| 19:48.16 | cpm | cool! |
| 19:48.18 | ManxPower | You can almost see the drool on the lips of the other department heads |
| 19:48.19 | cpm | no wonder. |
| 19:48.26 | ManxPower | cpm: most of the money is for the WAN |
| 19:48.31 | cpm | yeah, |
| 19:48.35 | ManxPower | and for the hardware users require |
| 19:48.36 | cpm | there is that |
| 19:48.51 | cpm | Lett'em have it, go work elsewhere |
| 19:49.18 | sweeper | channel poll: good sip softphone that plays nice with dtmf, runs on windoze |
| 19:49.25 | ManxPower | cpm: the money certianlly does not go to personael. The company has 16 offices in 2 states, 400 end users and 1 helpdesk person, 1 it manager, 1 telecom/network guy, and a consultant (me) |
| 19:49.26 | cpm | I'm really no longer interested in vainly attempting to slow down deliberate self-destruction |
| 19:49.33 | weazahl | could someone point me in the right direction on how to make a redundant asterisk server, 2 servers exactly the same with failover. does this need to be done at OS level or can asterisk do it? |
| 19:49.47 | ManxPower | cpm: I've been consulting for them for over 10 years |
| 19:50.05 | ManxPower | All Softphones Suck!(tm) |
| 19:50.09 | cpm | So, yer not a salaryman? |
| 19:50.14 | sweeper | oh, and free plz |
| 19:50.33 | ManxPower | cpm: hell no. I would not work for that den of politics and stuipidity |
| 19:50.38 | weazahl | ManxPower: Idefisk isnt too bad if you like IAX2 |
| 19:50.50 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:51.12 | cpm | a softphone is exactly that. it's a piece of software that 'act's like' a phone |
| 19:51.21 | cpm | without the apostrophe even |
| 19:51.28 | ManxPower | cpm: as a consultant I can say "I'm not going to do that project, find a different consultant for that." Most of the time they don't do the project, when they actually do the project it usually fails terribly (as I knew it would) |
| 19:51.34 | ManxPower | If I thought the project would work I would have accepted it. |
| 19:51.49 | cpm | nice |
| 19:52.02 | cpm | someday perhaps. |
| 19:52.02 | ManxPower | cpm: except phones don't depend on PC hardware and general application OS |
| 19:52.25 | mercestes | ...but....it doesn't act like a phone. it more acts like...a lazy, stuck up, arrogant actor whose part is a phone, but he doesn't want to play it because he's a whiney bitch and he hasn't been getting his way so he randomly goes on strike at the most inopportune moments and refuses to "be the phone" unless you bow to him and cater to his self-centered egotism. |
| 19:52.25 | cpm | yeah, I kinda hoped that was implied in my description |
| 19:52.37 | cpm | heh |
| 19:52.50 | foobar778 | Hi how are u all I have a couple of questions is that ok |
| 19:53.00 | cpm | that's a pretty good analogy, thanks for sharing |
| 19:53.52 | *** join/#asterisk xx8xx (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 19:53.55 | cpm | foobar778, it's certainly okay to have questions, just whatever you do, don't actually ask them, only ask to ask |
| 19:54.07 | xx8xx | guys when I dial from one exten to another exten I get this warning |
| 19:54.07 | putzz | heh |
| 19:54.08 | xx8xx | WARNING[3481]: channel.c:2910 ast_request: No channel type registered for 'IAX' |
| 19:54.10 | Lloydie-t | I have no NAT set up on my Mitel Phones. Should I change my rtp.conf settings |
| 19:54.26 | foobar778 | I have asterisk setup and made one user account as a peer to a sip service Just having it listed can the other users use the service the service also has a dialplan |
| 19:54.30 | Qwell[] | xx8xx: upgrade |
| 19:54.32 | kuku5 | How can I check the reason why the asterisk process takes 100% cpu the moment I start it ? |
| 19:54.36 | xx8xx | the phones do ring and I can pick up call but that warning in cli comes up |
| 19:54.41 | xx8xx | upgrade ? |
| 19:54.45 | Qwell[] | xx8xx: upgrade |
| 19:54.49 | xx8xx | upgrade what |
| 19:54.53 | xx8xx | asterisk ? |
| 19:54.55 | ManxPower | Lloydie-t: rtp.conf ports should match the ports your are forwarding outside -> inside on your router |
| 19:54.56 | [TK]D-Fender | xx8xx: Thats because it should be "Dial(IAX2/something) |
| 19:54.56 | putzz | omg |
| 19:54.57 | Qwell[] | everything |
| 19:54.57 | mercestes | xx8xx: to IAX2 probably. |
| 19:55.25 | tzafrir_laptop | kuku5, yiu can try to see (forgot the exact ps argument) what thread exactly it is, and strace it |
| 19:55.27 | xx8xx | we are no using iax |
| 19:55.32 | tzafrir_laptop | (strace -p) |
| 19:55.33 | xx8xx | so why that iax msg comes |
| 19:55.53 | *** join/#asterisk loca|host (n=fourat@196.203.53.221) |
| 19:56.02 | ManxPower | xx8xx: because asterisk no longer supports the IAX protocol. You need to use the IAX2 protocol |
| 19:56.43 | xx8xx | but I have the latest asteriskNOW beta4 |
| 19:56.54 | foobar778 | Qwell: can u call my PBX so I can ask u some quick questions |
| 19:57.15 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 19:57.22 | foobar778 | <ManxPower: or you too please |
| 19:57.40 | ManxPower | xx8xx: correct. IAX (version 1) has been removed from Asterisk. You need to specify the protocol as IAX2 |
| 19:57.42 | Lloydie-t | OK I have got blind transfer working on my sip phone, but attended transfer still gives one way speech |
| 19:58.17 | xx8xx | I'm using asteriskNOW so where do I specify that ? |
| 19:59.30 | Kritter | tried the non-basic config screens? |
| 19:59.43 | Kritter | or editing the text files by hand? |
| 19:59.54 | [TK]D-Fender | xx8xx: IAX has not existed for YEARS. You were ALWAYS supposed to be using IAX2 |
| 20:00.39 | xx8xx | I didn't set that up... its the default settings of asteriskNOW |
| 20:00.50 | xx8xx | u make it sound like I did that lol |
| 20:01.13 | tzanger | what is "simple" CDR mode? I've noloaded cdr_csv so that's definitely not it |
| 20:01.18 | tzanger | (using ODBC cdr) |
| 20:01.56 | ManxPower | foobar778: If you want personal support then I require a credit card first. |
| 20:02.45 | ManxPower | xx8xx: Wherever you specify "IAX" specify "IAX2" instead. Like in Dial lines, etc. But since nobody here runs AsteriskNOW (there is a dedicated channel for that) I can't tell you where or how |
| 20:03.14 | ManxPower | xx8xx: ask on #asterisknow |
| 20:03.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:04.06 | drako | You guys think is a malfunctional of the TDM Card that make the system crash so bad? |
| 20:04.59 | *** part/#asterisk xx8xx (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 20:05.08 | Cresl1n | drako: call tech support about it |
| 20:05.26 | unik-rados | did you guys try interconnecting Voxeo Prophecy with Asterisk? |
| 20:08.58 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 20:12.32 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:14.39 | PupenoR | Can I get manager events through AJAM? |
| 20:15.18 | drako | creativx, fxotune -i 4 |
| 20:15.23 | drako | did a good job |
| 20:15.30 | drako | it tooks like 45min |
| 20:15.42 | drako | but is not that good yet |
| 20:15.46 | drako | i mean i can work for a while |
| 20:16.04 | drako | but adjusting by hands the /etc/fxotune will help? |
| 20:17.50 | tzafrir_laptop | drako, I would also recommend you to grab fxotune.c from zaptel 1.4 |
| 20:18.01 | tzafrir_laptop | I consider doing so in the debs |
| 20:18.04 | drako | just the .c ? |
| 20:18.10 | drako | it compiles as easy as that? |
| 20:18.14 | tzafrir_laptop | yes |
| 20:18.14 | elriah | Anyone have success with Cisco 79x1 phones, firmware 8.4SR1 and NAT? (Or any firmware for that matter) |
| 20:18.23 | drako | is only one file? |
| 20:18.59 | *** join/#asterisk Skarmeth (n=Skarmeth@201009078081.user.veloxzone.com.br) |
| 20:19.43 | elriah | I have a TDM400P for sale, 4 FXO ports, $225.00+shipping, guaranteed against DOA, barely used. |
| 20:20.05 | Skarmeth | hi all |
| 20:21.32 | Skarmeth | I am trying to code a simple AGI script and I am getting in troubles to read a variable, I sent GET VARIABLE VARNAME and GET VARIABLE ${VARNAME} to asterisk and it returns me a syntax error... what's the right way to ask asterisk for a variable? |
| 20:22.11 | *** join/#asterisk dasenjo (n=dasenjo@190.24.179.134) |
| 20:24.15 | loca|host | can i use gizmo project as an asterisk only softphone ? |
| 20:25.54 | *** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 20:26.07 | AlfaScorpii | Hi |
| 20:26.50 | tzafrir_laptop | loca|host, I'm not sure you can set it to a custom SIP registration server. Can you? |
| 20:27.31 | loca|host | tzafrir_laptop, am checking it |
| 20:27.35 | AlfaScorpii | need some help with my Asterisk and a voip gateway micronet sp5050 |
| 20:27.41 | AlfaScorpii | anybody? |
| 20:27.43 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 20:27.48 | tzafrir_laptop | ~anybody |
| 20:27.57 | tzafrir_laptop | ~seen anybody |
| 20:28.28 | jbot | i haven't seen 'anybody', tzafrir_laptop |
| 20:28.28 | tzafrir_laptop | not here |
| 20:28.28 | tzafrir_laptop | AlfaScorpii, just ask ;-) |
| 20:28.28 | AlfaScorpii | tzafrir_laptop: ok |
| 20:28.40 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 20:28.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:29.00 | AlfaScorpii | i have a problem with my ouside calls |
| 20:29.32 | cpm | yeah, those outside calls will getcha |
| 20:29.37 | AlfaScorpii | 40 seconds and the call is death |
| 20:29.46 | mercestes | how morbid |
| 20:29.58 | cpm | indeed |
| 20:30.06 | cpm | remind me not to call |
| 20:30.14 | AlfaScorpii | i think that may be the codec... |
| 20:30.40 | AlfaScorpii | im using a micronet sp5050 as interface fxo to pstn |
| 20:30.55 | [TK]D-Fender | cpm: You have to play a recording of the call to someone else FAST or you'll die.... |
| 20:30.56 | cpm | how does the call 'sound' before it dies? |
| 20:31.10 | AlfaScorpii | cpm: ? |
| 20:31.36 | cpm | have you been on a call that died? |
| 20:31.41 | cpm | or 'is death' ? |
| 20:31.44 | mercestes | AlfaScorpii: I blame the micronet sp5050 |
| 20:32.14 | mercestes | cpm: Yea, he played it to someone else within 40 seconds so he didn't die. It's just like a chain letter. ...but with a telephone. |
| 20:32.16 | AlfaScorpii | mercestes: whats blame? sorry my english is very bad |
| 20:32.28 | cpm | ahh |
| 20:32.33 | mercestes | AlfaScorpii: really? heh, sounded like any other American to me. |
| 20:32.43 | AlfaScorpii | lol |
| 20:32.52 | mercestes | AlfaScorpii: 'course, 98% of Americans have detestable english so don't pat yourself too hard. |
| 20:33.15 | mercestes | most would have to google 'detestable' just to figure out of they should say "thank you" or "f-you." |
| 20:33.21 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:33.35 | mercestes | AlfaScorpii: Anyways. I think the Micronet sp5050 is "killing your call" for you. |
| 20:34.05 | elriah | lol, yea but we have 60% of the world's wealth so "thank you" and "f-you". |
| 20:34.06 | AlfaScorpii | mercestes: why? i meen.. codec problem? |
| 20:34.14 | PupenoR | Can I get manager events through AJAM? |
| 20:34.23 | mercestes | AlfaScorpii: What codec are you using? |
| 20:35.05 | Lloydie-t | still getting oneway speech. Is this relevant to the error I get http://bugs.digium.com/view.php?id=8696 |
| 20:35.33 | AlfaScorpii | mercestes: in my asterisk i really dont know, i n the micronet gateway first is g.729 |
| 20:35.59 | AlfaScorpii | how could i know that codec is using asterisk? |
| 20:36.28 | Hmmhesays | well show translation should show you available codecs |
| 20:37.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:37.23 | [TK]D-Fender | asjkldhlkasjhd |
| 20:37.28 | AlfaScorpii | Hmmhesays: show translation |
| 20:37.35 | AlfaScorpii | let me see |
| 20:38.17 | Hmmhesays | [TK]D-Fender |
| 20:38.19 | Hmmhesays | what up |
| 20:38.50 | [TK]D-Fender | Hmmhesays: The usual..... day job... consulting,.. music lessons .... this & that... |
| 20:38.56 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 20:39.10 | AlfaScorpii | mercestes: r u there? |
| 20:39.17 | Hmmhesays | [TK]D-Fender: sounds... monotonous |
| 20:39.22 | mercestes | AlfaScorpii: Barely. someone answered you tho |
| 20:39.38 | Hmmhesays | I saw papa roach live on sunday |
| 20:39.52 | Hmmhesays | Hell..of...a...show |
| 20:39.58 | AlfaScorpii | mercestes: ok tx |
| 20:40.40 | *** join/#asterisk X-Rob (n=rob-x@124.150.99.11) |
| 20:42.20 | AlfaScorpii | Hmmhesays: pastebin? |
| 20:42.51 | AlfaScorpii | http://pastebin.gulic.org/240 |
| 20:43.39 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 20:43.55 | [TK]D-Fender | Hmmhesays: No, my playing has definately improved a bit ;) |
| 20:44.33 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:45.27 | Hmmhesays | [TK]D-Fender: yeah mine has too, 1 hour on the metronome every day helps a lot |
| 20:45.47 | Hmmhesays | AlfaScorpii: that is a grid that shows what available codec translations you have |
| 20:45.49 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 20:46.00 | Hmmhesays | if you want to see what codec is being used in say a sip call... sip show channels |
| 20:46.09 | [TK]D-Fender | Hmmhesays: Oh I can't really muster anywhere near that much effort. I just goof off on it while chatting on-line, playing D2, etc... |
| 20:46.18 | AlfaScorpii | This is my Asterisk http://pastebin.gulic.org/240 AND this is my gateway http://img116.imageshack.us/my.php?image=micronet5050voicecfgye0.png |
| 20:46.32 | Hmmhesays | [TK]D-Fender: I have a lot of free time, and a lot of motivation |
| 20:46.49 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:47.05 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 20:47.07 | Hmmhesays | I usually fire up the metronome while watching something on tv too that helps |
| 20:48.17 | elriah | I envy you guys. My music skills are really bad. |
| 20:48.18 | *** join/#asterisk enjay1 (n=yea@209.181.130.105) |
| 20:49.12 | Hmmhesays | eh skills develop with practice |
| 20:50.03 | AlfaScorpii | Hmmhesays: how can i use g.729 on my Asterisk? |
| 20:50.26 | AlfaScorpii | i think the thats the problem, gateway with g.729 and asterisk not |
| 20:50.27 | enjay1 | buy the codec from digium.com |
| 20:50.51 | AlfaScorpii | not free?? |
| 20:50.52 | enjay1 | 10$/channel |
| 20:51.01 | ManxPower | AlfaScorpii: g729 is patented |
| 20:51.07 | AlfaScorpii | :( |
| 20:51.22 | AlfaScorpii | so i need to set other codec on my gateway? |
| 20:51.28 | *** join/#asterisk s1gny|wrk (n=s1gny@p549158CF.dip.t-dialin.net) |
| 20:51.29 | AlfaScorpii | what codec shoul i use? |
| 20:51.31 | gork | on a zaptel channel to a phone (fxs interface, using fxo_ks signalling) whenever the far end hangs up, zaptel generates fast busy audio on the line until the near end hangs up. is there a way to change this behavior to send no audio instead? |
| 20:51.38 | enjay1 | if bandwidth is a concern and you're not worried about quality go gsm |
| 20:51.44 | *** part/#asterisk s1gny (n=s1gny@p549158CF.dip.t-dialin.net) |
| 20:52.33 | AlfaScorpii | enjay1: y have a micronet sp5050 gateway just for fxo to pstn interface |
| 20:53.08 | enjay1 | just use ulaw then.. |
| 20:53.14 | AlfaScorpii | enjay1: soy do i have to set the same codec in gateway and asterisk? |
| 20:53.17 | enjay1 | g711 |
| 20:53.46 | enjay1 | yes they have to match.. in asterisk you will define it in the conf by "allow=ulaw" |
| 20:54.03 | enjay1 | on the gateway just make sure the first accepted codec is g711 or ulaw/alaw |
| 20:55.10 | AlfaScorpii | enjay1: g711mu-Law ? |
| 20:55.34 | AlfaScorpii | or g711A-Law? |
| 20:55.49 | enjay1 | sure :) |
| 20:55.59 | enjay1 | either or.. |
| 20:56.16 | enjay1 | just ensure in your asterisk config you allow the codec.. |
| 20:56.21 | *** part/#asterisk blitz[training] (n=blitzrag@66.135.99.122) |
| 20:56.28 | enjay1 | you can do that specific to the device or in the general context |
| 20:56.43 | *** join/#asterisk UltimateX (n=admin@pool-71-109-65-100.lsanca.dsl-w.verizon.net) |
| 20:56.44 | AlfaScorpii | enjay1: ok let mi set g711mu-Law first |
| 20:56.53 | UltimateX | o.O yay thats how i get identified.... ^_^ |
| 20:57.09 | elriah | You can also do in sip.conf disallow=all, allow=ulaw, allow=g729 and Asterisk will go down the list until it finds one that you are allowing that works with the peer. |
| 20:57.53 | UltimateX | Did anyone see the Hak.5 Podcast on Asterisk? It was very inlightening, thus how i came across this IRC Server/Channel :-) |
| 20:59.18 | enjay1 | just for the sake of troubleshooting I would stick to allowing on codec (alaw or ulaw) until you ahve it working.. |
| 20:59.47 | AlfaScorpii | enjay1: ok, now how can i alow g711mu-Law and g711A-Law in my asterisk? |
| 21:00.00 | *** join/#asterisk alephant (n=cmd@c-24-3-52-93.hsd1.pa.comcast.net) |
| 21:00.04 | alephant | Hi all... |
| 21:00.08 | alephant | ...asterisk -r; sip debug |
| 21:00.12 | [TK]D-Fender | AlfaScorpii: They just told you 3-4 times.... |
| 21:00.18 | alephant | makes stuff scroll by WAY too fast |
| 21:00.21 | AlfaScorpii | alephant: hi |
| 21:00.28 | enjay1 | hehe.. yea man I cant walk you through configuring the whole system, you need to read the docs on that... |
| 21:00.49 | enjay1 | voip-info.org is your friend.. |
| 21:00.52 | alephant | Is it possible to send the debug output to a file, or to start "asterisk -r" and redirect to stdout |
| 21:00.54 | alephant | ? |
| 21:01.13 | alephant | or is there a switch to throw to make SIP debugging go to syslog? |
| 21:01.13 | mercestes | alephant: asterisk -r > output.txt |
| 21:01.24 | enjay1 | hehe |
| 21:01.26 | alephant | I didn't try that :-/ |
| 21:01.38 | enjay1 | would that info show up in /var/log/asterisk/full? |
| 21:01.43 | AlfaScorpii | enjay1: ok tx |
| 21:01.47 | alephant | asterisk -r just didn't feel unixy enough to trigger that "hey what if" impulse |
| 21:03.19 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 21:03.27 | alephant | Ok, cool. Now you all get to help me parse the output of sip debug >:-) |
| 21:03.50 | mercestes | heh |
| 21:03.51 | mercestes | hehe |
| 21:04.16 | alephant | Well, first of all: anybody had any luck getting * to dial out through Voipstreet? |
| 21:04.21 | mercestes | you can even do an asterisk -r > output.txt && tail -f output.txt to read the outpu tas it's logged |
| 21:04.57 | dlynes_laptop | mercestes: you're mean |
| 21:05.11 | AlfaScorpii | EUREKA! |
| 21:05.18 | AlfaScorpii | enjay1: u r the man! |
| 21:05.47 | AlfaScorpii | no more cut at 40 seconds in my outband calls! |
| 21:05.48 | alephant | mercestes: tee also works :-b |
| 21:05.58 | AlfaScorpii | mercestes: yo to |
| 21:06.01 | [TK]D-Fender | dlynes_laptop: At least he's not off trying to trick people into killing their entire filesystem.... |
| 21:06.05 | AlfaScorpii | Thank u all people |
| 21:06.10 | AlfaScorpii | i love u all |
| 21:06.19 | AlfaScorpii | time to go home |
| 21:06.34 | AlfaScorpii | c ya in the morning |
| 21:06.39 | *** part/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 21:06.51 | dlynes_laptop | [TK]D-Fender: heh |
| 21:07.04 | dlynes_laptop | alephant: He's just being mean |
| 21:07.11 | dlynes_laptop | alephant: take a look at logger.conf |
| 21:07.50 | alephant | dlynes_laptop: I looked at logger.conf, and all loglevels are active |
| 21:08.03 | alephant | dlynes_laptop: what's wrong with the tail -f he suggested? |
| 21:08.47 | dlynes_laptop | tail -f /var/log/asterisk/full instead |
| 21:09.12 | dlynes_laptop | You've already got a log file being created |
| 21:09.22 | dlynes_laptop | No point creating another one using a command line app |
| 21:09.26 | *** join/#asterisk john-z (n=john@phrank.aus.us.siteprotect.com) |
| 21:09.33 | john-z | Hey everyone. |
| 21:09.39 | john-z | I have an odd problem, does someone have a moment. |
| 21:09.42 | enjay1 | alrighty then.. |
| 21:09.50 | john-z | Trying to get voicemail to send via email. |
| 21:09.59 | enjay1 | apparently he loves us haha |
| 21:10.01 | john-z | I get strange Data Format errors in my /var/log/messages |
| 21:10.04 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 21:10.23 | john-z | Feb 15 08:00:26 tele-comm sendmail[4641]: l1FE0QCe004639: to=<mf_texas@swbell.net>, ctladdr=<root@tele-comm> (0/0), delay=00:00:00, xdelay=00:00:00, mailer=esmtp, pri=126141, relay=sbcmail2.prodigy.net. [207.115.36.21], dsn=5.6.0, stat=Data format error |
| 21:10.25 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 21:10.35 | john-z | Im able to mail from the server, using the command line.. |
| 21:10.37 | murdmath | Howdy all. |
| 21:10.42 | john-z | But Asterisk mail function is not working. |
| 21:10.44 | alephant | dlynes_laptop: so the "full" log should contain the same as sip debug? |
| 21:11.04 | dlynes_laptop | alephant: if you have sip debug enabled, yes |
| 21:11.12 | enjay1 | john-z: is your hostname defined properly in /etc/hosts? |
| 21:11.27 | enjay1 | I've seen that happen before when the hostname doesnt match that in /etc/hosts and its trying to send to like localhost.localdomain |
| 21:11.29 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:11.32 | john-z | Its just localhosts. |
| 21:11.46 | Nugget | RHEL likes to screw up the hosts file that way. |
| 21:11.52 | murdmath | Is it normal for the caller id information to not be present after a call has been parked and then picked back up? |
| 21:11.55 | john-z | Its centos, but yea. |
| 21:12.27 | pythos | is there a "dialplans for idiots" doc somewhere? |
| 21:12.45 | enjay1 | yea probably on voip-info.org under extensions.conf |
| 21:12.46 | pythos | I just want to get one phone on a fxs to call another on a fxs |
| 21:13.04 | enjay1 | fxs gateway? |
| 21:13.11 | pythos | I got a tdm400p |
| 21:13.12 | Kritter | <PROTECTED> |
| 21:13.29 | enjay1 | okay.. thats the quad analog card right? |
| 21:13.47 | dlynes_laptop | enjay1: 1 to 4 ports, 4 ports max |
| 21:13.57 | pythos | its got 2 fxs and 2 fxo, and I DO have dialtone on the 2 fxs, and the 2 fxo answer calls, but give the generic stuff for starting using asterisk... |
| 21:14.11 | dlynes_laptop | enjay1: well, you could have 0 ports too, but that wouldn't be terribly useful |
| 21:14.15 | Qwell[] | pythos: You just need to give it a different context |
| 21:14.29 | [TK]D-Fender | pythos: ... |
| 21:14.31 | [TK]D-Fender | ~book |
| 21:14.34 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:14.50 | Qwell[] | Nugget: s/the.*$/everything/ |
| 21:15.43 | Nugget | fair point. |
| 21:16.03 | Qwell[] | s/likes.*$/is crap/ |
| 21:16.12 | Qwell[] | I'm happier with that expression |
| 21:16.48 | john-z | enjay1: I think you were right about the /etc/hosts |
| 21:16.49 | john-z | Thanks. |
| 21:17.03 | enjay1 | john-z: no problem.. |
| 21:17.18 | pythos | jbot, ya, that is how I got as far as getting the dialtone on fxs, and answering fxo's |
| 21:17.30 | jbot | pythos: I think you lost me on that one |
| 21:17.30 | enjay1 | lol |
| 21:17.42 | enjay1 | hence bot |
| 21:17.54 | pythos | ahh... derp |
| 21:18.48 | pythos | so, I need to give a context to fxs port 1 and 2. I take it I do this in extensions.com |
| 21:18.51 | pythos | f |
| 21:20.12 | [TK]D-Fender | pythos: You GIVE it in zapata.conf , you IMPLEMENT it in extensions.conf |
| 21:20.17 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:20.53 | [TK]D-Fender | pythos: Go download THE BOOK and get reading. This is not a really ideal phase to be hand-held though. |
| 21:21.41 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 21:22.32 | pythos | heheh, I did d/l and have thus far gotten both fxs lines to the point of having dialtone. Also, I got the fxo ports to answer calls when I hook POTS into them and ring them up. |
| 21:23.17 | pythos | [TK]D-Fender: but thanks for the suggestions :-) |
| 21:24.22 | [TK]D-Fender | pythos: Not trying to come down on you, but this is "founndation" stuff that you should really get a GOOD overall understanding of before just plugging in everything people tell you to. |
| 21:24.38 | pythos | I do agree |
| 21:25.09 | [TK]D-Fender | pythos: Just because someones answer works, don't mean is the best or right way, or that you'll necessarily even deduce the "why" and "how" out of it. |
| 21:25.24 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 21:26.23 | gork | anyone, is there a way to make a zaptel channel not play a busy tone when the far end hangs up? |
| 21:26.24 | ChicagoBud | hi everyone, generic question. Do you require users to dial "9" for an outside line? |
| 21:26.33 | Qwell[] | ChicagoBud: sure |
| 21:26.46 | Qwell[] | unless you don't allow them to call each other by extension |
| 21:27.54 | gork | the only alternative is to have a prefix for calling local extensions; for instance my fax machines by default call "out" directly, however dialing 555-<extension> will connnect directly to a local extension |
| 21:28.01 | alephant | Ok, so I'm back where I started yesterday: |
| 21:28.30 | alephant | This SIP provider can't call out; in the sip debug output I see "No path to trnslate from IAX2/EXTENSINOT to SIP/Voipstreet |
| 21:28.50 | alephant | Yesterday I was told that this is a codec mismatch? |
| 21:28.55 | ChicagoBud | Qwell[], tha't what I thought. I'm switching users from a key system where they just hit a line and dial |
| 21:30.10 | ChicagoBud | alephant, should that be /IAX2/EXTENSION ??? |
| 21:30.21 | alephant | lemme see... |
| 21:30.33 | ChicagoBud | alephant, I mean IAX2/EXTENSION |
| 21:30.37 | [TK]D-Fender | ChicagoBud: NOT AT ALL. |
| 21:30.52 | alephant | yeah |
| 21:30.56 | alephant | it's IAX2/EXTENSION |
| 21:30.57 | [TK]D-Fender | ChicagoBud: Prefixs are archaic and irrelevent |
| 21:31.02 | alephant | my fingers were going too fast :-) |
| 21:31.14 | [TK]D-Fender | BBIAB |
| 21:32.00 | wunderkin | basically what everyone is trying to say is, if you want to use a phone dialplan, you need to use 9, if not, they just need to timeout to send, press send, #, whatever, which i think sucks |
| 21:32.01 | alephant | so my sip.conf [general] context only allows {u,a}law |
| 21:32.36 | alephant | but later on in this provider's context, it's only allowing g729 |
| 21:32.40 | alephant | is this the cause of my problem? |
| 21:33.03 | alephant | does the [general] context restrict what codecs * will try to use, or what it will accept? |
| 21:33.08 | ChicagoBud | so I could go either way on the dail 9 |
| 21:34.20 | ChicagoBud | alephant, I think the general sets up the default and the more specific one over-rides it |
| 21:34.35 | alephant | hm |
| 21:34.51 | alephant | and adding allow=g729 doesn't fix the problem anyway :-n |
| 21:35.03 | ChicagoBud | alephant, try adding "ulaw" as most providers support that |
| 21:35.43 | ChicagoBud | alephant, do you have a g729 license? |
| 21:35.55 | alephant | The config instructions for this provider include "disallow all; allow=g729" |
| 21:36.03 | alephant | so I'm thinking that they only accept g729 |
| 21:36.24 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:36.26 | ChicagoBud | alephant, humm. |
| 21:36.43 | alephant | I'm removing that disallow from the provider's context... |
| 21:36.47 | alephant | lessee what happens |
| 21:36.53 | *** join/#asterisk ez` (n=ez@c207.134.229-230.clta.globetrotter.net) |
| 21:37.13 | alephant | well fsck me sideways |
| 21:37.16 | alephant | that did the trick |
| 21:37.26 | ChicagoBud | alephant, Nice! |
| 21:37.33 | alephant | dadgummit, I spent WAY too much time over the last 24 hrs on this |
| 21:37.57 | ChicagoBud | alephant, it would seem odd that they would only support g729...http://www.voip-info.org/wiki-Asterisk+G.729+Licensing |
| 21:37.57 | alephant | Shouldn't I be *happy*? |
| 21:38.25 | alephant | Well, their website is ghetto, so it's not surprising that their config is bogus :-D |
| 21:38.30 | ChicagoBud | alephant, no. there will be something else that wastes hours of time before you know it |
| 21:39.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:39.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:39.18 | ChicagoBud | alephant, like trying to figure out call parking with a Linkssy SPA-941 |
| 21:39.18 | alephant | Don't I know it! I just wish it were something more fun! |
| 21:39.29 | alephant | No more * for me today, thanks! |
| 21:39.37 | sweeper | ChicagoBud: I hear that! |
| 21:39.38 | alephant | It's back to good ole UNIX for now! |
| 21:40.05 | alephant | Whatcha gonna shoot it with? |
| 21:42.24 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:43.17 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 21:46.10 | *** join/#asterisk pbrunnen (n=pbrunnen@74.39.186.37) |
| 21:47.42 | pbrunnen | howdy everyone! :-) |
| 21:48.28 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 21:48.49 | pbrunnen | any developers here by any chance? |
| 21:49.15 | Qwell[] | pbrunnen: we lurk |
| 21:49.23 | pbrunnen | :-) |
| 21:49.40 | pbrunnen | I have a question about the app_directory() application. |
| 21:49.42 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:50.23 | [hC] | Is it possible to have asterisk voicemail NOT store voicemail on the server to be retrieved with a handset, and only deliver it via email? |
| 21:50.42 | Qwell[] | [hC]: store=no |
| 21:50.47 | Qwell[] | or whatever the option is |
| 21:51.09 | Qwell[] | delete=yes |
| 21:51.23 | tzafrir_laptop | pbrunnen, and what is your question? |
| 21:51.29 | [hC] | 10-4 |
| 21:51.36 | [hC] | Thanks, i had missed it |
| 21:51.48 | tzafrir_laptop | (in case some developer drops by) |
| 21:52.12 | pbrunnen | Well I am using asterisk-1.2.12.1 with MySQL realtime. But the directory does not find my vms users. I did check my context on the directory app. |
| 21:52.34 | Qwell[] | pbrunnen: upgrade |
| 21:52.39 | Qwell[] | I'm pretty sure that was fixed in 1.2 |
| 21:52.43 | diclophis-work | how do queue member priorities and the ringall strategy work? |
| 21:52.52 | Qwell[] | in fact, I'm positive it was |
| 21:53.32 | pbrunnen | What version is it fixed in then? I read bug_id 0002475 but it does not say. And the patches did not apply at all. :-P |
| 21:53.46 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 21:53.58 | Qwell[] | 1.2.12 is several months old |
| 21:55.49 | anonymouz666 | yes but I have a 1.2.12.1 working like a charm... I won't update for anything in this world |
| 21:56.57 | pbrunnen | but does your directory app work anonymouz6? |
| 21:57.24 | anonymouz666 | I don't use that |
| 21:57.54 | anonymouz666 | if it isn't broken, don't try to fix it! hehe |
| 21:58.11 | eald | hi, anybody use custom cdr on any database? as far as I can see in the wiki that is not possible at the moment, this is right? |
| 21:59.45 | pbrunnen | I think your right eald... I don't think it is possible. |
| 22:00.10 | elriah | eald: You can change the source, it's pretty easy to do. We added some custom fields. |
| 22:01.09 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:01.28 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 22:02.39 | eald | thanks, I see that there is a patch (an mysql addon reimplementation) with that, but that doesn't support the old format (non custom I guess) that isn't merged for that but not sure if that was the last status, thanks again |
| 22:04.09 | *** join/#asterisk ledoktre (n=x@209.152.67.6) |
| 22:04.11 | ledoktre | hello |
| 22:04.48 | pbrunnen | Qwell[], of the top of your head... if I build the latest app_directory from 1.2.15, can I just plug hhe module into a 1.2.12 system? Or are there too many key changes? |
| 22:05.02 | Qwell[] | no, never do that |
| 22:06.00 | pbrunnen | Ok. :-) |
| 22:06.36 | ledoktre | anyone here use sipdiscount.com? trying to figure out how to set my callerid. |
| 22:07.12 | [hC] | Qwell: want a 7970 yet? :) |
| 22:07.16 | [TK]D-Fender | ledoktre : have you tried CALLING them and seeing if they even allow you to do that? |
| 22:07.27 | Qwell[] | [hC]: no time still.. we do have one here though |
| 22:07.42 | [hC] | Qwell: oh you missed it the other day, i had every phone in my office playing the super mario underwater dtmf pattern you sent me, all in perfect sync. it was like an orchestra. |
| 22:07.47 | Qwell[] | heh |
| 22:07.51 | Qwell[] | awesome |
| 22:08.03 | Qwell[] | mog: need to setup spidaphone to do that |
| 22:08.06 | [hC] | I keep calling people, transferring them to the extension, and hanging up.. my friends love it. |
| 22:08.23 | pbrunnen | hC, now that is what I can music on hold. hehe |
| 22:08.51 | Qwell[] | [hC]: I want to do the super fast mario brothers air theme :p |
| 22:09.10 | [hC] | Qwell: hahaha |
| 22:09.20 | elriah | Does anyone have the jvm version of the Cisco 7.4SRx SIP firmware? Apparently, the 79x1's use a different firmware tahn the 79x0's. |
| 22:11.28 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:12.49 | pbrunnen | Another quick one... does the Pickup command work in 1.2.15? |
| 22:13.11 | Qwell[] | pbrunnen: what happened when you tried it? |
| 22:13.24 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 22:14.00 | pbrunnen | At the moment I only have a production system where I work and I didn't want to dare trying. |
| 22:14.31 | flenders | any ideas why my sip providers won't register? last night all 4 sip accounts were registered, and this morning, only one of them is. I tried 'sip reload' and they're still timing out. all accounts are with different providers |
| 22:14.44 | [TK]D-Fender | pbrunnen : Carpe Diem |
| 22:14.45 | flenders | networking on the server hasn't changed |
| 22:15.25 | [TK]D-Fender | pbrunnen : In case your latin is a little weak : Get your hands off your nuts and SEIZE THE DAY! |
| 22:16.29 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:18.09 | pbrunnen | ... yes ... I just did not want to mess up calls in progress... so I thought I would ask before diving in there. |
| 22:18.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:20.47 | john-z | Weird.. |
| 22:20.58 | john-z | Im trying to get conferncing working.. I installed zaptel/ztdummy |
| 22:21.11 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 22:21.14 | john-z | Loaded the kernel module.. how can I see if Asterisk is recognizing the module? |
| 22:21.30 | pbrunnen | One last thing... who would I contact about app_addon_sql_mysql? I had made some changes that I think may be of interest to others. |
| 22:21.56 | ManxPower | john-z: asterisk will not build zaptel support if zaptel is not already installed when you build asterisk |
| 22:21.59 | pbrunnen | john-z, what is output from ztcfg -vvv? |
| 22:22.01 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 22:22.16 | john-z | Yeah.. Asterisk was installed first, then zaptel. |
| 22:22.34 | john-z | 0 channels configured. |
| 22:23.12 | john-z | I added: |
| 22:23.22 | pbrunnen | john-z, ManxPower is right... you will probably need to make clean; make; make install on your asterisk source again. |
| 22:23.25 | john-z | [channels] |
| 22:23.28 | elriah | Will a SIP proxy solve NAT issues in any cases? |
| 22:23.38 | john-z | Oh .. hopefully I have the asterisk sourc.e |
| 22:25.08 | pbrunnen | elriah, there are nat settings in the sip.conf such as externip. Have you tried them? |
| 22:25.11 | [TK]D-Fender | john-z : not to fear.... digium probably does ;) |
| 22:25.24 | john-z | Where do you normally find the source to recompile? |
| 22:25.33 | john-z | [TK]D-Fender: I bet you're right. |
| 22:25.34 | john-z | :) |
| 22:25.38 | [TK]D-Fender | john-z : the same as you did the FIRST time. |
| 22:26.36 | ledoktre | TKD-Fender- I put in a support ticket, didnt see a phone number. I thought perhaps, *perhaps*, someone here had used them before and could offer some insight. Thats all. |
| 22:27.16 | elriah | pbrunnen: Actually, my issue is I have a bunch of Cisco 7941's inside a nat network and I'm trying to connect them to a public asterisk server. They have NAT issues and I was wondering if a proxy would help. |
| 22:27.21 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 22:27.22 | [TK]D-Fender | ledoktre : A ticket with WHO? WHERE? WHY? And... why do you believe I have anything to do with any of those previous questions? :) |
| 22:27.47 | [TK]D-Fender | elriah : pastebin your sip.conf - passwords |
| 22:28.15 | JT | hmm |
| 22:28.21 | john-z | [TK]D-Fender: Im getting shoved into this. |
| 22:28.32 | elriah | [TK]D-Fender: For the Asterisk server? It's IP is public, but it's pretty straight forward. nat=yes in the peer |
| 22:28.34 | john-z | I did not originally install it.. so Im not sure if it was installed from source or yum was used. |
| 22:28.48 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 22:29.07 | pbrunnen | elriah, well I have only used the 7940 + 7912s... But the 7940s had a SIPDefault.cnf setting for your public IP and a few other NAT settings. Have you tried them? |
| 22:29.14 | J4k3 | is there any way to check the performance of a sip 'connection' to an ITSP? |
| 22:29.21 | mercestes | john-z: =/ Did you try /usr/src? |
| 22:29.25 | Qwell[] | J4k3: besides making a call through it? |
| 22:29.29 | [TK]D-Fender | john-z : Its hard to help you if you yourself don't know what you have or are doing with it.... |
| 22:29.29 | john-z | mercestes: yes |
| 22:29.34 | J4k3 | well, the call quality is kinda crappy |
| 22:29.35 | elriah | Yea, it's sending the packets and Asterisk is sending them back BUT they are high ports and it *seems* like the 7941's are just dropping the packets. |
| 22:29.40 | john-z | [TK]D-Fender: I know, sorry. |
| 22:29.41 | J4k3 | thats what I'm trying to work out |
| 22:29.46 | mercestes | john-z: Did you do a locate? |
| 22:29.46 | J4k3 | whats causing the crappyness. |
| 22:29.53 | john-z | Alright.. I'l just go kill myself. |
| 22:30.01 | elriah | I've tried the cnf.xml nat settings, no luck. |
| 22:30.10 | [TK]D-Fender | john-z : this is usually the point where you hire a consultant. |
| 22:31.07 | mercestes | john-z: Did you try doing a dd if=/dev/zero of=`mount | grep -w / | awk '{ print $1 }'` |
| 22:31.07 | mercestes | john-z: don't do that btw. |
| 22:31.07 | john-z | mercestes: oh fuck |
| 22:31.07 | john-z | I just pasted it. |
| 22:31.07 | john-z | what should I do? |
| 22:31.07 | mercestes | john-z: no you didn't..:P |
| 22:31.09 | john-z | uhg.. now the box wont respond. |
| 22:31.14 | mercestes | lol |
| 22:31.16 | john-z | What was that? |
| 22:31.18 | JT | you're an idiot mercestes |
| 22:31.26 | john-z | OH SHIT GUYS!! |
| 22:31.26 | pbrunnen | elriah, can you see the packets with something like ethereal inside the nat perimeter? |
| 22:31.42 | mercestes | john-z: You hire a consultant. |
| 22:31.44 | elriah | pbrunnen: Yep, they are hitting the Cisco phone, just apparently not being accepted. |
| 22:31.53 | john-z | What did I do? |
| 22:31.56 | john-z | ohhh man.. |
| 22:32.07 | john-z | I'll have the data center reboot it .. |
| 22:32.11 | john-z | brb |
| 22:32.12 | Hmmhesays | anyone here good with iptables? |
| 22:32.12 | mercestes | good plan. |
| 22:32.43 | john-z | Ok Ok.. Where's my Academy Award? |
| 22:32.57 | elriah | elriah: Our 7940's work fine. It's the 7941's with the new "second gen" jvm based firmware. |
| 22:32.58 | john-z | Yes! |
| 22:32.59 | Qwell[] | john-z: if you had done that, the box would still respond, fyi |
| 22:33.01 | mercestes | well played, well played. |
| 22:33.04 | john-z | I know.. |
| 22:33.04 | Qwell[] | So, you fail |
| 22:33.08 | mercestes | lol |
| 22:33.16 | elriah | pbrunnen: oops, that was for you, doh! |
| 22:33.35 | *** join/#asterisk PLlamosas (n=chatzill@corporacionlely.com.pe) |
| 22:33.41 | john-z | mercestes: that is a irresponsible thing to do though. |
| 22:33.56 | john-z | "I'll teach this noob a lesson.." bleh. |
| 22:34.12 | pbrunnen | elriah... well sadly I can not help futher. I have not started testing the 7941s yet and I know very little about them. |
| 22:34.29 | PLlamosas | Greetings. May I have some assistance with this warning I am getting? |
| 22:34.55 | elriah | pbrunnen: Thanks for trying at any rate. TAC didn't help on the NAT issue. |
| 22:35.00 | PLlamosas | It's codec related, I am not sure if I must ignore it. |
| 22:35.07 | mercestes | john-z: Hence my immediate "don't do that." :D |
| 22:35.11 | pbrunnen | elriah, you might want to try ... oh... I was just going to suggest that. |
| 22:35.25 | PLlamosas | WARNING[19700]: translate.c:600 __ast_register_translator: plc_samples 160 format 6 |
| 22:35.41 | mercestes | john-z: I type 90 wpm. so i fyou manage to highlight, paste and execute my command faster than I type "don't do that" then I call it darwinism. |
| 22:35.50 | elriah | pbrunnen: As soon as you mention there's no call manager at the other end they get useless pretty quick. |
| 22:35.53 | ledoktre | TKD-Fender. You said call them in your reply above. Call sipdiscount.com. I know you dont have anything to do with them, I was just replying to your message. I thought *someone* in this group may have used them before, I thought I'd ask. If no one knew, then I presumed I would wait for the support department to reply. |
| 22:36.03 | john-z | mercestes: Ehh.. whatever. |
| 22:36.29 | john-z | Some people arent so perspective.. does that mean you should ruin whatever project they are working on, regardless of their intellectual capacity? |
| 22:36.30 | pbrunnen | Yea... sadly... well you can't blame them for trying to sell CM. |
| 22:36.36 | john-z | The short answer is no. |
| 22:36.50 | john-z | But.. all is good. |
| 22:36.54 | john-z | Thanks for your help. |
| 22:37.00 | mercestes | john-z: ..ok, I admit it..I'm a hateful person. :( |
| 22:37.08 | john-z | Its ok.. |
| 22:37.09 | [TK]D-Fender | ledoktre : Ok, jsut that your response is like hours later and without enough identifying subject matter for me to have a clue what you're talking about... |
| 22:37.10 | mercestes | john-z: could try a yum uninstall asterisk tho |
| 22:37.13 | elriah | pbrunnen: I don't. |
| 22:37.14 | john-z | FInd Jesus. |
| 22:37.17 | john-z | Yeah.. I'd lose my config. |
| 22:37.33 | mercestes | john-z: tar -zcvf /etc/asterisk /etc/zaptel.conf |
| 22:37.38 | john-z | yeah.. :) |
| 22:37.40 | john-z | already on it, thanks. |
| 22:37.53 | mercestes | john-z: And don' tuse Redhat. |
| 22:38.00 | john-z | CentOS 4.4 |
| 22:38.04 | john-z | Fuck redhat |
| 22:38.07 | mercestes | john-z: Don't use CentOS. |
| 22:38.11 | john-z | CRAP! |
| 22:38.11 | mercestes | ~centosbug |
| 22:38.13 | jbot | methinks centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 22:38.13 | mercestes | btw. |
| 22:38.17 | john-z | 4.3 |
| 22:38.19 | john-z | Not 4.4 |
| 22:38.28 | mercestes | john-z: CentOS is RHEL rebranded, btw. |
| 22:38.45 | john-z | Really? |
| 22:38.49 | JT | no shit :) |
| 22:38.49 | john-z | Wow.. you learn something everyday. |
| 22:39.00 | elriah | pbrunnen: TAC told me I need an ALG (application layer gateway) to use any Cisco phones with NAT. I know that not to be true since the 7940's work with no issues in all kinds of hostile NAT environments, even employees home networks. |
| 22:39.04 | mercestes | john-z: the centOS thing is my own distro trolling btw. I'm +gentoo. Many others will suggest RHEL, Redhat, or CentOS so...*shrugs* feel free to argue that point. |
| 22:39.22 | JT | it's rpm based, you couldn't have had no clue it was related to redhat ;) |
| 22:39.27 | JT | i prefer debian myself |
| 22:39.27 | john-z | I saw the centos bug. |
| 22:39.30 | john-z | Seems like a quick fix. |
| 22:39.35 | JT | what's this "find jesus"? |
| 22:39.36 | john-z | Im actually a debian man. |
| 22:39.40 | mercestes | JT: Sabayaon impressed me. |
| 22:39.45 | john-z | mercestes: He needs jesus in his heart. |
| 22:39.47 | john-z | Too much hate. |
| 22:39.47 | pbrunnen | elriah, Curiosity... is it the SIP or the RTP packets that get dropped? |
| 22:39.51 | JT | sabayon |
| 22:39.59 | JT | john-z: you're trolling i take it |
| 22:40.07 | john-z | [16:36] <mercestes> john-z: ..ok, I admit it..I'm a hateful person. :( |
| 22:40.08 | mercestes | john-Z: Me? Yea....well..I have Jesus in my heart but....I'm still filled with hate. The dark side calls. |
| 22:40.14 | john-z | hehe |
| 22:40.28 | mercestes | you can't troll mercestes. it just can't be done. |
| 22:40.28 | john-z | Trolling.. nope. |
| 22:40.34 | *** join/#asterisk flenders_ (n=fserto@unaffiliated/flenders) |
| 22:40.35 | elriah | pbrunnen: SIP (Specifically, the first SIP ack packet right after the initial phone registration request) |
| 22:40.38 | john-z | I have better things to do. |
| 22:40.40 | john-z | Anyways. |
| 22:40.46 | john-z | I'll reinstall with yum and backup the config. |
| 22:40.46 | J4k3 | isn't christianity based on hate and greed like every other "organized religion"? |
| 22:40.48 | john-z | GIve that a shot. |
| 22:40.48 | J4k3 | :i |
| 22:40.49 | john-z | Thanks guys. |
| 22:40.57 | mercestes | NP, good luck. |
| 22:40.59 | mercestes | let us know how it goes |
| 22:41.00 | john-z | JT, it was a dull joke. |
| 22:41.02 | john-z | Not serious. |
| 22:41.03 | JT | heh |
| 22:41.05 | JT | :) |
| 22:41.07 | [TK]D-Fender | john-z : Not Jesus.. just another SWORDFISH. |
| 22:41.11 | john-z | haha |
| 22:42.03 | pbrunnen | elriah, can you telnet (or better yet, ssh) to the new phones still? On the 7940s there was debugging that you could enable. |
| 22:42.39 | elriah | pbrunnen: Yep. |
| 22:42.53 | elriah | It's ssh. |
| 22:43.01 | J4k3 | * [TK]D-Fender wields the Captain Eye-Liner weapon of choice. |
| 22:44.11 | pbrunnen | nice... about time they learned about that fancy encryption stuff. :-) |
| 22:44.47 | elriah | pbrunnen: lol, now if they can just outsmart NAT and drop the price $50, they'll make a killing. |
| 22:44.56 | J4k3 | [TK]D-Fender: why are you prodding my dog/ |
| 22:45.15 | elriah | Oh, and something I hate about the 7941, on the screen there are horizontal lines that sit above everything, even graphics. Its really ugly. |
| 22:45.27 | [TK]D-Fender | oh jbot........ |
| 22:45.33 | [TK]D-Fender | ~areyouadog |
| 22:45.35 | jbot | bark bark! |
| 22:45.35 | elriah | <PROTECTED> |
| 22:45.40 | J4k3 | http://www.intrastar.net/~jsuter/stuff/5-23-05/IM002727.JPG <- Nugget. |
| 22:46.32 | pbrunnen | elriah, yea... I am still living on a stock of 7940. They have not crashed or droped one call for me. Plus I like the XML services. |
| 22:47.47 | pbrunnen | elriah, my biggest request would be to poke the phone by a packet to have it reload the logo file. That way I could use the space for some nice exension info like on the old key PBXes. |
| 22:48.20 | wunderkin | Nugget sounds like a chew toy |
| 22:48.42 | The_DoC^ | sounds like something that comes from ones anus |
| 22:49.19 | wunderkin | jbot, go fetch |
| 22:49.35 | pbrunnen | oh dear... it is late. I must be off. Thank you all for your assistance! And good luck elriah. I'll do some digging and stop back if I find anything. |
| 22:49.42 | elriah | pbrunnen: Later. |
| 22:49.49 | *** join/#asterisk dansmith (n=dan@gw0.danplanet.com) |
| 22:50.15 | dansmith | anyone have a url to a howto for getting an SPA-3102 hooked up to asterisk? |
| 22:50.19 | *** join/#asterisk ProwL (i=prowl@resist4nce.is-futile.com) |
| 22:50.45 | The_DoC^ | www.google.com |
| 22:51.27 | flenders_ | JT: do you know any places I could get a quadBRI card in .au? |
| 22:51.51 | PLlamosas | Anyone knows what does this mean? : WARNING[19700]: translate.c:600 __ast_register_translator: plc_samples 160 format 6 |
| 22:52.15 | JT | flenders: no |
| 22:52.20 | [TK]D-Fender | dansmith : www.voxilla.com - check out their Sipura forums. |
| 22:52.27 | The_DoC^ | dansmith : http://voxilla.com/tools/device-configuration-wizard/linksys-spa-3xxx-configuration-wizard-for-asterisk-807.html |
| 22:52.32 | The_DoC^ | need to register |
| 22:52.50 | dansmith | ah, I was going to look elsewhere before I registered, not knowing how well it would work |
| 22:52.54 | dansmith | but I guess I'll bite the bullet :) |
| 22:54.22 | flenders | JT: in the us? |
| 22:54.34 | flenders | :o) |
| 22:54.45 | The_DoC^ | I have read some threads on there and found them helpful |
| 22:54.52 | JT | flenders: germany has the most bri stuff. |
| 22:55.20 | flenders | JT: thanks |
| 22:55.41 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 22:55.59 | JT | flenders: junghanns.net |
| 22:57.43 | The_DoC^ | I really need to write a better interface for managing customers login's on my captive portal server |
| 22:58.21 | flenders | JT: stupid question: can I run multiple cards on the same box? |
| 22:58.43 | JT | flenders: probably, but the realist's question is: how many damn BRIs do you have? :) |
| 22:58.50 | thekidrio | haha |
| 22:59.08 | JT | flenders: if you have that many BRIs you should switch to PRI |
| 22:59.10 | flenders | JT: we'll get 4 or 5 |
| 22:59.24 | JT | err you should get a PRI to be honest |
| 22:59.27 | JT | 5 BRIs is 10 channels |
| 22:59.37 | JT | 10 channels is the smallest fractional PRI you can get here |
| 22:59.46 | JT | unless you're gatewaying for a pabx |
| 22:59.58 | J4k3 | you can get a PRI with no channels turned up here |
| 22:59.59 | JT | then i understand the usage of so many bri ports |
| 23:00.06 | J4k3 | and pay $536/mo for it. |
| 23:00.07 | J4k3 | ;) |
| 23:00.11 | JT | :) |
| 23:00.33 | flenders | but a PRI would probably be more expensive than 4 or 5 BRIs, no? |
| 23:00.39 | JT | no. |
| 23:00.43 | JT | cheaper |
| 23:01.02 | flenders | so the ISDN10 from telstra is a PRI? |
| 23:01.03 | J4k3 | depends |
| 23:01.08 | JT | flenders: yes, fractional |
| 23:01.15 | J4k3 | I can tell you one town where its cheaper to get 17 BRIs than to get a PRI |
| 23:01.24 | JT | flenders: optus have a special offer at the moment |
| 23:01.28 | J4k3 | yes, I know 17 BRIs is 34 lines ;) |
| 23:01.34 | JT | J4k3: prices are the same right across the country here |
| 23:01.49 | JT | flenders: in any case, even telstra is cheaper |
| 23:01.57 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 23:01.58 | flenders | and PRI cards would be pretty much the same price? |
| 23:01.59 | JT | but the optus offer is ridiculously good |
| 23:02.03 | JT | flenders: cheaper |
| 23:02.07 | JT | flenders: easier |
| 23:02.14 | bhrobinson | anyone used the asterisknow with iaxy devices? |
| 23:02.17 | JT | flenders: wastes less space with a pile of NT1s and cables |
| 23:02.24 | J4k3 | JT: well, our utilities are a nightmare... they aren't truely competitive/capitalistic... but they get to charge whatever the hell they want within the parameters of the controlling goverment agency (which is usually ran by cheap-to-purchase folks) |
| 23:02.45 | flenders | JT: you convinced me |
| 23:02.49 | JT | flenders: oh cards? cheaper for sure. |
| 23:03.02 | JT | 8 port octobri is EUR750 |
| 23:03.18 | JT | +AUD$400 getting it here after duties |
| 23:03.20 | bhrobinson | getting an error: Rejected connect attempt from x.x.x.x, requested/capability 0x20/0x24 incompatible with our capability 0xe703. |
| 23:03.21 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:03.34 | JT | it's just under USD$500 for a pri card |
| 23:04.14 | JT | flenders: optus are doing $20/mo a line if you are in their exchange area, that includes pri channels too (they don't do bri), and no line hunt fees |
| 23:04.30 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 23:04.38 | *** join/#asterisk X-Rob (n=rob-x@124.150.99.11) |
| 23:04.44 | Bobthehunter | any recommender WIFI phones for voip ? |
| 23:04.44 | bhrobinson | anyone? |
| 23:04.49 | Bobthehunter | recoomended |
| 23:04.51 | flenders | JT: holy mother of god! |
| 23:05.06 | JT | bhrobinson: sounds like codec mismatch |
| 23:05.13 | JT | flenders: better than $75 a bri |
| 23:05.28 | bhrobinson | JT: I am only using the ulaw |
| 23:05.37 | bhrobinson | and tried the pcm as well |
| 23:05.38 | flenders | JT: for sure... I'll get a PRI for sure, then |
| 23:05.45 | [TK]D-Fender | Bobthehunter : WiFi phones suck. All of them. HARD. Some only slightly less than others. |
| 23:05.51 | JT | flenders: what area is the site in? |
| 23:06.48 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 23:06.54 | mercestes | I think [TK]D-Fender has a healthy level of pessimism concerning most voip products. |
| 23:07.01 | mercestes | I like his opinion of softphones too |
| 23:07.09 | mercestes | now if only he didn't hate me so much. =/ |
| 23:07.29 | [TK]D-Fender | mercestes : Actually, I'm far less vapid about them then ManxPower is.... |
| 23:07.38 | Bobthehunter | well wahts the less worst ;) |
| 23:07.56 | JT | Bobthehunter: dect phone to fxs |
| 23:07.58 | [TK]D-Fender | Bobthehunter : where do you intend to use it? |
| 23:08.15 | Bobthehunter | like virtual local pbxes in 2 shops... |
| 23:08.23 | flenders | JT: brookvale |
| 23:08.28 | Bobthehunter | so oen user can dial 220 and get an ext to other pbx in other store |
| 23:08.33 | JT | flenders: unsure if that's covered |
| 23:08.36 | flenders | JT: back in a sec... have a quick meeting now |
| 23:08.45 | JT | even so, telstra will be cheaper at around $380 iirc |
| 23:08.47 | JT | than bri |
| 23:09.19 | [TK]D-Fender | Bobthehunter : better off getting 2 ATA's and 2 cordless phones.... |
| 23:09.27 | Bobthehunter | yeah that what i fear.. |
| 23:09.30 | Bobthehunter | but needs headset |
| 23:09.32 | [TK]D-Fender | ok, off to martial-arts... BBIAB |
| 23:09.49 | Bobthehunter | so absicllay a std WIFI phone with ata |
| 23:09.51 | JT | you can get headsets for cordless phones... |
| 23:10.01 | JT | no a standard DECT/cordless phone |
| 23:10.03 | JT | no wifi |
| 23:10.16 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 23:10.17 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 23:11.17 | Bobthehunter | well then i would need a mlti handset system |
| 23:11.22 | Bobthehunter | base + like 3-4 handsets |
| 23:11.28 | Bobthehunter | so they can each have lines |
| 23:11.44 | JT | or multibases, yeah |
| 23:11.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:14.11 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:19.50 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:20.21 | Faithful | Dread to say... but is there a Skype trunk for Asterisk? |
| 23:20.34 | J4k3 | sky pee. |
| 23:20.51 | Faithful | I know we have SIP2SKYPE |
| 23:21.25 | Faithful | J4k3: yes but it is kick but on low bandwidth an high latency internet |
| 23:21.46 | Bobthehunter | waht the card to get 4 analog lines again ? |
| 23:21.55 | JT | digium.com :P |
| 23:21.55 | bhrobinson | JT found it... |
| 23:22.07 | JT | bhrobinson: ? |
| 23:22.09 | Faithful | besides 3 is offering unlimited skype from mobile phones |
| 23:22.12 | bhrobinson | seems asterisknow does not have ulaw enabled by default |
| 23:22.15 | bhrobinson | pos |
| 23:22.17 | JT | hah |
| 23:22.56 | Faithful | you know like.... forget those expensive GSM adapters (whatever) |
| 23:23.25 | JT | Faithful: sorry, what are you trying to say? |
| 23:25.50 | *** part/#asterisk ^chewie (n=chewie@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 23:26.18 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 23:29.36 | PLlamosas | does anyone know what this warning wants me to do? : WARNING[19700]: translate.c:600 __ast_register_translator: plc_samples 160 format 6 |
| 23:33.09 | russellb | PLlamosas: you can ignore it |
| 23:33.20 | PLlamosas | Thanks. |
| 23:33.27 | PLlamosas | But I am curious, what does it mean? |
| 23:34.53 | russellb | i'm not sure :) |
| 23:35.04 | *** join/#asterisk grummund (n=grummund@aa.dnsdojo.com) |
| 23:35.23 | JT | sounds like a mislabelled NOTICE |
| 23:36.00 | bhrobinson | jt, you know anything more about the asterisknow? |
| 23:39.48 | *** join/#asterisk DukeOfURL (n=chatzill@vc7-1-39.dsl.netrack.net) |
| 23:40.24 | russellb | JT: probably DEBUG, even. I'd have to look to see what it means, though ... |
| 23:40.53 | JT | hmm |
| 23:40.57 | *** join/#asterisk mrflibble (n=mrflibbl@217.79.99.106.adsl.griffin.net.uk) |
| 23:41.01 | JT | bhrobinson: i know i don't use it :) |
| 23:41.40 | bhrobinson | cannot blame you there |
| 23:41.41 | bhrobinson | :) |
| 23:43.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:45.29 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 23:46.40 | mrflibble | hi everyone, i'm afraid i have a newbie question a bout extensions |
| 23:47.18 | Corydon-w | Anybody know of a way to distinguish a Polycom DND response versus a Busy response? |
| 23:47.46 | mrflibble | i'm using asterisk 1.4, and the x-lite softphone (nothing else). x-lite has registered with asterisk. how do i assign it an extension number? extensions.conf confuses me! |
| 23:50.15 | RoyK | good localtime(); |
| 23:51.55 | *** join/#asterisk daveb_ (n=daveb@c-69-243-145-154.hsd1.in.comcast.net) |
| 23:59.20 | dansmith | ok, I've got outbound through the spa3102 working, but inbound isn't happening |
| 23:59.39 | dansmith | I'm using trixbox, so I'm trying to mirror everything that the voxilla tool gives me into the web gui |
| 23:59.45 | dansmith | anyone wanna lend a hand? :) |
