| 00:00.24 | *** join/#asterisk mentat (~Mentat@68.63.120.21) |
| 00:01.39 | nesys | harryvv if you have any suggestion ... :) |
| 00:04.44 | nesys | the iax2.fwdnet.net is down? |
| 00:06.50 | fugitivo | anything to do videoconferences? |
| 00:10.20 | *** join/#asterisk donavan (~donavan@4wx.net) |
| 00:10.24 | nesys | register => fwdnum:pass@iax2.fwdnet.net |
| 00:10.29 | nesys | that's ok, isn't it? |
| 00:10.50 | Juxt | hmm when i do sip show peers |
| 00:11.02 | Juxt | it shows status unmonitored |
| 00:11.29 | nesys | Juxt use qualify=yes on peers contexts (sip.conf) |
| 00:13.40 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
| 00:13.47 | nesys | Juxt it works? You'll see something like "OK (x ms)" |
| 00:14.10 | Juxt | yeah it does |
| 00:14.15 | Juxt | now it shows OK 46 ms |
| 00:14.32 | nesys | the link is good |
| 00:14.38 | *** join/#asterisk danalien (~danalien@danalien.user) |
| 00:14.54 | nesys | you have a jitter problem? |
| 00:15.13 | Weezey | if I have an SPA-3000 which connects to asterisk on both the FXS and FXO sides and I make a call that goes out via asterisk, does it go out and then back to the same unit? |
| 00:16.11 | Weezey | also, does anyone have an open MeetMe I can call via IAX? I want to see how it sounds. |
| 00:16.16 | *** join/#asterisk syb-grrr (~syb_hmm@207.107.243.226) |
| 00:17.07 | syb-grrr | hi~ |
| 00:17.13 | JerJer[mobile] | preliminary "managed DNS lookup" support <-----very nce |
| 00:17.37 | Weezey | mobile? How so? |
| 00:18.36 | Juxt | nesys: yes |
| 00:18.46 | Juxt | it seems that the voice skips a bit |
| 00:18.53 | Juxt | i don't have that problem with i connect to an iax provider |
| 00:19.25 | nesys | Juxt wich iax provider? |
| 00:19.34 | Juxt | nufone |
| 00:20.14 | nesys | k ... mmm, have you tryed with another sip provider? |
| 00:20.21 | Juxt | no not yet |
| 00:20.45 | nesys | sip debug says nothing special? |
| 00:21.08 | Juxt | nope |
| 00:21.22 | Juxt | in fact it goes thru super clean, no errors or warnings |
| 00:21.37 | Juxt | also with sip i hear this light "hum" int he background where with iax i don't |
| 00:21.56 | nesys | codec? |
| 00:22.06 | Juxt | i switched everything to gsm |
| 00:22.15 | Juxt | both iax and sip |
| 00:22.39 | nesys | have you a low bandwidth link? |
| 00:22.49 | Juxt | no it's a 4 mbit link |
| 00:22.57 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
| 00:23.03 | Juxt | ok i see something whacky |
| 00:23.16 | Juxt | my sip provider status just jumped to 417ms |
| 00:23.38 | Essobi | Nice. |
| 00:23.51 | Essobi | You got latency foo' |
| 00:23.56 | Juxt | nesys, i am off to the gym but i will pick your brain when i get back or whnever i catch you. thank you! |
| 00:24.01 | Essobi | That's going to kill your BW. |
| 00:24.07 | Weezey | Juxt: how do you see your ms? |
| 00:24.17 | Juxt | qualify=yes in sip.conf |
| 00:24.27 | Weezey | cool |
| 00:24.49 | nesys | Juxt now I go to bed :) here 2:24 am :) |
| 00:24.54 | Juxt | oh ok |
| 00:25.08 | Juxt | doing traceroute now |
| 00:25.10 | nesys | but try with another codec |
| 00:25.21 | nesys | moment |
| 00:25.56 | nesys | I use alaw |
| 00:26.22 | Juxt | hmm i will try that |
| 00:26.33 | Juxt | but i see an ungodly number of hops between me and that provider |
| 00:26.36 | Juxt | that might explain it |
| 00:27.01 | Juxt | good nite |
| 00:27.07 | *** part/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 00:27.11 | nesys | how many? how much latency? |
| 00:27.21 | Essobi | Juxt go grab matt's trace route |
| 00:27.23 | Essobi | mtr.. |
| 00:27.45 | *** join/#asterisk bbledsoe (nobody@dhcp-69-43-0-252.pitbpama-max5.dialup.citynet.net) |
| 00:28.04 | Weezey | I want a sexy headset. |
| 00:28.15 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 00:29.02 | nesys | Essobi what's mtr? |
| 00:29.09 | bbledsoe | anybody have any ideas what sort of date formatting a d-link ATA might use in the SIP header? |
| 00:30.46 | nesys | there's someone connecting on FWD via IAX2? |
| 00:30.50 | Weezey | matt's trace route |
| 00:31.00 | nesys | Weezey interesting |
| 00:31.03 | Sedorox | I am |
| 00:31.17 | nesys | Sedorox have you problems now? |
| 00:31.34 | Weezey | nesys: never heard of it, I'm just guessing by what SOB said. |
| 00:31.48 | nesys | Sedorox my registration is rejected :( |
| 00:31.55 | *** join/#asterisk Schism (~schism@cpe-024-074-101-230.carolina.res.rr.com) |
| 00:32.10 | Sedorox | umm |
| 00:32.11 | nesys | Weezey I'll see that |
| 00:32.12 | Sedorox | let me check |
| 00:32.20 | chaoscon | Sedorox: I'm sitting in the console btw |
| 00:33.07 | Sedorox | I noticed |
| 00:33.22 | *** join/#asterisk Cinen (~srash@cpe-065-188-184-163.triad.res.rr.com) |
| 00:33.29 | Sedorox | odd..... |
| 00:33.33 | Sedorox | *ponders* |
| 00:33.42 | chaoscon | hmmm very odd |
| 00:34.54 | Sedorox | I think its connected |
| 00:35.01 | chaoscon | lol |
| 00:35.20 | nesys | Sedorox are you registered via IAX on FWD now? |
| 00:35.43 | Sedorox | 65.39.205.121:4569 589476 64.251.71.178:4569 60 Registered |
| 00:35.44 | Sedorox | Si |
| 00:35.44 | Sedorox | yes |
| 00:36.31 | nesys | today a lot of problems for me about IAX :) I would try that, but IAXtel is flapping, FWD reject my registration :( |
| 00:36.33 | nesys | LOLù |
| 00:36.46 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.139) |
| 00:37.10 | Sedorox | we droped iaxtel |
| 00:38.13 | nesys | have you got any IAX free provider on your white list? :) |
| 00:38.32 | Sedorox | Ummm |
| 00:38.39 | Sedorox | I'm only linked with fwd right now |
| 00:38.40 | Sedorox | thats free |
| 00:39.06 | nesys | yep ... but debug says nothing, and it rejects my registration |
| 00:39.17 | Sedorox | got everything setup according to their howto? |
| 00:39.31 | nesys | I use register => fwdnum:pass@iax2.fwdnet.net |
| 00:39.57 | *** join/#asterisk drbrown (~chatzilla@user-0cdvec3.cable.mindspring.com) |
| 00:40.15 | chaoscon | Sedorox: did you see the error that popped up? |
| 00:40.18 | Sedorox | register => 589476:<password>@iax.fwdnet.net |
| 00:40.20 | Sedorox | yes |
| 00:40.27 | nesys | ah |
| 00:40.28 | chaoscon | hehe |
| 00:40.32 | nesys | iax,fwdnet.net |
| 00:40.36 | Sedorox | hehe |
| 00:40.38 | Sedorox | could be why |
| 00:40.39 | nesys | and not iax2.fwdnet.net |
| 00:40.48 | nesys | ? |
| 00:41.05 | harryvv | sed you use xlite |
| 00:41.18 | Sedorox | nesys: correct |
| 00:41.22 | chaoscon | iax. and iax2. point to the same thing |
| 00:41.33 | Sedorox | harryvv: yes |
| 00:41.34 | harryvv | need somone here who has setup xlite on outside a network calling into a * network |
| 00:41.37 | Sedorox | not personally.. but once |
| 00:41.44 | Sedorox | eh? |
| 00:41.59 | Wazb | hi all |
| 00:42.09 | Sedorox | I've had xlite clients here in my dorm.... connecting to our * server that is colo'd |
| 00:42.14 | nesys | harryvv I don't understand, sorry |
| 00:42.40 | drbrown | hello |
| 00:42.41 | harryvv | yea I have a xlite > * nat > nat > xlite done anything like that? |
| 00:42.44 | nesys | connection through nat? |
| 00:42.51 | harryvv | yup |
| 00:42.59 | Wazb | i am getting chan_oss.c:269 sound_thread, any suggestion |
| 00:43.18 | harryvv | betwen the nats of course is the internet |
| 00:43.19 | nesys | normally no problems there |
| 00:43.19 | harryvv | :) |
| 00:43.28 | nesys | with x-lite |
| 00:43.30 | Sedorox | ummm |
| 00:43.33 | Sedorox | with sip |
| 00:43.36 | Sedorox | nat causes problems |
| 00:43.39 | Sedorox | get firefly |
| 00:43.41 | Sedorox | and use IAX |
| 00:43.41 | nesys | no with x-lite |
| 00:43.42 | Sedorox | then try |
| 00:43.47 | harryvv | nesys so you configured the server and xlite client end on a remote end? |
| 00:44.05 | nesys | no here |
| 00:44.20 | nesys | but I think nat=yes on server is enough |
| 00:44.23 | Weezey | ohhhhh, I'm downloading, that explains a lot about why that last call was so craptacular. |
| 00:44.24 | harryvv | tell me about it |
| 00:44.25 | harryvv | :) |
| 00:45.03 | nesys | harryvv Have you seen in sip.conf the specific x-lite configuration? |
| 00:45.10 | harryvv | I can call out on xlite to iax.cc or all internal soft/hardphones but friend to friend with nats in between. |
| 00:45.33 | harryvv | yes |
| 00:46.13 | harryvv | mine is pretty much the same. |
| 00:46.16 | nesys | Sedorox could you help me about iax-fwd in pvt? |
| 00:46.32 | nesys | harryvv ok ... and the problem is? |
| 00:47.44 | Sedorox | hehe |
| 00:47.53 | Sedorox | nesys: sure |
| 00:47.58 | nesys | nice :) thanks |
| 00:48.02 | Sedorox | tho is you follow the howto's.. it should work |
| 00:48.17 | nesys | I've followes that ... |
| 00:48.21 | nesys | followed |
| 00:48.30 | nesys | probably a mistake :( |
| 00:48.31 | harryvv | nesys check your msg |
| 00:48.37 | nesys | yep |
| 00:48.50 | drbrown | I keep getting an error when I try to start my asterisk server |
| 00:49.04 | Sedorox | drbrown: paste it |
| 00:49.29 | drbrown | Apr 6 15:57:29 WARNING[1138]: Unable to open IAX timing interface: Permission denied |
| 00:49.40 | Sedorox | hmmmm |
| 00:49.48 | drbrown | I have gotten ztdummy to compile and install properly |
| 00:49.49 | Sedorox | you running iax as a seperate user? |
| 00:49.53 | Sedorox | er |
| 00:49.56 | Sedorox | * as a sep uer |
| 00:49.57 | Sedorox | user* |
| 00:50.02 | drbrown | no |
| 00:50.33 | Sedorox | pastebin what happens when you start it.. the entire thing.. not just that line |
| 00:50.35 | Sedorox | ~pastebin |
| 00:50.36 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 00:52.06 | drbrown | http://pastebin.ca/8976 |
| 00:52.45 | Sedorox | thats everything that shows up? |
| 00:52.57 | drbrown | yes |
| 00:53.10 | Sedorox | hmmmmm |
| 00:53.12 | Sedorox | doesn't seem right |
| 00:53.24 | drbrown | I'm sorry that's what shows up in the messages log |
| 00:53.29 | Sedorox | I'm not sure what it is.. maybe someone else can help.. sorry |
| 00:53.30 | Sedorox | ok |
| 00:56.42 | drbrown | I reposted on pastebin what happens when I start it in real tim |
| 00:56.47 | drbrown | http://pastebin.ca/8977 |
| 00:57.04 | drbrown | time* |
| 00:57.52 | Sedorox | and it starts |
| 00:58.41 | drbrown | yeah, but when I type commands I get nothing |
| 00:58.58 | Sedorox | like.,..? |
| 00:59.10 | drbrown | stop now |
| 00:59.19 | chaoscon | you starting with -c? |
| 00:59.48 | drbrown | no asterisk -vvvp |
| 00:59.52 | drbrown | no asterisk -vvvvp |
| 01:00.42 | Sedorox | you need to start with -c |
| 01:00.44 | Sedorox | to have a console |
| 01:00.46 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.165) |
| 01:00.46 | Sedorox | and do commands |
| 01:00.49 | drbrown | ok |
| 01:00.52 | chaoscon | :) |
| 01:02.05 | drbrown | I guess it doesn't matter if the timing device is not working? |
| 01:02.55 | Sedorox | I didn't see that error on the second pastebin |
| 01:03.13 | drbrown | I think it only shows up in the message log |
| 01:03.55 | Sedorox | dunno |
| 01:04.46 | drbrown | i know I havn't been able to get a single iax client to connect, but it could be another config error, this is my first setup |
| 01:07.57 | *** part/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 01:13.17 | bbledsoe | anybody know anything about d-link ATAs? |
| 01:13.34 | shido6 | whats up? |
| 01:13.49 | shido6 | drbrown, pastebin.ca your iax.conf |
| 01:13.54 | shido6 | maybe its a newbie mistake |
| 01:14.36 | robl^ | you can't paste to .ca.. its against the export laws.. you need a passport with a DNA sample to paste :) |
| 01:14.38 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 01:15.01 | Sedorox | robl^: not yet :-p |
| 01:16.26 | *** join/#asterisk captrb (~crozierm@64.65.134.42) |
| 01:18.52 | *** join/#asterisk captrb (~crozierm@64.65.134.42) |
| 01:19.19 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:19.29 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 01:19.34 | captrb | i'm trying to set the outbound cid info based on my sip extension |
| 01:19.45 | captrb | does anybody know of a variable with that info? |
| 01:20.07 | captrb | such that I could cut it out and use it in setCIDNum? |
| 01:20.43 | robl^ | is there an easy / good way to export / import data from the Asterisk database (not realtime/odbc)? |
| 01:21.32 | tessier | Anyone know anything about vocal? Can it do anything that asterisk cannot? |
| 01:21.58 | robl^ | captrb: not really. CLID is sent between the first and second ring. no other time |
| 01:22.35 | captrb | robl^: hrm. so there is no way to set CID differently for each extension? |
| 01:22.52 | robl^ | oops |
| 01:22.55 | captrb | i think i'm going about this the wrong way... |
| 01:23.29 | robl^ | captrb: never mine. ignore that. I am not sure what I was answering. |
| 01:23.34 | drbrown | http://pastebin.ca/8978 |
| 01:23.38 | drbrown | sorry for the delay |
| 01:23.58 | captrb | robl^: ack |
| 01:24.06 | robl^ | captrb: you can use the logic in your dial plan to set the CLID.. or you can just default to the caller id set in sip.conf for each phone |
| 01:24.43 | captrb | robl^: oh really? I tried setting something in fromuser, but it didn't set it. |
| 01:24.57 | captrb | robl^: do you know, is that the wrong variable to assign the cid to? |
| 01:26.09 | robl^ | captrb: nah... in your phone definitions do something like" callerod="Rob"<4332 |
| 01:26.09 | robl^ | > |
| 01:26.33 | ManxPower | Don't use quotes in callerid |
| 01:26.36 | robl^ | ignore the return and put a space between " & < |
| 01:27.25 | robl^ | ManxPower: no quotes?? I've had it in my sip.conf since ,0.7x and I copied from jtodd's example :) |
| 01:27.36 | *** join/#asterisk dave_mwi_ (~dave_mwi@adsl-068-153-207-210.sip.bct.bellsouth.net) |
| 01:27.48 | ManxPower | robl^, *most* of the time they don't cause a problem. Most of the time. |
| 01:27.56 | ManxPower | I'll bet you don't have any Cisco phones do you, robbyt |
| 01:28.00 | ManxPower | robl^ too |
| 01:28.27 | robl^ | ManxPower: all my phones are cisco 7960s |
| 01:28.32 | robl^ | and it works fine |
| 01:28.37 | dave_mwi_ | anyone know why all my mysql cdr userfield values would be limited to 239 characters? The column is a 'text' column...not even a varchar...is there some kind of internal limit on the size of the string? |
| 01:29.12 | ManxPower | robl^, then at least some versions of the firmware won't ring if callerid has quotes |
| 01:29.15 | dave_mwi_ | I'm compiling the string with SetCDRUserField and AppendCDRUserField commands... |
| 01:29.22 | ManxPower | ~google site:lists.digium.com cisco ring quotes |
| 01:29.46 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 01:29.55 | robl^ | ManxPower: I've had no trouble and used every firmware since 6.1 |
| 01:32.47 | syb-grrr | simple question from a newbie |
| 01:32.59 | syb-grrr | regular modems are supported by Asterisk, correct? |
| 01:33.04 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 01:34.20 | harryvv | yea if you call out |
| 01:34.38 | drbrown | eveyone have a good night |
| 01:34.53 | drbrown | I appreciate the help |
| 01:35.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:36.25 | Essobi | I got a few 60's too. |
| 01:36.43 | Essobi | Never had any not rin |
| 01:36.48 | Essobi | ring even |
| 01:38.22 | ManxPower | syb-grrr, no. |
| 01:38.30 | ManxPower | syb-grrr, Let me repeat that: NO! |
| 01:38.44 | ManxPower | Essobi, I could be srong about the vendor |
| 01:39.22 | drumkilla | :w |
| 01:39.22 | harryvv | unless there is a asterisk driver made for it. |
| 01:39.27 | drumkilla | oops :) |
| 01:39.50 | harryvv | even then the elelectroncics might not work |
| 01:39.56 | harryvv | electronics :) |
| 01:42.51 | *** join/#asterisk qwerp (~abc@60.48.82.153) |
| 01:42.55 | qwerp | harlo.. |
| 01:42.56 | harryvv | if a context= is missing from [context] in sip.conf or its out of place would it generate a error of cannot find extension context 'default' even if default is not a context used? I am getting that three times a min for no reason. |
| 01:43.07 | qwerp | i just face some really weird problem. |
| 01:43.13 | harryvv | everything is context=users |
| 01:44.00 | qwerp | i have a * box with ip A, and a gateway with ip B, when gateway goes down, all the sip peers soes crazy. thay can't make anny calls. |
| 01:44.00 | captrb | robl^: thanks |
| 01:44.27 | qwerp | and when i do a "sip show peers" the status shows that the sip users are "lagged" |
| 01:44.35 | robl^ | captrb: did it work? |
| 01:45.22 | qwerp | do i have to set the gateway of the sip peers to point to * also or i can have a different gateway.. |
| 01:45.53 | *** join/#asterisk _zigo__ (~ogiz@m6.net81-64-48.noos.fr) |
| 01:46.24 | _zigo__ | Hey, what have changed in the priority, what should I put instead of "n" ? |
| 01:49.36 | Essobi | anyone used cdr_custom lately? |
| 01:50.31 | Essobi | I want to have the final dnis in my cdr's where I'm dialing more then one at a time. |
| 01:50.51 | ManxPower | _zigo__, I have no idea what you just said. |
| 01:51.08 | *** join/#asterisk cc (~cc@byte.fedora) |
| 01:51.24 | _zigo__ | I've update from asterisk 1.0 to 1.0.7, and my extension.conf does not work anymore. |
| 01:51.37 | _zigo__ | It complains each time the priority is "n" and not numbers... |
| 01:51.46 | drumkilla | 'n' was never in any version of 1.0 |
| 01:51.56 | robl^ | priority 'n' was only in head, I thought |
| 01:52.00 | drumkilla | it is |
| 01:52.03 | _zigo__ | Never mind... |
| 01:52.07 | ManxPower | _zigo__, "n" priority was NEVER in 1.0.x |
| 01:52.17 | _zigo__ | Maybe it was an older version.. |
| 01:52.25 | drumkilla | not there either :) |
| 01:52.30 | drumkilla | you must have been running head |
| 01:52.41 | robl^ | _zigo__: its only in development/beta versions. not in stable. |
| 01:52.58 | _zigo__ | Ha, I used the CVS version. |
| 01:53.00 | ManxPower | I'm SO tired. |
| 01:53.10 | ManxPower | _zigo__, thought you could fool us, huh? |
| 01:53.19 | _zigo__ | So I should add the numbers byexten => 600,1,Playback(demo-echotest) ; Let them know what's going on |
| 01:53.19 | _zigo__ | exten => 600,n,Echo ; Do the echo test |
| 01:53.22 | _zigo__ | Sorry |
| 01:53.23 | robl^ | have _zigo__ flogged! |
| 01:53.24 | _zigo__ | mistake ! :) |
| 01:53.25 | qwerp | when i do a "sip show peers" what is the meaning of lagged on the status section? |
| 01:53.32 | _zigo__ | (wrong key) |
| 01:53.38 | dave_mwi_ | does anyone know if the length of the value you can pass through SetCDRUserField is truncated...say around 239 chars? |
| 01:53.46 | _zigo__ | You seems to be specialists ! :) |
| 01:53.49 | ManxPower | Qwell, It means the device took longer than qualify= (yes=2000) to respond to a SIP OPTIONS request. |
| 01:54.12 | ManxPower | dave_mwi_, It would not suprize me. CDR is for call records, not for storing novels. |
| 01:54.14 | Qwell | hmm |
| 01:54.33 | _zigo__ | Well, I tried to switched versions because it was not working anymore. |
| 01:54.54 | _zigo__ | It didn't accept inbound calls from broadvoice since 3 weeks. |
| 01:55.11 | *** join/#asterisk zilas (~1@adsl-19-106-35.asm.bellsouth.net) |
| 01:55.18 | ManxPower | dave_mwi_, It's prolly defined in the source what the max length is and you can change it and recompile. |
| 01:55.21 | _zigo__ | Now it does again, and maybe I'm wrong with versions numbers, by the way... |
| 01:55.30 | zilas | hello |
| 01:55.40 | dave_mwi_ | Manx: ok, I'll do some looking for that |
| 01:55.43 | *** join/#asterisk Asskick (~Asskick@red-corp-200.76.225.19.telnor.net) |
| 01:55.57 | ManxPower | _zigo__, "show version" will show you, oddly enough, the version. |
| 01:56.09 | _zigo__ | ManxPower: thanks ! :) |
| 01:56.13 | Asskick | guys anyone has the sip firmware for the cisco 7960 ?? |
| 01:56.29 | qwerp | is there any specific issue that will result in the "lagged" issue? coz usually i don't have that problem. it just come out when there is some problem with my gateway.. |
| 01:56.33 | _zigo__ | By the way, I'll just write down numbers instead of "n", I don't care it's not the dev version anymore, I just want it to work!!! |
| 01:56.35 | _zigo__ | :) |
| 01:56.44 | ManxPower | Asskick, I do. |
| 01:56.49 | Qwell | qwerp: It means you're lagged... |
| 01:57.00 | ManxPower | Qwell, network latency is the most common |
| 01:57.17 | Qwell | He needs to change his nick :p |
| 01:57.18 | qwerp | so, main issue is on the network. |
| 01:57.28 | qwerp | so i need to check on the network.. right? |
| 01:57.31 | captrb | robl^: don't know yet, getting ready to look into it (fires elsewhere) |
| 01:57.49 | Asskick | ManxPower could u eamil it to me ? |
| 01:57.49 | ManxPower | high network latency will increase the time it takes to get back the OPTIONS response and phone are already pretty slow in responding to that. |
| 01:57.58 | Asskick | i just bought 2 cisco phones |
| 01:58.05 | ManxPower | Asskick, I could, but then I would be comiting copyright infringement and could GO TO JAIL. |
| 01:58.16 | ManxPower | Asskick, Cisco firmware is NOT FREE. |
| 01:58.23 | ManxPower | It's like $120 |
| 01:58.28 | captrb | laaame |
| 01:58.34 | Asskick | how come it aint free ?? |
| 01:58.36 | Qwell | ManxPower: Thats per phone, right? |
| 01:58.43 | Qwell | Asskick: because it costs them money to make |
| 01:58.45 | ManxPower | Asskick, Call up Cisco and ask them. |
| 01:58.45 | captrb | Polycom doesn't give theirs out to end users either, but its obtainable |
| 01:58.48 | ManxPower | Qwell, yes. |
| 01:58.52 | Qwell | ouch |
| 01:58.53 | captrb | doesn't cost money regardless |
| 01:58.57 | Qwell | thats a bit excessive :p |
| 01:59.07 | Asskick | 300 bux for a phone that wont be able to be used for asterisk |
| 01:59.11 | ManxPower | captrb, the difference is that polycom does not sell their firmware. You are supposed to get it for free from your reseller. |
| 01:59.18 | Qwell | Asskick: You can too use it with asterisk |
| 01:59.29 | Essobi | ManxPower committing. ;) |
| 01:59.32 | captrb | ManxPower: sucks when you bought them off the internet though. |
| 01:59.38 | zilas | ztcfg: damm......line 0: Unable to open master device '/dev/zap/ctl' |
| 01:59.43 | Essobi | captrb TRUE THAT |
| 01:59.48 | ta[i]nted | have u guys tried yuxin products? |
| 01:59.49 | Qwell | zilas: modules loaded? |
| 01:59.55 | ManxPower | captrb, Yes. It's a good lesson to teach people to do more research next time. |
| 02:00.03 | qwerp | is it adviceable to set asterisk on a class B network? |
| 02:00.06 | zilas | I think they are lets try again |
| 02:00.13 | qwerp | or should i just stick back on a class C network? |
| 02:00.22 | *** part/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 02:00.23 | ManxPower | qwerp, Um, networks really don't have classes anymore. |
| 02:00.25 | harryvv | ztcfg -v zilas |
| 02:00.43 | captrb | ManxPower: well, i found the firmware and saved $600... |
| 02:00.57 | qwerp | ManxPower, y do u say so? |
| 02:01.02 | ManxPower | captrb, The wiki has links to downloadable Polycom formware |
| 02:01.30 | ManxPower | qwerp, this google search is for you: |
| 02:01.35 | ManxPower | ~google CIDR glossary |
| 02:01.43 | ManxPower | ~google CIDR RFC |
| 02:01.50 | Qwell | CIDR <3 |
| 02:01.52 | _zigo__ | I realy think there MUST be a "maake deinstall" in Asterisk... |
| 02:01.57 | _zigo__ | Is it planned ? |
| 02:02.11 | fearnor | zigo: so we can tell that to retards who ask silly questions here? |
| 02:02.17 | ManxPower | _zigo__, let us know when you have posted the patch |
| 02:02.26 | _zigo__ | :) |
| 02:02.40 | ManxPower | Well, make install in 1.0.7 or later DOES tell you what you need to do to downgrade from CVS-HEAD to 1.0.7 |
| 02:02.59 | _zigo__ | I'm doing this: http://www.gplhost.com/?rub=softwares&sousrub=dtc, and it's enough work ! :P |
| 02:03.09 | qwerp | okie. |
| 02:03.12 | fearnor | ok, i have a retarded question |
| 02:03.17 | fearnor | i have polycoms ip500 |
| 02:03.22 | qwerp | thanz for all the replys folks. |
| 02:03.25 | *** join/#asterisk iq (~iq@70-59-163-109.omah.qwest.net) |
| 02:03.25 | fearnor | when turned on, they display polycom logo |
| 02:03.25 | *** part/#asterisk qwerp (~abc@60.48.82.153) |
| 02:03.29 | fearnor | and then screen goes blank blank |
| 02:03.41 | fearnor | anyone experienced anything like that? |
| 02:04.02 | _zigo__ | ManxPower: Are you of Asterisk's dev team ? |
| 02:04.16 | fearnor | should i assume those polycoms are DOA? |
| 02:04.22 | captrb | fearnor: I have them, but they don't do that |
| 02:04.25 | ManxPower | _zigo__, No. If I was employed by Digium or an official developer I would have to be nice to people. |
| 02:04.37 | captrb | fearnor: does it should the "setup" menu option? |
| 02:04.44 | fearnor | cap: no. blank screen. |
| 02:04.46 | _zigo__ | ManxPower :) |
| 02:04.50 | fearnor | as in, logo, then nothing. |
| 02:04.58 | captrb | fearnor: more than one does it? |
| 02:05.11 | ManxPower | _zigo__, Abusing users that deserve it is one of the joys of helping the people that need help. |
| 02:05.13 | fearnor | 4 of dem |
| 02:05.19 | captrb | fearnor: used or new? |
| 02:05.22 | fearnor | could be my PoE switch |
| 02:05.30 | fearnor | hrm, bought from voipsupply, bought as new :) |
| 02:05.34 | ManxPower | fearnor, Plug them into the wall power |
| 02:05.43 | fearnor | do you know voltage they need? |
| 02:05.45 | captrb | fearnor: yeah, really |
| 02:05.50 | ManxPower | fearnor, VoipSupply has been good to me in the past. |
| 02:05.52 | fearnor | 12V? |
| 02:05.55 | captrb | fearnor: I'm not using PoE |
| 02:06.09 | ManxPower | fearnor, You didn't get the power supplies with them? |
| 02:06.12 | fearnor | cap: what voltage does it need? voipsupply didn't ship me wall wart |
| 02:06.15 | fearnor | nope |
| 02:06.19 | captrb | fearnor: try it with wall power and without them plugged into the network |
| 02:06.21 | fearnor | just non-poe cable but no wallwart |
| 02:06.26 | captrb | fearnor: oh |
| 02:06.32 | ManxPower | fearnor, Ah. Call them up and order a wall wart. |
| 02:06.36 | fearnor | what voltage is wall wart should be? |
| 02:06.39 | fearnor | 12v? |
| 02:06.42 | ManxPower | That way you'll have at least one power supply. |
| 02:06.58 | ManxPower | fearnor, If you use a non-approved power supply you void the warrenty and can blow up the phone. don't do it. |
| 02:07.04 | fearnor | doh |
| 02:07.09 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 02:07.13 | ManxPower | Just say no! |
| 02:07.13 | Sedorox | fearnor: sure you ordered the phone with the wallwort.. if you did... email/call them |
| 02:07.43 | captrb | fearnor: 12vdc 400mA lps |
| 02:07.44 | fearnor | sedorox: well...it was supposed to be 'like new' :) |
| 02:07.50 | fearnor | cap: gracias |
| 02:08.05 | Sedorox | yea.. but to cisco |
| 02:08.11 | Sedorox | 'new' doesn't include wallwart |
| 02:08.17 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 02:08.21 | fearnor | true enough |
| 02:08.26 | Sedorox | so you would have to look on the site where you ordered it to see if it includes it |
| 02:08.36 | Sedorox | most of the time they do.. but you never know.. coulda ordered the other one |
| 02:08.38 | fearnor | muchos gracias |
| 02:08.57 | captrb | fearnor: mine came with them... but no PoE cable. must be different package |
| 02:09.15 | ManxPower | fearnor, You know that polycom has different PoE cables for Cisco .vs. 802.3af PoE, right? |
| 02:09.28 | fearnor | yeah |
| 02:09.31 | fearnor | i have proper af cable |
| 02:09.31 | captrb | I'm REALLY impressed with the IP500's so far |
| 02:09.42 | fearnor | but a ghetto af switch |
| 02:09.52 | ManxPower | Sedorox, Cisco is one of the vew vendors that does NOT include a wall wart |
| 02:09.56 | fearnor | i actually have non-af cables too |
| 02:10.13 | ManxPower | fearnor, And polycom doesn't do PoE in the phone, it does it in the cable. |
| 02:10.21 | Sedorox | hmmm |
| 02:10.23 | *** part/#asterisk bbledsoe (nobody@dhcp-69-43-0-252.pitbpama-max5.dialup.citynet.net) |
| 02:10.35 | ManxPower | Well, for the 300 and 500. The 600 has PoE built in. |
| 02:10.38 | captrb | only ip600's, I think. |
| 02:10.52 | fearnor | thanks, i'm aware |
| 02:11.29 | Asskick | so what u guys recommend for a cisco phone.. use skinny support or rather try to get a sip firmware for it to connect to asterisk? |
| 02:11.58 | *** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm) |
| 02:12.16 | Slainte | anyone know what this means? |
| 02:12.17 | Slainte | WARNING[1133]: PRI: !! No channel map, no channel, and no ds1? What am I supposed to identify? |
| 02:12.30 | *** join/#asterisk iq[tablet] (~iq@70-59-163-109.omah.qwest.net) |
| 02:13.35 | *** join/#asterisk ross_cav (~ross_cav@60-240-47-244.tpgi.com.au) |
| 02:14.16 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:14.21 | captrb | Slainte: don't know, sorry. |
| 02:14.52 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 02:14.52 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:15.11 | captrb | does anybody happen to use a PRI T1 with the bchannels split by a Adtran 600R? |
| 02:15.21 | wdatkinson | Can anyone give me a hand setting up the follow-me config from voip-info.org? |
| 02:15.32 | *** join/#asterisk jmac (~dj@pc-24-181-187-85.sbi.ct.charter.com) |
| 02:18.15 | Wazb | i am getting chan_oss.c:269 sound_thread, can anyone please tell how can i resolve this error |
| 02:21.18 | ManxPower | Asskick, I suggest you sell it back on ebay and get a polycom or, if you are a cheap bastard, get a SIPura SPA-841, but be sure to upgrade the (free) formware. |
| 02:21.48 | ManxPower | Wazb, other process may be using asterisk, or you are running asterisk as non-root |
| 02:21.52 | *** join/#asterisk |nix (~inix@202.148.164.48) |
| 02:21.53 | *** join/#asterisk Silik0n (~krice@rso.suspicious.org) |
| 02:22.03 | jmac | you wouldn't be talking about the clone fxo cards would you? |
| 02:22.11 | Slainte | Manx no idea about my PRI error above? |
| 02:23.00 | Supaplex | what did professor google suggest? |
| 02:23.00 | jmac | NFBBCA 02 |
| 02:23.03 | |nix | hi, i would want all calls of channel 8 to go do extention 102, and i wrote a gotoif string that don't seem to be working.. would anyone be kind enuf to help on this? |
| 02:23.05 | |nix | exten = s,1,GotoIf($[${CHANNEL:4:2} = 4]?local-extensions,102,1:) |
| 02:23.22 | Slainte | Supa, only thing was one person said they changed their motherboard for a similar error |
| 02:23.45 | Qwell | |nix: ${CHANNEL:4:2}? |
| 02:24.00 | |nix | Qwell: i even tried CHANNEL:8:1 |
| 02:24.28 | |nix | i've got 3 tdm04b on my PBX |
| 02:24.36 | Qwell | Is that a valid variable syntax? |
| 02:24.40 | Juxt | is there some sort of software with which i could track ping latency over a period of say 24 hours and log it? |
| 02:24.41 | Wazb | i am running Asterisk as root |
| 02:25.06 | *** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3972900.sympatico.ca) |
| 02:25.08 | Juxt | well i guess i could just do ping <host> >> log.txt |
| 02:25.14 | Slainte | Juxt, what interval do you want? |
| 02:25.21 | ManxPower | jmac, in my professional opinion you have something seriously fucked up. |
| 02:25.25 | Juxt | like every 10 minutes? |
| 02:25.30 | Slainte | BigBrother, Nagios, |
| 02:25.38 | Slainte | MRTG, Cricket, Cacti |
| 02:25.43 | *** join/#asterisk blankman (~blankman@c-24-61-108-24.hsd1.nh.comcast.net) |
| 02:25.49 | Qwell | |nix: What does ${CHANNEL:4:2} return? |
| 02:25.51 | blankman | Hey guys ... |
| 02:26.02 | Qwell | Seems like it would return ...a string with a length of... negative 2 |
| 02:26.03 | ManxPower | jmac, put your /etc/zaptel.conf and /etc/asterisk/zapata.conf on pastebin.ca |
| 02:26.19 | blankman | So, I have a strange issue with Global Crossing ... and I wanted to know if anyone else has had this problem. |
| 02:26.36 | Slainte | Blankman, yes they ar ecrap and I lsot a bunch in their stock dive |
| 02:26.43 | jmac | manx: not sure what you're talking about, i was just kinda bsing |
| 02:26.53 | _zigo__ | I used to have the following on the CVS version, how can I do same with the stable one ? |
| 02:26.53 | _zigo__ | exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN:0}@sip-account1,30) |
| 02:26.53 | _zigo__ | exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN:0}@sip-account2,30) |
| 02:27.06 | |nix | Qwell: i'll check that out |
| 02:27.09 | |nix | but any quick fix? |
| 02:27.20 | blankman | I need to "delay" the pulse dialing of the e and m wink ... namely ... I am using a te410 with em_w |
| 02:27.25 | ManxPower | jmac, Sorry, wrong person. |
| 02:27.25 | marc324 | root locus |
| 02:27.28 | |nix | i'm just keen on having channel 8, or zap/g4 calling a specific extension |
| 02:27.32 | jmac | no worries |
| 02:27.34 | |nix | straight |
| 02:27.42 | ManxPower | Slainte, in my professional opinion you have something seriously fucked up. |
| 02:27.47 | ManxPower | Slainte, put your /etc/zaptel.conf and /etc/asterisk/zapata.conf on pastebin.ca |
| 02:27.54 | Slainte | Okily dokily |
| 02:27.55 | zilas | ztcfg -v : ztchanconfig failed on channel 1: invalid argument (22). ???? |
| 02:27.58 | blankman | When the GC t1 goes off-hock the digium card is pulsing the digits ... problem is that GC still hasn't completed the wink ... |
| 02:28.04 | _zigo__ | (I hope I don't abuse too mutch asking stupid questions...) |
| 02:28.16 | ManxPower | blankman, play with prewink |
| 02:28.17 | Qwell | |nix: What does ${CHANNEL} return? NoOp it |
| 02:28.39 | blankman | ManxPower, okay, I tried that ... but I can't figure out what to do with it :-( |
| 02:28.51 | |nix | Qwell: i'm new with this asterisk thingni, sorry but i don't really know what you mean |
| 02:29.03 | Qwell | |nix: README.variables |
| 02:29.06 | Qwell | read that, then comes back |
| 02:29.09 | |nix | ok |
| 02:29.10 | |nix | thanks |
| 02:29.20 | zilas | did you forget that FXS interfaces are configured with FXO signalling and FXO interfaces use FXS signalling. What the hell that suppose to mean? |
| 02:29.23 | blankman | I set it to: prewink=200 ... cause they said they needed 200ms form the time they "go off hook" ... it does take it's values as ms right? |
| 02:29.41 | Qwell | zilas: exactly what it says... |
| 02:29.55 | Slainte | Mank, http://pastebin.ca/8981 zttool shows no errors, all modules load fine |
| 02:30.02 | blankman | ManxPower, so prewink=200... |
| 02:30.10 | Slainte | Manx sorry |
| 02:30.18 | zilas | qwell: smart :) :) :) |
| 02:31.25 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 02:31.34 | blankman | I also tried to play with txwink, rxwink, and start ... all to know with no luck ... I also tried to set a few www in the dial to make it "hold" the dial .. but he system is still just going off hook and dialing ... |
| 02:31.38 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-217.modem.logical.net) |
| 02:31.49 | ManxPower | blankman, I was thinking more like increase by %30 over the devault. |
| 02:32.11 | ManxPower | blankman, We had a problem where our provider was expecting longer winks than what we were sending. |
| 02:32.19 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 02:32.55 | ManxPower | blankman, We needed: |
| 02:32.56 | ManxPower | wink=270 |
| 02:32.56 | ManxPower | rxwink=270 |
| 02:32.58 | Carp1 | damn, I'm trying to install Asterisk on a new box and its a pain. |
| 02:33.01 | Carp1 | downlaoded gcc |
| 02:33.08 | Carp1 | but gcc needs like 3 other files to install |
| 02:33.13 | Carp1 | 3 other packages* |
| 02:33.18 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 02:33.24 | dan2 | bkw_: ping |
| 02:33.25 | ManxPower | Carp1, Um, use the package management your distro provides. |
| 02:33.37 | Carp1 | I don't know much linux at all. |
| 02:33.41 | Carp1 | I'm on RH8. |
| 02:33.50 | Qwell | Why so low? |
| 02:33.57 | Carp1 | 56k :( |
| 02:33.58 | Qwell | old...whatever |
| 02:34.06 | ManxPower | Carp1, I sugest you spend a couple of hundred $ on books from www.ora.com and amazon.com |
| 02:34.14 | blankman | ManxPower, okay .. I will try that one :-) but I still am curious ... they are getting our wink, it is just that we are "waiting" for it before we dial ... |
| 02:34.23 | jmac | could anyone recommend a good a@h article aside from kerry garrison's piece? i'm a little stumped |
| 02:35.03 | ManxPower | blankman, I think tzanger may bave been the one to help me with my winking problems. |
| 02:36.37 | Qwell | Don't message me |
| 02:36.49 | *** part/#asterisk dave_mwi_ (~dave_mwi@adsl-068-153-207-210.sip.bct.bellsouth.net) |
| 02:36.50 | blankman | ManxPower, okay, I will wait till he is on ... he is in NZ right? |
| 02:37.01 | ManxPower | blankman, Toronto\ |
| 02:37.10 | ManxPower | ~seen tzanger |
| 02:37.12 | jbot | tzanger is currently on #asterisk (3h 2m 50s). Has said a total of 1112 messages. Is idling for 2h 43m 3s |
| 02:37.29 | Slainte | ManxPower, http://pastebin.ca/8981 zttool shows no errors, all modules load fine |
| 02:37.42 | ManxPower | blankman, copiece helped me with something too, I think it was with my Kewlstart problems. |
| 02:37.49 | blankman | ManxPower, ... toronto ... NZ same both part of the crown ;-) |
| 02:38.11 | blankman | ~seen copiece |
| 02:38.13 | jbot | i haven't seen 'copiece', blankman |
| 02:38.16 | ManxPower | Slainte, what does zttool say? |
| 02:38.26 | Qwell | jesus people |
| 02:38.28 | Slainte | no alarms, all is good |
| 02:38.29 | Qwell | Don't message me |
| 02:38.40 | Qwell | That doesn't mean a SECOND person can message me...wtf |
| 02:38.51 | |nix | woops |
| 02:38.53 | |nix | sorry |
| 02:38.59 | |nix | Qwell: i've did the noop |
| 02:39.00 | |nix | <PROTECTED> |
| 02:39.00 | |nix | Apr 7 10:37:38 WARNING[16723]: pbx.c:1599 pbx_extension_helper: No application 'NoOp,GotoIf' for extension (incoming, s, 1) |
| 02:39.00 | |nix | <PROTECTED> |
| 02:39.00 | |nix | <PROTECTED> |
| 02:39.01 | |nix | Apr 7 10:37:38 DEBUG[16723]: chan_zap.c:2420 zt_answer: Took Zap/8-1 off hook |
| 02:39.03 | |nix | Apr 7 10:37:38 DEBUG[16723]: chan_zap.c:1327 zt_enable_ec: Enabled echo cancellation on channel 8 |
| 02:39.05 | |nix | Apr 7 10:37:38 DEBUG[1 |
| 02:39.09 | Qwell | ... |
| 02:39.10 | blankman | ManxPower, on a different note ... it seems odd that we can "hold" up a dial some how no? |
| 02:39.17 | Qwell | ManxPower: He's all yours |
| 02:40.00 | *** join/#asterisk mog_home (~mog_home@146.229.181.169) |
| 02:40.16 | ManxPower | going to buy booze. bibiaw |
| 02:40.28 | ManxPower | coppiece is in HK |
| 02:40.40 | _zigo__ | Is there (by some magical chance) some doc somewhere telling the differences on the extension.conf form CVS and the 1.0.7 ? ... I think I'm dreaming, I don't do it for my own projects... |
| 02:40.59 | blankman | ManxPower, you know what debounce and start are for in the zapata.conf? |
| 02:41.50 | blankman | ~seen coppiece |
| 02:41.52 | jbot | blankman: i haven't seen 'coppiece' |
| 02:42.08 | NormAst | ~NormAst |
| 02:42.39 | blankman | bkw_ ... you don't know of away to "hold" the dial on the zap channel do you? |
| 02:42.42 | Carp1 | I know! |
| 02:42.44 | Carp1 | I will try yum |
| 02:42.52 | Carp1 | I couldnt think of the name for like 2 days lol. |
| 02:42.53 | Slainte | blankman, what are you trying to do? |
| 02:42.55 | Qwell | Carp1: RH8 doesn't have yum |
| 02:43.07 | Carp1 | I'm downloading it. |
| 02:43.07 | blankman | Get the digital t1 to work with GC ... |
| 02:43.12 | Carp1 | I used it on RH8 before. |
| 02:44.03 | Slainte | Blankman do you need a pause? |
| 02:44.10 | Slainte | like 2 seconds? |
| 02:44.10 | blankman | They are saying (and I belive them from "listening" to the dialing on the t-bird) that the system is getting the off hook and immediately out pulsing the number ... not waiting for the wink to end. |
| 02:44.27 | blankman | Slainte, nope... 200ms would do it :-) |
| 02:44.36 | Slainte | blankman very easy to do. |
| 02:44.39 | blankman | Slainte, I did try the w |
| 02:44.44 | ManxPower | blankman, I don;t know |
| 02:44.52 | blankman | ManxPower, Thanks :-) |
| 02:45.05 | blankman | ManxPower, once I know I will update the wiki. |
| 02:45.15 | Slainte | and the w did not help you? |
| 02:45.20 | blankman | Slainte, is that what you were thinking? |
| 02:45.20 | KristinG | thank goodness this day is about over |
| 02:45.33 | blankman | Slainte, the w in the dial? |
| 02:45.46 | Slainte | Well I have another way you can stick the pause in |
| 02:46.42 | KristinG | t-berds are fun :) |
| 02:49.55 | iq[tablet] | Hi, any document/link to starting asterisk development? CLI commands, etc. ? Just want to see if I can do anything useful :) |
| 02:52.37 | blankman | iq[tablet], you can look at the www.asterisk.org. |
| 02:53.29 | iq[tablet] | blankman, I do every day :) |
| 02:56.07 | blankman | So, the links there will give you what you need ... I think ... |
| 02:57.03 | iq[tablet] | blankman, thanks ... I'll do that |
| 02:57.09 | _zigo__ | The CVS version seemed to work a lot better than the stable one... :( |
| 02:59.02 | iq[tablet] | _zigo__, whats wrong with stable? |
| 02:59.21 | iq[tablet] | _zigo__, did you install stable on top of head? |
| 02:59.41 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 02:59.42 | _zigo__ | iq[tablet]: I'm afraid I did, because I didn't find any uninstall script. |
| 02:59.57 | iq[tablet] | _zigo__, were you getting some module error ;) ? |
| 03:00.10 | _zigo__ | iq[tablet]: No, that part is done already ! :) |
| 03:00.12 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 03:00.29 | _zigo__ | I'm have 2 same section in my extension.conf |
| 03:00.39 | _zigo__ | (copy of them) |
| 03:00.46 | _zigo__ | And they don't produce the same result !!! |
| 03:00.55 | iq[tablet] | _zigo__, okay, most people (including myself) forget to remove modules |
| 03:01.07 | _zigo__ | Am I dreaming or totaly disy ??? |
| 03:01.33 | *** join/#asterisk tessier (~treed@222.253.81.147) |
| 03:01.53 | iq[tablet] | :) |
| 03:02.26 | _zigo__ | http://pastebin.ca/8985 |
| 03:02.31 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 03:02.34 | Juxt | hello again |
| 03:02.39 | _zigo__ | The first one doesn't work, it's with broadvoice. |
| 03:02.50 | Juxt | how would i find out which codec has been negotiated for a sip connection? |
| 03:02.54 | Juxt | i did sip debug |
| 03:02.59 | _zigo__ | It goes DIRECTLY to my phone... |
| 03:02.59 | Juxt | but i can't see it, am i missing something |
| 03:03.04 | _zigo__ | The second one does... |
| 03:03.30 | Qwell | _zigo__: Does broadvoice go to s? |
| 03:03.42 | _zigo__ | Yes. |
| 03:03.59 | _zigo__ | But it does not play the welcome message... |
| 03:04.01 | Qwell | It goes straight to voicemail, or what? |
| 03:04.12 | _zigo__ | No, straight to Dial(à |
| 03:04.13 | Qwell | erm |
| 03:04.14 | _zigo__ | () |
| 03:04.41 | znoG | question. I have asterisk running on my linux FW (iptables) and i've allowed ports 5060(tcp/udp) as well as the RTP ports (10000 -> 20000). A SIP user connects to my * and they can hear me, but i can't hear them. What settings should I be playing with? |
| 03:04.48 | Qwell | is s,5(dial),Dial() valid? |
| 03:04.58 | _zigo__ | Ha... |
| 03:05.07 | _zigo__ | That might be remaining of the CVS version !!! :) |
| 03:05.25 | _zigo__ | Too bad... |
| 03:05.26 | _zigo__ | :( |
| 03:05.27 | Qwell | What is it supposed to do? |
| 03:05.35 | *** join/#asterisk hawaiianphoneguy (~mdarnell@66.135.226.125) |
| 03:05.45 | _zigo__ | I've copy/past that from an example on the voip-info.org wiki... |
| 03:05.53 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 03:05.54 | _zigo__ | (I'm not sure, but I think it's from there...) |
| 03:06.29 | Qwell | yeah, but what does it do? |
| 03:07.12 | _zigo__ | I don't realy know. |
| 03:07.16 | _zigo__ | :( |
| 03:07.23 | Slainte | you need to have a Command form the * command list, after the priority in a context |
| 03:07.29 | Slainte | (Dial) is not a cmd |
| 03:07.40 | Slainte | command from, |
| 03:09.00 | blankman | Hey, anyone know what featdmf is in the zapata.conf? I know it is a signaling ... but what kind? |
| 03:09.17 | _zigo__ | It does same even without the (Dial) ... :( |
| 03:09.17 | blankman | ~seen tzanger |
| 03:09.19 | jbot | tzanger is currently on #asterisk (3h 34m 57s). Has said a total of 1112 messages. Is idling for 3h 15m 10s |
| 03:09.34 | NormAst | in band DTMF broken in CVS head? |
| 03:14.19 | Juxt | are there any soft phones that support g729? |
| 03:14.57 | Slainte | Anyone compile * on Debian. I have an error configure: error: termcap support not found |
| 03:15.30 | _zigo__ | Slainte: I'm trying to compile on my woody currently. |
| 03:15.49 | _zigo__ | Slainte: saying app_queue.c:279: warning: unnamed struct/union that defines no instances |
| 03:15.52 | _zigo__ | to me... :( |
| 03:16.19 | _zigo__ | No, not that one... |
| 03:16.33 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 03:16.59 | _zigo__ | http://pastebin.ca/8986 |
| 03:18.52 | *** join/#asterisk moy (~moy@201.138.195.87) |
| 03:18.52 | iceyp | anyone here using LCDial? |
| 03:19.02 | iceyp | calling card app |
| 03:19.12 | *** join/#asterisk danfrey (user@24.229.232.63.res-cmts.mtp.ptd.net) |
| 03:19.24 | Slainte | iceyp it is not a calling card app. it is a Least Cost Dialing routine |
| 03:19.32 | danfrey | are there any quicknet pros here? |
| 03:20.01 | Essobi | Mmm. Anyone have an idea how to get c->cid.dnid into cdr_custom? |
| 03:20.04 | dsmouse | anyone know much about how sip works? not works with asterisk, but works low-level... |
| 03:20.17 | danfrey | I have a phonejack lite isa and would like to know how to use ulaw with it |
| 03:20.23 | Juxt | i used lcdial for a while but chose to make my own LCR in FastAGI |
| 03:21.00 | Essobi | Juxt Why? |
| 03:21.14 | Juxt | i had more extensive routing that it supported |
| 03:21.44 | Essobi | lol |
| 03:21.48 | *** join/#asterisk tessier (~treed@222.253.79.76) |
| 03:21.53 | Essobi | Did you really read the code? |
| 03:22.08 | Juxt | no i didn't read the code |
| 03:22.22 | Juxt | there was more than 1 reason tho |
| 03:22.26 | Juxt | i run on postgres |
| 03:22.32 | iceyp | can anyone suggest a good calling card app? |
| 03:22.33 | Essobi | I did. you can make it do anything you want. |
| 03:22.55 | Essobi | In fact.. I recoded lcdial to do quite a bit more then it does standardly. |
| 03:23.18 | Juxt | that's cool man |
| 03:24.08 | znoG | question. I have asterisk running on my linux FW (iptables) and i've allowed ports 5060(tcp/udp) as well as the RTP ports (10000 -> 20000). A SIP user connects to my * and they can hear me, but i can't hear them. What settings should I be playing with? |
| 03:24.11 | Essobi | And I don't use pgres. |
| 03:24.25 | Slainte | znoG, nothing to do with the ports |
| 03:24.37 | reallost1 | grr... asterisk behind nat... |
| 03:24.37 | Juxt | well i chose to run pgsql on production for more than 1 reason |
| 03:24.44 | Juxt | but write-ahead-logging is the primary one |
| 03:24.44 | Slainte | znoG, it is the sip.conf file, make sure you have the proper type set i.e friend peer etc. |
| 03:24.51 | Essobi | *SHRUG* |
| 03:24.56 | Essobi | I do fine with mysql. |
| 03:24.57 | Slainte | cheack the wiki for sip.conf and look at the type parameter |
| 03:25.15 | Essobi | 30+ million records in one table.. |
| 03:25.52 | Essobi | Besides.. I know how to tune queries/kernels/mysql, so it fits my bill. |
| 03:26.08 | Damin | ~say Touch my volume.. |
| 03:26.10 | jbot | Touch my volume.. |
| 03:26.21 | Juxt | hey everyone is free to chose their posion :-) |
| 03:26.26 | znoG | Slainte: its set to friend. they can register fine, its just making calls they're not heard, but they can hear fine. |
| 03:26.29 | Juxt | i like my triggers, subselects, etc. |
| 03:26.32 | _zigo__ | Slainte: Are you using woody ? |
| 03:26.39 | znoG | reallost1: asterisk isn't behind NAT in my case. It's on the firewall |
| 03:26.45 | Slainte | zigo, sarge |
| 03:27.00 | *** part/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 03:27.00 | *** part/#asterisk danfrey (user@24.229.232.63.res-cmts.mtp.ptd.net) |
| 03:27.01 | Slainte | znog, can they actually make a call? |
| 03:27.03 | znoG | reallost1: not sure if you were aiming what you said at me. :) |
| 03:27.05 | fugitivo | mysql is for personal webpages, hehe |
| 03:27.07 | _zigo__ | Slainte: I bet it's less trouble... |
| 03:27.12 | reallost1 | znoG, no I'm having problems with a client who has asterisk behind a firewall and wants remote sip clients. |
| 03:27.21 | _zigo__ | Slainte: Right ? |
| 03:27.24 | Slainte | zigo I am having a horrible time. It is all borken and I dont knwo why |
| 03:27.27 | znoG | Slainte: yep that's what im saying, they can make calls. It's just they're not heard. |
| 03:27.34 | znoG | reallost1: ah, looked into a SIP proxy? |
| 03:27.38 | reallost1 | znoG, the problem is similar though, the calls connect, but nobody is heard. |
| 03:27.42 | _zigo__ | Slainte: Did you had some trouble with app_queue.c ? |
| 03:27.49 | Slainte | znoG is it NATted? |
| 03:27.52 | Slainte | zigo no |
| 03:27.58 | reallost1 | znoG, they are running a cheap linksys firewall. |
| 03:28.05 | znoG | reallost1: in my case (I'm on the asterisk side) the other person can hear me, i can't hear them. |
| 03:28.10 | _zigo__ | Slainte: Mine here didn't want to compile... |
| 03:28.22 | reallost1 | znoG, maybe I can switch them out to a better firewall. |
| 03:28.27 | znoG | Slainte: as i said, no. the remote user is not behind NAT and on my side, * is not behind NAT. |
| 03:28.30 | Essobi | Juxt Triggers are nice. Subselects are in mysql. |
| 03:28.41 | Slainte | zigo, mine compiled fine yesterday, I did an apt-get upgrade at some point and now, it complains about termcap |
| 03:28.41 | Essobi | triggers are in 5.. but I wouldn't run that in production. |
| 03:28.50 | _zigo__ | Now it's CURL... :( |
| 03:28.54 | znoG | reallost1: yeah, otherwise it might get tricky |
| 03:29.21 | Slainte | znoG, I hav enot scrolled up to see everything you have said, sorry. Regardless, turn the iptables off to prove it is not the iptables |
| 03:29.51 | Essobi | Juxt I like merged table views. :) |
| 03:30.01 | hawaiianphoneguy | anyone know how to prompt a SIP user to enter an account code when they dial long distance? |
| 03:30.12 | Essobi | znoG sounds like your other party has an rtp problem.. fire up tethereal and see if it ever gets to you |
| 03:30.25 | _zigo__ | Slainte: I have the choice with 1/ CVS sources that don't compile anymore with my woody and 2/ an * 1.0.7 that wont accept my extension.conf and 3/ an old CVS version that don't accept Broadvoice inbound calls anymore... :( |
| 03:30.33 | _zigo__ | I'm having a very hard time too !!! |
| 03:30.58 | Slainte | zigo, why wont the 1.0.7 take your extensions.conf? |
| 03:32.16 | _zigo__ | Slainte: With same 2 parts of my extension.conf, it doesn't produce the same result, I don't know why ! |
| 03:34.16 | Qwell | _zigo__: RUATA |
| 03:34.35 | _zigo__ | Qwell: what ??? |
| 03:34.43 | Qwell | replace user and try again |
| 03:35.17 | _zigo__ | What users ? |
| 03:35.22 | Qwell | You. |
| 03:35.32 | _zigo__ | :))) |
| 03:35.45 | _zigo__ | I'm a better coder than admin, sorry ! :) |
| 03:37.45 | TheEmperor | anyone know how to configure zaptel.conf and zapata.conf for an e1? |
| 03:41.01 | *** part/#asterisk moy (~moy@201.138.195.87) |
| 03:41.11 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 03:41.23 | DEEZED | does anyone here use iax.cc/sixtel? |
| 03:41.43 | file[laptop] | slowly am I going crazy |
| 03:42.02 | *** join/#asterisk tessier (~treed@222.253.76.53) |
| 03:48.07 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 03:48.36 | Qwell | file[laptop]: I'll get you a job at my work...quicken it up a bit |
| 03:48.52 | file[laptop] | Qwell: too late ... *SNAP* |
| 03:53.04 | Corydon76-home | file[laptop]: woohoo... kinky... |
| 03:54.29 | file[laptop] | ooh la la |
| 03:54.56 | Wazb | where i can find all conf file entries in * |
| 03:57.02 | Carp1 | . /etc/asterisk |
| 04:00.39 | Wazb | sorry i want to ask is there any file which contains all conf filename entires |
| 04:01.18 | Slainte | your asterisk.conf is the start and you can call other files as needed. But certain modules look for certain files |
| 04:01.22 | Slainte | if you load the module you need the file. |
| 04:01.31 | Slainte | for example meetme module need the meetme.conf |
| 04:01.39 | Slainte | sip module needs sip.conf |
| 04:03.15 | Wazb | thanks |
| 04:05.48 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 04:08.24 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 04:10.50 | *** part/#asterisk marc324 (~marc32344@69-90-36-26.dsl.teksavvy.com) |
| 04:12.06 | *** join/#asterisk marc324 (~marc32344@69-90-36-26.dsl.teksavvy.com) |
| 04:13.20 | blankman | Hey, what is the zapata in the cvs for ... meaning why is it there if you don't need it to build? |
| 04:13.32 | blankman | There is no read me on ... |
| 04:13.37 | *** join/#asterisk sudhir492 (~sudhir@4.8.141.4) |
| 04:14.30 | sudhir492 | Hi all |
| 04:15.37 | *** join/#asterisk Cinen (~srash@cpe-065-188-184-163.triad.res.rr.com) |
| 04:19.55 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 04:21.06 | ariel_ | sudhir492, hello |
| 04:21.35 | ariel_ | blankman, you don't but if you need to dial via a pots line or complite ztdummy for timing meetme you need the zapata |
| 04:21.48 | sudhir492 | hello ariel_ |
| 04:27.22 | ariel_ | slow night |
| 04:27.56 | Sedorox | si |
| 04:32.49 | sudhir492 | yes, very slow night |
| 04:34.02 | sudhir492 | Is it possible to have an asterisk box dual hosted |
| 04:34.14 | Sedorox | dual hosted? |
| 04:34.27 | sudhir492 | I mean have two IP address on two separted LA |
| 04:34.30 | sudhir492 | LAN |
| 04:34.45 | marc324 | do you need dual cpu, for running two te410 cards |
| 04:35.04 | sudhir492 | I need two network interfaces. |
| 04:35.37 | blankman | ariel_, thanks ... so it is only if you don't have a zap card right? |
| 04:35.51 | Sedorox | sudhir492: what would prevent you from putting two nicks in the box? |
| 04:36.29 | ariel_ | blankman, no you need it if you have a zaptel and also if you need to use ztdummy for meetme and iax2 trunking. |
| 04:36.40 | sudhir492 | Sedorox: Nothing would prevent me from that. However, I remember reading somewhere that Asterisk has problem in those cases |
| 04:36.56 | Sedorox | I wouldn't think so... |
| 04:37.00 | blankman | ariel_, I guess I am confussed ... what is ztdummy used for? |
| 04:37.23 | sudhir492 | Sedorox: That is quite encouraging ! |
| 04:37.29 | ariel_ | timing for meetme and iax2 trunks when you don't have a zaptel hardware installed like a x101p |
| 04:37.46 | Sedorox | lol |
| 04:38.05 | Sedorox | from looking at it from a networking point of view.. it shouldn't cause any problems |
| 04:38.16 | Sedorox | unless you have a shitload of phones on it.. and you don't have a fast enough box... |
| 04:39.19 | blankman | ariel_, thanks. |
| 04:39.25 | sudhir492 | Of course. If one does not have fast enough box, the box will choke with 1 NIC too :-) |
| 04:39.31 | Sedorox | true |
| 04:40.10 | Sedorox | this is odd |
| 04:40.18 | Sedorox | exten => _2[60-89],1,Dial(IAX2/stormy@mercury/${EXTEN},,rtT) |
| 04:40.21 | Sedorox | doesn't wanna work |
| 04:40.28 | blankman | So, before I go off and wade through all the zaptel code ... does anyone on list right now, know of a way to make the em_w setting for the zaptel (te410) wait for the response wink from the provider? |
| 04:40.31 | Sedorox | thought you could do that... |
| 04:41.11 | sudhir492 | Is there a network cable for Polycom 500 that does not need power supply. In case one has 802.3af compliant PoEthernet switch. |
| 04:41.57 | sudhir492 | The current cable is a big PITA. The cable is not standard, at the same time does not obviate the need of a power supply either |
| 04:42.18 | blankman | Sedorox, try _2[6-8]X |
| 04:43.26 | Qwell | Sedorox: I think yours is 6, 0 through 8, OR 9 |
| 04:43.37 | Sedorox | hmmm |
| 04:43.43 | Sedorox | I want... |
| 04:43.48 | Qwell | 60 through 89? |
| 04:43.53 | Sedorox | 260-289 to goto one server |
| 04:44.00 | Sedorox | 230-259 to another server |
| 04:44.06 | Qwell | yeah, probably gonna be a little hack |
| 04:44.07 | Sedorox | 200-229 to another server |
| 04:44.08 | Sedorox | etc.. |
| 04:44.11 | Qwell | Would be useful though |
| 04:44.40 | Qwell | might not be too difficult to code up a new syntax for that... |
| 04:44.40 | Sedorox | blankman's thing worked |
| 04:44.42 | Sedorox | :) |
| 04:44.50 | blankman | Sedorox, so your need to use: _2[6-8]X, _2[3-5]X and _2[0-2]X |
| 04:44.54 | Sedorox | yea |
| 04:45.12 | Sedorox | now I know... |
| 04:45.18 | Qwell | I was thinking more like 20-35, 36-58, 59-92... |
| 04:45.22 | blankman | Sedorox, so now you can fix my problem right ;-) |
| 04:45.22 | Sedorox | I was thinking it would go 60-89... then I just looked at the docs again |
| 04:45.24 | sudhir492 | Sedorox _2[0-2]X, _2[3-5]X, _2[6-8]X should work |
| 04:45.34 | Sedorox | I wunno about the wink |
| 04:45.35 | Qwell | Would be nice to be able to do [60:89] or something |
| 04:45.36 | Sedorox | dunno* |
| 04:45.39 | Sedorox | hehe |
| 04:45.54 | Sedorox | Qwell: no.. I'm going by 30 |
| 04:46.00 | Sedorox | including 0's |
| 04:46.02 | Sedorox | making it esy |
| 04:46.02 | Qwell | Sedorox: yeah.. |
| 04:46.04 | Sedorox | easy |
| 04:46.14 | Sedorox | sudhir492: yup.. we got it.. ehhe thanks tho :-p |
| 04:46.21 | chaoscon | well somewhat |
| 04:46.22 | chaoscon | :P |
| 04:46.25 | Sedorox | lol |
| 04:46.27 | chaoscon | 290 still doesn't work |
| 04:46.38 | Sedorox | 'cause I didn't finish yet |
| 04:46.39 | Sedorox | duh |
| 04:46.39 | Sedorox | :-p |
| 04:46.43 | chaoscon | :P |
| 04:46.47 | Qwell | chaoscon: 290-299 = _29X |
| 04:46.52 | Sedorox | geez.. be patient man! |
| 04:46.59 | chaoscon | Qwell: Sedorox is our VoIP tech ;) |
| 04:47.09 | blankman | ~seen tzanger |
| 04:47.10 | jbot | tzanger is currently on #asterisk (5h 12m 48s). Has said a total of 1112 messages. Is idling for 4h 53m 1s |
| 04:47.11 | Qwell | What happens when you want to span 290-319? |
| 04:47.47 | Qwell | I'm trying to make a case for spans like that, somebody help me out :P |
| 04:48.09 | Sedorox | well.. the last 10 90-99.. I have for globals.. then 300-XXX is for external dialing... |
| 04:48.10 | Sedorox | but.. |
| 04:49.09 | Sedorox | _[2-3][91]X. |
| 04:49.09 | Sedorox | ? |
| 04:49.28 | Qwell | not quite, heh |
| 04:49.33 | Sedorox | hehe |
| 04:49.37 | Sedorox | seems like it would sork |
| 04:49.50 | blankman | Qwell, myExten,1, _2[9]X,1,Goto(myExten,1), _3[0,1]X,1,Goto(myExten,1) |
| 04:49.57 | Qwell | 210, 211, 21..., 290, 291, 29..., 310, 311, 31... |
| 04:50.34 | Qwell | blankman: yeah, perhaps |
| 04:50.50 | Sedorox | I didn't know you could double up like that... |
| 04:50.52 | Sedorox | hmmm |
| 04:50.59 | Qwell | like what? |
| 04:51.05 | blankman | Qwell, the best way is to make it a macro ... |
| 04:51.16 | Sedorox | how he had it.. with _2[9] and _3[0,1] |
| 04:51.25 | Qwell | Sedorox: Thats two different lines |
| 04:51.43 | blankman | Qwell, that way you can set the macro exit result and control your DP better. |
| 04:51.47 | Sedorox | well he had a , after ).. so I figured it was the same :-p |
| 04:51.50 | Sedorox | anyway |
| 04:52.01 | Sedorox | gotta fix the extentions before chaoscon chews me a new one |
| 04:52.06 | chaoscon | haha |
| 04:52.12 | chaoscon | now I wouldn't do that |
| 04:52.13 | blankman | Sedorox, oh ... sorry that was my short hand for you need three extension in the plan. |
| 04:52.14 | chaoscon | nah* |
| 04:52.55 | Sedorox | hehe |
| 04:53.01 | Sedorox | yea.. I figured that what he would have to do... |
| 04:53.04 | Sedorox | chaoscon: should work now |
| 04:53.14 | chaoscon | yay |
| 04:53.19 | blankman | Sedorox, basically, make the extensions as abstract as possible then send the specific ones to them :-) |
| 04:53.37 | Sedorox | yea |
| 04:53.43 | Sedorox | well we have three servers |
| 04:54.06 | Sedorox | the three different sections of 30 numbers goes to each server |
| 04:54.17 | blankman | After a time you will learn to like PGSQL app ... makes life much easier for the extension :-) |
| 04:54.17 | Sedorox | the last 10 are for globals (vmail. meetme.. etc) |
| 04:54.19 | Sedorox | so... |
| 04:54.21 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 04:54.23 | Sedorox | lol |
| 04:54.40 | chaoscon | Sedorox: will be MUCH better when its centralized ;) |
| 04:54.42 | Wazb | any idea how can i use gatekeeper for H323 |
| 04:55.11 | blankman | K. well I have to go all ManxPower if your still on thanks for your help earlier. |
| 04:55.20 | Sedorox | chaoscon: well it will be staying that way :-p |
| 04:55.28 | blankman | Slainte, I will let you know how it goes, thanks to. |
| 04:55.40 | Sedorox | just maybe the one context expanded into the other |
| 04:55.43 | Slainte | np |
| 04:56.33 | Sedorox | I see chaoscon |
| 04:56.34 | Sedorox | :-p |
| 04:56.39 | Sedorox | fuck |
| 04:56.40 | Sedorox | brb |
| 04:56.41 | Sedorox | ops... |
| 04:56.43 | Sedorox | brb |
| 04:56.53 | chaoscon | lol |
| 04:57.17 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:58.05 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 04:58.16 | Sedorox | I have to remove that logout button from my menu |
| 04:58.29 | chaoscon | lol |
| 04:59.04 | chaoscon | well you will have a whole week to play without me buggin ya :P |
| 04:59.31 | Sedorox | eh |
| 04:59.46 | Sedorox | it should be done once I finish fixing the error (very easy to fix) |
| 04:59.52 | Sedorox | and the matching extention |
| 04:59.59 | Sedorox | which yours is done.. so... |
| 05:01.50 | Sedorox | chaoscon: that extention that you tried.. should work now |
| 05:02.21 | chaoscon | :) |
| 05:02.31 | Qwell | spying on your calls :p |
| 05:02.40 | chaoscon | we both watch the console |
| 05:02.45 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 05:02.51 | Qwell | with zapbarge I mean |
| 05:02.53 | odie_flocon | Hey alll. |
| 05:03.01 | Sedorox | lol |
| 05:03.06 | Sedorox | we run -vvvvvc in a screen |
| 05:03.16 | Qwell | hmm |
| 05:04.06 | Qwell | Is it possible to connect more then one console to * at a time? asterisk -r |
| 05:04.28 | Qwell | looks like you can |
| 05:04.28 | Sedorox | dunno |
| 05:05.08 | odie_flocon | you have more then one copy of * running on your pc at one time? |
| 05:05.28 | Qwell | no, the same instance |
| 05:05.30 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
| 05:05.40 | afrosheen | mazeltov! |
| 05:05.46 | odie_flocon | so why would you need 2 consoles? |
| 05:05.58 | Qwell | odie_flocon: Why not? |
| 05:06.12 | Qwell | I leave one connected, I'm at work, I want to check things out. |
| 05:06.19 | Qwell | If I was limited to 1, I'd be stuck |
| 05:06.27 | Sedorox | not with screen :-p |
| 05:06.29 | Sedorox | but anyway |
| 05:06.44 | florz | ... nur with kill =:-) |
| 05:06.45 | Qwell | screen isn't always ideal |
| 05:06.49 | florz | s/nur/nor/ |
| 05:06.49 | odie_flocon | ok now you've given a reason why. |
| 05:07.48 | odie_flocon | have you tried multiple connections? |
| 05:08.06 | Qwell | yeah, it worked fine |
| 05:09.12 | odie_flocon | ok |
| 05:09.37 | odie_flocon | I want to monitor * |
| 05:09.44 | odie_flocon | kinda like a heartbeat monitor. |
| 05:11.43 | Sedorox | Hmmmm |
| 05:11.55 | Sedorox | seems that FWD won't do new registers today.. but are fine for currents... |
| 05:12.08 | afrosheen | odie_flocon: what, you don't trust it? ;0 |
| 05:12.09 | Sedorox | viLeR: is the second person I've talked to today having problems registering... |
| 05:12.12 | Sedorox | I'm fine.. |
| 05:12.24 | odie_flocon | actually, |
| 05:12.34 | odie_flocon | I have a small house install. |
| 05:12.51 | odie_flocon | and I want to keep this thing up and running all the time. |
| 05:13.21 | odie_flocon | I do trust * |
| 05:13.26 | afrosheen | odie_flocon: if you're using zaptel hardware just set a cron job to make it reboot nightly, it'll work forever |
| 05:13.43 | odie_flocon | reboot the machine nightly? |
| 05:13.48 | afrosheen | yeah |
| 05:13.55 | afrosheen | zaptel driver bugs are plaguing our server |
| 05:14.10 | odie_flocon | hmm. |
| 05:14.15 | afrosheen | really starting to piss me off, I'm going to get rid of zaptel eventually |
| 05:14.19 | odie_flocon | this is why I want a heartbeat monitor. |
| 05:14.25 | Sedorox | hmmmm |
| 05:14.30 | florz | *CLI> show uptime |
| 05:14.30 | florz | System uptime: 5 weeks, 9 hours, 26 minutes, 45 seconds |
| 05:14.33 | odie_flocon | problem is I need 8 analog ports. |
| 05:14.39 | Sedorox | I don't have any problems with zaptel |
| 05:14.43 | florz | yeah, with zaptel - zaphfc only, though |
| 05:14.45 | odie_flocon | so I can't get away with zaptel. |
| 05:15.02 | afrosheen | Sedorox: I wish I didn't |
| 05:15.17 | afrosheen | one time the second card was giving static and not picking up on all channels |
| 05:15.30 | afrosheen | the other time the first card woudn't pick up the phone |
| 05:15.36 | Sedorox | hmmm |
| 05:15.38 | afrosheen | stuff like this makes me look bad at work :( |
| 05:15.39 | Sedorox | I only have one.. so... |
| 05:15.54 | Sedorox | why don't you get a TMD400 and get two FXO modules |
| 05:15.59 | Sedorox | would probably be better then two X100p's |
| 05:16.07 | afrosheen | I have 2 tdm400's |
| 05:16.10 | afrosheen | and I hate them both |
| 05:16.24 | Sedorox | lol |
| 05:16.26 | Sedorox | sell to me :-p |
| 05:16.26 | Sedorox | j/k |
| 05:16.38 | chaoscon | sell to me instead :P |
| 05:16.40 | afrosheen | they're cursed cards I tell you |
| 05:16.42 | chaoscon | lol |
| 05:16.42 | odie_flocon | that's my plan. |
| 05:17.47 | odie_flocon | hmmmm. |
| 05:18.01 | odie_flocon | I need a reliable solution. |
| 05:18.16 | afrosheen | hey, it may work for you, just spritz with holy water after you get it |
| 05:18.21 | odie_flocon | this guy want's to buy 6 WIFI sip handsets. |
| 05:18.21 | chaoscon | lol |
| 05:18.34 | afrosheen | wifi sip...*shudder* |
| 05:19.02 | Sedorox | afrosheen: ahaha |
| 05:19.08 | odie_flocon | his current phones interfere with his wireless network. |
| 05:19.10 | afrosheen | that and the meetme delay..oy vey |
| 05:19.36 | odie_flocon | so he want's to buy 6 wifi phones. |
| 05:19.44 | odie_flocon | and 6 analog phones. |
| 05:19.49 | odie_flocon | plus a door phone. |
| 05:20.20 | afrosheen | no wifi |
| 05:20.26 | odie_flocon | wants to be able to answer the door while he's on the elevator. |
| 05:20.29 | afrosheen | get him a handful of iaxy's and normal portable phones |
| 05:20.31 | odie_flocon | why no wifi? |
| 05:20.48 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 05:20.51 | odie_flocon | problem is normal portable phones interfere with his pc's. |
| 05:20.55 | afrosheen | bah |
| 05:21.03 | afrosheen | why would they do that |
| 05:21.15 | afrosheen | wireless network with his pc's? |
| 05:21.17 | odie_flocon | same range as his wireless G network. |
| 05:21.23 | afrosheen | then he's using the wrong phones |
| 05:21.29 | odie_flocon | well yeah. |
| 05:21.38 | Sedorox | anyone use the Aastra 480e (PT-480e) ? |
| 05:21.39 | afrosheen | cordless phones can be purchased in a variety of mhz/ghz formats |
| 05:21.43 | odie_flocon | but he saw these wifi sip phones. and want's them. |
| 05:21.50 | afrosheen | who makes them |
| 05:22.35 | odie_flocon | Hitachi |
| 05:22.39 | afrosheen | oh really |
| 05:22.44 | odie_flocon | yeah. |
| 05:22.55 | odie_flocon | they are like 400.00 usd each. |
| 05:23.03 | odie_flocon | and he wants 6 |
| 05:23.18 | afrosheen | http://james.seng.sg/files/public/hitachi-sipphone.jpg |
| 05:23.20 | afrosheen | those are sweet |
| 05:24.11 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 05:24.11 | afrosheen | too bad they cost way too much |
| 05:24.16 | afrosheen | maybe next year... |
| 05:24.17 | odie_flocon | I know |
| 05:24.28 | odie_flocon | so anyhow. I'm buying 6 in about a month |
| 05:24.45 | afrosheen | is this for some dude's house or a business? |
| 05:24.51 | odie_flocon | hosue |
| 05:24.53 | odie_flocon | house |
| 05:25.06 | florz | .o( business house ) |
| 05:25.09 | afrosheen | he must have some serious grip |
| 05:25.11 | odie_flocon | no house |
| 05:25.31 | odie_flocon | the guy cut my bro a cheque for 27,000.00 last week. |
| 05:25.38 | Sedorox | 0_o |
| 05:25.48 | Sedorox | give me... 10% of that please!!!! |
| 05:25.57 | Qwell | I'll take 1% :P |
| 05:25.58 | odie_flocon | his only comment was that would buy a car. |
| 05:26.16 | chaoscon | Sedorox: eventually ;) |
| 05:26.19 | Sedorox | lol |
| 05:26.28 | odie_flocon | hehe |
| 05:26.34 | Sedorox | anyone have experience with Aastra phones? |
| 05:26.37 | odie_flocon | good so it does have a headset plugin. |
| 05:27.31 | odie_flocon | I was worried about it not having a headset plugin. |
| 05:28.01 | odie_flocon | nope I havn't Sedorox. |
| 05:28.16 | odie_flocon | right now I'm working on his sytem. |
| 05:28.17 | afrosheen | Sedorox: strictly polycom so far |
| 05:28.24 | Sedorox | ok |
| 05:28.27 | odie_flocon | ohh how's the polycoms afro? |
| 05:28.32 | Sedorox | the Aastra phone's look nice |
| 05:28.35 | Sedorox | yea.. how are they? |
| 05:28.39 | afrosheen | polycoms rock |
| 05:28.45 | afrosheen | we have all IP500's |
| 05:28.46 | odie_flocon | I'm looking at getting the ip600's |
| 05:28.53 | afrosheen | we have probably 3 of the 600's |
| 05:28.55 | Sedorox | for me.. it will probably come down to either Polycom's IP600 or that Aastra |
| 05:29.00 | afrosheen | the speakerphone is very hard to beat |
| 05:29.04 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:29.05 | odie_flocon | I bet. |
| 05:29.12 | odie_flocon | polycom make the best phones. |
| 05:29.15 | Sedorox | hmmm |
| 05:29.19 | afrosheen | full duplex, which impresses compared to their old nortel system with half duplex speakerphones |
| 05:29.20 | Sedorox | how's the screen on them? |
| 05:29.34 | afrosheen | screen quality seems to vary somewhat but the 600's are a beauty |
| 05:29.42 | odie_flocon | good. |
| 05:29.43 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.197) |
| 05:29.43 | Sedorox | hmmm |
| 05:29.51 | odie_flocon | they are like 301.00 each |
| 05:30.01 | afrosheen | the 500's are almost as good for half the money |
| 05:30.15 | odie_flocon | they are 245 for the 500's |
| 05:30.19 | afrosheen | only real advantages to the 600 are 3 extra 'lines' and a higher-res screen. |
| 05:30.30 | afrosheen | no no no |
| 05:30.32 | Sedorox | hehe |
| 05:30.34 | afrosheen | 500's are never that high |
| 05:30.34 | Sedorox | voipsupply.com |
| 05:30.39 | odie_flocon | hmm. |
| 05:30.41 | odie_flocon | ok. |
| 05:30.45 | afrosheen | we bought ours for $167 each |
| 05:30.45 | Sedorox | <PROTECTED> |
| 05:30.46 | Sedorox | SIP VoIP Business Phone |
| 05:30.46 | Sedorox | <PROTECTED> |
| 05:30.47 | odie_flocon | that's my price in $cad |
| 05:30.51 | afrosheen | oh cad |
| 05:30.54 | afrosheen | yeah that's a good price then |
| 05:31.00 | Sedorox | thats in US |
| 05:31.07 | afrosheen | uh that's too high then |
| 05:31.16 | PTG123 | get the cisco instead of the polycom i have both |
| 05:31.22 | odie_flocon | yeah and 301.00 cad for the IP600's |
| 05:31.32 | afrosheen | yeah vendor lockin and licensing fees RULE |
| 05:31.51 | Qwell | vendor lockin? |
| 05:31.58 | odie_flocon | Cisco is putting out a PBX system. |
| 05:32.05 | odie_flocon | with voicemail. |
| 05:32.12 | Qwell | afrosheen: please qualify that last statement... |
| 05:33.11 | afrosheen | you're stuck pulling teeth trying to get cisco firmware images if you go Sip with cisco phones |
| 05:33.18 | odie_flocon | I pay about $245 Cad, for the IP500, and 301 Cad, for the IP600's. |
| 05:33.32 | Qwell | afrosheen: as opposed to? |
| 05:33.34 | odie_flocon | yeah. but * supports skinny now |
| 05:33.36 | Qwell | being stuck with polycom firmware? |
| 05:33.53 | afrosheen | the only way you have a legal right to that firmware is with a service contract or other payout |
| 05:34.14 | afrosheen | whereas anyone with polycom phones can get the firmware anytime at no additional cost from the suppliers |
| 05:34.27 | Qwell | That doesn't mean you're any more "locked in" |
| 05:34.34 | odie_flocon | hmm |
| 05:34.49 | afrosheen | if you buy a service contract, you're locked in |
| 05:34.58 | afrosheen | in my opinion |
| 05:35.03 | afrosheen | you're not just 'buying a phone' |
| 05:35.07 | Qwell | How are you not locked in with a polycom, or a sipura, or any device? |
| 05:35.34 | afrosheen | none of those other phones require BS service contracts |
| 05:35.50 | Qwell | I'm talking about your "locked in" statement. |
| 05:36.08 | foobos | anyone experienced very choppy voicemail recordings? |
| 05:36.08 | Qwell | yeah, if you want SIP on a cisco, you have to deal with licensing, I'm not worried about that |
| 05:36.31 | afrosheen | I consider licensing a type of vendor lock in, that's what I'm trying say |
| 05:36.48 | Qwell | So, its just an opinion? |
| 05:37.34 | afrosheen | it's the way I see it, my interpretation of lockin |
| 05:37.56 | afrosheen | if it didn't require a contract and cisco handed out the firmware to anyone who bought a phone, it'd be different to me |
| 05:38.28 | afrosheen | at any rate, from what I've seen, some cisco phones are nearly rebranded polycoms anyway |
| 05:39.01 | *** join/#asterisk netMonkey (~netMonkey@s8.http-tunnel.com) |
| 05:40.37 | Qwell | I don't even want to know what you're basing that off of |
| 05:40.56 | afrosheen | nortel has done it in the past with their 3 way conferencing phones |
| 05:41.14 | Qwell | oh, I see...because Nortel does it, you think Cisco is too...got it |
| 05:41.46 | afrosheen | well where else would they get IP phones from, you think they have a Cisco factory? |
| 05:41.54 | PTG123 | ok well my cisco blows away my polycom |
| 05:42.04 | PTG123 | in quality, way it feels when holding it, features, and interface |
| 05:42.17 | Qwell | afrosheen: kindly message me when you're done spreading crap, so I can unignore you |
| 05:42.27 | Sedorox | then report on it |
| 05:42.27 | Sedorox | :-p |
| 05:42.28 | afrosheen | what a jerk |
| 05:42.36 | PTG123 | hah |
| 05:43.05 | PTG123 | just fyi for everyone usually there are 2 or 3 factories in korea that make most electronics like that.. however the designs are usually done by the company, the factories are contracted |
| 05:43.16 | PTG123 | korea, china, etc |
| 05:43.23 | afrosheen | taiwan, you name it |
| 05:43.32 | afrosheen | same goes for laptops or anything else really |
| 05:43.35 | Qwell | its almost always the design that makes a product good |
| 05:43.42 | PTG123 | actually laptops are usually built in us |
| 05:43.46 | PTG123 | components are made elsewhere |
| 05:43.53 | PTG123 | yah the design is what matters |
| 05:43.57 | PTG123 | and in the case of phones the firmware |
| 05:44.03 | Sedorox | dell's are assembled in mexico... |
| 05:44.10 | Qwell | mmm, the firmware helps, but thats not 100% |
| 05:44.23 | Qwell | PTG123: like you were saying with the ciscos, "the way it feels when holding it" |
| 05:44.27 | Sedorox | wiki go down? |
| 05:44.34 | TomL | what what what? |
| 05:44.40 | Qwell | PTG123: That sounds kinda disturbing out of context BTW |
| 05:44.40 | TomL | someone going down? :O |
| 05:44.45 | Sedorox | 0_o |
| 05:45.58 | PTG123 | haha |
| 05:46.06 | PTG123 | my cisco feels soft against my skin :) |
| 05:46.12 | PTG123 | and the way it massages my ears.... :) |
| 05:46.15 | Qwell | ... |
| 05:46.18 | Qwell | :p |
| 05:46.40 | Sedorox | note to self... don't touch PTG123's phones |
| 05:46.43 | PTG123 | i am really tempted to buy all thise broken 7960s on ebay and restore them right now |
| 05:46.53 | Sedorox | hmmm |
| 05:47.03 | Sedorox | if you do.. and they work.. let me know.. and if you didn't defile them |
| 05:47.04 | Sedorox | :-p |
| 05:47.15 | PTG123 | hah why you wanna buy some 7960s? :) |
| 05:47.23 | afrosheen | broken? but they feel soooo good ;) |
| 05:48.02 | Sedorox | I dunno... |
| 05:48.06 | Sedorox | just want a good ip phone |
| 05:48.11 | Sedorox | I have a Budgetone 100 right now... |
| 05:48.52 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 05:49.39 | Qwell | PTG123: broken how? |
| 05:49.44 | afrosheen | have you guys seen the new iaxy's? |
| 05:49.56 | afrosheen | they look like little cobra heads |
| 05:50.14 | Sedorox | pic? |
| 05:50.20 | PTG123 | alot of different things |
| 05:50.23 | afrosheen | it's on digium's site..hang on |
| 05:50.28 | PTG123 | i am taking apart mine now to see how easy they are to fix |
| 05:50.34 | Qwell | heh |
| 05:50.43 | afrosheen | http://www.digium.com/index.php?menu=iaxy |
| 05:50.53 | Sedorox | yea |
| 05:50.55 | Sedorox | your right... |
| 05:51.21 | afrosheen | we're lucky we ordered ours a week ago or we wouldn't be able to get one now |
| 05:52.29 | Sedorox | umm |
| 05:52.34 | afrosheen | has anyone implemented a secretary phone yet? |
| 05:52.34 | Sedorox | voipsupply should still sell them |
| 05:52.46 | Sedorox | no. but been thinking about it.. |
| 05:52.46 | Sedorox | hehe |
| 05:53.04 | afrosheen | we're going to need one soon and the closest thing we've found is the sipura 220 |
| 05:53.29 | afrosheen | ack make that the snom 220 |
| 05:54.27 | knight_ | anyone using BroadVoice with G729? |
| 05:55.07 | afrosheen | Sedorox: it's this one with the optional panel http://www.telephonyware.com/telephonyware/tw00128.html?mv_pc=A00015 |
| 05:55.35 | Sedorox | yea |
| 05:55.53 | Slainte | afrosheen I have the polycom 600 working as a secretary phone, with some XML stuff. |
| 05:56.08 | afrosheen | Slainte: how do you handle call presence? |
| 05:56.28 | afrosheen | we want the sec. to be able to see if lines are already busy from the led's or whatever like a regular phone |
| 05:57.08 | PTG123 | ok just succesfully fully disassembled and put back together the 7960 |
| 05:57.13 | PTG123 | Sedorox: you really want one |
| 05:57.30 | Slainte | afrosheen, I use a small script that watches the sip extensions and voicemail, and then logs to a database, every 6 seconds the web page refreshes and a php grabs the info from the database |
| 05:57.35 | afrosheen | PTG123: are they all breaking the same way or do they have weird issues |
| 05:57.45 | Qwell | PTG123: How much you planning on selling them for? |
| 05:57.52 | PTG123 | they all have a different issue |
| 05:57.54 | Sedorox | ptblank: if you get one fixed.. let me know what you want for one |
| 05:57.56 | Sedorox | er |
| 05:57.57 | PTG123 | i figure its gonna take 2 phones to make one |
| 05:57.57 | Sedorox | PTG123: |
| 05:58.17 | PTG123 | so probably gonna be between $110-$140 each |
| 05:58.27 | PTG123 | the more people that want one though the cheaper they would get for us |
| 05:58.30 | PTG123 | which is why i asked |
| 05:58.44 | Sedorox | hmmmm |
| 05:58.46 | Sedorox | possibly... |
| 05:58.48 | afrosheen | Slainte: we could use FOP for that |
| 05:58.56 | Sedorox | depends if I have the money at the time I guess |
| 05:59.16 | afrosheen | the sec will have a pc, I guess she'll have to keep FOP open but transfers may be tricky |
| 05:59.18 | PTG123 | auction closes in 12 hours |
| 05:59.18 | PTG123 | :) |
| 05:59.26 | Sedorox | lol |
| 05:59.36 | Slainte | I could not convince the sec to use a PC, she kept closing hte window |
| 05:59.44 | Slainte | I had to do it XML on the phone |
| 05:59.49 | Slainte | stupid cow |
| 06:00.15 | afrosheen | is she one of those potato chip crunching secretarys |
| 06:01.00 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 06:01.15 | Slainte | no, just a stupid cow, thats all, just a stupid bovine |
| 06:01.16 | afrosheen | coz we don't have ours yet, but when we do, I'll force her to use FOP or hit the road |
| 06:01.22 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 06:01.25 | ellvis | hi people |
| 06:01.35 | afrosheen | hi hellvis |
| 06:01.38 | PTG123 | yay phone still works :) |
| 06:02.12 | ellvis | i am testing IAX calling and i am getting "Raw Hangup 172.16.30.25:4569, src=3, dst=8903", where can be the problem? |
| 06:02.29 | Sedorox | FOP? |
| 06:02.33 | ellvis | except the fact that between chair and keyboard:) |
| 06:02.36 | afrosheen | flash operator panel |
| 06:02.37 | Slainte | flash operator panel |
| 06:02.50 | afrosheen | it's getting better daily |
| 06:02.58 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 06:03.39 | Sedorox | hmmm |
| 06:04.42 | afrosheen | Slainte: explain a little more about how your system works, sounds interesting |
| 06:05.14 | Slainte | I have a perl script based on monastary (available via the wiki GUI section) |
| 06:05.28 | afrosheen | ok |
| 06:05.35 | Slainte | it interconnects with the terminal, and contiiously asks who is on or off, and who has voicemail |
| 06:05.41 | afrosheen | right |
| 06:05.49 | Slainte | it then updates the database via perl, |
| 06:06.02 | afrosheen | what database |
| 06:06.03 | Slainte | and the php web page does a simple query, creates a small table, |
| 06:06.14 | Sedorox | Apr 6 23:05:16 WARNING[84033]: pbx.c:1889 ast_pbx_run: Channel 'IAX2/iaxfwd@FWD/2' sent into invalid extension 's' in context 'ss-in', but no invalid handler.... hmmm |
| 06:06.20 | Slainte | a simple SQL one that can run on the same server |
| 06:06.34 | afrosheen | ok |
| 06:06.38 | Slainte | like 5 tables |
| 06:06.54 | Slainte | there is a meta refresh tag to ahve it refresh every 6 seconds |
| 06:07.01 | afrosheen | so it does a query of asterisk, pushes that data to a database, the php script builds an xml table then dumps it to the phone? |
| 06:07.10 | Slainte | yup |
| 06:07.27 | afrosheen | well so what does the phone do with the xml file, uses it's microbrowser to show her something? |
| 06:07.42 | Slainte | yes micro browser on the polycom 600 |
| 06:07.48 | Slainte | IP600 |
| 06:08.13 | afrosheen | I haven't seen it in action yet |
| 06:08.24 | afrosheen | although we have some 600's we don't use the microbrowser at all |
| 06:09.16 | Slainte | it works well.I find the contrast is uneven on all our phones |
| 06:09.19 | Slainte | very irritating |
| 06:09.31 | PTG123 | yah the screen for the polycoms suck |
| 06:09.33 | afrosheen | yeah like I was saying earlier, the screens vary somewhat |
| 06:09.36 | Slainte | you drop 500 bux on a phone and the screen is crap |
| 06:09.41 | PTG123 | go with cisco |
| 06:09.43 | afrosheen | who spent 500 bucks |
| 06:09.53 | afrosheen | ours were barely over 275 |
| 06:09.55 | Slainte | in bermuda everything has 33% duty plus the shipping |
| 06:10.01 | afrosheen | ohh snap |
| 06:10.15 | afrosheen | I guess they're lucky to make it through the triangle |
| 06:10.27 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 06:10.31 | _zigo__ | 500 bucks for a PHONE ! I'll keep my budgetone... :) |
| 06:10.47 | afrosheen | cisco's don't support xml do they? |
| 06:10.49 | Slainte | we are not using cisco because we are a cisco partner, but only securiy and wireless specialty. To other voice people on the island. If we sell the cisco phones we would get murdered from our channel partner |
| 06:10.54 | Slainte | yes they do |
| 06:10.59 | afrosheen | all models? |
| 06:11.30 | Slainte | th 7906 is the only one I know of |
| 06:12.30 | afrosheen | hmm |
| 06:12.47 | afrosheen | has anyone gotten custom rings to work on the polycoms |
| 06:12.51 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:13.06 | afrosheen | I've uploaded and played with the configs a million times and I can't ever select the ring wavs I upload |
| 06:13.20 | Slainte | never tried |
| 06:13.44 | afrosheen | the snom 220 has a very, very nice configuration manager built into it |
| 06:13.52 | afrosheen | kinda wish polycom would catch up |
| 06:14.39 | Slainte | they said they are only gong to give their channel partners access to the config apps |
| 06:14.51 | Slainte | I think Apple should come out with a few IP phones |
| 06:15.16 | afrosheen | just make some frosted white cisco's and slap an apple logo on them |
| 06:15.42 | Slainte | well cisco steals all the speaker and mic technology from polycom |
| 06:17.01 | afrosheen | don't say that out loud |
| 06:17.07 | afrosheen | Qwell is still here |
| 06:17.08 | ellvis | :) |
| 06:19.08 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:20.58 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.254) |
| 06:32.37 | Sedorox | turning in |
| 06:32.37 | Sedorox | night |
| 06:33.13 | Slainte | gnight |
| 06:33.28 | *** join/#asterisk Zilas (~info@c-24-30-75-206.hsd1.ga.comcast.net) |
| 06:34.32 | *** part/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 06:41.09 | ellvis | anyone can help with ztdummy compilation problems? |
| 06:41.20 | *** join/#asterisk |nix (~inix@202.148.164.48) |
| 06:46.29 | knight_ | anyone using BroadVoice with G729? |
| 06:46.55 | knight_ | I've heard about successes, but havent been able to reproduce that. |
| 06:47.59 | wildcard0 | any wireless experts on? |
| 06:48.47 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:48.52 | Zilas | what would you offer for sending sms out of asterisk that would be free decision? |
| 06:51.08 | *** join/#asterisk abracsas (~abuono@217.9.64.150) |
| 06:56.54 | *** join/#asterisk `Rage (admin@office.vilerage.us) |
| 06:59.04 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 07:02.02 | *** join/#asterisk tainted_ (~tainted@adsl-69-108-108-201.dsl.irvnca.pacbell.net) |
| 07:02.53 | *** join/#asterisk ikey (ikey@202.54.37.184) |
| 07:06.01 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 07:06.01 | *** join/#asterisk rpr_ (~ricardMad@195.53.197.70) |
| 07:07.51 | rpr_ | Hi to all. I need help to troubleshooting an E1 connection. How can I verify the phicial connection? |
| 07:08.43 | *** join/#asterisk juice (~juice@mo-205-240-40-98.dyn.sprint-hsd.net) |
| 07:08.47 | wildcard0 | loopback? |
| 07:08.48 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:08.56 | rpr_ | Could be. |
| 07:09.04 | wildcard0 | no i mean verify it via loopbac |
| 07:09.04 | wildcard0 | k |
| 07:09.08 | wildcard0 | it should be able to detect itself |
| 07:10.43 | rpr_ | Can I diference between phisical and frame errors in any log? |
| 07:11.29 | wildcard0 | hmm. that's more difficult. physical errors often manifest themselves as either completely non-functional devices or (rarely) masquerade as other errors |
| 07:11.52 | wildcard0 | frame errors may indicate a physical problem, but most often a physical problem will result in a completely non-working connection |
| 07:12.03 | *** join/#asterisk outsidefactor (barf@203-206-247-72.dyn.iinet.net.au) |
| 07:12.55 | rpr_ | MY connection doesnt work in any mode. But I've not any tool to knowif the wire is broken or if is a configuration error. |
| 07:13.11 | Slainte | rpr, are you using zttool |
| 07:13.26 | rpr_ | Where can i find this tool? |
| 07:13.37 | Slainte | it is part of the zaptel package |
| 07:13.48 | wildcard0 | rpr, start with putting a loopback connector on the end of the wire and ask your provider if they can see the loop |
| 07:13.55 | wildcard0 | then you know if it's a physical error or not |
| 07:14.11 | Slainte | you can set a software loop with zttool |
| 07:14.43 | wildcard0 | that works too |
| 07:14.44 | rpr_ | must i make with any parameter to see zttool? |
| 07:15.13 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:18.41 | tainted_ | how do i tell if a client has a PRI line or a regular T1 line |
| 07:20.12 | Slainte | tainted is it used for internet right now, voice or both? |
| 07:20.44 | tainted_ | both |
| 07:20.53 | RestLessGemini | rpr : just type zttool and it will comeup |
| 07:21.31 | Slainte | then it is probably channelised, you need to ask your local loop provider what channels they have setup |
| 07:22.38 | tainted_ | Slainte is there a page that shows the different signalling types used by different pbx manufacturers? |
| 07:23.14 | Slainte | the signaling is depdendent on the provider. Look at the wiki for the zapata.conf its a good start |
| 07:23.34 | tainted_ | thanks |
| 07:23.48 | Slainte | np |
| 07:26.47 | tainted_ | Slainte this is a general question.. if they use a lucent pbx and lucent phones.. will i need to replace the phones or can * replace just their pbx |
| 07:28.05 | Slainte | I seriously doubt you would be able to get the lucent phones to work on an * solution, without the lucent PBX in the mix somewhere. |
| 07:29.14 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.105) |
| 07:29.14 | tainted_ | can i stick * between the lucent pbx and their line? |
| 07:29.20 | Slainte | there line to the teloc? |
| 07:29.22 | Slainte | telco? |
| 07:29.23 | tainted_ | yea |
| 07:29.48 | tainted_ | i don't think they want to overhaul their existing $$$$ pbx |
| 07:30.15 | Slainte | There are setups like that yes, and it can be a good way to slowly move people to IP sets. You will need to have configuration control of the Lucent unit. |
| 07:30.43 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 07:31.09 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 07:32.34 | RoyK | hmmmmm |
| 07:32.46 | *** join/#asterisk GhostXz (ikusher@CPE000d88a9ac16-CM001225419a6c.cpe.net.cable.rogers.com) |
| 07:32.51 | RoyK | I have this queue with sip and gsm phone members |
| 07:33.50 | RoyK | is it possible to use agents or something to have the gsm members 'log out'/removed in case they need to switch their phone off? if we don't the telco will terminate the call with "subscriber not available" :P |
| 07:33.55 | GhostXz | whats the website? |
| 07:33.57 | GhostXz | for this |
| 07:34.14 | Slainte | www.asterisk.org |
| 07:34.28 | RoyK | Sláinte! |
| 07:34.32 | GhostXz | any screeenshots:P |
| 07:34.44 | RoyK | ??? |
| 07:34.46 | GhostXz | wait |
| 07:34.49 | GhostXz | whats a PBX? |
| 07:34.50 | GhostXz | :| |
| 07:34.56 | RoyK | GhostXz: you don't want screenshots from a terminal |
| 07:34.59 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 07:35.02 | RoyK | ~pbx |
| 07:35.04 | jbot | somebody said pbx was a Private Branch eXchange |
| 07:35.15 | RoyK | GhostXz: telephony... |
| 07:35.18 | GhostXz | what exactly dose it do o.o |
| 07:35.20 | GhostXz | uhh |
| 07:35.22 | GhostXz | right.. |
| 07:35.22 | Zeeek | Tell a phony |
| 07:35.24 | RoyK | ~docs |
| 07:35.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 07:35.27 | RoyK | ~rtfm |
| 07:35.28 | jbot | rtfm is probably read the f*cking manual... try asking me about "FAQ" |
| 07:35.34 | RoyK | ~lart GhostXz |
| 07:35.45 | GhostXz | ;o |
| 07:36.05 | GhostXz | i thought pbx was like one of thos media player things.. |
| 07:36.08 | Zeeek | then they're surpsied to see so few women here :) |
| 07:36.10 | RoyK | ~lart GhostXz |
| 07:36.16 | GhostXz | ;o |
| 07:36.25 | GhostXz | ~lart RoyK |
| 07:39.39 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 07:40.14 | luke-jr_ | GhostXz: Asterisk is, to make it simple, a phone server. It handles phone calls instead of eg webpages. |
| 07:40.15 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:41.22 | RoyK | is it possible to use agents or something to have the gsm members 'log out'/removed in case they need to switch their phone off? if we don't the telco will terminate the call with "subscriber not available" :P |
| 07:41.33 | GhostXz | a "phone server" |
| 07:41.34 | GhostXz | o.o |
| 07:42.45 | luke-jr_ | GhostXz: for example, if I call someone, my phone dials a # which goes to my system running Asterisk. Asterisk routes the call to some random VoIP company that routes it over the POTS network to the destination |
| 07:43.10 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 07:43.34 | luke-jr_ | well, not quite randomly selecting a company, but... you get the idea ;) |
| 07:47.47 | RoyK | ~pots |
| 07:47.48 | jbot | pots is, like, Plain Old Telephone Service as in "Old Analogue Crap" |
| 07:47.49 | RoyK | ~pstn |
| 07:47.50 | jbot | from memory, pstn is Public Switched Telephone Network |
| 07:48.00 | RoyK | ~lart himself |
| 07:48.25 | ellvis | jbot: database for jokes |
| 07:48.57 | GhostXz | i bet i cant use a cell phone ;p |
| 07:48.57 | RoyK | ~jbot? |
| 07:49.13 | RoyK | GhostXz: asterisk doesn't have any chan_gsm, no :P |
| 07:49.20 | luke-jr_ | GhostXz: depends on WiFi coverage in your area |
| 07:49.23 | GhostXz | oooo |
| 07:49.25 | GhostXz | i know |
| 07:49.29 | RoyK | http://karlsbakk.net/mp3fun/asterisk-installation.wav |
| 07:49.29 | GhostXz | i can have 2 phone lines |
| 07:49.44 | GhostXz | and i have em both hooked into a box |
| 07:49.58 | GhostXz | when then when my cell phones 1 |
| 07:50.03 | RoyK | GhostXz: I have 120 phone lines (4 PRIs) connected to each box :P |
| 07:50.13 | GhostXz | it connects to the other phone line and dials a number |
| 07:50.21 | GhostXz | then i talk to the person o.o |
| 07:50.27 | luke-jr_ | GhostXz: ... why? =p |
| 07:50.30 | GhostXz | i dont know |
| 07:50.35 | GhostXz | that would cost alot o money |
| 07:50.35 | luke-jr_ | lol |
| 07:50.35 | GhostXz | lol |
| 07:50.40 | RoyK | nice way to throw away money |
| 07:50.52 | GhostXz | that is uselesss |
| 07:50.55 | GhostXz | o.o |
| 07:50.57 | luke-jr_ | GhostXz: Or you could get rid of the PSTN/POTS lines |
| 07:51.05 | luke-jr_ | GhostXz: and replace em w/ a VoIP solution |
| 07:51.08 | knight_ | anyone using BroadVoice with G729? ;) |
| 07:51.08 | GhostXz | i still dont get wha ti can do |
| 07:51.46 | luke-jr_ | GhostXz: disconnect the PSTN/POTS lines and connect the Asterisk system to your home's phone line |
| 07:52.04 | luke-jr_ | GhostXz: and then your home's phones go to the Asterisk system |
| 07:52.14 | luke-jr_ | GhostXz: then route the phone call over the internet |
| 07:52.31 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 07:52.35 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 07:52.46 | GhostXz | ( is lost ) |
| 07:52.53 | *** part/#asterisk outsidefactor (barf@203-206-247-72.dyn.iinet.net.au) |
| 07:53.00 | luke-jr_ | GhostXz: you can make phone calls over the internet |
| 07:53.09 | GhostXz | right.. |
| 07:53.26 | GhostXz | why do i need a phone line then |
| 07:53.27 | GhostXz | :| |
| 07:53.32 | luke-jr_ | you don't =p |
| 07:53.37 | GhostXz | ohhhhh |
| 07:53.42 | GhostXz | k |
| 07:53.52 | GhostXz | can VoIP's phone real phone #'s? |
| 07:54.01 | luke-jr_ | if you get termination service |
| 07:54.26 | luke-jr_ | and vice-versa if you get origination service |
| 07:54.47 | GhostXz | so basicaly when i pick up meh phone and dial a number the pbx will connect to the voip service, and dial the # from there and yeah |
| 07:55.01 | luke-jr_ | sure |
| 07:55.11 | GhostXz | cool |
| 07:55.12 | luke-jr_ | or it can do some other fancy stuff first :) |
| 07:55.18 | GhostXz | like? |
| 07:55.35 | luke-jr_ | like change *01 to a longer # |
| 07:55.45 | luke-jr_ | (speed-dial sortof) |
| 07:55.47 | shido6 | flush a toilet |
| 07:55.55 | shido6 | page you when certain system take a dump |
| 07:56.04 | luke-jr_ | shido6: has someone actually written a script for that? |
| 07:56.05 | shido6 | page all phones and do price checks on aisle 7 |
| 07:56.08 | shido6 | voicemail |
| 07:56.10 | shido6 | conferencing |
| 07:56.15 | GhostXz | o |
| 07:56.16 | shido6 | play interactive voice response menus |
| 07:56.24 | shido6 | give you the weather |
| 07:56.30 | shido6 | unlock a security door |
| 07:56.40 | shido6 | Calling card business |
| 07:56.44 | GhostXz | so when someone calls like myself and punch in some passwds i can make it run xamp /music/* |
| 07:56.50 | shido6 | termination/oringination/ ITSP |
| 07:56.59 | luke-jr_ | GhostXz: or just setup music on hold :) |
| 07:57.05 | GhostXz | loll |
| 07:57.12 | GhostXz | can u make it talk :| |
| 07:57.21 | luke-jr_ | Theoretically |
| 07:57.32 | luke-jr_ | Some people have done it :) |
| 07:57.46 | RoyK | shido6: what do you want to know? |
| 07:57.56 | Slainte | the only thing it cant do is wipe your ass, but if you get an IP module for your toilet, it can flush it for you |
| 07:57.56 | RoyK | ~lart shido6 for /MSG people |
| 07:58.47 | GhostXz | so i can like call and make it say "Current play list" then list some songs then i hit a number and it will play it :| |
| 07:59.15 | luke-jr_ | GhostXz: sure |
| 07:59.21 | GhostXz | sweet |
| 07:59.28 | newl | shido6: Would you happen to know why when A number calls B number, B diverts, A takes the CDR hit, not B? (presuming test bed of two extensions on the same asterisk daemon) |
| 07:59.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 07:59.55 | GhostXz | well the only problm is i still want to use my normal phone :P |
| 08:00.02 | GhostXz | wait |
| 08:00.06 | GhostXz | what the fuck am i talkinga about |
| 08:00.14 | luke-jr_ | GhostXz: That's why you connect the server to your existing POTS wiring |
| 08:00.24 | GhostXz | POTS? |
| 08:00.30 | luke-jr_ | -pots |
| 08:00.36 | luke-jr_ | ~pots |
| 08:00.37 | jbot | i heard pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 08:00.37 | newl | Plain Old Telephone System. |
| 08:00.38 | shido6 | newl, are you in need of a billing solution for asterisk? :) |
| 08:01.06 | luke-jr_ | GhostXz: your "normal phone" |
| 08:01.07 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 08:01.08 | shido6 | Slainte, call us at 248-724-VoIP |
| 08:01.09 | GhostXz | Plain Old Telephone System |
| 08:01.13 | GhostXz | i c |
| 08:01.50 | GhostXz | how would it connect |
| 08:01.57 | luke-jr_ | various ways |
| 08:01.59 | newl | shido6: No. I'd just like to know if I should log a bug report if this is the expected behavior, or if there is something I'm not seeing on the wiki or in the docs relating to diversions and proper CDR recording. :) |
| 08:02.22 | luke-jr_ | I use a Linksys PAP2-NA (hard to find) |
| 08:02.33 | luke-jr_ | Digium sells expensive PCI cards, IIRC |
| 08:02.35 | PTG123 | very easy to find |
| 08:02.41 | luke-jr_ | PTG123: NAs? |
| 08:02.51 | PTG123 | i think the soyo w/ 4 ports are the best deal going right now though |
| 08:02.53 | PTG123 | luke-jr_: yah |
| 08:03.14 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:03.18 | PTG123 | of course i am an authorized linksys sales agent or whatever :) |
| 08:03.42 | luke-jr_ | PTG123: can you sell em w/o service? =p |
| 08:04.11 | PTG123 | hah i suppose i could, but whats the point :) |
| 08:04.19 | shido6 | I have 200 PAP2-Na's sitting right here |
| 08:04.23 | shido6 | I need to move them all |
| 08:04.27 | shido6 | thinking about $70 each |
| 08:04.33 | PTG123 | shido6: why you switching to something else? |
| 08:04.39 | GhostXz | I bet you want to give poor ol' GhostXz a free one |
| 08:05.01 | shido6 | sure I'll send you one send billing@nufone.net 80 bucks and I'll send you one with 10 bucks of service |
| 08:05.02 | PTG123 | i just got one of the new wireless gateways from linksys today.. can't wait to try it out :) |
| 08:05.02 | shido6 | :) |
| 08:05.31 | GhostXz | how can i do this without buying something.. ;p |
| 08:05.54 | luke-jr_ | GhostXz: you'll need some hardware to connect POTS lines... |
| 08:06.05 | GhostXz | a dialup modem :P |
| 08:06.08 | luke-jr_ | GhostXz: otherwise, you could use microphone & headset on a comp |
| 08:06.48 | luke-jr_ | I think it might be possible w/ WinModems, but realise they won't put out the power needed by many POTS phones |
| 08:06.58 | luke-jr_ | Wireless POTS phones should work tho |
| 08:07.08 | GhostXz | ic |
| 08:07.17 | GhostXz | aka cordless phone |
| 08:08.57 | shido6 | xten xlite and a noise cancelling mic with onboard dsps like the plantronics headset |
| 08:09.23 | luke-jr_ | eww @ xlite junk |
| 08:09.38 | luke-jr_ | GhostXz: PAP2-NA info @ http://www.linksys.com/products/product.asp?prid=651&scid=38 |
| 08:10.09 | luke-jr_ | GhostXz: Make sure that if you get one, it's a -NA model. If it's not NA/unlocked, you won't be able to configure it |
| 08:11.13 | *** join/#asterisk snitter (snitt@a84-0-177-239.adsl-pool.axelero.hu) |
| 08:16.55 | snitter | hi |
| 08:20.13 | tainted_ | PTG123 u are linksys sales rep? |
| 08:22.54 | *** join/#asterisk Dibbler (~Dibbler@zidane.pi-net.net) |
| 08:23.45 | *** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es) |
| 08:26.14 | shepherd | asterisk can make use of multiple processors right? |
| 08:27.38 | Zgarbi | what's wrong with new libpri? I cannot compile |
| 08:28.31 | shepherd | what distro? |
| 08:29.09 | Zgarbi | today cvs |
| 08:29.23 | Zgarbi | just updated |
| 08:29.34 | shepherd | what distro of linux? |
| 08:29.57 | Zgarbi | fedora core development |
| 08:30.04 | shepherd | hmmm! |
| 08:30.10 | shepherd | that would be odd |
| 08:30.21 | shepherd | it usually works on fedora first |
| 08:30.37 | Zgarbi | pri_facility.c: In function 'asn1_name_decode': |
| 08:30.38 | Zgarbi | pri_facility.c:190: warning: pointer targets in passing argument 1 of 'strlen' differ in signedness |
| 08:30.43 | Zgarbi | pri_facility.c: In function 'rose_number_digits_decode': |
| 08:30.44 | Zgarbi | pri_facility.c:237: warning: pointer targets in passing argument 1 of 'strlen' differ in signedness |
| 08:30.44 | Zgarbi | make: *** [pri_facility.o] Error 1 |
| 08:31.01 | shepherd | i'll cvsupdate and see if i can get it working |
| 08:31.11 | Zgarbi | ok |
| 08:31.42 | Zgarbi | give me then PM, ok? |
| 08:32.00 | shepherd | ok |
| 08:32.04 | Zgarbi | 10xs |
| 08:32.24 | GhostXz | i think i should post this |
| 08:32.26 | GhostXz | http://ikush.com/pwned.txt |
| 08:40.45 | RestLessGemini | ping |
| 08:40.53 | shepherd | pong? |
| 08:41.07 | *** join/#asterisk jwitte (~jwitte_su@port-212-202-101-206.static.qsc.de) |
| 08:41.10 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 08:42.08 | Mimmus | hi, I'm trying meetme application but I get a 'unable to open psuedo channel - trying device' warning. Why? I don't use 'zap' |
| 08:43.10 | shido6 | then use ztdumm |
| 08:43.11 | shido6 | y |
| 08:43.15 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:43.15 | shido6 | or x100p |
| 08:43.37 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 08:45.35 | Mimmus | shido6: how need I config ztdummy? |
| 08:51.27 | Zgarbi | I have using linux2.6 kernel, and make in zaptel make config |
| 08:52.07 | Zgarbi | so then in init.d u will have zaptel. so start service zaptel will start ztdummy |
| 08:52.53 | Mimmus | ummmm.... |
| 08:52.56 | Mimmus | [chan_zap.so]Apr 7 10:51:52 WARNING[23144]: loader.c:258 ast_load_resource: /u |
| 08:52.57 | Mimmus | sr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_suspend_acknowledge |
| 08:52.57 | Mimmus | Apr 7 10:51:52 WARNING[23144]: loader.c:440 load_modules: Loading module chan_z |
| 08:52.57 | Mimmus | ap.so failed! |
| 08:53.16 | Mimmus | I made modprobe ztdummy first |
| 08:53.39 | Zgarbi | which kernel are u using? |
| 08:54.00 | Mimmus | 2.6.8-2-386 (Debian unstable) |
| 08:54.14 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 08:55.08 | Zgarbi | try to compile zaptel with make linux26 |
| 08:55.29 | facek_ | what can i do with that problem |
| 08:55.29 | Mimmus | no, no: I used * Debian package |
| 08:55.29 | facek_ | ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
| 08:55.30 | facek_ | Failed to register zone 'United States / North America': No data available |
| 08:55.31 | facek_ | ? |
| 08:56.09 | Mimmus | Point is that the meetme then works, even if I get this error |
| 08:56.42 | Zgarbi | ops... then I cannot help |
| 08:57.16 | *** part/#asterisk eye69 (magnus@upcore.net) |
| 08:57.35 | Mimmus | Zgarbi: weel, it's only a warning |
| 08:58.30 | Zgarbi | just I as usual user want to say how I have compile ztdummy |
| 08:58.43 | Zgarbi | zaptel, not a ztdummy |
| 09:02.00 | *** join/#asterisk opsys (~aa@adsl-065-006-173-010.sip.mia.bellsouth.net) |
| 09:03.33 | opsys | Any bug Marshels on line???? |
| 09:04.30 | shepherd | i'm like 1/2 a marshel :) |
| 09:05.09 | shepherd | but not a real marshal |
| 09:05.52 | opsys | I placed a bug in the bug tracker, I placed it under the wrong catagory. I wanted to know if it could be moved to core asterisk |
| 09:06.57 | *** join/#asterisk fenlander_ (~neils@82.152.81.57) |
| 09:08.31 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:12.50 | facek_ | can sb help me little with sip ? |
| 09:13.08 | opsys | what do you need with sip? |
| 09:13.49 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 09:13.54 | zoa | um here |
| 09:13.58 | zoa | i will move it |
| 09:14.01 | zoa | what is the id ? |
| 09:14.17 | shepherd | 0003977 |
| 09:14.23 | facek_ | opsys *CLI> Apr 7 11:12:12 NOTICE[6838]: chan_sip.c:7988 sip_poke_noanswer: Peer 'inezk' is now UNREACHABLE! |
| 09:14.51 | shepherd | i need to give myself admin rights tomorrow and i could have done it :/ |
| 09:15.14 | zoa | moved it |
| 09:15.27 | opsys | thanks zoa and shep |
| 09:16.02 | zoa | np |
| 09:16.03 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 09:17.46 | *** join/#asterisk tainted- (~tainted@adsl-69-108-108-189.dsl.irvnca.pacbell.net) |
| 09:20.25 | Zeeek | <PROTECTED> |
| 09:20.52 | *** join/#asterisk rlarsen (~richard@194.248.136.69) |
| 09:21.35 | rlarsen | Can anyone help me with agent/queues and transfer ? |
| 09:24.32 | Zeeek | the sun is coming out |
| 09:26.10 | rlarsen | Is it possible to use the "sip transfer"-button in my xten softphone insted of '#' for transfering queue calls ? |
| 09:28.31 | Zeeek | should be if it's not X-Lite |
| 09:28.39 | rlarsen | nope eyebeam |
| 09:28.53 | Zeeek | what happens when you try? |
| 09:29.07 | RestLessGemini | xpro also has a transfer button |
| 09:29.08 | rlarsen | hangup , but '#' works |
| 09:29.32 | Zeeek | I don't know what the transfer button does on the X-Tens |
| 09:29.46 | Zeeek | it's grayed out on X-Lite IIRC |
| 09:29.58 | rlarsen | with incomming none "queue calls" the transfer button works. |
| 09:30.25 | *** join/#asterisk Muttley1976 (~michael@195-144-078-077.stat.sdsl.xs4all.be) |
| 09:31.10 | Muttley1976 | Does somebody already worked with RFC3311 (or the SIP UPDATE method ?) |
| 09:31.25 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:31.31 | RoyK | ehlo |
| 09:32.13 | *** join/#asterisk Alex1 (~chatzilla@ARennes-202-1-5-67.w81-48.abo.wanadoo.fr) |
| 09:32.19 | Alex1 | hello |
| 09:33.01 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 09:34.13 | Alex1 | french here ? |
| 09:34.19 | Zeeek | sort of |
| 09:34.27 | Zeeek | 'sup? |
| 09:34.27 | Alex1 | cà d ? |
| 09:34.38 | Alex1 | fine thx |
| 09:34.55 | Alex1 | got a problems with an sjphone |
| 09:36.15 | Zeeek | I never got sjphone to work |
| 09:36.21 | Zeeek | some people love it |
| 09:36.38 | Zeeek | I used X-Lite, which worked. THen I used a bunch of IAX clients |
| 09:36.54 | Alex1 | ok lol |
| 09:36.56 | Alex1 | thx anyway :) |
| 09:37.15 | Zeeek | use wengo |
| 09:37.22 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 09:37.28 | Alex1 | wengo ? |
| 09:37.36 | Zeeek | http://www.wengo.fr |
| 09:37.41 | Zeeek | SIP client for windows |
| 09:37.42 | Alex1 | thanx |
| 09:38.52 | facek_ | Zeeek can you little help me with SIP. i am unable to call clients by SIP/name.. i am only able to call SIP/ip_Addr |
| 09:39.25 | Zeeek | are they registered? |
| 09:39.27 | Alex1 | do u know a free software ? |
| 09:39.35 | Zeeek | wengo is free |
| 09:39.39 | Alex1 | we need to be registered no ? |
| 09:39.43 | Zeeek | so is X-Lite |
| 09:39.55 | Zeeek | Alex, I don't know, but the software is open source |
| 09:40.17 | Alex1 | okay |
| 09:40.26 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:40.42 | Alex1 | we work on lan here |
| 09:41.30 | Zeeek | use X-Lite, it works great |
| 09:41.50 | Alex1 | it's a linux one ? |
| 09:42.00 | Zeeek | they have one, yes |
| 09:42.05 | Alex1 | ok |
| 09:42.06 | Zeeek | or so I've heard |
| 09:42.45 | darkskiez | is three any conference sip phones other than the polycom ? |
| 09:42.56 | *** join/#asterisk msupino (~msupino@gateway.sd.com) |
| 09:43.02 | *** part/#asterisk rlarsen (~richard@194.248.136.69) |
| 09:43.11 | facek_ | how to change the confiugration of xlite |
| 09:43.24 | msupino | Anyone knows how to group together two isdn cards , so when i use the Dial() command, it will use the first availble line from the 4 availble ? |
| 09:43.32 | Zeeek | facek_ X-Lite has a manual |
| 09:44.42 | Delvar | msupino: Spans and groups |
| 09:44.55 | facek_ | Zeeek yes, but conf key is disabled |
| 09:45.38 | Zeeek | facek_ I've never heard of that before. I you sure you're not hitting the wron button? |
| 09:46.04 | facek_ | NO |
| 09:46.13 | Zeeek | no, you're not sure? |
| 09:46.13 | msupino | delver: i am using fritz with CAPI, more info please |
| 09:46.20 | facek_ | i am not sure |
| 09:46.23 | opsys | facek_:!! CONF is for CONFerance |
| 09:46.24 | facek_ | where should i have that button |
| 09:46.25 | facek_ | ? |
| 09:46.37 | Zeeek | It looks like tools |
| 09:46.38 | opsys | the men if the button with the lines in the middle |
| 09:46.58 | opsys | the menu is the button with the lines in the middle |
| 09:47.23 | facek_ | i have |
| 09:47.25 | facek_ | LINES 1 2 3 |
| 09:47.31 | facek_ | and no menu |
| 09:47.41 | Zeeek | facek_ save some time. Try every button |
| 09:47.56 | facek_ | Zeeek i am trying and trying |
| 09:48.12 | Zeeek | http://faq.nikotel.com/index.php?sid=622076&aktion=artikel&rubrik=018001&id=185&lang=en |
| 09:48.16 | Zeeek | Look at this: |
| 09:48.27 | Zeeek | it's from google - I typed two word into google |
| 09:48.44 | Zeeek | configure x-lite |
| 09:49.49 | facek_ | Zeeek sorry. i have other xlite |
| 09:49.50 | Zeeek | facek_ Better yet: http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite |
| 09:49.53 | facek_ | no w i download this |
| 09:49.53 | Alex1 | the sjphone cant loged on asterisk |
| 09:50.12 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 09:50.35 | Zeeek | Alex1 http://www.voip-info.org/wiki-SJphone |
| 09:52.01 | msupino | delvar : found a solution, thanks |
| 09:52.20 | facek_ | fuck |
| 09:52.26 | Zeeek | now, now... |
| 09:52.35 | facek_ | Zeeek can you give me xlite.. the server have a problem with download |
| 09:52.52 | Zeeek | I don't have it here. Go to X-Ten and download the latest version |
| 09:52.56 | facek_ | ;] |
| 09:52.58 | facek_ | ok |
| 09:53.09 | facek_ | Zeeek can you little help me. i am making a simple solution. |
| 09:53.11 | RoyK | "how are xxx cards compared to digium cards" "buy digium, because they're so nice, giving us asterisk" |
| 09:53.12 | RoyK | sic |
| 09:53.37 | Zeeek | X-Lite can be downloaded here: http://www.xten.com/index.php?menu=products&smenu=download |
| 09:53.39 | facek_ | i have two peers on SIP. i allow everyone to call out.. and i want to make a good with BUSY or unavilable |
| 09:53.43 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:53.49 | facek_ | and i want enable to call for that people from world. |
| 09:53.58 | facek_ | and then.. give them access to makle a transfer |
| 09:54.15 | Alex1 | no french here ? |
| 09:54.31 | Zeeek | 1/2 French |
| 09:54.57 | Mimmus | serious problem: chan_capi doesn't compile anymore with latest * CVS, even if I apply the special patch |
| 09:54.57 | zyke | any one familiar with the error - chan_sip.c:7506 handle_request: Unable to create/find channel ? |
| 09:55.06 | zyke | is that a codec issue? |
| 09:57.23 | *** join/#asterisk emitrax (~emitrax@ingnatdyn33.unime.it) |
| 09:59.14 | emitrax | Is there anybody that use cisco phone 7940 with asterisk? |
| 09:59.14 | facek_ | i have that problem |
| 09:59.15 | facek_ | ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
| 09:59.15 | facek_ | Failed to register zone 'United States / North America': No data available |
| 09:59.29 | facek_ | but in indicatiosn i have a [us] wiht description 'United States / North America' |
| 10:04.17 | Mimmus | why is not possible to integrate chan_capi in main * CVS? |
| 10:06.08 | newl | Call diversions are now getting recorded correctly in CDR. =) |
| 10:06.22 | clive- | mimmus, how did you manage to get chan_capi going with CVS head in the end? |
| 10:08.09 | *** join/#asterisk tessier (~treed@222.253.76.53) |
| 10:08.31 | Mimmus | clive-: for europea<n users like me, capi is a channel like others |
| 10:09.08 | *** join/#asterisk ]expic (~Inferna@194.158.51.171) |
| 10:09.20 | clive- | my question is, how easy was it to make chan_capi work with cvs head |
| 10:09.28 | facek_ | clive- can you hep me with transfers? |
| 10:09.49 | RestLessGemini | which other fxo/fxs interface cards asterisk support other then asterisk? |
| 10:09.56 | ]expic | anybody had the problem with chan_h323 and no audio? i see that asterisk gets RTP traffic but doesn't resend it, i am doing sip<->h323 converting |
| 10:10.01 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 10:10.31 | Mimmus | clive-: sorry for my bad english, I don't understand very well: chan_capi worked untile some days ago, applying a patch |
| 10:11.09 | Mimmus | clive-: I found it on the wiki and lbarth on this channle helped me to apply to main * CVS |
| 10:11.13 | *** join/#asterisk Kaneda_ch (~Kaneda@pc90.geneva.ch.psi.com) |
| 10:11.37 | Kaneda_ch | Hi |
| 10:11.49 | Kaneda_ch | I'm new to SIP so don't flame :) |
| 10:12.04 | Kaneda_ch | I would like to do the following setup |
| 10:12.10 | clive- | mimmus sounds like to me I will need some expert help when I get brave enough to try it:) |
| 10:12.18 | ]expic | anybody had the problem with chan_h323 and no audio? i see that asterisk gets RTP traffic but doesn't resend it, i am doing sip<->h323 converting |
| 10:12.31 | Kaneda_ch | SoftPhone <-> asterisk <-> Cisco 3640 <-> PBX <-> PSTN |
| 10:12.51 | Kaneda_ch | first question: is that a good idea ? |
| 10:13.22 | Mimmus | clive-: oh, yes: in fact, I joined this channel this morning because I'm unable to recompile it with latest, current CVS! |
| 10:13.39 | Mimmus | clive-: I'm re-trying just now... |
| 10:13.52 | clive- | lol...you mean CVS changed since your last time? |
| 10:14.58 | ]expic | Communication from sip to sip and h323 to h323 is working. |
| 10:14.58 | ]expic | When i now call from the siphone (three tested) the h323 phone (also |
| 10:14.58 | ]expic | three tested) the connection is coming up and everything seems to be ok |
| 10:14.58 | ]expic | (no errors, no debug info). But there is no audio in both directions. |
| 10:14.58 | ]expic | Also when i call voicemail, i hear nothing one the h323 phone. |
| 10:15.11 | Mimmus | clive-: CVS changes always (peraphs... I'm not an expert...) |
| 10:15.32 | RestLessGemini | which other fxo/fxs interface cards asterisk support other then asterisk? |
| 10:15.40 | RestLessGemini | sorry |
| 10:15.40 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 10:15.44 | RestLessGemini | which other fxo/fxs interface cards asterisk support other then digium? |
| 10:16.35 | Mimmus | clive-: for instance, it looks for channel_pvt.h that is not in CVS anymore! |
| 10:18.09 | clive- | Mimmus, sounds like a lot of time is required...I wish Klaus would make a new chan_capi version |
| 10:18.48 | Mimmus | clive-: it would be better, possibly with FAX support! |
| 10:21.01 | clive- | is that why you are updating cvs? |
| 10:21.43 | Mimmus | clive-: no, I'm struggling with Astersik 'eternal' bug 2687 |
| 10:21.59 | clive- | what is that one? |
| 10:22.25 | emitrax | does anyone know what kind of account I need with cisco.com in order to download the SIP firmware for the cisco 7940 phone? |
| 10:22.37 | Mimmus | clive-: SIP not RFC compliant :-) |
| 10:22.38 | *** join/#asterisk ckruetze (~nospam@i3ED65FFE.versanet.de) |
| 10:22.58 | Mimmus | clive-: serious problems if phones are not directly connected to Asterisk |
| 10:23.19 | Mimmus | clive-: anyway, I was able to recompile chan_capi... uffff... |
| 10:23.23 | abracsas | emitrax: you have to buy something from cisco to get the account to download |
| 10:24.45 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 10:30.00 | *** join/#asterisk nextime (~nextime@213-140-22-64.fastres.net) |
| 10:31.13 | knight_ | anyone aware of any programs for any cellular phone that will set call forwarding when it detects a bluetooth device? |
| 10:31.39 | knight_ | i get crappy cell service in my house, and i want my calls automatically forwarded to my voip lines when i'm home |
| 10:37.24 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 10:37.47 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.143) |
| 10:38.17 | RoyK | is there a way to keep dynamic queue members past restarts? |
| 10:41.29 | *** join/#asterisk Jas_Williams (~jas_willi@host217-44-216-142.range217-44.btcentralplus.com) |
| 10:45.43 | foobos | royk, with cvs-head yes. there's persistent option |
| 10:46.06 | facek_ | can somebody help me with transfers? |
| 10:46.31 | RoyK | foobos: any idea if this could be hard to backport? |
| 10:46.57 | foobos | royk, no idea, i'm just an user of that feature |
| 10:46.59 | facek_ | RoyK ? |
| 10:47.05 | RoyK | jau |
| 10:47.32 | *** join/#asterisk cc (~cc@byte.fedora) |
| 10:51.00 | facek_ | RoyK can you help me with transfers? |
| 10:51.15 | RoyK | don't think so... |
| 10:51.22 | RoyK | if just pressing * doesn't workk |
| 10:51.23 | RoyK | :P |
| 10:51.46 | facek_ | RoyK work. but how can i specificy which numbers fot which people are transfering |
| 10:51.59 | facek_ | RoyK where to specif a default context for transfers? |
| 10:53.38 | RoyK | don't remember |
| 10:53.41 | RoyK | rtfw |
| 10:53.55 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:54.19 | Mother__ | greetings |
| 10:54.31 | Mother__ | anyone here that has played with OpenWRT? |
| 10:54.32 | facek_ | RoyK o am looking on voip-info, but unable to find |
| 10:54.36 | msupino | anyone using early dial with Grandstream and asterisk ? i am getting strange results when tring |
| 10:55.10 | Mother__ | I've got it running, but when I try to ipkg the asterisk package, it sits doing nothing when it gets to the 'configuring asterisk...' part |
| 10:55.53 | Mother__ | asterisk will start, but the configs are not present, this is on a 2.2 WRT54G with experimental OpenWRT |
| 10:56.09 | facek_ | RoyK should i use mgcp.conf? |
| 11:00.51 | knight_ | heh |
| 11:00.57 | knight_ | guess i have to write an app to do it |
| 11:01.31 | RoyK | facek_: for what? |
| 11:02.40 | facek_ | RoyK for enable transfers |
| 11:03.01 | *** join/#asterisk RaYmAn-Bx (user@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 11:03.20 | RoyK | ~mgcp |
| 11:03.21 | jbot | hmm... mgcp is Media Gateway Control Protocol |
| 11:04.02 | facek_ | RoyK so where? where cani specificy parameters for tansfers? and for dtmf sending by softphone and hardware hpone |
| 11:04.05 | facek_ | ? |
| 11:04.51 | RoyK | ~docs |
| 11:04.52 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:04.56 | RoyK | ~wiki? |
| 11:04.58 | jbot | i guess wiki is http://www.voip-info.org |
| 11:04.58 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 11:05.04 | foobos | facek, why not read the tutorials on www.voip-info.org instead of trying the shotgun approach to problem solving |
| 11:05.05 | RoyK | ~lart facek_ for not rtfm |
| 11:05.25 | Mother__ | lol |
| 11:05.51 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.164) |
| 11:06.31 | facek_ | foobos i am reading |
| 11:06.39 | facek_ | but i want to specifyc another context for transfer |
| 11:06.43 | facek_ | and i dont know how to do that |
| 11:07.09 | foobos | well i can tell you that mgcp has nothing to do with it |
| 11:08.33 | facek_ | foobos now i know. but still dont knwo how to specifcy another context |
| 11:12.17 | *** join/#asterisk markak2 (~twist@ndn-165-130-34.telkomadsl.co.za) |
| 11:12.23 | markak2 | afternoon all |
| 11:12.32 | jluk | afternoon |
| 11:13.21 | clive- | howzit markak |
| 11:13.54 | markak2 | i have a hassle that is not critical but annoying. asterisk no problems what so ever at this point. it jut seems to be taking a long time to answer incoming calls on the zap channels. i have tried searching for ways to speed this up but am falling short. it is currently answering after around seven rings i would prefer it to answer immidiately |
| 11:15.58 | markak2 | anyone ideas ? |
| 11:16.19 | jluk | immediate=yes |
| 11:16.25 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 11:16.32 | jluk | do you have distinctive ring setup |
| 11:17.14 | markak2 | honestly i dont know. doubt it if it has to be done manualy. |
| 11:17.27 | clive- | whats your interface? |
| 11:17.45 | markak2 | digium fxo card |
| 11:20.59 | PoWeRKiLL | hi |
| 11:22.17 | *** part/#asterisk Kaneda_ch (~Kaneda@pc90.geneva.ch.psi.com) |
| 11:22.21 | markak2 | hi |
| 11:23.32 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 11:24.01 | *** join/#asterisk Steve_DL (~Steve_DL@eth87.tas.adsl.internode.on.net) |
| 11:24.06 | markak2 | guys any idea where i might search to sort this out ? |
| 11:24.15 | markak2 | not so much where as to for whatr ? |
| 11:24.55 | clive- | mark 7 rings? |
| 11:25.13 | markak2 | never mind just got the immidate=yes jluk : thanks |
| 11:27.26 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 11:30.38 | clive- | mark did that work? |
| 11:31.09 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 11:32.04 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:32.22 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 11:35.14 | *** join/#asterisk wadeh (~wadeh@4.22.175.195) |
| 11:38.18 | rpr | One question to ALL: Is zapata driver compatible with 64 bits linux? |
| 11:40.58 | markak2 | clive : no immediate is for fxs handsets connected and tells asterisk whether to get the dialled number then dial over the fxs channel or just give you the channel and pass what you dial straight to the channel |
| 11:41.19 | markak2 | busy working on the distinctivering option |
| 11:41.47 | Jas_Williams | markak2, do you user caller id ? |
| 11:43.46 | *** join/#asterisk wired (~a@81.168.114.236) |
| 11:43.54 | wired | whats asterisk? |
| 11:44.23 | Jas_Williams | * ;-P |
| 11:44.27 | tzanger | I have to say that's probably the newbiest question I've ever seen here |
| 11:44.32 | tzanger | wired: www.asterisk.org |
| 11:45.45 | wired | hahaha |
| 11:46.03 | wired | it it like php or more of a "desktop" language like c |
| 11:46.22 | wired | oh |
| 11:46.24 | wired | read up |
| 11:46.26 | wired | nvm |
| 11:46.30 | wired | thought it was a new lang |
| 11:46.31 | zoa | rpr: yes |
| 11:48.28 | *** join/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 11:48.31 | *** part/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 11:54.52 | *** join/#asterisk dreamcode (~iancu@81.181.199.39) |
| 11:58.15 | *** join/#asterisk pino (~z@host41-28.pool21345.interbusiness.it) |
| 11:58.33 | dreamcode | why doesn't MusicOnHold start when i put someone on hold ? |
| 12:01.24 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 12:02.11 | Slainte | dreamcode, musichold.conf |
| 12:02.50 | dreamcode | it's ok an extension like : exten= 12,1,MusicOnHold().. works |
| 12:03.28 | dreamcode | my problems seems to be.. that aserisk does not detect when someone is onhold |
| 12:03.49 | Slainte | wiki for musichold |
| 12:03.52 | dreamcode | should i reinstall it ? |
| 12:04.18 | Slainte | its a dead dog, yes reinstall, adn while you are at it put the computer back in its box and send it back to the vendor, telling them you dont know how to read :) |
| 12:04.36 | Slainte | sorry I have been up all night working on a problem. |
| 12:04.45 | Slainte | the wiki has a great musichold section |
| 12:05.54 | dreamcode | man i read . i think all about musiconhold.. but i didn't found any thing about my problem |
| 12:06.10 | dreamcode | :( |
| 12:06.46 | dreamcode | if i run * with debug.. i don't have any line refering to aplication MusicOnHold .. to start or to stop |
| 12:08.47 | dreamcode | Scenario is:1) A call B 2) B answer ,then push Flash 3) shouldn't be A on Hold ? |
| 12:13.01 | knight_ | wow, Live Blackjack on Golden Palace is fun :) |
| 12:13.08 | knight_ | you see a live video feed of the cards in play |
| 12:13.21 | knight_ | and there's a reader that detects the cards dealt |
| 12:14.41 | *** join/#asterisk danalien (~danalien@danalien.user) |
| 12:19.01 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 12:19.19 | ellvis | re |
| 12:20.13 | tzanger | love that post to -users |
| 12:20.15 | tzanger | "I am a big fan of both curry and Asterisk, but have not as of yet found a way to combine my loves." |
| 12:20.17 | ellvis | as i am not abble to modrobe ztdummy, should i jump under some truck or go to swim to the lake with a rock around my neck? |
| 12:20.32 | tzanger | ellvis: that isn't the suggested course of action for problems of this nature |
| 12:21.04 | ellvis | tzanger: ah, ok, i'll try some other chanel :) |
| 12:21.30 | riksta | you could have a pot of curry on the stove, with one of those automated switches, hooked up to asterisk |
| 12:21.35 | tzanger | ztdummy wants a very specific type of USB host. You probably don't have it. |
| 12:21.35 | riksta | and you could call the stove on the way home from work |
| 12:21.36 | inspired | is G723 included in Asterisk? |
| 12:22.07 | inspired | or do I need to pay a royalty? |
| 12:22.55 | ellvis | tzanger: yes, i found it |
| 12:23.25 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:30.56 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 12:31.32 | *** join/#asterisk jmav (~jmav@201.243.76.158) |
| 12:31.39 | jmav | Hello all |
| 12:34.07 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 12:39.06 | facek_ | hi |
| 12:47.48 | Zeeek | "America Online (AOL) customers in 40 select cities have "Got VoIP!" Thursday, AOL began to roll out the beginnings of what will become a nationwide Voice over Internet Protocol (VoIP) phone service called AOL Internet Phone Service. " |
| 12:49.33 | jmav | I have a question ..... when i make calls with my granstream 286 using the zap card everithing works fine .... but if i try to connect from one grandstream to another grandstream the conection its not good anyone having this problem ? And if I use a service like stanaphone also have the same problem |
| 12:55.14 | snitter | what does that 'its not good' mean? |
| 12:55.52 | *** join/#asterisk jmac (~dj@pc-24-181-187-85.sbi.ct.charter.com) |
| 12:56.08 | RoyK | Apr 7 14:47:43 WARNING[8637]: Unable to create RTP session: Address already in use |
| 12:56.08 | RoyK | Apr 7 14:47:43 WARNING[8637]: No RTP ports remaining |
| 12:56.10 | RoyK | WTF? |
| 12:56.25 | RoyK | beleive me, I don't have 2000 active calls through this box |
| 12:56.29 | Zeeek | not enough rtp ports? |
| 12:57.10 | snitter | uh, omg |
| 12:58.12 | RoyK | Zeeek: 2000 should be enough... |
| 12:58.22 | RoyK | unless asterisk doesn't release them, that is |
| 12:59.16 | Zeeek | exactly |
| 13:00.27 | jmav | <PROTECTED> |
| 13:00.30 | zoa | hehe |
| 13:01.09 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 13:01.34 | RoyK | hm |
| 13:01.35 | RoyK | SIP Debugging Enabled for IP: 80.202.22.245:50197 |
| 13:01.35 | miller7 | anyone here ever managed to make spandsp work to send out a fax? |
| 13:01.40 | RoyK | what is that port number used for? |
| 13:02.45 | zoa | thats a remote sip signalling port |
| 13:05.03 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 13:05.15 | *** join/#asterisk tugalone (~tugalong@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 13:05.47 | Essobi | miller7 I have |
| 13:06.40 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 13:07.34 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 13:07.41 | ManxPower | *coffeegasam* |
| 13:07.59 | RoyK | zoa: I have a problem with the client |
| 13:08.06 | RoyK | I get REGISTERs but not INVITEs |
| 13:08.09 | RoyK | for some reason |
| 13:08.09 | RoyK | :P |
| 13:08.40 | zoa | i also have a problem asterisk has issues under my heavy load :/ |
| 13:09.00 | zoa | Apr 7 15:55:54 WARNING[22335]: pbx.c:2010 ast_pbx_start: Failed to create new channel thread |
| 13:09.00 | zoa | Apr 7 15:55:54 WARNING[22335]: chan_iax2.c:2765 ast_iax2_new: Unable to start PBX on IAX2/agent918@10.0.1.136:4569/443 |
| 13:10.21 | markak2 | has anyone here played with the indications.conf file. if i change the default from us to za would this help with ring detection and busy detection gl,obaly on asterisk |
| 13:11.25 | RoyK | no |
| 13:11.31 | RoyK | it's only for playing stuff iirc |
| 13:12.45 | Essobi | markak2 FXO lines? |
| 13:13.13 | Essobi | *SHRUG* |
| 13:13.31 | zoa | i also think its only for playing audio |
| 13:13.53 | Essobi | What's format is the easiest to transcode from to G711 and 729 for app_playback? |
| 13:14.15 | jmac | anyone have any thoughts as to why genzaptelconf -s -d would generate some ugly errors? |
| 13:14.21 | Essobi | I believe the actual indication code/configs is related to the zaptel drivers, ehh? |
| 13:14.27 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 13:14.42 | Essobi | miller7 Most of the time. some faxes have a problem. |
| 13:15.00 | Essobi | I have super clean lines too, thou. :) |
| 13:15.19 | miller7 | Essobi: I can receive faxes fine but I can't send no matter what I do |
| 13:15.26 | miller7 | I have 2 * boxes and both are the same |
| 13:15.27 | Essobi | To whom? |
| 13:15.46 | miller7 | ? |
| 13:15.52 | Essobi | Oh, umm. are you sending from one * to another *? |
| 13:15.58 | miller7 | For test, yes |
| 13:16.00 | Essobi | Across IP? |
| 13:16.07 | miller7 | no, across TDM |
| 13:16.32 | jmav | i am having problems when conecting 2 voip grandstream 286 over asterisk |
| 13:16.33 | Essobi | Hmm. Sounds like a rx/tx loss or echo cancellation is killing it. |
| 13:16.52 | jmav | anyone can help me |
| 13:16.52 | miller7 | the tx only? |
| 13:17.00 | Essobi | miller7 But that's just a guess.. |
| 13:17.02 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
| 13:17.12 | Essobi | I mean rx/tx gain loss on the TDM |
| 13:17.16 | miller7 | ah |
| 13:17.31 | miller7 | have you ever tried to txfax to the same * box via TDM? |
| 13:17.43 | Essobi | Umm. Yea. |
| 13:17.50 | miller7 | does it work? |
| 13:17.54 | Essobi | Oh did you disable the /tmp disk writes? |
| 13:18.01 | miller7 | nope |
| 13:18.09 | miller7 | normal distro installation |
| 13:18.11 | Essobi | That'll kill the performance especially if you got slow disk |
| 13:18.15 | *** join/#asterisk Prowler (~Stephen@24-116-250-78.cpe.cableone.net) |
| 13:18.18 | Essobi | "distro"? |
| 13:18.22 | miller7 | linux gentoo |
| 13:18.24 | markak2 | essobi : sorry fxs lines |
| 13:18.29 | Essobi | What version of spandsp/* are you runing? |
| 13:18.31 | markak2 | essobi : sorry fxo lines yes |
| 13:18.44 | miller7 | latest spandsp |
| 13:18.52 | miller7 | asterisk is 1.0.2 I think on the PRI box |
| 13:18.57 | miller7 | yep |
| 13:18.58 | Essobi | markak2 I'm pretty sure that's ni the libpri stuff somewhere. |
| 13:19.05 | Essobi | miller7 that's your problem. |
| 13:19.14 | miller7 | u sure? |
| 13:19.21 | Essobi | miller7 use -head and the newest spandsp |
| 13:19.34 | Essobi | the new one is much much much better the it's previos ones. |
| 13:19.48 | jmac | any thoughts as to why my tdm extensions can't seem to dial anything, but my pc-phone can ring the extensions? |
| 13:19.50 | Essobi | I think there's still a tiny problem with canon fax machines, because they are simply stupid. |
| 13:19.53 | miller7 | well, these are both production boxes so I can't easily put the -head |
| 13:20.12 | Essobi | miller7 True enough. Been there, done that. |
| 13:20.18 | miller7 | but I will go that way to test |
| 13:20.23 | miller7 | at least that's something |
| 13:20.36 | miller7 | back to cvs :P |
| 13:21.22 | miller7 | at least, if I sent you a tiff file, can you see if you can fax it? because i did some conversion from windows prn to tiff? |
| 13:22.00 | Essobi | miller7 just back all your current versions up first.. makes it a lot easier. |
| 13:24.17 | MikeJ[Laptop] | are there sip firmware's for the 7940g/7960g cisco phones, I know there are for the onese without the g... but was not sure about the new phone |
| 13:24.18 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 13:24.38 | Essobi | Aight, meeting time. |
| 13:25.07 | Essobi | MikeJ[Laptop] G and non G's only differ on the PoE spec and the pictures on the buttons. |
| 13:25.23 | Essobi | and the weight of the plastic on the shell. |
| 13:25.43 | *** join/#asterisk olivier_ (~olivier_@82.127.99.32) |
| 13:25.43 | rpr | Sorry, I repeat my previous question: Is Zaptel / asterisk compatible with Linux 64? Are there any special consideration? Are there any document that I must read? |
| 13:26.07 | Essobi | rpr AFAIK, there is no problems with it. |
| 13:26.08 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 13:26.09 | ariel_ | hello everyone |
| 13:26.22 | miller7 | Essobi: sorry, got disconnected |
| 13:26.33 | miller7 | what did I miss after you said about backup? |
| 13:26.59 | jmac | <Essobi> Aight, meeting time. |
| 13:27.19 | jcollie | rpr: do you mean running asterisk on x86_64, ia64, or ppc64? |
| 13:27.43 | MikeJ[Laptop] | Essobi, cool, thank you |
| 13:27.47 | rpr | Jcollie: x86_64 |
| 13:28.15 | Essobi | There are G729 codecs from digium for x86_86 |
| 13:28.22 | jcollie | no, i think that there are people running asterisk on x86_64 successfully |
| 13:28.51 | Essobi | I've heard of people running it too.. just make sure your mobo will work with any planned hardware and you're GOLD, JERRY, PURE GOLD. |
| 13:29.15 | MikeJ[Laptop] | does anyone have a working tiff to pdf to email script handy... the one I have is not working cleanly |
| 13:29.23 | MikeJ[Laptop] | for fax to email |
| 13:29.40 | Essobi | "script"? |
| 13:29.54 | Essobi | You seriously can't write shell? |
| 13:30.06 | tzanger | shell |
| 13:30.07 | tzanger | I just did |
| 13:30.09 | tzanger | see, it's easy |
| 13:30.23 | Essobi | You're so mean tzanger |
| 13:30.24 | rpr | jcollie: I had think that i could had problems whit kernel driver of zaptel hardware. |
| 13:30.33 | Essobi | Oh wait.. that's me. :) |
| 13:31.40 | jmac | sorry to repeate, but any thoughts as to why my tdm extensions can't seem to dial anything, but my pc-phone can ring the extensions? |
| 13:32.07 | *** join/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 13:32.10 | rpr | I if there are not special considerations, tomorrow I will try it with one opteron processor. |
| 13:32.17 | tzanger | jmac: probably because the dialplan that your zaptel devices are dumping in to doesn't have anything useful in it |
| 13:32.22 | miller7 | jmac: perhaps your dialplan is wrong? |
| 13:32.23 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 13:32.45 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 13:32.55 | jmac | i knew being a lowly *@h user would get me in trouble quickly |
| 13:33.06 | queuetue | Is there any way to reuse a bunch of digital phones with asterisk? (IX-12KTD-2 , etc) |
| 13:33.16 | DrWho17 | Is there such a thing as sip echo cancellation in asterisk |
| 13:33.34 | Makenshi | i'm running asterisk on xeon em64t cpus with x86_64 kernel, no problems |
| 13:33.50 | miller7 | queuetue: Yes, sell them and buy IP phones instead :P |
| 13:33.54 | DrWho17 | I've recently started egressing calls via sip to a Lucent TNT, instead of out zap channels, and picked up an echo to the caller |
| 13:34.14 | queuetue | miller7, That's "plan a" already - but I thought I'd check. :) |
| 13:34.28 | DrWho17 | pbx -> t1 cross -> asterisk -> sip -> tnt -> pstn |
| 13:34.33 | queuetue | DrWho17, http://www.voip-info.org/tiki-index.php?page=Asterisk+echo+cancellation ? |
| 13:34.35 | miller7 | queuetue: you can also use them as a nice way to bring the * box higher so you don't have to bend |
| 13:34.42 | DrWho17 | queuetue: yea been there |
| 13:34.51 | DrWho17 | it only talked about zap channels |
| 13:35.29 | Slainte | DrWho, read the article on ciscos site about echo. There is a link on the wiki for it |
| 13:35.30 | DrWho17 | pbx -> t1 cross -> asterisk -> zap -> pstn works fine |
| 13:35.53 | queuetue | miller7, That line did not get the laugh from the client we both expected. :) |
| 13:36.23 | miller7 | oh well... |
| 13:38.26 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 13:38.40 | markak2 | does anyone know what this is popped up on my CLI first time i have seen it |
| 13:38.42 | markak2 | Saved useragent "Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26" for peer 2008 |
| 13:39.11 | queuetue | miller7, are the interfaces on digital phones standardized? Are they public? |
| 13:39.19 | miller7 | markak2: some avaya perhaps? |
| 13:39.20 | jcollie | markak2: debug message from SIP i think |
| 13:39.30 | miller7 | queuetue: I have no idea |
| 13:39.40 | markak2 | miller7 : ahh thanks yes 1 avaya hardphone |
| 13:39.53 | miller7 | markak2: you're welcome |
| 13:39.56 | jcollie | queuetue: no, each vendor has a different proprietary standard |
| 13:40.40 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:41.27 | Mimmus | hi, I have a problem with app_capiFax.c crashing Asterix: does anyone know who is its author? |
| 13:41.53 | miller7 | Mimmus: doesn't the source mention him? |
| 13:42.24 | Mimmus | miller7: no, I found it as a patch for chan_capi |
| 13:42.37 | jmac | i may have unwittingly used a PCI v2.1 mb, if in fact i did, would my tdm cards not work at all or act strangely? |
| 13:42.39 | miller7 | then you have to google it perhaps |
| 13:44.09 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:44.46 | Mimmus | miller7: peraphs he is cas at accld.com (Carl Sempla) |
| 13:45.01 | miller7 | Mimmus: I have no idea |
| 13:45.08 | miller7 | perhaps he is |
| 13:45.15 | miller7 | perhaps he's not |
| 13:46.48 | _THEEND_ | someone uses cisco ip phones? |
| 13:47.18 | bjohnson | miller7: some digital phones have converters you can buy for voip usage .. I think Nortel for example has one. End result is you can buy a voip phone for the same price |
| 13:47.20 | *** join/#asterisk mct1 (~malcolmct@pcp0010478837pcs.hamden01.ct.comcast.net) |
| 13:47.46 | miller7 | bjohnson: I'm sure you can |
| 13:48.57 | jsharp | Glorp |
| 13:49.05 | bjohnson | miller7: in the end you can point out that the current waste of investment was due to the use of a proprietary system encouraged by typical hardware vendors and that VOIP hardware that follows public standards will be adaptable to other hardware in the future that follows the same standards. A feature that is not possible in proprietary systems |
| 13:49.22 | danalien | is it possible to 'software cross' the zaptel+bristuff driver? What I mean, is control what signal goes to what pin - instead of having to slit a kabel and 'hardware cross' it :-) |
| 13:49.33 | miller7 | bjohnson: I agree |
| 13:49.50 | DrWho17 | Slainte: so are you pointing me in the direction of a level mismatch? |
| 13:50.59 | *** join/#asterisk moy (~kvirc@201.135.98.129) |
| 13:51.11 | Slainte | DrWho17: It is important to understand how echo is created, You upgraded the TAOS on your TNT lately? |
| 13:52.02 | *** join/#asterisk devi-o (~dev@gw.01063telecom.de) |
| 13:52.08 | devi-o | hi everyone |
| 13:52.55 | DrWho17 | Slainte: no it's 10.1.1 |
| 13:53.00 | blankman | Hey guys. |
| 13:53.50 | devi-o | is there someone who successfully compiled the mysql addon module for asterisk-realtime, i am failing because of a mutex statement which is undeclared in the lock.h - include |
| 13:54.02 | devi-o | host OS is linux |
| 13:54.50 | DrWho17 | Slainte: so since it's the caller hearing the echo, I probably should bump the output-pad on the t1's |
| 13:55.03 | DrWho17 | the called-party doesn't get any echo |
| 13:55.19 | DrWho17 | I can't find any echo canceller settings on the tnt itself |
| 13:57.47 | devi-o | hm when is it the best time to ask such questions in here ? :) |
| 13:58.06 | miller7 | after lunch |
| 13:58.08 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 13:58.12 | jcollie | when someone that knows the answer is around |
| 13:58.23 | DrWho17 | devi-o: I have compiled it, and use it |
| 13:58.40 | devi-o | ok. |
| 13:58.46 | *** part/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:58.53 | DrWho17 | last time was 2 weeks ago CVS |
| 13:59.00 | devi-o | the sources are fresh here |
| 13:59.16 | devi-o | so, you had no trouble ? |
| 13:59.22 | RoyK | DrWho17: cvs co -D 'a fortnight ago'? |
| 13:59.51 | DrWho17 | well, I manually applied the patch to the makefile, but other then that it worked fine |
| 14:00.08 | blankman | So, does anyone on currently have a T1 using em_w signaling to a provider? |
| 14:00.22 | DrWho17 | blankwan: I am doing that also |
| 14:00.24 | devi-o | ill blame it on the weather for today.. ill see tommorrow :) |
| 14:00.46 | blankman | DrWho17, what did version of * are you running? |
| 14:00.56 | *** part/#asterisk donavan (~donavan@4wx.net) |
| 14:01.02 | DrWho17 | haha, all kinds of versions |
| 14:01.19 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 14:01.22 | blankman | I am having a problem that for some reason * isn't waiting on the wink single from the provider for the out pulse. |
| 14:01.40 | DrWho17 | try increasing the wink |
| 14:01.49 | ManxPower | can anyone resolve www.sipura.com ? |
| 14:01.56 | Essobi | wink wink |
| 14:01.59 | devi-o | lol |
| 14:02.07 | Essobi | Nope.. she still isn't picking me up. |
| 14:02.07 | RoyK | $ host www.sipura.com |
| 14:02.08 | RoyK | www.sipura.com has address 66.43.93.101 |
| 14:02.15 | Essobi | Mayday mayday, |
| 14:02.22 | devi-o | youre already down |
| 14:02.24 | devi-o | :P |
| 14:02.28 | Essobi | damn |
| 14:02.32 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 14:02.32 | ManxPower | RoyK, Thanks. Must be a local issue then |
| 14:02.35 | Mimmus | hi, does anyone knows why channel_pvt.h is not in CVS anymore? |
| 14:02.59 | blankman | DrWho17, I have tried to mess with the different settings, but I can't seem to get the system to wait for their wink. Instead it just starts to out pulse the number when the hook state trans. occurs. |
| 14:03.22 | blankman | Digium says that the system waits for the wink ... but looking at it on t-bird, it doesn't ... |
| 14:03.37 | blankman | DrWho17, did you have this issue on yours? |
| 14:03.49 | Essobi | AND SO FROM THE ASHES OF THE PRIVATE CHANNEL STRUCT, RIZE TEH WINK! WINK WINK. |
| 14:03.59 | Essobi | I need more coffee. |
| 14:04.03 | DrWho17 | signalling=em_w |
| 14:04.03 | DrWho17 | rxwink=300 |
| 14:04.14 | DrWho17 | works, I've never touched it since then |
| 14:04.23 | *** join/#asterisk MattB2 (~mattb@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 14:04.26 | blankman | DrWho17, yeah, tried that did work for me ... who is your provider? |
| 14:04.30 | Essobi | What version of *? |
| 14:04.31 | DrWho17 | Verizon |
| 14:04.47 | blankman | DrWho17, I have GC ... |
| 14:04.54 | DrWho17 | Asterisk CVS-HEAD-10/08/04-23:56:27 |
| 14:04.58 | MikeJ[Laptop] | Essobi, I figured out my issue |
| 14:05.00 | Essobi | there was some discussion a while back about e&m missing winks in some cases.. can't remembers what thou |
| 14:05.05 | devi-o | are most of you real techies workin at the on or another telecom provider ? |
| 14:05.16 | MikeJ[Laptop] | it was that it is a bad idea to work on scripts of 4:30 am |
| 14:05.17 | MattB2 | hi all... not sure if this is the proper place to ask... How is the phone socket on the X100P supposed to be used? When we plug a phone in with a line attached, the phone just rings constantly! Card is a clone not a Digium one - could be the propblem? |
| 14:05.21 | Essobi | MikeJ[Laptop] lol.. admission is the first step to recovery. |
| 14:05.36 | devi-o | father i sinned |
| 14:05.43 | devi-o | i installes windows a thousand times |
| 14:06.05 | Essobi | MikeJ[Laptop] Yea.. Drunken kung-foo is way funner then tired kung-foo. |
| 14:06.07 | MikeJ[Laptop] | when you mistakenly delete one letter of a variable in one place, it does not know you *mean* the same thing where you refer to that variable |
| 14:06.14 | MikeJ[Laptop] | hehe |
| 14:06.17 | Essobi | ;) |
| 14:06.26 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 14:06.27 | devi-o | but it was without a licence ... so u are forgiven he spoke to me |
| 14:06.54 | ellvis | anyone can recomend some IAX phone for windows? |
| 14:07.07 | ellvis | i tried diax, but i'd like to "play" with another one |
| 14:07.16 | MikeJ[Laptop] | elvis, sure, testcall.exe |
| 14:07.18 | ellvis | and google said not much about it... |
| 14:07.26 | blankman | DrWho17, of the zaptel guru's know what the start, prewink and debounce are exactly? |
| 14:07.29 | Mimmus | hi, does anyone knows why channel_pvt.h is not in CVS anymore? |
| 14:07.41 | Essobi | I havn't used a single iax phone on windows.. save firefly that crashed my box when I installed it. |
| 14:07.45 | Essobi | Mimmus I answered you. |
| 14:07.45 | blankman | ellvis, firefly |
| 14:07.53 | antifuchs | Mimmus: the pvt structure was removed |
| 14:08.02 | Essobi | Mimmus in -dev but you left. |
| 14:08.14 | Mimmus | Essobi: sorry, I missed it |
| 14:08.19 | ellvis | ok, thanks, i'll take a look |
| 14:08.25 | devi-o | oh there are more * related channels ?! |
| 14:08.25 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnbm.dialup.mindspring.com) |
| 14:08.33 | Essobi | I had terrrrible luck with firefly |
| 14:08.44 | blankman | Essobi, it is windows. |
| 14:08.59 | Mimmus | Essobi: I'm coming back to -dev, hoping in a replay from you! |
| 14:09.04 | *** part/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 14:09.18 | *** join/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
| 14:09.32 | blankman | ~seen tzanger |
| 14:09.34 | jbot | tzanger is currently on #asterisk (14h 35m 12s). Has said a total of 1122 messages. Is idling for 37m 17s |
| 14:09.39 | ellvis | well, i also experienced not much luck with firefly |
| 14:09.47 | ellvis | but will probably try once again |
| 14:11.22 | Darwin[laptop] | asterisk is now working on fbsd 5.4 |
| 14:11.39 | Gand_DJ | I used firefly (still do for freshtel / verbiage) |
| 14:11.53 | Gand_DJ | don't recall having issues linking iax -> * |
| 14:14.17 | *** join/#asterisk cjrm (~cjrm@81-178-22-214.dsl.pipex.com) |
| 14:14.22 | cjrm | hi people |
| 14:14.48 | Darwin[laptop] | grr no audio |
| 14:15.08 | ariel_ | Darwin[laptop], what you spoke too soon? |
| 14:15.32 | Darwin[laptop] | well before it was crashing the kernel |
| 14:15.44 | cjrm | I'm interested in using asterisk to do 3rd party call control. I want to do it as cheaply as possible. What hardware will I need??? Could I use two bog standard modems? |
| 14:15.47 | Darwin[laptop] | I updated the zaptel and it stopped crashing |
| 14:15.58 | Darwin[laptop] | but now no audio |
| 14:16.11 | Darwin[laptop] | centos |
| 14:16.27 | cjrm | I only need to manage a single call at a time. |
| 14:16.43 | ariel_ | Darwin[laptop], you don't want to know about it. It's a RHEL repackaged OS |
| 14:17.26 | ariel_ | cjrm, anything thing would work if your not transcoding. |
| 14:17.26 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 14:17.51 | ariel_ | but modems are not support really only the X101p card or clones of them for fxo ports |
| 14:18.16 | Darwin[laptop] | no I had not heard about it |
| 14:18.19 | cjrm | ariel_: what do you mean, transcoding? |
| 14:18.54 | ariel_ | Darwin[laptop], look at www.centos.org great iso |
| 14:18.54 | malverian | cjrm, You can get a W1000P for 7$ on ebay. |
| 14:19.02 | malverian | cjrm, That's what I'm using and it works like a charm. |
| 14:19.22 | *** join/#asterisk kenshinblade (~maccheron@host37-9.pool80105.interbusiness.it) |
| 14:19.54 | *** join/#asterisk phpboy (~sj@tpr-165-249-135.telkomadsl.co.za) |
| 14:19.56 | phpboy | hey guys |
| 14:19.57 | Gand_DJ | $7, but like $15 shipping... lol |
| 14:19.58 | ariel_ | cjrm, more info will be needed from you on what you want to do. |
| 14:20.06 | ariel_ | phpboy, hello |
| 14:20.08 | phpboy | do you need the zaptel drivers installed for linux aswell |
| 14:20.09 | phpboy | ? |
| 14:20.25 | phpboy | to get the digium card working with Linux |
| 14:20.28 | phpboy | I mean |
| 14:20.30 | phpboy | with asterisk |
| 14:20.40 | ariel_ | if you have a zaptel card yes |
| 14:20.56 | phpboy | zeptel == digium drivers |
| 14:20.56 | kenshinblade | hi. could someone help me with SetCdrUserField action in manager API? I've found no docs about it and I didn't manage to ude it properly so far |
| 14:20.57 | phpboy | no? |
| 14:21.37 | ariel_ | phpboy, what are you trying to do? |
| 14:22.31 | bjohnson | what is a w1000p? |
| 14:22.36 | cjrm | ariel_: All I want to do is have a computer establish a call between 2 mobile phones. I figured the cheapest way would be to buy two modems, setup two voice calls and pipe the audio between them with some magic on my linux box. It's just for a demonstration so I don't need to handle more than one call at a time. |
| 14:23.07 | bjohnson | cjrm: the cheapest way would be to call from cell phone 1 to cell phone 2 |
| 14:23.11 | ariel_ | bjohnson, misstype |
| 14:23.12 | Gand_DJ | lol |
| 14:23.18 | cjrm | I figured asterisk might be able to do 'the magic' |
| 14:23.28 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 14:23.29 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 14:23.30 | cjrm | bjohnson: Can't do that. |
| 14:23.59 | bjohnson | then go voip .. you don't need to tie into pstn at all .. they will do it |
| 14:24.03 | cjrm | bjohnson: I want the computer to establish the call between the two cell phones. |
| 14:24.19 | Gand_DJ | You'd need cellphone terminals then? |
| 14:24.38 | Gand_DJ | 1 to broadcast to the cellphones.. and then program cellphones to connect to the terminal? |
| 14:24.50 | bjohnson | in N.A. it would be approx $0.02 per minute x the 2 calls (plus whatever cell phone per minute costs for your plan apply) |
| 14:25.16 | devi-o | or program something into the api that triggers the call via another cellphone api (at serial or irda or bluetooth ports) ? |
| 14:25.27 | bjohnson | my understanding of gsm terminals is that they talk to the gsm provider .. not act as a gsm provider |
| 14:25.54 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 14:26.29 | cjrm | bjohnson: N.A? |
| 14:26.37 | *** join/#asterisk Zebble (~Zebble@66.207.107.50) |
| 14:26.38 | Gand_DJ | North America |
| 14:27.02 | Gand_DJ | You could use gsm terminal, but then you're paying for 4 connections |
| 14:27.06 | Mimmus | antifuchs: you wrote me that pvt structures was removed; is it difficlut to patch an application still using them? |
| 14:27.09 | Gand_DJ | if they talk to the provider |
| 14:27.26 | jmac | maybe meetme |
| 14:27.52 | ariel_ | cjrm, in any case asterisk does not work with just any modem. |
| 14:27.58 | bjohnson | cjrm: north america |
| 14:29.04 | cjrm | ariel_: yeah... So I see :) ... If I have two phone lines for outgoing calls, what do I need to do to get asterisk to call two numbers and link the calls together? |
| 14:29.24 | cjrm | What hardware do I need? And what software do I need to write :) |
| 14:29.45 | antifuchs | Mimmus: AFAIK, it isn't |
| 14:29.47 | ellvis | bye |
| 14:29.48 | bjohnson | cjrm: for cell phones .. the advantage of voip really is to pick a plan with lots of minutes, don't worry about LD costs, and create a dial in system for the cell users to call in to dial out LD calls |
| 14:30.16 | bjohnson | cjrm: you need 2 fxo ports .. through pci cards or ATAs |
| 14:30.21 | devi-o | i have to quit.. cU guys |
| 14:30.25 | devi-o | and gals :P |
| 14:30.29 | antifuchs | Mimmus: the cvs log of the change has a reference to the bug number where they explain how to rework your modules |
| 14:30.33 | ariel_ | cjrm, either a TDM02b card that has two FXO ports or two X101P cards. any celeron or Piii with 256mg ram will do for 2 connections |
| 14:30.46 | jmac | i may have unwittingly used a PCI v2.1 mb, if in fact i did, would my tdm cards not work at all or act strangely? |
| 14:30.58 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:30.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.10 | Darwin[laptop] | ok fixed |
| 14:31.10 | bjohnson | cjrm: but to do what I "think" you want to do .. 1. you do not need any hardware 2. there are cheaper and easier ways to do it then using an asterisk box |
| 14:31.10 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 14:31.26 | Darwin[laptop] | i had to turn off g729 in sip.conf |
| 14:31.28 | Mimmus | antifuchs: emmmm... I'm not so able to retrieve such beast from CVS! |
| 14:31.35 | Darwin[laptop] | then got audio on fbsd |
| 14:31.41 | cjrm | bjohnson: I'm not interested in making calls from the cellphones. I need my software (and maybe asterisk) to establish a call between the two phones. |
| 14:31.54 | RoyK | can someone explain how sms works? I have this dect phone that allows me to send sms. does that open a modem connection or how does it do it? it wants a central number (one for recv and one for send) |
| 14:31.57 | cjrm | bjohnson: like what? |
| 14:32.16 | antifuchs | Mimmus: you could use a bit more self-confidence |
| 14:32.18 | bjohnson | cjrm: well .. I guess I'll stop telling you that you don't need hardware since you don't want to listen |
| 14:32.40 | Gand_DJ | I guess the modem would dial into a main sms station, pass on the sms info, and the sms station would forward to the proper place |
| 14:32.52 | Mimmus | antifuchs: in other words, reading the man page? |
| 14:33.06 | cjrm | bjohnson: I am listening, I'm just not sure whether you undestand what I want? :) |
| 14:33.07 | Zeeek | RoyK wiki |
| 14:33.13 | antifuchs | Mimmus: "cvs log channels/chan_sip.c" isn't that hard (: |
| 14:33.24 | RoyK | Zeeek: voip-info? |
| 14:33.29 | Zeeek | ya |
| 14:33.36 | Mimmus | antifuchs: I have not the minimal experience with CVS |
| 14:33.42 | Mimmus | antifuchs: thank you |
| 14:33.43 | Darwin[laptop] | wow nice to have * back |
| 14:33.54 | ariel_ | I am out for a while. I got to see if I can get some work. (Need to feed the family) |
| 14:33.55 | cjrm | bjohnson: Is there a web-based service for 3rd party call control that you know of? |
| 14:34.23 | Zeeek | RoyK and then this one http://www.automated.it/asterisk/sms.html |
| 14:35.10 | Gand_DJ | I noticed that when I loaded *@home, it has hostname of asterisk.local |
| 14:35.19 | Gand_DJ | is there a way to have it interface with a domain? |
| 14:35.25 | Gand_DJ | The linux CenOS part |
| 14:35.26 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmtm.dialup.mindspring.com) |
| 14:35.35 | Mimmus | antifuchs: OK, bug #3573 |
| 14:35.39 | Gand_DJ | That way email part works right |
| 14:36.18 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 14:36.19 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 14:37.49 | antifuchs | right. that one. |
| 14:38.09 | Gand_DJ | hrm.... I get this everytime I load *@home (end of loading).... "** /var/www/html/panel/op_buttons_custom.cfg not readable... skipping" |
| 14:38.22 | Gand_DJ | Also says "sip/200 in position 2" |
| 14:38.36 | Gand_DJ | don't recall that happening when I manually installed * sometime last year |
| 14:42.19 | phpboy | where can I get a copy of a zaptel rpm? |
| 14:43.09 | newl | holy crap, Mandrakesoft is to soon be known as Mandriva. gadzooks what a bloody stupid name! |
| 14:43.17 | blankman | Hey, anyone know where this function is declared? Meaning which file ... I need to figure out its "new" signature since it hasn't been updated in either cvs head or stable for the app_sql_postgres.c |
| 14:43.24 | blankman | pbx_builtin_setvar_helper(struct ast_channel *chan, char *name, char *value); |
| 14:44.38 | moy | why dont you try grep command to find out? |
| 14:45.04 | blankman | moy ... I did ... but it is extened every where and I am not a regex expert :-( |
| 14:45.17 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 14:45.36 | cjrm | Whats the cheapest way of getting a computer to phone two phones and then join the calls together? |
| 14:45.57 | Darwin[laptop] | anyone having issues with grandstream phones and the message button not working with the latest flash |
| 14:46.13 | newl | cjrm: For the two phones to call one another sans computer. :) |
| 14:46.34 | DrWho17 | he newl |
| 14:46.59 | newl | heya DrWho |
| 14:46.59 | moy | blankman: :p ...... sorry then, i dont have idea of what file is, the only files i have needed are the agi ones |
| 14:47.27 | cjrm | newl: No, the computer to call two phones and then link the two calls together, so both parties can talk to each other. |
| 14:47.33 | Gand_DJ | yeay... got something from fedex arriving today.. I think it might be my free copy of Office 2003 Premium :) |
| 14:47.47 | newl | cjrm: You mean ala 3-way calling? |
| 14:48.01 | Gand_DJ | 100% legal |
| 14:48.18 | DrWho17 | how does asterisk handle PIC codes? |
| 14:48.19 | Darwin[laptop] | MS office is junk use openoffice |
| 14:48.31 | newl | Darwin[laptop]: No problems with the message button here with the latest flash. |
| 14:48.38 | Darwin[laptop] | or koffice |
| 14:48.47 | Darwin[laptop] | hmm |
| 14:48.55 | langals | Hi there...anyone have experience with using meetme? |
| 14:48.59 | Darwin[laptop] | newl mine stopped working |
| 14:49.02 | langals | I have a few questions on it |
| 14:49.07 | DrWho17 | apparently I'm supposed to include PIC codes for our long distance provider |
| 14:49.23 | *** join/#asterisk The_Ball (~alex@static-112.35.240.220.dsl.comindico.com.au) |
| 14:49.24 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 14:49.35 | jsharp | Need to prepend the PIC before dialing an outgoing call? |
| 14:49.41 | cjrm | newl: no, I mean 3rd party call control. Picture the scenario. Some software decides that 2 people should speak to each other. It calls person A and person B and joins the calls together. |
| 14:49.49 | The_Ball | does anybody know if a user using the normal version of firefly will be able to call a asterisk server? |
| 14:50.11 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 14:50.23 | Uther_P | e |
| 14:50.27 | hajekd | hi, someone here is using voipjet? |
| 14:50.46 | hajekd | they said: "We now fully support CallerID to USA, Canada and most European countries. Some other ones, too! |
| 14:51.21 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 14:52.45 | firestrm | hey.. um is the command to grab stable not: cvs checkout -r v1_0_rc_2 asterisk ? at least thats what asterisk docs says it should be.. but cvs cant find the tg |
| 14:52.54 | Nugget | firestrm: no, that's not it. |
| 14:53.02 | Nugget | where do the asterisk docs say that? |
| 14:53.19 | Nugget | "cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 14:53.26 | firestrm | Nugget http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN284 |
| 14:53.32 | Nugget | thanks |
| 14:53.33 | bjohnson | cjrm: info on the on the wiki. hardware needed: none if you use a voip provided |
| 14:53.37 | bjohnson | cjrm: info on the on the wiki. hardware needed: none if you use a voip provider |
| 14:53.46 | Gand_DJ | The_Ball, I don't think so |
| 14:53.51 | Gand_DJ | you have to use 3rd party versio |
| 14:54.28 | The_Ball | Gand_DJ, ok, i knew the third party version works, i was just curios if the normal version would be able to call astreisk |
| 14:54.45 | Gand_DJ | I think normal is locked to freshtel only |
| 14:54.56 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:54.56 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:55.24 | cjrm | bjohnson: cheers :) |
| 14:55.27 | The_Ball | gambolputty, this is from their FAQ: What protocols does Firefly support? Firefly uses an enhanced version of IAX by default, however standard IAX and SIP are also fully supported. |
| 14:55.39 | The_Ball | Gand_DJ, eh, that was for you |
| 14:55.43 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:55.52 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:55.52 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:56.13 | Gand_DJ | Hrm, if you can setup domain & stuff in normal firefly, it might work |
| 14:56.25 | Gand_DJ | I've never played with it.. I have 3rd party version on my pc |
| 14:57.54 | cjrm | bjohnson: So I should be able to establish two VoIP calls from asterisk to a VoIP service provider who will then route them to two cell phones??? |
| 14:58.32 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.164) |
| 14:59.23 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 14:59.34 | Gand_DJ | You can try this...... setup a DID for *, use 1 cellphone to call DID, setup * to allow authenticated calling outbound, and then once you call *, call out to the next cellphone (or have * auto-forward to cellphone 2) |
| 14:59.56 | Gand_DJ | Or..... without a DID, call cellphone 1, and then 3way to cellphone 2. |
| 15:00.03 | Bonbon | has anyone developed a windoze app which can be used with asterisk to transfer calls to other people? |
| 15:00.23 | Darwin[laptop] | yeah xten |
| 15:00.34 | Bonbon | no, like a recptionist console |
| 15:00.49 | Gand_DJ | AMP? |
| 15:00.50 | Nugget | why do you call it "windoze"? it makes you look like an OSShole. |
| 15:01.01 | Gand_DJ | not windows based. but it would wokr |
| 15:01.02 | Bonbon | ha ha |
| 15:01.03 | Gand_DJ | work |
| 15:01.07 | Bonbon | AMP? |
| 15:01.09 | Nugget | and yeah, asternic.org is a decent solution for that. |
| 15:01.26 | *** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
| 15:01.42 | Bonbon | ah, right, you used it? |
| 15:02.33 | bjohnson | don't be a poo Nugget |
| 15:03.50 | Bonbon | Nugget: can you transfer calls without using the mouse? |
| 15:04.12 | Nugget | why don't you look and see for yourself if it would work for your needs? |
| 15:04.24 | jmac | for simple 1x4 systems, would you guys recommend just using asterisk@home with AMP, etc? |
| 15:05.34 | Bonbon | ok, thanks. What about AMP? |
| 15:05.35 | *** join/#asterisk chap (~chap@adsl-66-137-149-194.dsl.rcsntx.swbell.net) |
| 15:06.04 | pif | kram: mantis doesn't send passwords to new accounts |
| 15:06.46 | pif | where do I report that bug if I can't login to the BTS? |
| 15:07.00 | bjohnson | ahh .. a gui that doesn't require you to use a mouse. Don't see that requested much anymore |
| 15:07.21 | mutilator | hey another person trying to defraud me on autotrader! I'm 2 for 2 now |
| 15:07.53 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 15:08.48 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:10.23 | Zilas | what would you guys offer for sending sms out of asterisk that would be a free decision? |
| 15:10.30 | pif | any digium personel around? |
| 15:12.30 | Mimmus | UAO! I successfully patched app_capiFax.c, removing any reference to channel_pvt.h |
| 15:15.45 | Gand_DJ | http://www.voip-info.org/wiki-Asterisk+cmd+Sms |
| 15:15.48 | Gand_DJ | maybe? |
| 15:17.39 | mutilator | anyone recommend any load balancing appliances? |
| 15:20.43 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.206) |
| 15:21.00 | Zilas | grand_dj: does it work in usa on bellsouth lines to t-mobile??? |
| 15:21.22 | Gand_DJ | Not sure. wasn't aware * could do sms :) |
| 15:21.36 | Gand_DJ | I just searched google and found that |
| 15:22.14 | Zilas | there is another option http://www.bayhamsystems.com/asterisk.html |
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| 15:37.55 | Pinhole | Anybody have any luck with sphinx2? |
| 15:38.47 | djMax | I must be missing something obvious, but shouldn't _#1XXX match #1001? |
| 15:38.52 | DrWho17 | ok, CIC codes |
| 15:39.15 | *** join/#asterisk josealvers (~root@200.97.28.142) |
| 15:40.04 | Blissex | djMax: depends on the order in which matches are done... |
| 15:40.10 | josealvers | hello there.. please help me.. i installed asterisk using voicepulse connect.. i configured iax.conf and extensions.conf correctly.. but when I try to call using DIAX, i receive: Apr 7 12:30:43 NOTICE[18293]: chan_iax2.c:6983 socket_read: Rejected connect attempt from 192.168.8.55, request '1@outgoing' does not exis |
| 15:40.45 | djMax | what's strange is that I don't even see the call coming into asterisk. Shouldn't there be a debug level that shows an attempted extensions match? |
| 15:40.51 | *** join/#asterisk BrianR___ (brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 15:41.00 | djMax | (but I do see things like #2 coming in) |
| 15:41.10 | DrWho17 | djMax: asterisk shows all calls that come into it |
| 15:41.31 | chap | djMax: what is your verbose level? |
| 15:41.42 | djMax | 18 |
| 15:41.42 | DrWho17 | at least sip/mgcp/h323/zap show up for me |
| 15:42.31 | GiabboO | can anybody help me with the stripped initial 0 of my incoming calls ? |
| 15:43.26 | djMax | yeah, "sip debug" shows the call, but not normal debugging/verbosity |
| 15:43.42 | Pinhole | Anybody see anything that AOL is providing with their VoIP service that can't be done with *? |
| 15:44.02 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 15:44.33 | BuckRogers | GOOOOOD Morning |
| 15:45.03 | djMax | ok, I think I sort of see what's happening. I have a Sipura that has a dialplan of #X. So it's trying to call #1 even though I pressed #1001. Question is where those three digits went. |
| 15:45.18 | josealvers | Apr 7 12:30:43 NOTICE[18293]: chan_iax2.c:6983 socket_read: Rejected connect attempt from 192.168.8.55, request '1@outgoing' does not exist |
| 15:45.42 | josealvers | why this error happens? |
| 15:45.58 | zoa | because your extension does not exist |
| 15:46.18 | josealvers | but I created them.. and edite them correctly |
| 15:47.21 | josealvers | --> /etc/asterisk/extensions.conf |
| 15:47.26 | mogorman | hey anyone know how to allow more than one call to be placed to the same sip user on a snom 220 |
| 15:47.40 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
| 15:47.51 | Moc | morning/afternoon |
| 15:48.17 | jcollie | hello |
| 15:48.40 | jcollie | so what IS up with Sipura's web pages? can't resolve their DNS |
| 15:48.56 | pino | jcollie: neither can i |
| 15:49.02 | *** join/#asterisk zipp (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 15:49.06 | GiabboO | ciao pino :) |
| 15:49.21 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 15:49.31 | pino | (ciao :) ) |
| 15:49.36 | GiabboO | di dove ? :D |
| 15:49.44 | jcollie | damnation, i have a client that just bought a couple 841s and i want to grab the latest firmware before I go install them tomorrow |
| 15:51.01 | *** join/#asterisk fugitivo (~ajf@201.255.99.196) |
| 15:52.24 | pino | jcollie: looks like they have no mirrors :( |
| 15:52.53 | jcollie | arghh! |
| 15:52.53 | *** join/#asterisk rhygin (audio07r@209.47.250.41) |
| 15:53.50 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 15:55.53 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 15:55.56 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 15:55.57 | Gand_DJ | anyone here have a minute to help test something with *? |
| 15:56.07 | Gand_DJ | You'd need an iax phone though |
| 15:56.14 | Gand_DJ | cuz of nat |
| 15:56.16 | MasterYoda | who wrote the say time agi command? |
| 15:56.42 | jcollie | anyone have a sample config for a sipura spa-841 |
| 15:56.44 | rhygin | just about to file a bug report for this but thought I bring it up in here in case anyone else has seen this... when using ${CALLERIDNUM} in extensions it strips '.' out of the CID but if you hit '#' it has the '.' |
| 15:58.20 | *** join/#asterisk adx (~adx@phlox.int.addix.net) |
| 15:58.21 | rhygin | Executing VoiceMailMain("SIP/domain.com-800c2c60", "firstnamelastname") / No username but # key pressed. Using CID 'firstname.lastname' |
| 15:58.31 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
| 15:58.40 | adx | moin @all |
| 15:58.51 | Gand_DJ | For someone to be able to remotely connet to my * box from the internet (like a softphone or something), do I need to forward any ports to the * server from the router? (sip & iax) |
| 15:59.15 | rhygin | Gand: is * nat'd or behind a router/firewall that port filters? |
| 15:59.26 | Gand_DJ | * has a 192 IP |
| 15:59.35 | Gand_DJ | behind linksys router |
| 15:59.44 | outtolunc | the first question should be using what protocol |
| 16:00.02 | Gand_DJ | Some using sip, some using iax |
| 16:00.09 | outtolunc | sip yes, iax no |
| 16:00.31 | rhygin | then you'd need to forward port 5060 UDP for SIP, 4569 for IAX2 |
| 16:00.52 | jcollie | rhygin don't forget RTP ports |
| 16:00.54 | djMax | anybody know where the rest of the digits go in a case like I mentioned? (sipura dialplan is set to #1S0, I dial #1001) |
| 16:01.40 | rhygin | jcollie: yep, i wasn't really going to get into it... nat'ing * (or any thing that uses random ports) is just messy |
| 16:02.07 | Gand_DJ | ok... so for iax, I just need to port forward 4569 UDP (not tcp) to * IP? |
| 16:02.09 | pino | djmax: i think they are just discarded |
| 16:02.14 | jcollie | yeah, but you can edit rtp.conf and limit the number of ports you need to open |
| 16:02.23 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-220-226.dsl.scarlet.be) |
| 16:02.24 | pino | what if you try #XXXXS0? |
| 16:02.56 | *** join/#asterisk GreyFoxx (greg@out.of.phaze.org) |
| 16:02.56 | djMax | then my other things (#2,#3 etc) will have to wait. I suppose I can do #1XXXS0, but that sucks. |
| 16:02.56 | outtolunc | gand i misread, if you * box is behind the nat, either put it in dmz or forward ports for everything you need |
| 16:02.59 | djMax | would be nice to have a way to say "match this, then let * do the rest" |
| 16:03.06 | rhygin | Gand: I haven't used a linksys router in ages but if you can you'd be better to make your * box the 'DMZ' host on the router |
| 16:03.21 | mgth | linksys will let you do a dmz |
| 16:03.26 | rhygin | that way anything that was allowed through and didn't have a destination map already would end up at the * box |
| 16:03.31 | Gand_DJ | I could do dmz, but for security I'd rather not (incase there's an open port on * to hack into) |
| 16:03.38 | mogorman | hey anyone know how to allow more than one call to be placed to the same sip user on a snom 220 |
| 16:04.13 | pino | djmax: DISA maybe is what you're looking for. |
| 16:04.49 | moy | Does anybody have used voiceXml succesfully with Asterisk? |
| 16:07.01 | adx | anyone knows nome news about the Digium Te110 E1/T1 card and the driver reboot Problem? |
| 16:08.11 | phpboy | when I run asterisk |
| 16:08.14 | phpboy | I get this error |
| 16:08.14 | Pinhole | moy, I haven't seen anything with asr work with *. |
| 16:08.36 | phpboy | Ouch ... error while writing audio data: : Broken pipe |
| 16:08.50 | phpboy | Warning, flexibel rate not heavily tested! |
| 16:08.58 | phpboy | how do I go about fixing such an error? |
| 16:09.11 | thetalon | moy we have a VoiceXML engine running as AGI |
| 16:09.17 | adx | i read something about the te110 in mail archives, but no really Solution. |
| 16:11.45 | Gand_DJ | I setup port forwarding on my * box for 4569. If someone likes, you can see if you can register via IAX to my * box. U = 301 / P = test456 |
| 16:14.35 | phpboy | Gand_DJ: can we use the IP ur connecting from? |
| 16:14.46 | Gand_DJ | sure |
| 16:14.52 | moy | Pinhole: Thanks :) i guess im going to continue googling around |
| 16:14.58 | Gand_DJ | you can also use rivendell.twinworld.ca |
| 16:15.05 | Gand_DJ | or irc.twinworld.ca |
| 16:15.23 | Gand_DJ | I think I have IRC dns running still |
| 16:15.24 | Gand_DJ | heh |
| 16:15.53 | pif | bkw_ ? |
| 16:17.14 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 16:20.08 | phpboy | Gand_DJ: what's a nice soft phone for windows |
| 16:20.10 | phpboy | that I can use? |
| 16:20.24 | Gand_DJ | Firefly does iax |
| 16:20.39 | Gand_DJ | I have that test acct setup for iax |
| 16:20.44 | Pinhole | phpboy: x-lite |
| 16:21.16 | Darwin[laptop] | grrr g729 is not working |
| 16:21.42 | Gand_DJ | what's so great about g729? |
| 16:21.54 | jsharp | Its a good quality low bit rate codec. |
| 16:21.55 | Pinhole | thetalon, do you have asr? |
| 16:22.19 | _asr | stop talking about me behind my back |
| 16:22.23 | _asr | my irssi keeps lighting up |
| 16:24.18 | Pinhole | how about: thetalon, do you have speech recognition working? |
| 16:24.40 | phpboy | I've installed zaptel drivers on Mandrake 10.1 |
| 16:24.45 | phpboy | via urpmi |
| 16:24.56 | phpboy | but I can't do a "modprob zaptel" |
| 16:25.04 | phpboy | says no such module |
| 16:25.07 | phpboy | how do I fix this? |
| 16:26.25 | pino | i've never used mandrake, but -- depmod -a perhaps? or are you using a custom kernel (and Mandrake only supplies drivers for the stock kernel)? |
| 16:27.30 | phpboy | I have a default mandrake install |
| 16:27.43 | phpboy | is there anyway I could go about installing the modules manually? |
| 16:27.55 | Darwin[laptop] | everyone at digium must be sleeping |
| 16:28.14 | Darwin[laptop] | yes modules.conf |
| 16:28.34 | Darwin[laptop] | <PROTECTED> |
| 16:28.44 | Darwin[laptop] | and then put in there what to loaad |
| 16:28.53 | pino | if you find the zaptel.o / zaptel.ko files, you can also try with insmod |
| 16:29.01 | phpboy | /etc/modules.conf ? |
| 16:29.12 | Darwin[laptop] | yes |
| 16:29.29 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 16:29.33 | Juxt | good afternoon |
| 16:30.00 | Gand_DJ | anyone able to register with my * box? :( |
| 16:30.08 | phpboy | I don't think the modules are installed on my box |
| 16:30.08 | phpboy | :/ |
| 16:30.09 | Gand_DJ | just need to know if port forwarding is setup correctly |
| 16:30.18 | phpboy | Gand_DJ: what's a nice soft phone for windows |
| 16:30.21 | phpboy | I'll help u test |
| 16:30.29 | Darwin[laptop] | gaimphone |
| 16:30.31 | Gand_DJ | Someone mentioned X-lite |
| 16:30.36 | Gand_DJ | but I think that uses SIP |
| 16:30.37 | Darwin[laptop] | or xten-lite |
| 16:30.38 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 16:30.41 | Gand_DJ | I setup the acct for IAX. |
| 16:30.50 | Gand_DJ | I know firefly does IAX |
| 16:30.52 | Darwin[laptop] | iaxcomm |
| 16:31.02 | phpboy | Darwin[laptop]: any suggestions? |
| 16:31.03 | Juxt | firefly is great |
| 16:31.04 | outtolunc | firefly-thirdparty does iax |
| 16:31.08 | phpboy | well, other suggestions? |
| 16:31.10 | Juxt | except i can't find the damn g.729 dll |
| 16:31.12 | outtolunc | diax |
| 16:31.18 | Juxt | and when i try to make it crashes |
| 16:32.29 | Darwin[laptop] | whats the url for the firfly third party |
| 16:32.54 | Juxt | http://www.virbiage.com/firefly/download/firefly-thirdparty.exe |
| 16:32.55 | *** join/#asterisk rpr_ (~ricardMad@212.163.10.2) |
| 16:34.28 | phpboy | :/ |
| 16:35.44 | Darwin[laptop] | thnks |
| 16:37.55 | phpboy | does anybody know how to install the zaptel modules on Linux? |
| 16:37.57 | phpboy | so I can do a |
| 16:38.01 | phpboy | modprode zaptel |
| 16:38.02 | phpboy | ? |
| 16:38.13 | Juxt | did you read the instructions? it's there |
| 16:38.27 | phpboy | I installed it via urpmi |
| 16:38.35 | phpboy | didn't seem to have worked :/ |
| 16:39.06 | phpboy | it installed the zaptel program |
| 16:39.08 | Darwin[laptop] | wow no one is answering at digium |
| 16:39.10 | phpboy | but not the modules :/ |
| 16:39.25 | phpboy | Darwin[laptop]: u calling them or what? |
| 16:39.40 | robl^ | Darwin[laptop]: well. it IS lunchtime |
| 16:39.49 | MasterYoda | Darwin[laptop]: we are here... |
| 16:40.01 | djMax | anybody know of fixes for ast_waitstream_full in agi apps? |
| 16:40.19 | JunK-Y | djMax: whats wrong? |
| 16:40.35 | djMax | the wakeup call app keeps firing that |
| 16:40.41 | djMax | not EVERY time, but many times |
| 16:40.58 | Darwin[laptop] | Master Yoda I have issues with g729 for fbsd getting no audio |
| 16:42.07 | mogorman | Master Yoda me too |
| 16:42.26 | mogorman | are you calling support? |
| 16:42.32 | Darwin[laptop] | yes |
| 16:42.37 | rpr_ | phpboy: you need to compile and install zaptel edit the config file /etc/zaptel.conf load module zaptel and the specific module of your card. Then you can configure the asterisk's zapata.conf file and use zap channels in your asterisk configuration. |
| 16:42.41 | mogorman | we arent on the phone... |
| 16:42.43 | mogorman | or im not |
| 16:42.50 | mogorman | but g729 on bsd is not supported... |
| 16:42.56 | Darwin[laptop] | yes it is |
| 16:42.58 | mogorman | thus unsupported versions of g729... |
| 16:43.02 | Darwin[laptop] | they have compiledit now |
| 16:43.07 | mogorman | ? |
| 16:43.09 | Juxt | does anyone have the g729.dll for firefly compiled? |
| 16:43.29 | phpboy | rpr_: only trouble is |
| 16:43.32 | Darwin[laptop] | they have compiled it for fbsd now |
| 16:43.36 | MasterYoda | Darwin[laptop]: unfortunately we do not support using our g729 codecs on freebsd, although they are available... |
| 16:43.36 | mogorman | Codecs for FreeBSD 5.2.1 are made available in an unsupported format: |
| 16:43.37 | phpboy | zaptel module isn't working |
| 16:43.39 | Darwin[laptop] | its on the web page |
| 16:43.40 | mogorman | unsupported |
| 16:43.47 | phpboy | yeah |
| 16:43.52 | phpboy | I moved from FreeBSD to Linux |
| 16:43.54 | mogorman | but what is going on darwin |
| 16:44.00 | phpboy | zaptel drivers cause kernel panics |
| 16:44.01 | phpboy | :/ |
| 16:44.02 | Darwin[laptop] | well kram and I have wokred on it |
| 16:44.07 | MasterYoda | mogorman: is a luser |
| 16:44.12 | phpboy | rpr_: I modprode zaptel |
| 16:44.15 | phpboy | doesn't working |
| 16:44.17 | mogorman | MasterYoda WHHHE |
| 16:44.18 | phpboy | doesn't work |
| 16:44.19 | phpboy | so |
| 16:44.24 | Darwin[laptop] | it was working till we patched a load problem |
| 16:44.26 | phpboy | I'm trying to install it manually now |
| 16:44.41 | mogorman | ? |
| 16:44.44 | phpboy | and see what happens |
| 16:44.45 | phpboy | :/ |
| 16:45.17 | Darwin[laptop] | it was only loading about once every 4th to 5th boot |
| 16:45.23 | Darwin[laptop] | he fixed it |
| 16:45.31 | Darwin[laptop] | now I get no audio |
| 16:45.43 | Darwin[laptop] | lol |
| 16:45.58 | Gand_DJ | http://www.virbiage.com/firefly/download/g729.zip |
| 16:46.16 | Gand_DJ | ? |
| 16:46.21 | bkw_ | compiled |
| 16:47.19 | rpr_ | phpboy: for more information visit http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 16:47.24 | GiabboO | can I ask about RxFax problem ? |
| 16:47.43 | phpboy | I have learn't a HELL of a lot in the last 4 days |
| 16:47.45 | phpboy | with asterisk |
| 16:47.48 | phpboy | eish |
| 16:47.53 | phpboy | and I haven't even got it working yet |
| 16:47.59 | GiabboO | lol phpboy |
| 16:48.00 | phpboy | only started playing on Linux today |
| 16:48.01 | Darwin[laptop] | asterisk is a learnig curv |
| 16:48.02 | Gand_DJ | lol I got it working internally |
| 16:48.08 | rpr_ | phpboy: Do you use linux 2.4 or linus 2.6? |
| 16:48.21 | phpboy | Mandrake 10.1 |
| 16:48.31 | phpboy | kernel 2.6 |
| 16:48.32 | MasterYoda | dont use mandrake |
| 16:48.33 | GiabboO | can anybody help me with RxFax WARNING i get after fax is received ? |
| 16:48.41 | mogorman | its mandriva now |
| 16:48.41 | phpboy | if I do a 'uname -a' |
| 16:48.48 | phpboy | MasterYoda: I'm VERY new to linux |
| 16:48.51 | MasterYoda | mogorman: really? |
| 16:48.57 | MasterYoda | phpboy: don't use mandrake... |
| 16:48.58 | phpboy | so I'm gonna start as basic as possible |
| 16:49.02 | mogorman | MasterYoda go read slashdot... |
| 16:49.07 | GiabboO | thats good u start phpboy :) |
| 16:49.12 | mogorman | oh wait you dont read slashdot MasterYoda, luser |
| 16:49.14 | phpboy | MasterYoda: any reason, not to? |
| 16:49.25 | MasterYoda | phpboy: in my experience it does not work well |
| 16:49.29 | phpboy | I'm a FreeBSD guy myself |
| 16:49.32 | MasterYoda | phpboy: especially not with asterisk |
| 16:49.37 | phpboy | MasterYoda: all I need is to get asterisk working |
| 16:49.41 | MasterYoda | phpboy: oh, well then, go slackware, gentoo, or debian |
| 16:49.46 | phpboy | I hear it works famously on any Linux platform |
| 16:49.52 | MasterYoda | phpboy: you have unix experience... stay away from mandrake |
| 16:50.03 | Gand_DJ | I just installed the *@home package.. did everything for me :) |
| 16:50.24 | phpboy | great |
| 16:50.32 | MasterYoda | phpboy: I perfer debian, but you might like slack as I hear it is very unix like |
| 16:50.33 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 16:50.34 | phpboy | C compiler doesn't work properly |
| 16:50.44 | phpboy | eish |
| 16:50.46 | phpboy | now wait |
| 16:51.54 | phpboy | :< :< :< |
| 16:51.57 | phpboy | well |
| 16:52.05 | *** join/#asterisk Nix (~Nix@81.213.112.205) |
| 16:52.06 | phpboy | doesn't seem like I can build zaptel manually either |
| 16:52.08 | tzanger | whoa whoa whoa... just slow down kids |
| 16:52.12 | tzanger | don't make me separate you |
| 16:52.13 | GiabboO | NOTICE[5809]: channel.c:1764 ast_set_read_format: Unable to find a path from slin to unknown |
| 16:52.14 | phpboy | what's the next step gents? |
| 16:52.16 | Juxt | is there a way to find out which codec has been negotiated with a sip provider? |
| 16:52.19 | GiabboO | how do I correct this ? |
| 16:52.20 | Juxt | sip debug doesn't seem to say it |
| 16:52.24 | tzanger | phpboy: fix your C compiler |
| 16:52.35 | pino | phpboy: you should install the kernel sources, probably! |
| 16:52.48 | phpboy | tzafrir: how do I go about doing that in Linux? |
| 16:52.59 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 16:53.34 | rpr_ | phpboy: are you sure that the modules, teh kernel source and the kernel image taht you use are the same version? |
| 16:53.41 | Pinhole | Juxt: sip show channels |
| 16:53.44 | phpboy | erll |
| 16:53.45 | phpboy | well |
| 16:53.50 | phpboy | it's it brand new install |
| 16:53.52 | ariel_ | phpboy, I guess this is your first use of linux? Are you trying to setup an asterisk server for testing, production or learning? |
| 16:53.53 | phpboy | so I'd imagine so |
| 16:53.56 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 16:54.00 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 16:54.03 | phpboy | ariel_: learning |
| 16:54.11 | phpboy | so that I can get to know asterisk and Linux |
| 16:54.13 | phpboy | in one |
| 16:54.31 | ariel_ | phpboy, I would say you should look at Asterisk@home it a complete ISO with os and asterisk setup. |
| 16:54.34 | tzanger | phpboy: well it depends entirely on your system. |
| 16:54.45 | Juxt | thank you pinhole |
| 16:55.00 | phpboy | tzanger: Mandrake 10.1 default install |
| 16:55.44 | tzanger | phpboy: mandrake 10.1 default install should be able to build, I tink |
| 16:55.46 | *** join/#asterisk heison (~heison@ns1.somanetworks.com) |
| 16:55.47 | Nugget | linux is poo. :) |
| 16:55.51 | heison | ~seen JerJer |
| 16:56.01 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 39d 37m 6s ago, saying: 'mrgoby: sure'. |
| 16:56.01 | phpboy | :/ |
| 16:56.01 | ariel_ | phpboy, I think that mandrake has yum do yum install kernel-source |
| 16:56.05 | Wonka | ~seen JarJar |
| 16:56.06 | jbot | jarjar <~kevinsmit@ool-182f616a.dyn.optonline.net> was last seen on IRC in channel #kde, 157d 21h 29m 49s ago, saying: 'configure: error: missing argument to --prefix'. |
| 16:56.18 | phpboy | well |
| 16:56.23 | *** join/#asterisk _zigo__ (~ogiz@m6.net81-64-48.noos.fr) |
| 16:56.24 | phpboy | I'm upgrading the system |
| 16:56.30 | phpboy | hopefully that'll help |
| 16:56.31 | phpboy | :/ |
| 16:57.33 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 16:59.26 | Juxt | what dictates which codec gets selected when a sip connection is established? |
| 16:59.40 | *** join/#asterisk ckruetze (~ckruetze@i3ED65FFE.versanet.de) |
| 17:00.01 | ariel_ | Juxt, which version are you using? head or stable? |
| 17:00.14 | Juxt | head |
| 17:00.38 | ariel_ | codec is with disallow=all then allow=ulaw, allow=gsm. In head I think it's in the order placed in the sip.conf |
| 17:00.41 | Juxt | i am trying to force g.729 firefly -> asterisk > sip provider |
| 17:00.42 | phpboy | I'm gonna miss gym tonight because of this :< |
| 17:01.02 | ariel_ | phpboy, get your self the iso from asterisk at home and start there. http://asteriskathome.sourceforge.net/ |
| 17:01.23 | phpboy | what all is this? |
| 17:01.31 | _THEEND_ | how can i change language to a cisco phone? |
| 17:02.05 | Pinhole | Juxt, it also seems (in my experience) to depend on the preference of the sip phone as well. |
| 17:02.15 | ariel_ | Asterisk@home has the OS - CentOS great linux distro, has ASterisk, has AMP and can configure your zap ports even if you only use ztdummy. |
| 17:03.03 | Gand_DJ | ariel_, since I don't use a zap card / port.... do I have to manually install ztdummy? |
| 17:03.13 | phpboy | almost a turn key solution |
| 17:03.16 | Gand_DJ | I keep getting that ixxxx error |
| 17:03.17 | GiabboO | Apr 7 18:55:31 WARNING[5897]: app_rxfax.c:305 rxfax_exec: Unable to restore read format on 'Modem[i4l]/ttyI0' |
| 17:03.17 | phpboy | from the sounds of it |
| 17:03.25 | GiabboO | anybody have solution for this ? |
| 17:04.04 | ariel_ | phpboy, it's almost a turnkey solution. But you can learn and change all the settings yourself. |
| 17:04.16 | *** join/#asterisk loick (~loick@APuteaux-151-1-39-14.w82-124.abo.wanadoo.fr) |
| 17:04.27 | phpboy | of course |
| 17:04.30 | phpboy | which is what I want |
| 17:04.30 | ariel_ | Gand_DJ, my setup came with ztdummy already configured in aah. |
| 17:04.38 | phpboy | thanks a mill ariel_ |
| 17:04.42 | Gand_DJ | hrm |
| 17:04.45 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 17:04.57 | StealthMethod | any one heard of a predictive dialer....truedial |
| 17:04.58 | Gand_DJ | how can I tell if ztdummy is installed? |
| 17:05.00 | StealthMethod | old software |
| 17:05.20 | ariel_ | phpboy, there is also AMP which is a gui for configuration of asterisk in AAH and you can also get help with amp on the #amportal section here. |
| 17:05.23 | StealthMethod | truedial predictive dialer |
| 17:05.24 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 17:05.37 | StealthMethod | looking for help, if anyone familier , would greatly appreciate |
| 17:06.12 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 17:06.12 | ariel_ | Gand_DJ, if you do zttool it should tell you there |
| 17:06.37 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 17:06.59 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
| 17:07.12 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 17:07.15 | Gand_DJ | I show nothing in there |
| 17:07.21 | Gand_DJ | Alarms & Span |
| 17:07.22 | Gand_DJ | that is it |
| 17:08.23 | ariel_ | Gand_DJ, read this then and install it. http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy |
| 17:08.39 | Gand_DJ | maybe I deleted ztdummy from AMP... as I saw 1 zap channel in there and deleted it |
| 17:08.49 | shido6 | any r2 users? |
| 17:09.42 | phpboy | I get this error when running asterisk |
| 17:10.06 | phpboy | Ouch ... error while writing audio data: : Broken pipe |
| 17:10.06 | Gand_DJ | I see why ztdummy wasn't installed.... Virtual PC & Virtual Server doesn't recognize usb devices |
| 17:10.12 | phpboy | how can I fix this? |
| 17:10.21 | Gand_DJ | usb_uhci is not in lsmod |
| 17:10.24 | GiabboO | need help with RxFax module, please contact me in private thx |
| 17:10.44 | DrWho17 | anybody egressing out a Lucent TNT/APX? the TNT doesn't seem to send out the callerid I've specified in Asterisk |
| 17:10.54 | Gand_DJ | shido6: R2? |
| 17:10.55 | DrWho17 | while Zap channels hooked to the same trunks do |
| 17:11.51 | ariel_ | DrWho17, are you using ani settings in your dial string? |
| 17:11.56 | phpboy | !!!!!!!!!!! |
| 17:12.04 | phpboy | when trying to compile zaptel manually |
| 17:12.10 | phpboy | I get the following error |
| 17:12.34 | ariel_ | Dial(Zap/g1/${EXTEN}//a) |
| 17:12.40 | phpboy | You do not appear to have the kernel sources fro your current kernel installed. |
| 17:12.45 | DrWho17 | ariel_: no, and the TNT displays the callerid I've specified in it's monitoring of the call, it just get sent out to the pstn |
| 17:12.59 | phpboy | how do I install it? |
| 17:13.05 | GiabboO | phpboy, u need the kernel source under /usr/src and a sym link /usr/src/linux |
| 17:13.19 | *** part/#asterisk rhygin (audio07r@209.47.250.41) |
| 17:13.23 | DrWho17 | has to be something on the TNT ignoring dropping those things, but it is picking them up via sip |
| 17:13.27 | phpboy | I don't have that dir |
| 17:13.30 | GiabboO | phpboy, uname -a to get ur running kernel version |
| 17:13.34 | phpboy | looks like I'll have to install the src |
| 17:13.35 | phpboy | but how? |
| 17:13.50 | GiabboO | phpboy, then download the tar package of the kernel u are running now |
| 17:14.35 | ariel_ | phpboy, does your distro have yum or up2date ? |
| 17:14.43 | GiabboO | phpboy, untar it under /usr/src and make a symbolic link to like /usr/src/linux -> /usr/src/linux-2.x.x |
| 17:14.47 | DrWho17 | ariel_: exten => _XXXXXXX,4,Dial,SIP/BYEXTENSION@tnt1|60, |
| 17:14.51 | DrWho17 | pretty basic |
| 17:15.05 | ariel_ | DrWho17, add the a at the end |
| 17:15.11 | DrWho17 | hrm, ok |
| 17:15.29 | Gand_DJ | Hrm... When in the Operator Panel... when I click on the Lock, it asks for a security password.. what's the default pass? |
| 17:16.08 | GiabboO | i'm havin problem using ScanDsp, i get this error { app_rxfax.c:305 rxfax_exec: Unable to restore read format on 'Modem[i4l]/ttyI0' } after i receive a fax, takin a look on the asterisk-fax spool directory i see the TIFF image and its pretty good, but rxfax send me failure mail, anybody know why ? |
| 17:16.13 | DrWho17 | ariel_: where at, I'll clean up my extension lines too heh |
| 17:16.13 | ariel_ | Gand_DJ, good question I have never click there... |
| 17:16.14 | GiabboO | (spandsp) |
| 17:16.41 | ariel_ | DrWho17, exten => _XXXXXXX,4,Dial,SIP/BYEXTENSION@tnt1|60|a, |
| 17:16.46 | DrWho17 | ok |
| 17:18.44 | *** join/#asterisk MindChild (RntedMule@57.muca.pitt.washdctt.dsl.att.net) |
| 17:18.57 | ariel_ | Gand_DJ, passw0rd |
| 17:19.17 | DrWho17 | ariel_: heh no dice |
| 17:19.22 | Gand_DJ | heh |
| 17:19.23 | Gand_DJ | that worked |
| 17:20.21 | ariel_ | DrWho17, then you might have a problem with the configuration on the tnt |
| 17:20.21 | Gand_DJ | hrm.... when I refresh screen.... the lock becomes unlockd |
| 17:20.49 | ariel_ | Gand_DJ, you just reloaded it |
| 17:21.04 | ariel_ | it suppose to refresh it's self |
| 17:21.08 | MindChild | Ok, I am a total nothing when it comes to this stuff, so please try not to burn me to a crisp. I have a box full of old office phones from our old office. I think the system was Avaya Merlin (Im not sure that even matters). Could I use these phones somehow, rather then getting ATA adapters for analog phones? |
| 17:21.15 | Gand_DJ | heh..ok.. I haven't figured out the point of this panel yet |
| 17:21.31 | DrWho17 | ariel_: well, yes that was my suspicion as well (since out the same trunks hooked up to zapchannels it works perfectly) |
| 17:21.44 | *** join/#asterisk djMax (~djMax@dsl093-190-107.nyc2.dsl.speakeasy.net) |
| 17:22.13 | ariel_ | MindChild, are the phones analog or digital? |
| 17:22.13 | DrWho17 | that and the TNT shows the callerid->calledto correctly |
| 17:23.12 | Gand_DJ | anyone here have an iax softphone? Just need someone to see if my router is forwarding ports properly. |
| 17:23.26 | MindChild | ariel_: I believe digital. They use cable with an RJ45 connector, though Im sure they arent cat5 |
| 17:23.29 | Gand_DJ | U = 301 / P = test456 |
| 17:23.40 | ariel_ | MindChild, ebay them |
| 17:23.54 | MindChild | So I cant use them then...? |
| 17:24.07 | *** join/#asterisk dogz- (~bob@66.148.168.234.nw.nuvox.net) |
| 17:24.08 | ariel_ | MindChild, you catch on fast |
| 17:24.21 | Juxt | what does this mean: Apr 7 13:23:54 NOTICE[15835]: channel.c:1833 set_format: Unable to find a path from g729 to slin |
| 17:24.21 | Juxt | Apr 7 13:23:54 WARNING[15835]: channel.c:2263 ast_channel_make_compatible: Unable to set read format on channel SIP/rnktel-f49b to 256 |
| 17:24.30 | MindChild | I wasnt sure if you were implying it wasnt possible, or hinting its not worth the trouble |
| 17:25.18 | ariel_ | MindChild, both it's not worth the money and you can't directly use them with asterisk unless you use the Merlin pbx its self |
| 17:25.49 | MindChild | ok, thats what I figured. No wonder we got new phones when we got the new phone system |
| 17:26.33 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:26.42 | *** join/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 17:26.46 | *** part/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 17:27.03 | MindChild | Now, I assume there are phones that have an ethernet port, that can directly work with an astrisk PBX on the lan. Is there some definitive list somewhere? |
| 17:27.27 | shido6 | IP Phones |
| 17:27.28 | ariel_ | MindChild, use the wiki. |
| 17:27.33 | ariel_ | ~docs |
| 17:27.34 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:28.46 | MindChild | gracias |
| 17:29.33 | ariel_ | MindChild, Any time. |
| 17:30.46 | Gand_DJ | hrm.... anyone? I'm going to re-install *@home in a couple min |
| 17:30.51 | *** join/#asterisk owie (~owie_9i8@CPE0080c6e2e4c9-CM014100216061.cpe.net.cable.rogers.com) |
| 17:31.33 | festr_ | hello |
| 17:31.53 | PBXtech | anyone know this error: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) |
| 17:32.42 | Juxt | does anyone have Budgetone 102 ? |
| 17:33.49 | shido6 | yes |
| 17:33.52 | shido6 | my first ip phone |
| 17:33.54 | shido6 | still here |
| 17:33.57 | shido6 | whats up? |
| 17:34.07 | Juxt | well i need to get some phones for testing |
| 17:34.10 | Juxt | so i thought i'd get that one |
| 17:34.12 | *** join/#asterisk Elshar (~Elshar@ip205-68.oregonfast.net) |
| 17:34.28 | Juxt | while i am waiting on gxp-2000 |
| 17:35.06 | Gand_DJ | anyone have access to those X100P clones? Getting one off ebay is like highway robbery with shipping rates |
| 17:35.07 | festr_ | could someone help me with configuration of TE410P? ztcfg: 124 channels configured. zapata.conf: switchtype = euroisdn, signalling = pri_cpe, pridialplan=local, group = 1, channel => 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124. but when asterisk start -vvvgcd: Ignoring switchtype, Ignoring pridialplan, Signalling must be specified before any channels are etc.. i've successfully configured some single E1 cards. asterisk 1.0 |
| 17:36.00 | Juxt | how does it compare to say Sipura SPA-841 ? |
| 17:37.37 | dogz- | Hi, i moved to linux per bkw_ suggestion... I was having a terrible time getting asterisk to work with my two X100P cards till i ran "ztcfg" and now asterisk starts up like a champ... Being so new to linux im quite unsure how i can get ztcfg to run each time at startup... Can someone point me in the right direction? |
| 17:38.04 | Lee__ | dogz-: what distro? |
| 17:38.15 | dogz- | Gentoo |
| 17:38.30 | Lee__ | you're new to Linux and you picked Gentoo!!! |
| 17:38.35 | Lee__ | my god |
| 17:38.44 | Pinhole | Gentoo is the best/worst place to learn linux. |
| 17:38.54 | dogz- | Eh im a freebsd guy so its not to bad :) |
| 17:38.57 | Lee__ | that should be /etc/init.d/rc.local or /etc/rc.local |
| 17:39.18 | Lee__ | oh, never mind. Usually when someone says "new to Linux" they mean Windows |
| 17:39.48 | dogz- | any suggestions similar to the freebsd handbook? |
| 17:40.01 | festr_ | anyone here using TE410P? |
| 17:40.02 | Lee__ | www.gentoo.org has great docs |
| 17:40.46 | blankman | hey does anyone on know how the iax2 provision works? Where do I get more information on it ... like the format for templates etc? |
| 17:40.54 | Lee__ | so does www.debian.org for when you get tired of recompiling glibc because of an update to rm. |
| 17:40.58 | blankman | festr_, i am. |
| 17:41.50 | *** join/#asterisk cjohnson19401 (~no@pcp05736194pcs.norstn01.pa.comcast.net) |
| 17:42.15 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 17:42.33 | Juxt | Sipura SPA-841 vs BudgeTone 101 ? any opinions? |
| 17:43.21 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 17:43.59 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 17:44.35 | shido6 | how many iaxy's blankman |
| 17:44.37 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-26-145.d4.club-internet.fr) |
| 17:45.41 | Tall-guy | hey lads, is there a quick way to find out what version of "zaptel" I'm running? |
| 17:46.27 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:50.19 | Tall-guy | what is the weight of an unladen swallow? |
| 17:52.20 | Lee__ | Tall-guy: modinfo zaptel? |
| 17:52.49 | MindChild | What would be recommended as the cheapest, reliable Hard Phone? |
| 17:53.01 | *** join/#asterisk miguellinux (~miguellin@64.76.202.2) |
| 17:53.07 | miguellinux | Hi!!! |
| 17:53.27 | miguellinux | Please I need some help with broadvoice on * |
| 17:53.35 | denon | why not ask broadvoice? |
| 17:53.48 | PTG123 | broken 7960 cisco phones are going more then good ones usually go for on ebay right now, makes no sense. |
| 17:54.01 | PTG123 | i'm gonna start selling broken stuff |
| 17:54.13 | phpboy | LOL |
| 17:54.15 | phpboy | ;P |
| 17:54.19 | denon | PTG123: cant you get a cheap smartnet and send it in? |
| 17:54.29 | PTG123 | a cheap what? |
| 17:54.35 | denon | cisco service agreement |
| 17:54.48 | PTG123 | oh i doubt these things woul dbe covered |
| 17:54.50 | PTG123 | but no idea |
| 17:54.53 | PTG123 | like broken displays |
| 17:54.54 | PTG123 | broken cases |
| 17:54.55 | PTG123 | etc |
| 17:55.09 | denon | no clue, Ive never had a busted 7960 |
| 17:55.13 | Nugget | one of my 7960's locks up after a while and I had to write a script to reboot it every night. |
| 17:55.16 | denon | our employees dont kick their phones across the room |
| 17:55.19 | miguellinux | Broadvoice on asterisk, where I can get a hint? |
| 17:55.24 | Nugget | it's perfectly solid as long as I reboot it daily |
| 17:55.34 | PTG123 | nugget: weird any idea why? |
| 17:55.36 | Nugget | no clue |
| 17:55.52 | ikey | can any one help me to install two digium cards in single machine with 8 isdn pris |
| 17:58.26 | terrapen | nugget, did you try a new firmware? |
| 17:58.27 | *** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
| 17:58.34 | PBXtech | anyone know this error: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) |
| 17:58.48 | Nugget | yes, it's not firmware version specific. the phone has always done it |
| 17:58.58 | terrapen | <PROTECTED> |
| 17:59.01 | drumkilla | PBXtech: is one side running CVS head and the other stable? |
| 17:59.12 | PBXtech | umm yea actually |
| 17:59.18 | drumkilla | that'll do it |
| 17:59.28 | drumkilla | on the cvs head side, you need to have "trunktimestamps=no" |
| 17:59.32 | PBXtech | oh, doesnt affect the call though |
| 17:59.33 | drumkilla | I think that's the option ... |
| 17:59.35 | PBXtech | ok |
| 17:59.43 | *** join/#asterisk salimfadhley (~sal@host-83-146-34-206.bulldogdsl.com) |
| 17:59.54 | *** join/#asterisk jmac (~dj@pc-24-181-187-85.sbi.ct.charter.com) |
| 17:59.58 | drumkilla | doesn't affect the call? The call shouldn't work at all if you're seeing that message, heh |
| 18:00.03 | *** join/#asterisk iq (~iq@pc-628-018.omhq.uprr.com) |
| 18:00.08 | PBXtech | ok |
| 18:00.33 | PBXtech | i wasnt getting audio on CVS to CVS so i was trying from Stable to CVS |
| 18:00.48 | PBXtech | its odd sometimes it works sometimes it doesnt :/ |
| 18:00.58 | drumkilla | heh |
| 18:01.11 | drumkilla | well as soon as it uses trunking, it's not going to work if its using trunktimestamps |
| 18:01.25 | drumkilla | that was a backwards incompatible change |
| 18:01.54 | Darwin[laptop] | everything is workign execpt g729 on fbsd |
| 18:01.58 | Darwin[laptop] | this rocks |
| 18:02.07 | Nugget | yay |
| 18:02.08 | Juxt | well g729 is quite kick ass |
| 18:02.11 | Juxt | you will miss it |
| 18:02.20 | drumkilla | i thought 729 works on bsd now ... |
| 18:02.37 | jmac | i may have unwittingly used a motherboard not supporting PCI 2.1, if i had done so, would my TDM cards not work at all (not be recognized by the system), or behave strangely? |
| 18:02.38 | Darwin[laptop] | I am trying to reach kram about taking over the support for g729 on fbsd |
| 18:02.42 | PTG123 | um |
| 18:02.45 | PTG123 | g729 works great on fbsd |
| 18:02.54 | Darwin[laptop] | not on 5.4 |
| 18:03.01 | Darwin[laptop] | get no audio |
| 18:03.07 | PTG123 | works on 5.3 fine |
| 18:03.12 | PTG123 | did it authenticate ok darwin? |
| 18:03.28 | Darwin[laptop] | yes kram and I fixed that a few weeks ago |
| 18:03.40 | Darwin[laptop] | but on 5.4 we get no audio |
| 18:03.43 | Gand_DJ | Do you have to install something into the * server to properly support G729 in a softphone? |
| 18:03.48 | PTG123 | oh weird |
| 18:03.53 | PTG123 | you could always use intel libs |
| 18:04.21 | Gand_DJ | or does G729 work with * in it's default setup |
| 18:04.23 | MindChild | 5.4 ist even fina; |
| 18:04.43 | malverian | Has anyone tried using Comdial Impact phones with Asterisk? |
| 18:04.48 | drumkilla | Gand_DJ: passthrough works by default, for transcoding, you need a license from Digium |
| 18:05.15 | Darwin[laptop] | well everything works on 5.4 fine execpt for g729 . asterisk compiles and loads fine |
| 18:05.23 | Gand_DJ | transcoding? You mean to have * convert 1 codec into G729, or vise versa |
| 18:05.29 | drumkilla | Gand_DJ: yes |
| 18:06.09 | knight_ | anyone get g729 working with BroadVoice? |
| 18:06.20 | Darwin[laptop] | it works fine with bv |
| 18:06.36 | Darwin[laptop] | I had it working fine on 5.3 |
| 18:07.30 | Darwin[laptop] | but moved to 5.4 to work on keeping * working and stable |
| 18:07.53 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:08.17 | Darwin[laptop] | its my job in life |
| 18:08.32 | Darwin[laptop] | lol |
| 18:09.07 | terrapen | i hope that, someday, there are drivers for the T1 cards for FreeBSD |
| 18:09.54 | terrapen | ugh. i think i have an ear infection |
| 18:10.23 | Darwin[laptop] | there is |
| 18:10.27 | Darwin[laptop] | in the svn |
| 18:10.52 | terrapen | for which card? |
| 18:11.37 | festr_ | blankman: still here? |
| 18:13.15 | Darwin[laptop] | http://www.voip-info.org/tiki-index.php?page=FreeBSD%20zaptel |
| 18:13.44 | *** part/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 18:14.03 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-26-145.d4.club-internet.fr) |
| 18:14.10 | terrapen | beta |
| 18:14.13 | terrapen | hmmm |
| 18:14.19 | *** join/#asterisk asingh ([U2FsdGVkX@202.140.137.116) |
| 18:14.43 | pfn | wtf is wrong with people |
| 18:14.49 | pfn | people are paying >$100 for a broken cisco 7960??? |
| 18:14.52 | pfn | jesus christ people are morons |
| 18:15.04 | terrapen | welcome to ebay |
| 18:15.13 | terrapen | you can sell anything on there |
| 18:15.19 | terrapen | "Used Diapers - NR" |
| 18:15.26 | pfn | I just need a broken 7960 so I can replace my lcd |
| 18:15.39 | pfn | and try to see if I can make a hook-switch using my cs50 |
| 18:16.37 | PTG123 | they are paying > $150 |
| 18:16.38 | PTG123 | its stupid |
| 18:16.54 | pfn | maybe I should just throw my 7960 away and get an ip600 |
| 18:17.09 | terrapen | does gonzo IRC? |
| 18:17.14 | terrapen | pfn: nooooo |
| 18:17.42 | pfn | the lcd screen is busted, pretty useless without a replacement screen |
| 18:18.08 | terrapen | what did you do to it to break it? |
| 18:18.46 | pfn | got my wife mad at me... |
| 18:19.02 | terrapen | hahaha |
| 18:19.18 | terrapen | that happened to my father's laptop once |
| 18:19.23 | terrapen | she threw it in the street |
| 18:19.29 | pfn | heh |
| 18:19.39 | terrapen | because he insisted on taking it with them on their vacation |
| 18:19.56 | festr_ | my problem solved, i didnt make install in libpri |
| 18:19.56 | terrapen | my girlfriend knows better than doing something like that :P |
| 18:20.23 | tzanger | haha |
| 18:20.45 | PTG123 | i would be getting a new wife :) |
| 18:26.40 | dogz- | If everytime i try to dial a sip extension from my own, and recieve the following error "chan_sip.c:1398 create_addr: No such host: 1000", "unable to create channel 'sip'"... That would suggest that my extensions.conf file has problems correct |
| 18:34.02 | blankman | tzanger, have you looked that the zaptel em code closely? |
| 18:34.06 | *** join/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 18:34.45 | blankman | ManxPower said you helped him with his issues on it before ... and I wanted to know if you would work with me on it as well? |
| 18:35.21 | CoderCR | Hello all |
| 18:35.41 | blankman | Hey CoderCR |
| 18:36.14 | Hmmhesays | in living color |
| 18:36.22 | *** part/#asterisk GreyFoxx (greg@out.of.phaze.org) |
| 18:36.23 | Hmmhesays | oh wait, that was handyman |
| 18:36.58 | zoa | dogz that would meab your sip.conf has issues |
| 18:38.42 | dogz- | oh, thanks |
| 18:39.35 | dogz- | =] |
| 18:39.47 | tzanger | blankman: I have looked at it |
| 18:39.57 | tzanger | but again I do not understand where asterisk is winking |
| 18:40.01 | tzanger | I forget who told me it does |
| 18:41.40 | GiabboO | bye all |
| 18:42.01 | blankman | :-) So, the question that I have for you is when you where looking at it, did you remember seeing away to "hold" the dial for x milliseconds of time after the hookstate transition? |
| 18:42.34 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 18:43.32 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 18:43.43 | rvhi | hi |
| 18:44.13 | rvhi | in voicemail, there are folders. my customer thinks it is too confusing. can we only have two folders, |
| 18:44.17 | rvhi | new and saved? |
| 18:44.59 | blankman | rvhi, just change the gsm message to only anounce save and new. |
| 18:46.03 | *** part/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 18:49.41 | *** join/#asterisk SagoDan (silver@addictiveshells.net) |
| 18:50.46 | *** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net) |
| 18:50.51 | Shido | . |
| 18:51.58 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 18:52.36 | Tall-guy | any ADSI programming gods here? |
| 18:52.43 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 18:52.54 | Sedorox | Survey: Where are you located (State) and what is your current Gas price (Normal, 87) |
| 18:53.24 | *** part/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 18:53.31 | Tall-guy | sedorox: man, you don't wanna know (Canada) |
| 18:53.34 | Gand_DJ | Canada here..lol.. Manitoba... Gas is 89.0 cents / liter |
| 18:53.40 | paulc | Vancouver BC, it just hit C$1.00/liter |
| 18:53.42 | Tall-guy | Gand_dJ: 94.9 here buddy |
| 18:53.50 | Gand_DJ | 92.4 before the 3.5c off |
| 18:54.00 | Sedorox | hehe |
| 18:54.09 | Sedorox | mine here is upto $2.23/gal |
| 18:54.14 | wildgoose | How can I have a variable set depending on my starting context? The point is to have a standard dialplan, but to dial out via a different sip provider depending on the handset used (business/personal) |
| 18:54.19 | Gand_DJ | ouch |
| 18:54.41 | Tall-guy | sedorox: works out to $3.80 CDN/US Gallon....or about $3.15 US/US Gallon |
| 18:54.57 | Sedorox | yea |
| 18:55.01 | Sedorox | I knew canada was higher then us |
| 18:55.05 | StealthMethod | no one heard of truedial predictive dialer |
| 18:55.19 | Slainte | our gas is 2.05 US/LITRE so that is $8.20 a gallon |
| 18:55.19 | Tall-guy | sedorox: lotsa taxes on fuel here.... |
| 18:55.25 | Tall-guy | slainte: where u at? |
| 18:55.27 | Sedorox | :/ |
| 18:55.27 | firestrm | anyone want a good laugh? i was exploring an odd space that was walled off in a closet, i broke through the gyproc, and discovered a that someone had boarded over a bathroom.. :-D |
| 18:55.28 | Slainte | Bermuda |
| 18:55.40 | Gand_DJ | lol |
| 18:55.40 | Tall-guy | slainte: can I talk to you offline? |
| 18:55.51 | firestrm | only in victoria.. |
| 18:55.55 | Slainte | sure, but I reserve the right to ignore you :) |
| 18:56.00 | Tall-guy | ouch.... |
| 18:56.13 | Slainte | sure fire off a message |
| 18:56.18 | Weezey | Offline? Carrier pigeon? |
| 18:56.26 | Tall-guy | weezey: tin cans and string |
| 18:56.34 | Tall-guy | (but my soup cans are sip compatible) |
| 18:56.34 | firestrm | army ant ip protocol |
| 18:56.40 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 18:57.11 | bkw_ | asdf |
| 18:57.29 | mogorman | fdsa |
| 18:58.15 | Tall-guy | firestrm: you sure that room wasn't a water closet? :) |
| 18:58.42 | Sedorox | hmm |
| 18:59.13 | firestrm | Tall-guy, you might be onto something.. the guesses round here before i broke through were, dead bodies, treasure of al capone.. |
| 18:59.38 | Sedorox | lol |
| 19:00.23 | firestrm | i have never seen so much screwed up construction until i moved to Victoria B.C.. the building inspector must be on crack.. |
| 19:01.00 | Tall-guy | firestrm: my sister is building on pender island....plenty of odd things out there. |
| 19:01.08 | Sedorox | lol |
| 19:01.49 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 19:02.01 | firestrm | Tall-guy, thre really frightning thing is that for some reason in every house older than 10 years, all the electrical is switched neutral.. YIKES!! |
| 19:02.59 | Wazb | hi all |
| 19:03.16 | all | hello |
| 19:03.23 | firestrm | Wasb, wish i was.. |
| 19:03.38 | Wazb | where i can write environment varialbes so they remain permanent when i restart my Linux |
| 19:03.59 | *** join/#asterisk WebGuest (~WebGuest@pc-628-018.omhq.uprr.com) |
| 19:04.04 | firestrm | Wazb, scribe em onto the platter of you hard drive.. |
| 19:04.21 | firestrm | go ahead just open it up,, use a sharp awl.. |
| 19:04.26 | firestrm | :P |
| 19:04.27 | PTG123 | are there archives for mailing lists someplace |
| 19:04.58 | firestrm | Wazb, or you can try .bashrc |
| 19:05.29 | *** join/#asterisk zippp (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 19:05.35 | WebGuest | hi |
| 19:05.42 | firestrm | wish i was |
| 19:09.23 | Weezey | Joogle. |
| 19:10.05 | *** join/#asterisk Geraldoramos (~FullT@200.97.7.171) |
| 19:10.21 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 19:10.26 | Weezey | hmm, need to go buy a nice headset. |
| 19:10.29 | P-Chan | Hello. ^^ |
| 19:10.42 | Weezey | guh! I hate it when people cc you on an email and you have nothing to do with it. |
| 19:10.56 | lesouvage | tzafrir: I installed ampportal from http://tzafrir.org.il/rapid/APT.html using apt-get. When I want to enter the starting page of AMP the .php script isn't run, the browser tries to download it. What should I do to make AMP work with Rapid asterisk? |
| 19:10.56 | CoaxD | Weezey: Happens to me all the time |
| 19:10.57 | eKo1 | firestrm: eh, that only works if Wazb uses bash. |
| 19:11.10 | CoaxD | Weezey: And, joke lists. I hate joke lists. Why does everyone have to have a fscking joke list? |
| 19:11.12 | Weezey | I was part of the conversation before, you don't need to involve me anymore. |
| 19:11.22 | Weezey | www.weezey.com |
| 19:11.26 | P-Chan | I'm currently settinig up AMP w/ Asterisk on a virtual server and am having to manually adapt most everything for this environment. /sigh - I keep getting Autodial: Unable to open file, any ideas? |
| 19:11.36 | Weezey | is my antidote to sending jokes |
| 19:11.40 | Weezey | I get ' |
| 19:11.51 | Slainte | P-Chan enable debug on the logger.conf |
| 19:11.57 | Weezey | I get 'em and put them there, if you want to see them go, if not, don't. |
| 19:12.16 | CoaxD | weezey: Hehe |
| 19:12.19 | P-Chan | Slainte: ok, will do and test again, brb. |
| 19:13.09 | *** join/#asterisk drooth (~drooth@user-0cev8e9.cable.mindspring.com) |
| 19:13.46 | Weezey | What's the best way to log calls with asterisk? I need to track the time of calls. |
| 19:13.58 | Slainte | Weezy, look at the Wiki for billing |
| 19:13.58 | anti | does anyone know the differences between the cisco ip phone 7960 and 7960G, I know the button are internationalized, but other than that? |
| 19:14.11 | P-Chan | Slainte: Still get the same. I also get a bunch of Context 'from-internal' tries includes non-existant context 'from-internal-custom' and other of the same messages before the Autodial error |
| 19:14.15 | Slainte | * will do csv or SQL, in MySql, post, or sqlite |
| 19:14.36 | tzanger | P-Chan: do you have a "from-internal-custom' context in extensions.conf? |
| 19:14.46 | eKo1 | you forgot ODBC |
| 19:14.55 | tzanger | P-Chan: simple test: type "show diaplan from-internal-custom" in the asterisk CLI |
| 19:15.19 | Slainte | and ODBC :) |
| 19:15.19 | Weezey | Slainte: cool, I basically just want to know how to capture on and off hooks properly. |
| 19:15.29 | P-Chan | Unable to connect to remote asterisk - althrough it is running, I don't have 127.0.0.1 since I'm on a virtual server, maybe that has something to do with it? |
| 19:15.43 | Slainte | Weezy, it is automagic if you read the Wiki. Asterisk is not hard, You do need too read. |
| 19:16.22 | P-Chan | Slainte: Oh, I can get CLI if I start asterisk manually, amportal starts safe_asterisk |
| 19:17.10 | Slainte | thats a good start |
| 19:17.42 | P-Chan | Slainte: Dialplan doesn't exist... |
| 19:18.05 | Weezey | Slainte: that's very cool. All kinds of logs to parse now. thanks. |
| 19:18.32 | P-Chan | Slainte: from-internal exists, but no -custom. I'm gonna search the AMP sources for this internal-custom one |
| 19:20.09 | *** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 19:21.07 | tekjacob | anyone have a good clean way to light a MWI on a SIP phone when the phone registers to one * box and the Voicemail app lives on another? |
| 19:22.09 | P-Chan | Slainte: It takes care of a few of the missing dial-plans, checking for the others now too - ext-local, ext-group, ext-queues, outbound-allroutes |
| 19:23.27 | Weezey | tekjacob; what phone? |
| 19:23.47 | Weezey | wait, my possible solution won't help you. |
| 19:23.55 | tekjacob | Weezey mostly Polycom IP500 |
| 19:24.07 | tekjacob | Weezey Some Cisoco 7960 |
| 19:24.19 | *** join/#asterisk ikey (ikey@220.226.55.145) |
| 19:24.36 | P-Chan | Slainte: The extensions.conf in the amp sources contains: include => ext-local, but no actual file exists. :( |
| 19:24.42 | Slainte | P-Chan, thats the best way to do it. Jsut deal with the problems ones at a time. 90% off errors in * for new people, is in the extensions.conf |
| 19:25.05 | Slainte | thats fine, is it referenced from anywhere else, i.e in any calls? |
| 19:25.32 | P-Chan | exten => #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}o) |
| 19:25.42 | P-Chan | exten => *411,3,AGI(directory,general,ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}) |
| 19:26.14 | *** join/#asterisk toyosi (~gabri@216.90.111.178) |
| 19:26.50 | toyosi | hello people |
| 19:27.01 | P-Chan | Slainte: I hate to do this, but I think I'm going to resort to comparing my installatino to the one on the *@Home installation - a cheap way out I suppose. |
| 19:27.34 | tzanger | what's that amazing echo cancel howto doc that was on the lists? I can't find it and google's being unhelpful |
| 19:27.44 | Juxt | yay my metro-ethernet just arrived :-) |
| 19:27.52 | Sedorox | ?? |
| 19:27.54 | fearnor | yawn |
| 19:28.00 | fearnor | i'm tired from dropping bombs on asterisk-biz ;) |
| 19:28.03 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 19:28.09 | tzanger | bombs? |
| 19:28.17 | eKo1 | metro-ethernet? |
| 19:28.33 | fearnor | tzanger: slang. bombs/science means 'clue' ;) |
| 19:28.42 | Juxt | yeah 10 megabit of router-less traffic, a fiber hand off |
| 19:28.55 | tzanger | hahaha |
| 19:28.56 | fearnor | werd to metro fiber |
| 19:29.06 | fearnor | 'dropping science like bombs' |
| 19:29.07 | tzanger | youn're not supposed to drop bombs on -biz, you're supposed to collect contracts |
| 19:29.15 | fearnor | tzanger: yeah well |
| 19:29.31 | fearnor | i'm not interested in 99% of customers who would come from -biz |
| 19:29.43 | Weezey | .9 |
| 19:29.57 | CoaxD | 260 packages upgraded, 74 newly installed, 36 to remove and 5 not upgraded. |
| 19:29.58 | CoaxD | Need to get 135MB of archives. After unpacking 95.0MB will be used. |
| 19:29.58 | CoaxD | Do you want to continue? [Y/n] n |
| 19:30.01 | CoaxD | ow. |
| 19:30.16 | fearnor | disk space is cheap ktnx |
| 19:30.29 | CoaxD | fearnor: Yes, but tis a reasonably long download on my T1 :) |
| 19:30.34 | Juxt | scsi disk space isn't cheap |
| 19:30.40 | fearnor | it is cheap. |
| 19:30.42 | tzanger | 135M on a T1 isn't bad |
| 19:31.08 | CoaxD | tzanger: I get about 90k/sec if that |
| 19:31.12 | tekjacob | So no love on the two asterisk scenario (phones register to one - VM App on the other)? |
| 19:31.27 | tzanger | CoaxD: that's not a T1 then |
| 19:31.33 | CoaxD | uh |
| 19:31.50 | Juxt | hypothetical question: if i had to transcode from 1 codec to another, what would i use? |
| 19:31.56 | toyosi | I need some help with the snom phone |
| 19:31.56 | CoaxD | tzanger: Well i'm not sure then |
| 19:32.01 | CoaxD | tzanger: Maybe you can tell me what this means |
| 19:32.02 | CoaxD | Serial1 is up, line protocol is up |
| 19:32.02 | CoaxD | <PROTECTED> |
| 19:32.02 | CoaxD | <PROTECTED> |
| 19:32.03 | Juxt | provided i am currently running asterisk and nothing else |
| 19:32.06 | CoaxD | tzanger: kthx |
| 19:32.07 | tzanger | T1 should max out around double that |
| 19:32.21 | CoaxD | tzanger: Sure, it maxes out around that. :) |
| 19:32.23 | tzanger | 1544kbit = 193kByte |
| 19:32.33 | CoaxD | tzanger: It actually equals out to be about 170kbyte/sec |
| 19:32.38 | CoaxD | tzanger: (or a little more) |
| 19:33.23 | tzanger | yeah that's about right |
| 19:33.23 | CoaxD | tzanger: mathematically, you are correct |
| 19:33.23 | toyosi | I leave a voice mail and the phone displays a vmail button |
| 19:33.23 | tzanger | you need to take into account TCP/IP overhead |
| 19:33.23 | toyosi | I then press the button and get a fast busy |
| 19:33.23 | CoaxD | tzanger: Yeep |
| 19:33.23 | toyosi | please help! |
| 19:33.23 | *** part/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 19:33.27 | CoaxD | tzanger: The point is, I do indeed have a T1. The path between http.us.debian.org and us doesn't allow us to push the full amount |
| 19:33.49 | CoaxD | tzanger: Plus, i'm competing with local traffic |
| 19:33.49 | tzanger | CoaxD: use a different mirror then :-) |
| 19:34.03 | CoaxD | <PROTECTED> |
| 19:34.03 | CoaxD | <PROTECTED> |
| 19:35.28 | toyosi | is anybody using the snom 190? |
| 19:35.57 | CoaxD | Playing MPEG stream from The Little Mermaid - Kiss The Girl (Techno Remix).mp3 ... |
| 19:36.12 | Juxt | ahaha |
| 19:36.27 | Juxt | how about barbie girl rammstein cover |
| 19:36.57 | *** join/#asterisk tekjacob (~chris@c2.efb7d1.client.atlantech.net) |
| 19:37.22 | Juxt | i use shakatura in my music on hold, trips people out |
| 19:40.07 | PBXtech | whats the deal with IAX establishing a voice path but no audio on either side? doesnt it use that same port to voice path? |
| 19:40.31 | PBXtech | T1-IAX=IAX-T1 |
| 19:40.32 | MindChild | Can Astrisk just be a simple voicemail mechnism? Something I can use as an answering machine, and called into for messages when away? Or is there a more appropriate project for that? |
| 19:40.44 | PBXtech | [MindChild]: sure |
| 19:41.00 | bjohnson | MindChild: it can be used for that |
| 19:41.10 | Juxt | MindChil: you want to deploy a pc just to use it as an answering machine? |
| 19:41.23 | MindChild | Juxt: at the moment, yes |
| 19:41.25 | PBXtech | oh the horror |
| 19:41.26 | PBXtech | :) |
| 19:41.33 | Juxt | MindChild |
| 19:41.39 | MindChild | Im not finacially ready to nab some digiphones |
| 19:41.39 | bjohnson | of course .. it would work on a pc that was used for other stuff too |
| 19:41.40 | Juxt | you can just have one extension, s |
| 19:42.09 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 19:43.14 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 19:43.20 | MikeJ[Laptop] | ~seen juggie |
| 19:43.25 | jbot | juggie is currently on #asterisk (8d 12h 36m 47s). Has said a total of 60 messages. Is idling for 22h 45m 48s |
| 19:45.18 | bjohnson | MindChild: the emailed notifications are a nice feature for that purpose |
| 19:47.01 | PBXtech | whats the deal with IAX establishing a voice path but no audio on either side? doesnt it use that same port to voice path? |
| 19:47.09 | PBXtech | ne1 |
| 19:47.31 | bjohnson | yes |
| 19:47.40 | bjohnson | maybe codecs? |
| 19:49.12 | MindChild | Ok, this is probably the bonehead question of the day... rather then getting an ATA adapter and using an analog phone, is there some way to use a second modem, or even a serial card to plug a plain old phone into the Asterisk system? |
| 19:49.27 | *** join/#asterisk bah (048830696@ACABCEEB.ipt.aol.com) |
| 19:50.07 | PBXtech | tried ulaw and gsm |
| 19:50.27 | Juxt | MindChild: yes you can send a call to a zap interface easy |
| 19:50.54 | PBXtech | any other ideas? |
| 19:51.00 | *** join/#asterisk Veryhot (Veryhot@adsl-68-125-233-50.dsl.sndg02.pacbell.net) |
| 19:51.02 | MindChild | OK, you lost me. What is a "zap" interface? |
| 19:51.12 | PBXtech | are you Veryhot? |
| 19:51.15 | Juxt | a zaptel compatible voice modem |
| 19:51.26 | Veryhot | pbxtech: hi |
| 19:51.42 | PBXtech | oh a scale of 1 to 10 how hot are you |
| 19:51.55 | Veryhot | quick question about Asterisk@home, anyone able to get voipjet working with it? |
| 19:52.05 | MindChild | Im 11 |
| 19:52.55 | PBXtech | voipjet should work fine, follow the instuctions like they said |
| 19:53.04 | MindChild | Juxt: Ok, just to be straight, I can have one modem as the one that answers the phone line, and a mess of "zap" modems, with a phone hooked into them> |
| 19:53.29 | Juxt | yes |
| 19:53.31 | ariel_ | MindChild, a mess of zap modems? |
| 19:53.51 | Veryhot | pbxtech: I did, but it keep dial out as /voipjet/ I need xxxx@voipjet |
| 19:54.09 | Veryhot | pbxtech: work fine with voicepulse thought. |
| 19:54.10 | Juxt | MindChild: you're seriously limited to the number of pci slots in the server tho |
| 19:54.35 | MindChild | Juxt: I have multitudes of machines. PC hardware wont be an issue |
| 19:54.42 | *** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net) |
| 19:54.44 | cp5 | hi |
| 19:54.57 | MindChild | Ive got many multiproc pentium pros I got to put to use. They all have 2 PCI buses |
| 19:55.04 | cp5 | anyone know under what conditions i would get warnings in asterisk: "No D-channels available! Using Primary on channel anyway" |
| 19:55.14 | cp5 | i have a quad T1 card in there, two ports are CPE, two are net |
| 19:55.27 | cp5 | and all calls seem to drop at that time |
| 19:55.54 | Juxt | MindChild: sounds like you're building a tank out of scrap metal |
| 19:56.09 | MindChild | Juxt: I got lots of solutions looking for problems! |
| 19:56.27 | festr_ | just a question, when call come to my E1, and i make after Dial Busy application, no busy tone is generated to incomming call, how to debug this? or what to do? :) |
| 19:57.41 | MindChild | Sweet. So maybe I can finally do something with all of those modems |
| 19:59.04 | Slainte | cp5, I am having the exact same problem all day |
| 19:59.15 | Slainte | what build are you using? |
| 19:59.46 | tzanger | it was anthm who said that asterisk winks properly |
| 19:59.53 | tzanger | who was I talking to about tha tnow |
| 19:59.58 | tzanger | manxpower and someone else |
| 20:00.15 | Sedorox | blankman: |
| 20:00.42 | *** join/#asterisk dogz- (~bob@66.148.168.234.nw.nuvox.net) |
| 20:00.57 | cp5 | Slainte, what kind of card? are your configs net or cpe? |
| 20:01.15 | Slainte | cpe, T100P |
| 20:01.17 | cp5 | Slainte, 1.0.6 asterisk/zaptel |
| 20:02.11 | dogz- | does someone mind taking a look at my config files http://pastebin.ca/9046 , when attempting to make an outgoing call it informs me how it cant create channel type Zap... and im unsure what would be wrong with my zapata.conf |
| 20:02.19 | festr_ | what Busy application exactly does? |
| 20:03.31 | festr_ | And does this work for E1? and it is generating busy audio to the PRI channel? because when debbuging pri debug span 1 after exec. busy nothing will pass to the E1 |
| 20:03.44 | festr_ | or have i miss something? :) |
| 20:05.59 | bannerman | Ok! I'm getting closer to the root of my ringtone problem, I think. I have LiveVoip and Nufone, but prefer to use LiveVoip. Unfortunately, when I call my LiveVoip number I get no ringtone while Nufone correctly generates a ringtone. I signed up for a month of broadvoice and their asterisk instructions have me use dtmf=inband and dtmfmode=inband. I get no ringtone with broadvoice when using g729. Do I need to somehow force it to use rfc2833? |
| 20:06.16 | jontow | hmm, i want my dual ppro back |
| 20:06.21 | jontow | or that quad ppro that i almost had and never got :( |
| 20:06.29 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 20:07.05 | jontow | i think that makes me weird.. ill be quiet now :o |
| 20:07.25 | *** join/#asterisk Veryhot (Veryhot@adsl-68-125-233-50.dsl.sndg02.pacbell.net) |
| 20:07.49 | *** join/#asterisk marno (~marno@212-62-90-130.teleos-web.de) |
| 20:08.07 | marno | any who knwos one of this error?` |
| 20:08.10 | marno | Call failed to go through, reason 8# |
| 20:08.12 | *** join/#asterisk drbrown (~chatzilla@user-0cdvec3.cable.mindspring.com) |
| 20:08.16 | *** part/#asterisk thetalon (~toddl@66.179.151.216) |
| 20:08.27 | marno | Call failed to go through, reason 5 |
| 20:08.31 | drbrown | does anyone have any recomendations on cheap iax phones? |
| 20:08.37 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 20:08.39 | marno | Call failed to go through, reason 0 |
| 20:08.44 | marno | Cause not handled |
| 20:10.31 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 20:11.42 | drbrown | does anyone have any recomendations on cheap iax phones? |
| 20:13.43 | Veryhot | drbrown: there is a new company that make some iaxphone |
| 20:15.15 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 20:15.28 | Veryhot | drbrown: http://www.eezeephone.com/ |
| 20:15.37 | rvhi | is there a command to show which pri channels are being used? |
| 20:16.12 | jontow | show channels |
| 20:16.13 | jontow | ? |
| 20:16.52 | jontow | i mean.. that'll catch all channels in use |
| 20:16.58 | jontow | but you'll be able to see which Zap channels at least |
| 20:17.19 | jontow | zap show channels .. that'll also work |
| 20:17.21 | jontow | but it lists them all. |
| 20:17.31 | MindChild | "cheap" being relative |
| 20:17.46 | MindChild | maybe Im the only one in the world who doesnt think $70 is cheap |
| 20:17.52 | drbrown | veryhot: how good are the phones? |
| 20:18.33 | Veryhot | drbrown: haven't tried them yet |
| 20:18.36 | *** join/#asterisk file[laptop] (~file@mctn1-6952.nb.aliant.net) |
| 20:18.41 | jontow | mindchild; i prefer "less expensive" ;) because i agree with you.. |
| 20:18.49 | jontow | $70 isn't something i can just toss whenever i feel like it |
| 20:18.52 | Veryhot | drbrown: need to get one, but seem good price |
| 20:18.55 | jontow | although it'd be nice.. :) |
| 20:19.47 | drbrown | I was going to get a phone off of iaxtalk.com, but their site sucks. |
| 20:23.29 | Veryhot | drbrown: this are new iax phones |
| 20:23.51 | bannerman | So.. are there any "less expensive" iax phones? |
| 20:24.34 | *** join/#asterisk captrb (~crozierm@64.65.134.42) |
| 20:26.01 | Juxt | well as far as voip goes 70 bux is as cheap as it's gonna get |
| 20:26.23 | Juxt | i remember when cicso was the only contender, $300 was cheap then |
| 20:26.35 | MindChild | They should have some IAX phones in little electronics kits like you get at radio shack where you can piece together your own phone |
| 20:26.45 | Juxt | ahah |
| 20:26.55 | bjohnson | cheap is relative |
| 20:27.03 | Juxt | buy a decent microcontroller and write you own IAX stack |
| 20:27.17 | Juxt | you're still looking at about 30 bux for the controller |
| 20:27.27 | bjohnson | compared to a Nortel system where the phones are $200 + each .. even for a basic phone .. a basic wifi phone is cheap |
| 20:28.16 | *** join/#asterisk pmowry (~chatzilla@12.166.196.9) |
| 20:28.50 | drooth | does anyone know a VoIP fax provider? |
| 20:29.06 | drooth | ie: send fax via VoIP over the net |
| 20:29.13 | Juxt | droot: welcome to hell my friend |
| 20:29.19 | bjohnson | compared to buying a 2 line multi handset 5.8 GHz cordless system for $200 .. buying 1 line single systems on fxs ATAs is same cost but fuller feature (therefore better value for same cost .. equals cheaper) |
| 20:29.19 | Juxt | i just went thru this |
| 20:29.23 | Weezey | hah |
| 20:29.24 | drooth | you did? and? |
| 20:29.25 | Weezey | some faxes to some machines go fine. |
| 20:29.32 | Juxt | well yeah... g711 |
| 20:29.36 | Juxt | and it just might work |
| 20:29.44 | Weezey | yep |
| 20:29.49 | Juxt | you need to send fax to 9600 baud max |
| 20:29.56 | drooth | I thought that VoIP FAX would work! |
| 20:30.04 | drooth | how about a provider that I can try it with? |
| 20:30.07 | Juxt | most machines let you set max baudrate |
| 20:30.08 | Weezey | so did a whole lotta people. |
| 20:30.10 | drooth | I know in the USA there are some |
| 20:30.18 | Juxt | any provider that allows you to use g711 will work |
| 20:30.22 | Juxt | unless they do transcoding |
| 20:30.24 | drooth | ok |
| 20:30.26 | bjohnson | drooth: so why are you asking us if "you know" |
| 20:30.28 | Juxt | and send your traffic elsewhere |
| 20:30.39 | pmowry | Hello, does anyone know a SQL query to export phone info from a Cisco CallManager 3.2 system? I want to move a department to an asterisk system for testing. |
| 20:30.43 | Juxt | remember 9600 is the key, anything higher is a waste of time |
| 20:31.01 | drooth | bjohnson: I want to set up faxing with a VoIP company |
| 20:31.11 | drooth | but I don't know who I can call. |
| 20:31.31 | Juxt | drooth: if you find an ata device that supports t.38 let me know |
| 20:31.36 | Juxt | so far i found none |
| 20:31.48 | drooth | what is that device for? |
| 20:32.03 | Juxt | ata device is what lets you connect pots to voip |
| 20:32.19 | Juxt | so if you have a fax machine you'd connect it to it |
| 20:32.25 | drooth | ok |
| 20:32.55 | fearnor | juxt: linksys PAP supports |
| 20:33.12 | fearnor | drooth: we support t.38 |
| 20:33.21 | bjohnson | fearnor: are you certain? |
| 20:33.27 | Juxt | fearnor who are you with? |
| 20:33.31 | bjohnson | or does it just say supports fax |
| 20:33.36 | fearnor | bj: thats what calladvantage uses, and they do fax. |
| 20:33.43 | fearnor | just: pilosoft |
| 20:33.47 | bjohnson | my SPA units do not do t38 |
| 20:33.58 | dogz- | jontow: sup buddy |
| 20:34.07 | bjohnson | they adjust to ulaw when a fax is deteced |
| 20:34.08 | fearnor | bj: hrmmmmm |
| 20:34.10 | *** part/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 20:34.21 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-227-18.dsl.scarlet.be) |
| 20:34.25 | Weezey | bjohnson: I've sent faxes out those. |
| 20:34.32 | fearnor | bj: well, did you *try* t.38 with a gateway that supports t38 on them? |
| 20:34.34 | Juxt | uugh you can't buy linksys device unless you're a big cahuna |
| 20:34.56 | bjohnson | fearnor: there is no documentation or web setup pages that even mentions t38 |
| 20:34.57 | Weezey | bjohnson: they don't all get there, but I did use 14400 |
| 20:35.04 | bannerman | My plan is to just use an Internet fax service, the kind where you email your faxes and receive them via email |
| 20:35.09 | bjohnson | they do however have settings for ulaw |
| 20:35.23 | *** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es) |
| 20:35.36 | Weezey | bjohnson: http://www.voip-info.org/wiki-Asterisk+T.38+Bounty |
| 20:35.46 | drooth | bannerman: what internet fax service are you using. I think efax is overpriced. |
| 20:35.50 | bjohnson | bannerman: that is avoiding voip .. and seems to be the current method that is most dependable |
| 20:35.53 | Juxt | can you reprogram a previously configured linksys pap? |
| 20:36.22 | queuetue | Can anyone recommend a SOHO router thta will provide shaping/QOS? I'd prefer to not dedicate a general purpose PC for this. |
| 20:36.25 | bjohnson | Juxt: only if you have the admin password |
| 20:36.31 | bannerman | drooth: I haven't gone there yet. Still using PSTN for faxes. Trying to get my phones working right first. |
| 20:37.33 | queuetue | Is the WRT54G sufficient? |
| 20:37.42 | drooth | no |
| 20:37.48 | drooth | been there, done that |
| 20:37.50 | drooth | dont waste your time |
| 20:37.53 | drooth | build a linux router |
| 20:37.57 | drooth | pfsense or m0n0wall |
| 20:38.03 | Juxt | i concur with drooth |
| 20:38.17 | Weezey | what about cisco? |
| 20:38.23 | *** join/#asterisk PBXtech[mobile] (~upirc@wirelessdata-167-248.mycingular.net) |
| 20:38.32 | bjohnson | fearnor: look under admin, advanced, line 1 .. look at the fax settings (assuming pap2 web pages are similar to spa units) |
| 20:38.38 | queuetue | Hrm... What's a source for good, tiny, quiet, cheap pcs for the purpose? |
| 20:38.38 | terrapen | m0n0wall is the shit |
| 20:38.41 | Weezey | hmm, not really soho though. |
| 20:38.43 | bjohnson | fearnor: FAX Passthru Codec: g711u |
| 20:38.46 | terrapen | but OpenBSD+pf is better |
| 20:38.49 | fearnor | hrm |
| 20:38.54 | fearnor | bj: latest firmware? |
| 20:39.01 | bjohnson | fearnor: no mention of t.38 .. same thing in the user manual pdf from sipura |
| 20:39.12 | fearnor | er, we talking about same device? |
| 20:39.12 | terrapen | i use a Soekris for my firewall and a WRT54G for my AP |
| 20:39.14 | fearnor | PAP? |
| 20:39.18 | Weezey | nope |
| 20:39.19 | bjohnson | 2.0.13(GWg) |
| 20:39.27 | terrapen | the WRTs are great for wireless |
| 20:39.33 | terrapen | but sucky-sucky for anything else |
| 20:40.21 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 20:40.29 | bjohnson | fearnor: no we are not talking about the same device .. I am talking Sipura SPA 2000 and you are talking Linksys Pap2 |
| 20:40.42 | mog_home | soekris board makes a slick asterisk box |
| 20:40.50 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 20:41.17 | queuetue | terrapen, What specifically do I need ot buy at the soekris website? Does "board only" include power? Do I need two ethernet boards? |
| 20:41.39 | queuetue | Ah, i see "standard configs" now. |
| 20:41.42 | Sedorox | lol |
| 20:42.01 | bjohnson | fearnor: however .. everything I've read suggests that the linksys Pap2 IS the Sipura SPA 2000 |
| 20:42.50 | bjohnson | terrapen: running stock firmware? |
| 20:42.59 | bjohnson | terrapen: the wrt54g? |
| 20:44.53 | dogz- | woohoo fixed my problem with my x100, sorta... Had an irq conflict with both of them installed. Removed one and it works great |
| 20:47.35 | Juxt | dogz: check bios irq sharing |
| 20:47.36 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
| 20:47.44 | cjk | hi, there are several versions of spandsp |
| 20:47.48 | cjk | whcih one should i use |
| 20:47.57 | drbrown | any suggestions on ip phones? |
| 20:48.06 | marno | any who knwos one of this error?` |
| 20:48.08 | marno | Call failed to go through, reason 8# |
| 20:48.10 | dogz- | Juxt thanks for the suggestion will def try it out |
| 20:48.11 | marno | Cause not handled |
| 20:48.23 | marno | -# |
| 20:51.50 | mishehu | dogz-: it's normally better to have a tdm400 card with whatever modules you need instead of two separate x100p cards |
| 20:51.56 | mishehu | in the same machine |
| 20:52.09 | Juxt | mishehu: i have four x100ps with no issues |
| 20:54.34 | dogz- | yea i actually just bought a TDM400 off of digium |
| 20:54.40 | dogz- | just waiting for it to come in :) |
| 20:54.49 | shido6 | SWEET |
| 20:54.53 | captrb | if anybody has any experience with PRI(hdlc+voice), i could use a tip |
| 20:54.59 | shido6 | ooh |
| 20:55.00 | shido6 | hdlc |
| 20:55.10 | shido6 | sounds like a login with ssh for support question there |
| 20:55.12 | *** join/#asterisk jakepdev (~jakepdev@pool-68-163-51-71.phil.east.verizon.net) |
| 20:55.36 | captrb | that's what I thought.. |
| 20:55.43 | tzanger | captrb: fun sutff... the linux kernel keeps changing the HDLC interface every few kernel revs :-) |
| 20:56.04 | shido6 | yep |
| 20:56.15 | shido6 | it can be done - so dont get all freaked out |
| 20:56.18 | captrb | trying to use 2.6.11... is that stupid? |
| 20:56.27 | tzanger | not sure |
| 20:56.29 | *** join/#asterisk ToyKeeper (spanky@c-24-9-113-171.hsd1.co.comcast.net) |
| 20:56.31 | captrb | well, I'm trying to bypass eschelon's Adtran, which I suspect is causing problems |
| 20:56.34 | tzanger | but yes people are doing it, as shido6 suggests |
| 20:56.40 | tzanger | be calm |
| 20:56.48 | *** join/#asterisk crich1999 (~crich@212.122.40.196) |
| 20:56.53 | shido6 | thats pretty funny really |
| 20:56.59 | shido6 | almost the same engineering |
| 20:57.24 | shido6 | woooooosahhh |
| 20:57.45 | captrb | i configured the interfaces, but can't seem to ping anything. |
| 20:58.13 | captrb | (and can't reload the modules because the kernel oopses!) |
| 20:59.28 | captrb | now i'm looking for docs/utilitys (and maybe $upport :-) to track it down |
| 20:59.33 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 20:59.44 | wildgoose | How can I have a variable set depending on my starting context? The point is to have a standard dialplan, but to dial out via a different sip provider depending on the handset used (business/personal) |
| 20:59.45 | captrb | s/utilitys/utilities/ |
| 20:59.49 | PTG123 | anyone know off hand where asterisk stores voicemail? |
| 21:00.04 | riksta | it tells you on the cli |
| 21:00.08 | captrb | var/spool/voicemail/ |
| 21:00.21 | Nugget | var/spool/asterisk/voicemail/ |
| 21:00.37 | captrb | (sorry, i altered the layout) |
| 21:01.02 | Juxt | i like installing asterisk into /usr/local/asterisk |
| 21:01.08 | Juxt | easy to manage when there's more than 1 copy running |
| 21:01.19 | Nugget | /usr/local/ is a great habit to have. |
| 21:01.23 | `Rage | heh |
| 21:01.31 | `Rage | i never got into that habit :/ |
| 21:01.37 | Nugget | I hate being on linux boxes that stuff everything in root. |
| 21:01.37 | `Rage | my files are everywhere :/ |
| 21:01.40 | Nugget | it's such a mess |
| 21:01.41 | Juxt | when it's time to remove you just drop the folder |
| 21:01.44 | Juxt | no need to look around |
| 21:01.51 | `Rage | heh |
| 21:02.05 | Juxt | i think the whole spreading the wealth all over the filesystem thing is kinda weird |
| 21:02.10 | Juxt | i never understood it |
| 21:02.23 | Nugget | it's simple. put the base os at the root level and the site-specific stuff in /usr/local/ |
| 21:02.31 | Nugget | it makes upgrading easier and less dangerous |
| 21:02.37 | Nugget | and it makes backups a lot simpler |
| 21:02.41 | Juxt | tru dat |
| 21:02.47 | `Rage | that does make a lotta sense |
| 21:03.00 | `Rage | heh |
| 21:03.04 | captrb | wildgoosewildgoos |
| 21:03.36 | queuetue | I like having all of my configs in /etc, all of my variable data in /var, so I can put it on a faster disk. I like sharing /tmp with noatime and no quotas... |
| 21:03.40 | wildgoose | yeah? |
| 21:03.48 | captrb | wildgoose: sorry |
| 21:03.58 | Nugget | OS configs in /etc, site-specific configs in /usr/local/etc :) |
| 21:04.04 | Juxt | i usually move out /var stuff too |
| 21:04.06 | wildgoose | anyone answer my question on dialplans? |
| 21:04.07 | Juxt | via symlinks |
| 21:04.08 | captrb | wildgoose: why not have two contexts, then include the common stuff in each? |
| 21:04.18 | wildgoose | yeah, that's kind of what I am trying |
| 21:04.28 | wildgoose | But since the two contexts are basically identical |
| 21:04.45 | wildgoose | I want to just have a var at the top, and then I can include the subcontext and they will be the same |
| 21:04.48 | wildgoose | see the point? |
| 21:04.56 | Nugget | sounds like a good plan. what's your question? |
| 21:04.59 | Maveric | anyone from portugal here? |
| 21:05.09 | wildgoose | I tried just doing exten=>s,1,SetVar() |
| 21:05.20 | wildgoose | but that doesn't seem to set the var when a specific extension is rung |
| 21:05.39 | wildgoose | So basically, given phone A is on one context and B on a different context |
| 21:05.50 | *** join/#asterisk bimmerd00d (~Podunk121@68.92.185.130) |
| 21:05.59 | wildgoose | how can I have some var set that I can use to make my "Dial" command the same in both cases |
| 21:06.15 | bimmerd00d | is anyone here using Asterisk@Home that can give me some help? |
| 21:06.22 | L|NUX | ~google asterisk documentation |
| 21:06.22 | wildgoose | (ie dial out on a different sip provider on one context to the other, but the dial plan is otherwise identical) |
| 21:06.25 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 21:06.28 | nesys | hi folks |
| 21:06.37 | Nugget | instead of including the common context, you could do [homecontext] _.,1,SetVar(COWS=MOO) _.,2,Goto(commoncontext,${EXTEN},1) |
| 21:06.53 | Maveric | wildgoose what are you trying to do exactly? |
| 21:06.55 | wildgoose | Aha. I think that's the answer. Thanks |
| 21:06.57 | bimmerd00d | trying to connect asterisk via a SIP trunk to an external SIP provider to make calls. And i can't for the life of me get it to place a call |
| 21:06.59 | nesys | I've problems with FWD and IAX ... with SIP all works fine, with IAX2 my registration is rejected :( |
| 21:07.11 | Juxt | create a template context |
| 21:07.19 | Juxt | then create context1 |
| 21:07.21 | wildgoose | Basically I want a var like "SIPPROVIDER" which is set differently depending which phone is rung. Business or personal |
| 21:07.22 | Juxt | include the template context |
| 21:07.35 | Juxt | and add exteions _1xxx ... to that context that uses your provider |
| 21:07.36 | wildgoose | what is a template context...? |
| 21:07.56 | Juxt | just a context used as a template for your identical contexts |
| 21:08.07 | wildgoose | is it special? |
| 21:08.10 | Juxt | no |
| 21:08.27 | Juxt | just a regular context that you include to your context |
| 21:08.39 | Juxt | kinda like demo in the sample extensions.conf |
| 21:08.39 | wildgoose | I have the template context. I just need to work out how to get my var set differently when entering from one context or the next. I think Nugget gave me the answer though |
| 21:09.10 | Maveric | how are you making the decision |
| 21:09.15 | Maveric | to goto either or context? |
| 21:09.21 | Maveric | based on which phone was called? |
| 21:10.27 | wildgoose | yes |
| 21:10.38 | wildgoose | one phone for business, and one for personal |
| 21:10.47 | wildgoose | Dialplan lets me override the default. |
| 21:10.52 | Maveric | do they come in on different numbers? |
| 21:10.56 | wildgoose | yes |
| 21:11.13 | wildgoose | But it is outgoing calls I am interested in, not incoming |
| 21:11.18 | Maveric | via a pri or pstn? |
| 21:11.23 | wildgoose | sip |
| 21:11.24 | Maveric | ok |
| 21:11.35 | Maveric | so you have two different physical phones? |
| 21:11.40 | nesys | Could you advice a free IAX provider? |
| 21:11.59 | Juxt | nesys: ahaha |
| 21:12.00 | Juxt | FWD |
| 21:12.10 | wildgoose | basically for billing purposes I have two sip accounts at sipgate. My phone in my study defaults to dialing out on business line and the dect phones in the house default to personal account |
| 21:12.28 | wildgoose | I think the goto trick is what I need. Thanks |
| 21:12.44 | *** part/#asterisk tekjacob (~chris@c2.efb7d1.client.atlantech.net) |
| 21:12.53 | *** join/#asterisk tekjacob (~chris@c2.efb7d1.client.atlantech.net) |
| 21:13.02 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 21:13.17 | bimmerd00d | how can i configure asterisk at home to dial out using a sip account i have with a provider? |
| 21:13.19 | *** join/#asterisk mike8901 (mike8901@ool-4356f52f.dyn.optonline.net) |
| 21:13.43 | mike8901 | do you guys think vonage would work over a dialup connection(laptop connects to dialup and bridges over to vonage router) |
| 21:13.43 | nesys | Juxt ahahahahah ... it doesn't work |
| 21:13.49 | Maveric | wildgoose i'd prolly use two different contexts |
| 21:13.56 | mike8901 | cause i'm staying in st barths where calls to the us are a dollar a minute :/ |
| 21:14.06 | mike8901 | and there is no highspeed where i am staying |
| 21:14.30 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 21:14.47 | _SMP_ | anyone know what's up with nufone? |
| 21:14.49 | jsharp | mike8901: Should work. They use a good codec for it. |
| 21:15.01 | captrb | for TE110P with hdlc data channels, is this normal output from sethdlc-new? |
| 21:15.01 | mike8901 | jsharp: huh? |
| 21:15.05 | shido6 | _SMP_, whats up? |
| 21:15.05 | captrb | hdlc0: unknown interface 0x1 |
| 21:15.06 | bimmerd00d | mike8901: doubtful, i've used G.729a and it barely worked over a dialup connection. I'm almost positive vonage uses g.711u |
| 21:15.20 | mike8901 | hmmm |
| 21:15.33 | mike8901 | i can get wifi but its a 5 minute drive every time i need to make a call :S |
| 21:15.58 | Cheng29 | mike8901.. are you staying at a hotel? |
| 21:15.58 | nesys | Juxt it's strange ... with SIP all works fine, with IAX: 1)iaxtel, as you know, is flapping .. 2)fwd rejects my registration (but is correct, with SIP I'm registered) |
| 21:16.05 | bimmerd00d | mike8901: sounds like you need to go purchase a network adapter that allows you to connect a larger directional antenna to it ;) |
| 21:16.34 | Juxt | what is happening with iaxtel? i was considering getting an account there |
| 21:16.42 | mike8901 | cheng29: no, i am staying at a villa |
| 21:16.55 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 21:16.57 | mike8901 | bimmerd00d: it's a few miles line of site at best :/ |
| 21:17.11 | nesys | Juxt is flapping ... request sent ... auth sent .. registered ... request sent .. auth sent .. and so on |
| 21:17.12 | jero_sflphone_03 | hi |
| 21:17.29 | jero_sflphone_03 | if anyone interested...SFLphone 0.3 is out |
| 21:17.35 | bimmerd00d | mike8901: i fail to see the problem. I'm connected at home via 802.11 to my work, which is over 6 miles away using a large directional antenna with a signal amplifier |
| 21:17.40 | shido6 | mid April _SMP_ |
| 21:18.08 | mike8901 | bimmerd00d: did i mention that a) my wifi card bearly works b) it doesnt have an antenna jack and c) I dont want to spend more than $5 |
| 21:18.31 | shido6 | want me to send you one |
| 21:18.34 | *** part/#asterisk Veryhot (Veryhot@adsl-68-125-233-50.dsl.sndg02.pacbell.net) |
| 21:18.37 | mike8901 | as i understand it amplifers can cost over 200$ |
| 21:18.44 | bimmerd00d | mike8901: ahh well if you dont wanna spend any money, that's a diff story. If you are still here, see if you can dig up a dialup account to test with and try it out before you leave |
| 21:18.44 | captrb | ah loopback |
| 21:18.55 | bimmerd00d | mike8901: mine costed a whopping $29 |
| 21:19.01 | bimmerd00d | mike8901: it's a Linksys |
| 21:19.18 | *** join/#asterisk girabraz (~root@200.121.129.178) |
| 21:19.32 | mike8901 | bimmerd00d: is there any way i could setup an asterisk box to communicate with the phone system, and recompress the audio |
| 21:19.33 | bimmerd00d | mike8901: the antenna was about $50, required a little bit of wiring for the proper connectors, but it's solid |
| 21:19.43 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 21:19.55 | mike8901 | just outta curiousity which antenna? |
| 21:20.04 | bimmerd00d | mike8901: no, because it's all dependent on what codec Vonage is using |
| 21:20.21 | bimmerd00d | mike8901: i think it's a Hawking |
| 21:20.37 | mike8901 | bimmerd00d: i'm saying can i recompress it and send it over the internet again(yesi know it produces latency) |
| 21:20.39 | bimmerd00d | mike8901: honestly i've been through like 3 or 4 different brands |
| 21:20.45 | drooth | anyone know of a good eFax-like service? anyone tried any of these: http://www.iptelephony.org/GIP/providers/fax/ |
| 21:20.56 | bimmerd00d | mike8901: i suppose it's possible, i'm not sure how to set that up though |
| 21:21.04 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 21:21.14 | bimmerd00d | mike8901: new to asterisk, not new to voip :) |
| 21:22.27 | girabraz | hi people |
| 21:22.49 | girabraz | i need configurator my sip server proxy |
| 21:23.18 | Cheng29 | girabraz... you need help? |
| 21:23.35 | girabraz | Someone can give me some guidelines |
| 21:23.39 | girabraz | yes |
| 21:23.45 | girabraz | cheng 29 |
| 21:24.14 | girabraz | Newly and installed |
| 21:24.16 | girabraz | asterisk |
| 21:24.32 | jakepdev | hi |
| 21:24.39 | jakepdev | (for bkw's benefit) |
| 21:25.16 | *** join/#asterisk JimVanM (~jimvanm@Toronto-HSE-ppp3701421.sympatico.ca) |
| 21:25.35 | jakepdev | is there a way to disable devices without going into the bios? I have two zaptel cards that are sharing IRQs and digium tells me that the zaptel cards can't share IRQs |
| 21:25.55 | danalien | Loj' |
| 21:25.58 | jakepdev | for instance, I want to disable usb |
| 21:26.08 | danalien | any zaptel+bristuff-hacker around? |
| 21:26.15 | shido6 | jakepdev |
| 21:26.15 | girabraz | cheng 29 : You can give me a hand, I do not find a good manual for the configuration |
| 21:26.44 | jakepdev | yep |
| 21:27.58 | captrb | great. ifconfig hangs on zap hdlc0 interface |
| 21:28.16 | danalien | shido6: jakedev : ... was that an answer directed toward me? :-) |
| 21:28.21 | captrb | bugalicious |
| 21:28.35 | jakepdev | danalien - no |
| 21:28.49 | jakepdev | or maybe yes :) |
| 21:29.24 | jakepdev | i guess i'm a hacker in a sense... |
| 21:29.49 | captrb | thanks goodness for journaling filesystems, because the kernel won't be able to shutdown if it can't deactivate the interafaces |
| 21:29.57 | captrb | cleanly |
| 21:29.59 | jakepdev | dan - what's up? |
| 21:30.17 | PTG123 | whats the commandline to convert a wav to gsm for sox, anyone know off hand? |
| 21:30.33 | danalien | jakepdev: hacker : ..enough to solve this riddle? - "is it possible to 'software cross' the zaptel+bristuff driver? What I mean, is control what signal goes to what pin - instead of having to slit a kabel and 'hardware cross' it" :-) |
| 21:31.23 | jakepdev | nope - dan - don't know about that |
| 21:31.41 | jakepdev | i don't think that can be done |
| 21:31.41 | niZon | ~google sox gsm to wav |
| 21:32.25 | captrb | * is making sysreq my friend |
| 21:32.57 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:34.23 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:35.00 | *** join/#asterisk rob_- (~robb@matrix.netsoc.tcd.ie) |
| 21:36.04 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 21:36.36 | r0d3nt | Is there a way to transfer someone directly to voicemail ?? |
| 21:36.45 | cjk | hi, any spandsp experts here |
| 21:36.54 | r0d3nt | instead of ringing their extension for the duration ? |
| 21:37.03 | *** part/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 21:38.25 | BoRiS | r0d3nt(exten => s,1,Voicemail(u9999) )? |
| 21:39.06 | rob_- | I have two iax clients (iaxcomm) communicating through Asterisk over a lan. Is it normal that they should exchange a large number of packets in call setup? |
| 21:39.11 | bannerman | what's the difference between u<mailbox> and s<mailbox>? |
| 21:39.21 | *** join/#asterisk phpboy (~sj@tpr-165-249-135.telkomadsl.co.za) |
| 21:39.31 | BoRiS | u=unavailable message gets played... b=busy message gets played |
| 21:39.37 | *** part/#asterisk Geraldoramos (~FullT@200.97.7.171) |
| 21:39.37 | bannerman | Boris: thanks |
| 21:39.46 | BoRiS | :) |
| 21:41.18 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 21:41.30 | Wazb | hi all |
| 21:42.01 | SkySky | hi |
| 21:42.13 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 21:43.13 | phpboy | I get this error when I run asterisk |
| 21:43.27 | Wazb | i have a cisco which is pointing a phone number to Asterisk box , where i need to configure in Asterisk to accept and process those calls |
| 21:43.28 | r0d3nt | BoRiS, so i put something like that in my dialplan ? |
| 21:43.35 | phpboy | Ouch ... error while writing autio data: : Broken pipe |
| 21:43.42 | phpboy | how can I resolve this issue? |
| 21:43.55 | jakepdev | phpboy - check your zaptel config |
| 21:44.04 | phpboy | hmm |
| 21:44.05 | phpboy | ok |
| 21:44.27 | eKo1 | eh, how do you know the problem is zaptel? |
| 21:44.29 | jakepdev | there's usually a line above the one you saw that explains better what the error is |
| 21:46.00 | jakepdev | don't know for sure without seeing the other error messages above - but more times than not, that last error is zaptel mis-configuration |
| 21:46.10 | Wazb | i have a cisco which is pointing a phone number to Asterisk box , where i need to configure in Asterisk to accept and process those calls |
| 21:46.25 | phpboy | how do I unload and reload the zaptel configs? |
| 21:46.45 | jakepdev | phpboy - restart * |
| 21:46.49 | jakepdev | ztcfg -vv |
| 21:47.02 | phpboy | ah, ok |
| 21:47.06 | cjk | anyone here who got rxfax working |
| 21:47.07 | facek_ | phpboy ztcfg -s && ztcfg -vvv |
| 21:47.17 | cjk | i just get 8 kb files |
| 21:48.46 | Maveric | cjk i've had it working for awhile |
| 21:48.53 | phpboy | hmmm |
| 21:48.57 | phpboy | I get the following error |
| 21:49.15 | phpboy | line 0: Unable to open master device '/dev/zap/ctl' |
| 21:49.56 | phpboy | how can I address that issue? |
| 21:50.38 | *** join/#asterisk bjohnson (~bjohnson@ip169-172.dsl.istop.com) |
| 21:50.56 | captrb | phpboy: stupid question, but are you running as root? |
| 21:51.05 | captrb | phpboy: or asterisk user? |
| 21:51.12 | phpboy | root |
| 21:51.16 | Lee__ | anyone here using Voicepulse for termination with the Speex codec? |
| 21:51.38 | Lee__ | It's rejecting all my calls through it |
| 21:51.44 | captrb | phpboy: just checking. (step 1: is it plugged in. step 2: permissions?) |
| 21:52.02 | phpboy | it is plugged in |
| 21:52.04 | phpboy | but I see |
| 21:52.14 | phpboy | /dev/zapctl |
| 21:52.22 | phpboy | as opposed to /dev/zap/ctl |
| 21:52.23 | phpboy | :< |
| 21:52.27 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 21:53.11 | jakepdev | anyone in the Knockville, TN area? |
| 21:53.17 | jakepdev | Knocksville |
| 21:53.29 | Nugget | Knoxville. |
| 21:53.30 | captrb | phpboy: are you using devfs? |
| 21:53.39 | jakepdev | nugget thx :) |
| 21:53.43 | phpboy | I'm not sure |
| 21:53.44 | Lee__ | I like Knocksville better |
| 21:53.50 | phpboy | default install of Mandrake 10.1 |
| 21:53.53 | jakepdev | nugget - are you there? |
| 21:53.54 | Lee__ | the ville of hard knocks |
| 21:54.03 | jakepdev | or around there? |
| 21:54.06 | Nugget | no, but I can spell it. |
| 21:54.10 | jakepdev | tnx |
| 21:54.49 | captrb | phpboy: what linux distribution? |
| 21:55.04 | Chuji | Anyone use an ftp server to dish out Polycom config files? |
| 21:55.29 | captrb | Chuji: yes |
| 21:55.41 | Chuji | jakepdev : I'm about 90 miles from Knoxville |
| 21:55.54 | Chuji | captrb : Does it use a certain username and password? |
| 21:56.00 | Chuji | captrb : when it logs in? |
| 21:56.10 | captrb | Chuji: yeah, you set it in the phone |
| 21:56.19 | *** join/#asterisk Johnsie (~John@acs-24-154-32-12.zoominternet.net) |
| 21:56.20 | AmaDEE0_ | Does SIP work good behind SNAT, where all ports from a pub IP are forwarded to the same private IP. |
| 21:56.38 | phpboy | captrb: Mandrake 10.1 - kernel 2.6.8.1-12mdk |
| 21:56.47 | captrb | Chuji: menu/settings/network configuration/server menu/ftp user |
| 21:57.26 | Chuji | captrb : So I can't just plug a phone in and it find the ftp server via dhcp? |
| 21:57.27 | *** join/#asterisk madounet (~madounet@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 21:57.32 | Chuji | captrb : I have to config the phone first? |
| 21:57.34 | captrb | Chuji: when I configured a new phone, all I would initially change was dhcp boot server option, ftp user/pass, and create a default config |
| 21:57.50 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 21:57.52 | *** join/#asterisk ScaredyCat (~ScaredyCa@i169173.upc-i.chello.nl) |
| 21:57.54 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 21:57.56 | captrb | Chuji: yeah, you need to configure your dhcp server to give out the boot server IP |
| 21:57.57 | AgiNamu | "All Domestic terminations which exceed 20% of Intra-State traffic will be subject to a .0200 surcharge." |
| 21:58.06 | Chuji | captrb : and password? |
| 21:58.19 | AgiNamu | Can anyone explain wtf that means? This is on my domestic termination agreement. |
| 21:58.24 | captrb | Chuji: no, that must be hand configured int he phone initially (I think) |
| 21:58.35 | Chuji | captrb : That seems silly. |
| 21:58.40 | captrb | Chuji: I'm no expert, just did it a couple days ago |
| 21:58.56 | captrb | Chuji: well... you could use the default username and password |
| 21:59.00 | phpboy | ;< |
| 21:59.25 | captrb | Chuji: if it is not a secret. otherwise any dhcp client on your network would have automatic read/write access to your config server |
| 21:59.54 | captrb | Chuji: not that ftp is very secure anyway, but still... |
| 22:00.19 | captrb | Chuji: does that help at all? |
| 22:00.26 | eKo1 | AgiNamu: domestic from where? |
| 22:00.28 | Chuji | yeah, I understand it's not a secure practice, but it makes deploying the phones much easier |
| 22:00.31 | AgiNamu | eKo1 , USA |
| 22:00.40 | eKo1 | what state? |
| 22:00.43 | AgiNamu | I have the LATA raite sheet. Class 1 - 5 |
| 22:00.46 | captrb | Chuji: one tip: if you are using the dhcp server to send the boot server address |
| 22:00.50 | AgiNamu | all that shit. then at the bottom, it has that thing about intrastate |
| 22:00.54 | AgiNamu | i have no clue what that means. |
| 22:00.55 | captrb | Chuji: then you have to use option 150. |
| 22:01.06 | captrb | Chuji: for the standard dhcpd server from isc |
| 22:01.26 | Chuji | This would be from a ms dhcp server |
| 22:01.38 | captrb | Chuji: in the phone, you will need to change the boot server address from "option 66" to "Custom", and the alternate option number to "Option 150" |
| 22:01.56 | captrb | Chuji: then you want to use option 66 if it lets you |
| 22:02.03 | eKo1 | AgiNamu: If you terminate more than 20% of the calls terminated in your state, you get charged 0.02 $? |
| 22:02.06 | captrb | Chuji: so that you don't have to change the defaults on the phone |
| 22:02.24 | AgiNamu | wtf is my state ? :\ |
| 22:02.43 | eKo1 | beats me |
| 22:02.49 | robl^ | AgiNamu: Texas? |
| 22:02.57 | AgiNamu | i dunno. this is with a boston clec |
| 22:02.58 | AgiNamu | rnk |
| 22:03.02 | Chuji | captrb : what is option 150, I gues I'm confused |
| 22:03.23 | file[laptop] | AgiNamu: probably ANI based |
| 22:03.32 | eKo1 | So maybe it's ma? |
| 22:03.42 | captrb | Chuji: sorry. the dhcp protocol allows for addition information, basically name=value |
| 22:03.48 | AgiNamu | is this related to the DIDs I buy? |
| 22:03.51 | AgiNamu | I guess I'll call them up |
| 22:03.52 | captrb | Chuji: the name is either a stand name or a number. |
| 22:03.57 | AgiNamu | What's the different Class mean? |
| 22:04.09 | file[laptop] | call and see |
| 22:04.12 | AgiNamu | like, Class 5 is always 0.018. But class 1 is sometimes .006 |
| 22:04.19 | TedC | When doing an attended transfer using a SIP transfer, the group set on the inital call by the transferer is propagated to the transferee. Is this correct behavior? |
| 22:04.25 | captrb | Chuji: so when I say option 66 or 150, I mean that you can instruct your dhcp server to send a particular value to the client, key to the number |
| 22:05.16 | Chuji | captrb : Ok, makes sense |
| 22:05.43 | Wazb | can anyone tell me where conf file i need to configure in * when a call forward by CISCO |
| 22:06.43 | *** join/#asterisk blankman (~blankman@c-24-61-108-24.hsd1.nh.comcast.net) |
| 22:06.54 | blankman | Hey guys ... |
| 22:07.12 | captrb | I'm having no luck with 2.6.11 and HCLD. Going to try 2.4.27 |
| 22:07.14 | phpboy | well |
| 22:07.29 | phpboy | my fxo module is pluged into the 4th module port |
| 22:07.33 | phpboy | on my card |
| 22:07.43 | captrb | s/hcld/hdlc/ |
| 22:07.45 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 22:08.01 | blankman | simple question ... it looks from the code in app_voicemail.c that postgres is still supported the "old" way in cvs head ... but I can't seem to get into it after moving to the new version ... is it still supported or do we have to move ARA? |
| 22:08.18 | captrb | sethdlc-new seems to be... half done? has anybody experienced this? |
| 22:08.24 | robl^ | phpboy: put it in port 1. I had trouble if the modules weren't in order.. |
| 22:09.00 | phpboy | robl^: what kind of problems where u faced with? |
| 22:09.25 | *** join/#asterisk ivanfetch (~ifetch@gargamel.uts.du.edu) |
| 22:09.45 | robl^ | phpboy: driver issues mosty.. and they way asterisk tries to number the ports |
| 22:09.55 | phpboy | ah, I see |
| 22:10.06 | phpboy | so asterisk wouldn't start up in some cases? |
| 22:10.42 | robl^ | phpboy: it would complain that the zaptel interfaces didn't match the config file and refuse to start unless I disabled zap interfaces altogether |
| 22:10.45 | PTG123 | anyone here know how to play a gsm file and listen for them to push a button and do something if they do |
| 22:11.15 | phpboy | I think I'm going to swich over to the Asterisk OS |
| 22:11.22 | Cheng29 | phpboy.. yes |
| 22:11.23 | phpboy | it's too much of a mission in Mandrake |
| 22:11.42 | phpboy | waaay to much of a mission |
| 22:11.43 | phpboy | :< |
| 22:12.02 | DannyF | load knoppix and be up in 5 minutes |
| 22:12.14 | phpboy | ? |
| 22:12.26 | *** join/#asterisk mikeh720 (~mh720@c-24-0-113-5.hsd1.tx.comcast.net) |
| 22:12.47 | DannyF | bootable cd linux debian |
| 22:13.23 | tzanger | slackware, baby |
| 22:13.27 | DannyF | hehe |
| 22:13.28 | phpboy | I think the next distro I'll hit |
| 22:13.31 | phpboy | will be slackware |
| 22:13.38 | tzanger | phpboy: if you need any help just msg me |
| 22:13.43 | eKo1 | whatever floats your boat man |
| 22:13.44 | tzanger | I've converted probably a dozen people :-) |
| 22:14.00 | *** join/#asterisk Mark_Wales (~me@cpc3-swan1-4-0-cust224.swan.cable.ntl.com) |
| 22:14.00 | DannyF | <- debian fanatic ;) |
| 22:14.00 | phpboy | but that'll be after I've figured asterisk out on it's on OS |
| 22:14.02 | phpboy | etc |
| 22:14.08 | jakepdev | anyone ever have irq sharing issues with the zaptel cards? |
| 22:14.11 | DannyF | learning FC thou |
| 22:14.14 | robl^ | no.. Tao Linux or Debian! |
| 22:14.24 | fearnor | irq sharing issues? |
| 22:14.26 | eKo1 | centos |
| 22:14.31 | fearnor | its all simple. if you share irqs, you will have issues. |
| 22:14.33 | fearnor | done/done. |
| 22:14.40 | tzanger | The subGenius Must Have Slack |
| 22:14.46 | DannyF | *cough* |
| 22:14.51 | Mark_Wales | does anyone know if Asterisk can be installed on SLES9? |
| 22:14.51 | *** part/#asterisk ivanfetch (~ifetch@gargamel.uts.du.edu) |
| 22:15.07 | tzanger | Mark_Wales: I see no reason why it couldn't |
| 22:15.08 | jakepdev | right but - how is it possible 10 devices are sharing IRQ 5 - including the zaptel cards? |
| 22:15.13 | Mark_Wales | cheers |
| 22:15.17 | tzanger | jakepdev: it happens |
| 22:15.20 | jakepdev | that's what it says in lspci |
| 22:15.21 | eKo1 | If it's linux, then yes |
| 22:15.27 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:15.37 | Mark_Wales | just i was having issues installing it and once i asked the question i got the 'success' messasge! |
| 22:15.49 | tzanger | my home * box is also my SMB/NFS server (my media PC is netbooting) and the ethernet is sharing an IRQ iwth the TDM420P... no issues whatsoever |
| 22:16.10 | jakepdev | when I do lspci - it only shows IRQ 5 and 7 being used, why not take an open one? |
| 22:16.28 | tzanger | jakepdev: becaues your BIOS or motherboard is unable to allocate them properly |
| 22:16.32 | fearnor | jake: stop being ghetto and get a motherboard with IO-APIC |
| 22:16.33 | fearnor | thanksbye |
| 22:16.41 | tzanger | jakepdev: try using ACPI and/or the IO-APIC |
| 22:16.44 | eKo1 | hehe |
| 22:16.47 | jakepdev | farnor - this is a $3k server - let it go |
| 22:16.51 | TedC | actually, looking at the group thing with SIP transfers more, I'm actually seeing a channel with more than one group. |
| 22:17.10 | tzanger | fearnor: that doesn't necessarily help, some mobos just put every INTA on the same physical line and not even an IO-APIC can help you there |
| 22:17.13 | fearnor | jake: unwisely spent money. even 50$ motherboards manufactured in 2004 all have io-apic |
| 22:17.18 | fearnor | tzanger: true true true |
| 22:17.36 | tzanger | jakepdev: try ACPI, try compiling the IO-APIC support into the kernel, see what happens |
| 22:17.40 | fearnor | jake: try jiggering around cards |
| 22:17.44 | fearnor | see if that helps |
| 22:17.47 | jakepdev | only got 2 slots |
| 22:17.57 | fearnor | ugh |
| 22:18.00 | fearnor | what kinda mobo |
| 22:18.13 | eKo1 | a 2 slot 3K$ server?! |
| 22:18.27 | jakepdev | it's an HP PROLIANT DL360R04 G4 1U |
| 22:18.31 | fearnor | ekol: its hard to put >2 slots into a single U :) |
| 22:18.34 | tzanger | eKo1: 1U |
| 22:18.36 | tzanger | or 2U |
| 22:18.37 | fearnor | one may say, impossible ;) |
| 22:18.40 | eKo1 | oh, i see |
| 22:19.14 | eKo1 | that reminds me, i need an array controller from my compaq proliant storage system |
| 22:20.10 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 22:20.13 | shmaltz | anybody here knows how to coinfigure astpp? |
| 22:20.19 | tzanger | what is astpp |
| 22:20.25 | tzanger | tinkle technology :-) |
| 22:20.45 | *** part/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
| 22:21.01 | eKo1 | that's app_piss |
| 22:22.10 | Wazb | i am getting chan_h323.c:1130 setup_incoming_Call error , any help! |
| 22:22.24 | shmaltz | tzanger, asterisk Post Paid billing |
| 22:22.29 | tzanger | ahh |
| 22:23.19 | shido6 | _SMP_ your NuFone account is setup, thank you. |
| 22:23.47 | AgiNamu | How do you know it's not Pre Paid billing. |
| 22:23.52 | AgiNamu | s/./? |
| 22:24.27 | shmaltz | AgiNamu, b/c once you set up the first part and you go the the URL to conifgure it it says Post Paid |
| 22:24.36 | robl^ | shido6: hey, is it possible to setup a toll free fail-over for my OLD OLD NuFone account? My DIDs aren't even listed in the control panel. |
| 22:24.43 | shmaltz | ~astpp |
| 22:25.07 | tzanger | shido6: reminds me of the old alt.sex.passwords IRC channels of yore |
| 22:25.18 | tzanger | where the crackers would send similar messages |
| 22:25.27 | tzanger | Password sent to: shido6 |
| 22:25.49 | shmaltz | tzanger, you used to hang out on those channels ;) |
| 22:25.51 | robl^ | cheap ho sent to: tzanger |
| 22:26.00 | tzanger | robl^: no I take the expensive ones |
| 22:26.07 | tzanger | I have particular tastes |
| 22:26.19 | tzanger | shmaltz: yeah your nick looks familliar from there too |
| 22:26.28 | shmaltz | :) |
| 22:26.38 | shmaltz | you know what shmaltz means? |
| 22:26.48 | tzanger | nope |
| 22:27.06 | robl^ | tzanger: the woman in the leather dominatrix and Teletubbie outfit?? "Uh oh! Bad boy! Again! Again!" ?? |
| 22:27.20 | eKo1 | shmaltz is german right? |
| 22:27.20 | tzanger | robl^: hahaha |
| 22:27.31 | shmaltz | eKo1, nope, yiddish |
| 22:27.35 | shmaltz | ~yiddish |
| 22:27.36 | jbot | i guess yiddish is The language historically of Ashkenazic Jews of Central and Eastern Europe, resulting from a fusion of elements derived principally from medieval German dialects and secondarily from Hebrew and Aramaic, various Slavic languages, and Old French and Old Italian. |
| 22:27.47 | robl^ | ~burp |
| 22:27.48 | jbot | ACTION burps loudly |
| 22:28.04 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 22:28.08 | eKo1 | wasn't there a translator bot around? |
| 22:28.14 | shmaltz | http://dictionary.reference.com/search?q=shmaltz |
| 22:28.32 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
| 22:28.42 | tzanger | jbot, translate ich bin ein berliner from de to en |
| 22:28.53 | tzanger | hmm |
| 22:29.09 | tzanger | jbot, tr "good evening" from de en |
| 22:29.11 | cp5 | anyone know under what conditions i would get warnings in asterisk: "No D-channels available! Using Primary on channel anyway" |
| 22:29.17 | shmaltz | number 2 at this link is the intended defenition: |
| 22:29.18 | shmaltz | http://dictionary.reference.com/search?q=schmaltz |
| 22:29.24 | tzanger | cp5: it means the D channel's not up yet |
| 22:29.28 | eKo1 | jbot, translate de en scheisse |
| 22:29.47 | eKo1 | jbot, translate yi en schmaltz |
| 22:29.49 | RoyK | jbot: please lart eKo1 |
| 22:29.50 | shmaltz | ~ translate shmaltz |
| 22:30.04 | RoyK | jbot: lart eKo1 |
| 22:30.06 | tzanger | jbot, translate de en ich bin ein berliner |
| 22:30.15 | tzanger | I thought it could translate |
| 22:30.25 | tzanger | jbot, translate en fr polly waddle doodle |
| 22:30.27 | eKo1 | more like transliterate |
| 22:30.33 | tzanger | hahahaha |
| 22:30.38 | tzanger | HAHAHAHA |
| 22:30.41 | RoyK | jbot: tell tzanger to fuck off, please |
| 22:30.52 | tzanger | ~lart royk |
| 22:30.57 | tzanger | a moo? |
| 22:31.00 | tzanger | jesus |
| 22:31.05 | shmaltz | jbot, translate shalom |
| 22:31.28 | tzanger | he's certainly on your side tonight |
| 22:31.39 | RoyK | :) |
| 22:32.01 | cp5 | tzanger, the D channel is up. it happens in the middle of the day, a few lines are in use, and bam, that comes up and all lines are dropped |
| 22:32.20 | tzanger | cp5: has it always done this? |
| 22:32.32 | cp5 | not always, not sure if anything has changed |
| 22:32.34 | RoyK | jbot: translate Skål |
| 22:32.46 | cp5 | using a quad T1 card, two of the ports are CPE, two are NET. they all produce this problem |
| 22:32.51 | DannyF | cheers |
| 22:33.18 | tzanger | oh wow 2bct is in zaptel |
| 22:33.35 | *** join/#asterisk ZX81_Laptop (~ZX81@222-153-115-253.jetstream.xtra.co.nz) |
| 22:33.57 | AgiNamu | tzanger, does that mean everyone is now alloed to execute you? |
| 22:33.59 | tzanger | cp5: for shits and giggles, add "resetinterval=never" to zaptel.conf and restart |
| 22:34.12 | tzanger | AgiNamu: yeah I'm easy |
| 22:34.14 | tzanger | everyone has access |
| 22:34.16 | cp5 | what's that do? |
| 22:34.20 | AgiNamu | voip-ho |
| 22:34.32 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 22:34.51 | tzanger | cp5: zaptel will periodically (normally once an hour) restart any free B channels |
| 22:34.55 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 22:34.56 | tzanger | I wonder if you're having issues with that |
| 22:35.02 | cp5 | hmm ok |
| 22:35.15 | tzanger | since you can't give me more detailled data on what's happening this is just a shot in the dark |
| 22:35.34 | tzanger | nothing odd in dmesg or from zttool? no missed IRQs or anything? |
| 22:36.12 | bimmerd00d | What are the dependencies for Debian? |
| 22:36.26 | bimmerd00d | asterisk on debian, sorry |
| 22:36.52 | tzanger | debian is a distro, it has no other dependencies, save for your daily worship of RMS and to prefix GNU/ before every fourth word except on tuesdays, where it's every other word. |
| 22:37.12 | _SMP_ | hehe |
| 22:39.00 | *** part/#asterisk moy (~kvirc@201.135.98.129) |
| 22:39.32 | blankman | Hey guys ... is there away to turn more debugging on for the app_voicemail when trying to use postgres for the back end? It was working, after checking out the latest and greatest it can't find any of the accounts. Is there a way to see if it is still connecting? The CDR is connecting ... but not voicemail for some reason... |
| 22:39.46 | tzanger | blankman: turn up debug on postgres |
| 22:40.09 | RoyK | ~seen coppice |
| 22:40.14 | jbot | coppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 10d 13h 32m 32s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'. |
| 22:40.24 | blankman | tzanger, true ... I will see if I can figuer out how to do that ... any suggestions? |
| 22:40.28 | *** join/#asterisk jeffik (~jeffik@69.158.42.88) |
| 22:40.38 | tzanger | blankman: it's in postgresql.conf, pretty straightforward |
| 22:41.01 | blankman | tzanger ... k I will look it up thanks. |
| 22:43.51 | ZX81_Laptop | ~ping |
| 22:43.52 | jbot | pong |
| 22:44.13 | shmaltz | ~ ping www.yahoo.com |
| 22:44.15 | jbot | pong www.yahoo.com |
| 22:44.54 | RoyK | ~ping jbot |
| 22:44.56 | jbot | pong jbot |
| 22:45.11 | shmaltz | ~sex |
| 22:45.13 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 22:45.19 | shmaltz | ~pregnant |
| 22:45.21 | jbot | Yes, shmaltz, and it's your child. |
| 22:45.29 | shmaltz | ~age |
| 22:45.30 | jbot | I'm just born. |
| 22:45.33 | shmaltz | ~old |
| 22:45.34 | jbot | There are three ways of knowing you're getting really old: One is memory loss . . . . and I've forgotten the other two. |
| 22:45.48 | shmaltz | ~time |
| 22:45.49 | jbot | methinks time is 1 dimensional, or everlasting |
| 22:45.53 | shmaltz | ~date |
| 22:45.54 | jbot | Thu Apr 7 22:45:54 2005 |
| 22:46.02 | shmaltz | ~drink |
| 22:46.03 | jbot | ACTION chugs a big pitcher of ice-cold Kool-Aid |
| 22:46.12 | shmaltz | ~eat |
| 22:46.18 | shmaltz | ~fun |
| 22:46.19 | jbot | ACTION rolls on the floor, laughing |
| 22:46.30 | Chuji | ~botabuse |
| 22:46.31 | jbot | Stop tormenting me! |
| 22:46.32 | RoyK | ~lart shmaltz |
| 22:46.43 | Chuji | ~botabuse |
| 22:46.44 | jbot | Stop tormenting me! |
| 22:47.06 | shmaltz | *ouch* |
| 22:47.08 | L|NUX | can some one tell me good documentation about asterisk + video confrenceing ? |
| 22:47.22 | Chuji | ~rtfw |
| 22:47.23 | jbot | i guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 22:47.41 | L|NUX | well i did |
| 22:47.47 | L|NUX | but man there is not much info |
| 22:47.54 | shmaltz | L|NX, I'm not sure what you are looking for, if you cant find it on the wiki then add it to the wiki |
| 22:48.18 | Chuji | L|NUX : some stuff in the mailing list archives |
| 22:48.20 | L|NUX | well i am looking for installing and configuring video codecs |
| 22:48.49 | shmaltz | ~ google video confference site:voip-info.org |
| 22:49.07 | L|NUX | Chuji : just people says i did this that :) |
| 22:49.12 | L|NUX | i search alot on google :( |
| 22:49.25 | shmaltz | ~ google video conference site:voip-info.org |
| 22:49.27 | BuckRogers | hey there is a petition online to get iax on sipura boxes go to http://www.petitiononline.com/IAXPhone/ to make your voice heard |
| 22:49.42 | Chuji | ~google "video conference site:voip-info.org" |
| 22:49.43 | shmaltz | ~ google video conference site:lists.digium.com |
| 22:50.09 | Chuji | ~google "video conference site:lists.digium.com" |
| 22:50.15 | shmaltz | ~google video conference site:lists.digium.com |
| 22:50.52 | ZX81_Laptop | ~google |
| 22:50.53 | jbot | google is probably a search engine found at http://www.google.com/ |
| 22:50.53 | Chuji | L|NUX : It's not advanced much, you aren't going to find a whole lot about it out there |
| 22:51.01 | L|NUX | hmmm |
| 22:51.12 | ZX81_Laptop | ~google ;ls -al |
| 22:51.14 | ZX81_Laptop | :) |
| 22:51.33 | Chuji | ~google miserable failure |
| 22:51.35 | ZX81_Laptop | ~google %00hehe |
| 22:51.46 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 22:51.51 | ZX81_Laptop | ahaha whitehouse is a miserable failure |
| 22:51.53 | ZX81_Laptop | :) |
| 22:52.00 | Chuji | gw's bio |
| 22:52.04 | ZX81_Laptop | lol |
| 22:52.06 | ZX81_Laptop | so funny |
| 22:52.24 | RoyK | miserable failure is http://www.whitehouse.gov/president/gwbbio.html |
| 22:52.39 | RoyK | still |
| 22:52.53 | ZX81_Laptop | how? |
| 22:53.03 | RoyK | google for the phrase |
| 22:53.29 | RoyK | it's so true :) |
| 22:55.30 | Chuji | ~google "google bombs" |
| 22:56.54 | Chuji | Matt, you need to get Kevin's bio now that he's a digium dood |
| 22:57.07 | Chuji | You haven't put out any bios in a long time |
| 22:57.34 | Chuji | Or get Eric's |
| 22:58.19 | tzanger | manxpower works for digium? |
| 22:58.24 | Chuji | no |
| 22:58.42 | ZX81_Laptop | Yah I know |
| 22:58.43 | Chuji | but he's a regular face on here and mailing lists |
| 22:58.48 | ZX81_Laptop | been real busy |
| 22:58.51 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&cid=562&ncid=738&e=1&u=/ap/20050407/ap_on_hi_te/aol_internet_phone |
| 22:58.53 | Chuji | well, so are you tzanger |
| 22:58.59 | ZX81_Laptop | coming up to the end of this project |
| 22:59.02 | shmaltz | another thing I know will *not* work |
| 22:59.11 | tzanger | I'm a what? |
| 22:59.16 | Chuji | regular face |
| 22:59.33 | Chuji | you post a lot of replies on -users right? |
| 22:59.39 | tzanger | yeah |
| 22:59.41 | Chuji | tzanger = andrew right? |
| 22:59.45 | tzanger | and fuel a few flame fests too |
| 22:59.47 | tzanger | yeah |
| 23:00.04 | Chuji | haha |
| 23:00.06 | AgiNamu | andrew kolh...? |
| 23:00.08 | Chuji | you're no critch |
| 23:00.14 | tzanger | yes akohlsmith-asterisk@benshaw.com |
| 23:00.28 | tzanger | hahaha |
| 23:00.35 | tzanger | he's been really quiet lately |
| 23:00.42 | tzanger | I wonder if he's gonna blow soon |
| 23:00.47 | Chuji | haha |
| 23:00.55 | Chuji | he lives close to me |
| 23:00.55 | AgiNamu | lol |
| 23:01.07 | Chuji | I see him on our Linux user group a lot |
| 23:01.17 | blankman | hey don't suppose anybody knows when app_voicemail.c was changed to not support postgres do you? I need to go back to a version that has postgres support in it but the cvs logs aren't helping much :-) |
| 23:01.17 | tzanger | oh wow |
| 23:01.30 | tzanger | blankman: did it ever support PG out of the box? |
| 23:01.34 | tzanger | I thought that was an addon |
| 23:01.36 | Chuji | he's much more tame on our lug |
| 23:01.57 | *** join/#asterisk iq (~iq@65-103-166-184.omah.qwest.net) |
| 23:02.23 | AgiNamu | What is the correct IAX2 response when you get something you should not? |
| 23:02.32 | AgiNamu | like TXCNT or TXACC when there's no transfer in progress. |
| 23:02.33 | *** join/#asterisk phpboy (~sj@tbnb-165-211-45.telkomadsl.co.za) |
| 23:02.35 | blankman | tzanger, nope, you just change the make file to say use postgres ... that part is still in the make file for the apps, but the part that does the actuall "switching" is nolonger in the app_voicemail.c only the new ARA stuff use odbc ... which isn't ready yet. |
| 23:02.47 | phpboy | who suggested I use asteriskathome.iso |
| 23:02.47 | phpboy | ? |
| 23:02.48 | tzanger | hmm |
| 23:02.56 | tzanger | not me, I'm the slackware advocate |
| 23:03.08 | AgiNamu | working on adding better IAX2 support to the PA168 |
| 23:03.11 | BuckRogers | hey there is a petition online to get iax on sipura boxes go to http://www.petitiononline.com/IAXPhone/ to make your voice heard |
| 23:03.12 | Chuji | captrb : you still lurking? |
| 23:03.16 | tzanger | AgiNamu: nice! |
| 23:03.23 | AgiNamu | native transfers |
| 23:03.34 | denon | BuckRogers: yeah .. I know .. I wrote it.. <g> |
| 23:03.35 | AgiNamu | and with that in, it'll be easy to add call forwarding, attended rtansfers, etc. |
| 23:03.37 | captrb | Chuji: somewhat... fighting with PRI and hdlc :-) |
| 23:03.56 | AgiNamu | BuckRogers, screw Sipura. Support Centrality! |
| 23:03.56 | Chuji | uhhg, not on the same card are you? |
| 23:04.15 | BuckRogers | well im spreading the news denon |
| 23:04.15 | captrb | Chuji: yeah |
| 23:04.21 | denon | BuckRogers: good man .. |
| 23:04.25 | denon | today Sipura, tomorrow Cisco |
| 23:04.26 | denon | <G> |
| 23:04.28 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 23:04.28 | phpboy | I'm contimplating going to the office to try install asterisk at home |
| 23:04.36 | Chuji | captrb : did you get the directory.xml to work on your poly? |
| 23:04.38 | BuckRogers | screw cisco |
| 23:04.49 | shmaltz | anybody here knows where Micro$oft speech server comes in when it comes to PBX functions? |
| 23:04.55 | captrb | Chuji: um... I did once, but when I re-edited it, the changes didn't take. |
| 23:04.59 | BuckRogers | this is a revolution for the people |
| 23:05.00 | AgiNamu | shmaltz, it's more of an IVR system |
| 23:05.02 | captrb | Chuji: wait, I'll reboot and check. |
| 23:05.03 | AgiNamu | with awesome speech recognition |
| 23:05.07 | AgiNamu | as far as i can tell |
| 23:05.20 | shmaltz | so its just a good speech recognition program |
| 23:05.40 | captrb | Chuji: are you familiar with HDLC setup on 2.6? |
| 23:05.44 | shmaltz | how does it conncect to phones? Intel Dialogic? |
| 23:05.50 | AgiNamu | yea |
| 23:06.10 | AgiNamu | what Speech Server has going for it is a good solid system plus kick ass dev experience. |
| 23:06.15 | shmaltz | hmmmmmmm, which means that to use VoIP one would need something else as well |
| 23:06.24 | AgiNamu | if asterisk had a managed system for developing...... |
| 23:06.40 | bjohnson | then MS would copy it, change it, and claim it |
| 23:06.54 | shmaltz | bjohsnon, A++++++++++++++ |
| 23:07.34 | AgiNamu | bjohson, no, im afraid ASterisk is quite far behind as far as dev experience goes. |
| 23:08.08 | shmaltz | AgiNamu, its quite possible, but at the moment none of us needs M$ to claim it |
| 23:08.23 | AgiNamu | MS doesn't need to claim it. |
| 23:08.33 | AgiNamu | it's not like Asterisk is gonna publish an RFC on stuff |
| 23:08.46 | shmaltz | but they would if Asterisk had a good system for dev |
| 23:08.46 | AgiNamu | AND, I doubt any contributer can design a class library as well as MS. no offense, just i dont see it at all. |
| 23:08.59 | AgiNamu | If asterisk had a nice managed library for AGI and so on? Um, no, they wouldn't. |
| 23:09.06 | AgiNamu | cause they dont have a product to make it work with |
| 23:09.12 | AgiNamu | and it doesnt look like they ever will |
| 23:09.23 | Chuji | captrb : no, I'm not at all. are you using digium or sangoma card? |
| 23:09.30 | AgiNamu | they are quite happy pushing Exchange, Live Communications and throwing SIP around, while letting other people do the back end. |
| 23:09.35 | captrb | Chuji: yeah, for some reason it isn't updating the from 000000000000-directory.xml |
| 23:09.41 | AgiNamu | i.e., at VON, the best they can do is tell you to goto one of their PBX partners. |
| 23:09.43 | AgiNamu | they don't get it. |
| 23:09.45 | captrb | Chuji: it was initially working though |
| 23:09.50 | captrb | Chuji: digium |
| 23:10.05 | captrb | te110 |
| 23:10.15 | Chuji | captrb : not to rub salt in the wound, but I hear sangoma's hdlc is superior |
| 23:10.27 | Chuji | captrb : that is more of their strength |
| 23:10.53 | captrb | Chuji: ah. well, I'm really just trying to bypass an adtran 600r that I suspect is a POS. |
| 23:11.59 | captrb | my t1 is dropping all the time. i think that it is losing sync |
| 23:12.16 | Chuji | captrb : I can't get my phone to boot the directory either |
| 23:12.30 | Chuji | captrb : it's not even trying to pick it up |
| 23:12.45 | captrb | Chuji: same. it definitely worked at least once, because my phone has entries. |
| 23:12.48 | mw` | hi |
| 23:13.15 | mw` | why cant ethereal identify the rtp pakets sent by asterisk? |
| 23:13.24 | mw` | i only get "UDP" as protocol |
| 23:14.30 | mw` | i'm trying to do some regexp for layer7-filter on rtp |
| 23:14.52 | Chuji | captrb : did you rename it to macaddress-directory.xml? |
| 23:15.22 | captrb | Chuji: no, because I only wanted one for all the phones (unless they users update on their own) |
| 23:15.28 | RoyK | ~seen coppice |
| 23:15.31 | jbot | coppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 10d 14h 7m 49s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'. |
| 23:15.35 | captrb | Chuji: but maybe I have misunderstood the functionality |
| 23:17.38 | AgiNamu | ethereal decodes iax just fine |
| 23:17.48 | AgiNamu | that's how im writing this firmware code |
| 23:17.49 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 23:17.58 | AgiNamu | off of an ethereal capture |
| 23:19.40 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 23:20.28 | mw` | aginamu: hm when i make a voip call, i get a sip session and lots of udp packets, but nothing is identified as rtp |
| 23:20.35 | AgiNamu | sip sucks :) |
| 23:20.41 | AgiNamu | i dont use sip, so sorry, can't help. |
| 23:21.07 | *** join/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 23:21.54 | mw` | ah i think i got it, when i manually select rtp |
| 23:22.04 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 23:22.28 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 23:22.53 | l-fy | morning people |
| 23:23.01 | TechDawg | evening |
| 23:24.16 | tzanger | werd |
| 23:24.53 | l-fy | hey tzanger |
| 23:25.03 | tzanger | how are you this morning, iubito |
| 23:25.24 | l-fy | very well thank you |
| 23:25.36 | l-fy | i've decided to sue Digium for intelectual property :) |
| 23:25.42 | tzanger | oh excellent |
| 23:26.08 | chap | I've decided to sue you for suing them.. |
| 23:26.22 | l-fy | chap > but i have a previous art :) |
| 23:26.23 | l-fy | anyway |
| 23:26.32 | Juxt | i have a connection to an iax provider |
| 23:26.33 | tzanger | sorry chap, I've patented that. I'm suing you for using my patent idea without a license |
| 23:26.39 | TechDawg | Anyone care to remind me what deb package I need for /usr/include/openssl/ssl.h? |
| 23:26.41 | Juxt | but when i do iax2 show channels i do not see it listed |
| 23:26.43 | Juxt | what gives |
| 23:26.46 | *** join/#asterisk lethol (~lethol@201.129.88.242) |
| 23:27.28 | shmaltz | l-fy, whats your IP problem? |
| 23:27.38 | l-fy | i've discover that in libiax there is a part of the code (something like 4 lines) that is similar with my patch from my forked libiax version :) |
| 23:27.43 | Chuji | captrb : I just logged all of it's commands over ftp it never looked for a directory file |
| 23:27.43 | l-fy | o god |
| 23:27.51 | l-fy | no one notice when i make a joke? |
| 23:28.02 | tzanger | l-fy: those of us with a sense of humor got it |
| 23:28.04 | captrb | Chuji: just did the same, but also reset the phone config |
| 23:28.22 | l-fy | :) |
| 23:28.24 | l-fy | anyway |
| 23:28.35 | l-fy | those 4 lines or something are perfect like mine |
| 23:28.50 | l-fy | but guess what, because that's the logic way of doing that job |
| 23:28.58 | captrb | l-fy: gpl humor is very subte |
| 23:28.58 | Chuji | captrb : you using 300,500,600? |
| 23:29.01 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 23:29.02 | captrb | subtle |
| 23:29.13 | lethol | can someone tell why would an * box change the iax2 port from 4596 to another (1072) |
| 23:29.15 | tzanger | subtle humour is often lost on those it's presented to |
| 23:29.16 | captrb | Chuji: 500 |
| 23:29.50 | ariel_ | Juxt, iax2 show channels will only show you a channel if it's active. iax2 show peers or iax2 show registry will let you know more about the connection. |
| 23:30.27 | Juxt | yeah the channel was active |
| 23:30.29 | Juxt | and it didn't show up |
| 23:30.34 | ariel_ | lethol, it should not it's hard coded to 4569 |
| 23:30.52 | ariel_ | Juxt, what dis show channels do by it's self. |
| 23:30.58 | ariel_ | dis/did |
| 23:31.03 | tzanger | ariel_: no it's not... bindaddr and bindport is configurable |
| 23:31.10 | shmaltz | why do the sppamers think that I am not confident, and are trying to sell me Viagra? are these spammers female? |
| 23:31.16 | Juxt | nothing just epty |
| 23:31.26 | bjohnson | kissmyasterisk |
| 23:31.32 | Juxt | epty=empty |
| 23:31.43 | ariel_ | tzanger, not in stable it's not I was just looking at the code the iax was configurable. |
| 23:31.59 | Chuji | captrb : Ok, got it to load with mac-directory.xml |
| 23:32.04 | tzanger | ahh stable |
| 23:32.17 | captrb | Chuji: yeah, but that isn't what I want... |
| 23:32.36 | captrb | Chuji: maybe the phone knows it already "seeded" the directory |
| 23:33.53 | *** part/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 23:33.56 | lethol | ariel_, this is the only box doing this.. iax2 show reg will show Host as xxx.xxx.xxx.xxx:4569 and Perceived as xxx.xxx.xxx.xxx:1072 |
| 23:34.46 | lethol | all my other reg's will show OK to other * boxes |
| 23:38.36 | ariel_ | lethol, what is the service your showing this too? |
| 23:40.23 | lethol | ariel, its a box to box iax trunk.. all voip provider trunks Ive tried/used work fine |
| 23:40.58 | TechDawg | What's the problem here: /usr/bin/ld: cannot find -lssl |
| 23:41.20 | *** join/#asterisk Entegrity (~Entegrity@c-65-96-119-254.hsd1.ma.comcast.net) |
| 23:41.27 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:41.28 | TechDawg | I'm trying to compile asterisk-1.0.7 |
| 23:41.32 | tzanger | TechDawg: install openssl |
| 23:41.39 | ariel_ | argh I keep having network problems today. |
| 23:41.43 | Entegrity | anyone want to give me some advice? |
| 23:41.51 | TechDawg | But it is installed tzanger |
| 23:41.58 | tzanger | TechDawg: then install the -devel part of it |
| 23:41.59 | lethol | ariel_, thought u had left |
| 23:41.59 | Chuji | ~advice |
| 23:42.00 | jbot | hmm... advice is something for which you must pay attention. Many people get irritated if they have to repeat themselves |
| 23:42.04 | ariel_ | Entegrity, ask |
| 23:42.07 | tzanger | Entegrity: get a haircut, and get a new job |
| 23:42.22 | TechDawg | I cannot find the devel part in the deb package system. |
| 23:42.25 | ariel_ | lethol, my network keeps dropping off today. |
| 23:42.26 | Entegrity | Need asterisk for a SIP proxy w/ CCM integration. Using it to just proxy off to vonage. |
| 23:42.30 | tzanger | TechDawg: then don't use deb. :-) |
| 23:42.31 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 23:42.31 | Entegrity | What do I need for a system specs? |
| 23:42.37 | TechDawg | LOL |
| 23:42.37 | Entegrity | 3-4 phones... |
| 23:42.39 | Entegrity | nothing fancy. |
| 23:42.44 | TechDawg | not the answer I was looking for. |
| 23:42.45 | tzanger | Entegrity: an old P3 will suffice |
| 23:42.59 | Entegrity | with? ;P |
| 23:43.00 | tzanger | TechDawg: I dont' run debian, and it's obvious you don't have libssl |
| 23:43.02 | ariel_ | vonage arhg |
| 23:43.17 | TechDawg | BINGO, thanks tzanger |
| 23:43.32 | Entegrity | whats the best linux dist for asterisk? |
| 23:43.40 | tzanger | Entegrity: there is no best |
| 23:43.42 | ariel_ | Entegrity, if your just using it to connect 4 to 5 lines not much really. |
| 23:43.49 | Chuji | Entegrity : what are you accustomed to? |
| 23:43.51 | tzanger | I like slackware, others like FC2 or gentoo or debian or LFS or whatever |
| 23:43.55 | Entegrity | umm |
| 23:43.55 | ariel_ | CentOS 3.4 for me |
| 23:43.56 | Chuji | ~distro |
| 23:43.57 | jbot | well, distro is significa distribution (english) |
| 23:44.01 | Entegrity | red hat I guess |
| 23:44.03 | Entegrity | but I dont care which |
| 23:44.10 | lethol | ariel_, do u know if there is such thing as a limit to connections going thru iax/4569 |
| 23:44.11 | Chuji | Entegrity : then run FC3 |
| 23:44.11 | Entegrity | whatever is the most stable |
| 23:44.13 | Entegrity | etc |
| 23:44.22 | Chuji | Entegrity : Then run debian :) |
| 23:44.23 | ariel_ | Entegrity, CentOS is RHEL 3 |
| 23:44.47 | ariel_ | lethol, setgroup and checkgroup count |
| 23:44.53 | Entegrity | I'm not linux guru |
| 23:44.55 | Entegrity | just a hacker |
| 23:45.03 | Entegrity | so I'll go with debian I guess |
| 23:45.13 | Chuji | ariel_ : is centos 4 eq rhel4? |
| 23:45.22 | ariel_ | Chuji, yes |
| 23:45.25 | Entegrity | I have no clue what centos 4 rhel4 is |
| 23:45.26 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 23:45.28 | Entegrity | lol |
| 23:45.33 | ariel_ | but 3.4 is far more stable for now. |
| 23:45.35 | tzanger | Entegrity: me either |
| 23:45.39 | lethol | ariel_, can that have something to do with asterisk mving my 4569 port connection to 1072? |
| 23:45.42 | TechDawg | Doesn't really matter on the distro. Whatever works and can be secured easiest. |
| 23:45.50 | Chuji | centos is the Redhat clone |
| 23:45.54 | ariel_ | Entegrity, RedHat Enterprise |
| 23:45.55 | Entegrity | oh ok |
| 23:45.59 | Chuji | rhel = redhat enterprise linux |
| 23:46.09 | ariel_ | lethol, no |
| 23:46.12 | jakepdev | ~yellow alarm |
| 23:46.21 | ariel_ | jakepdev, problems |
| 23:46.28 | Entegrity | why go w/ enterprise? |
| 23:46.31 | Entegrity | is it more secure? |
| 23:46.32 | jakepdev | how bout rec? |
| 23:46.38 | jakepdev | no reds |
| 23:46.44 | ariel_ | Entegrity, works just plain works. |
| 23:46.47 | tzanger | ~yai |
| 23:46.57 | Entegrity | cool thx for the adviced |
| 23:47.03 | Entegrity | install is easy as well? |
| 23:47.05 | Chuji | Entegrity : well, redhat would say so you get a supported distro |
| 23:47.05 | tzanger | hmm jbot's on a smoke break |
| 23:47.20 | tzanger | or maybe his server's smoking |
| 23:47.20 | jakepdev | still tryin to figure out my query |
| 23:47.22 | jakepdev | :) |
| 23:47.26 | Entegrity | so lets see |
| 23:47.37 | Entegrity | p3 512mb ram? |
| 23:47.41 | Juxt | can someone explain this: channel.c:1833 set_format: Unable to find a path from g729 to slin ? |
| 23:47.43 | ariel_ | Red Hat is now a paid linux so there are some clones that are good. CentOS, White Box and Tao. I like CentOS due to it's yum servers are for me faster |
| 23:47.47 | jakepdev | ~server smoking |
| 23:47.54 | ariel_ | Entegrity, just fine |
| 23:47.57 | tzanger | Juxt: have you paid for a g729 license? |
| 23:48.14 | Entegrity | k |
| 23:48.38 | jakepdev | if it just goes into yellow occasionally, but rec and green again - is it still no good? |
| 23:49.07 | Juxt | tzanger: i'm using the dev one |
| 23:49.12 | *** join/#asterisk t3sture_ (~t3sture@user-24-214-152-32.knology.net) |
| 23:49.13 | tzanger | Juxt: what 'dev' one |
| 23:49.21 | tzanger | there is no 'dev' g729 codec for asterisk |
| 23:49.33 | ariel_ | jakepdev, it's not good it should stay OK all the time. |
| 23:49.35 | Juxt | http://www.voiceage.com/freeimplement.html |
| 23:49.36 | *** join/#asterisk bah (048830696@ACABCEEB.ipt.aol.com) |
| 23:49.42 | Juxt | oh |
| 23:49.53 | Juxt | i am just trying to get asterisk to pass thru g729 |
| 23:50.11 | tzanger | Juxt: well it'sobviously not passing through, it's trying to convert to slinear (probably because an endpoint is a zaptel device) |
| 23:50.16 | jakepdev | ariel - have you seen it do the yellow once in a while - what ended up to be the problem? |
| 23:50.44 | ariel_ | jakepdev, problem with timing sync and other reasons. But all of them not good. |
| 23:50.55 | Juxt | <PROTECTED> |
| 23:51.02 | jakepdev | k tnx |
| 23:51.14 | captrb | <PROTECTED> |
| 23:51.16 | tzanger | Juxt: does your Dial() command have tT or any othe rflags that would make asterisk have to 'listen in' on the audio stream? |
| 23:51.18 | captrb | or am I special? |
| 23:51.20 | ta[i]nted | Juxt do u have conf right? |
| 23:51.22 | ariel_ | Juxt, if your using sip. make sure you have canreinvite=yes |
| 23:51.35 | tzanger | captrb: you're special alright |
| 23:51.43 | Juxt | i am using 2 firefly phones in iax2 mode |
| 23:51.46 | tzanger | I've had a zaptel module oops once but that was once in a full year of operation |
| 23:52.17 | captrb | ztcfg -s or modprobe -r reliable hang the system |
| 23:52.39 | captrb | sysreq is no use at that point, totally hung |
| 23:53.21 | ariel_ | captrb, you have something configured wrong then. |
| 23:54.20 | captrb | ariel_: i really doubt that a misconfig would cause the kernel to hang so severly |
| 23:54.47 | ariel_ | captrb, how did you do the setup? |
| 23:55.35 | captrb | I co'd code from CVS, I compiled a kernel, compile the modules, and rebooted |
| 23:55.44 | captrb | with pretty standard /etc/zaptel.conf |
| 23:56.01 | Juxt | what is a reasonable codec t use, i am getting tired of trying to make g.729 work |
| 23:56.19 | ariel_ | so you did zaptel make clean , make, make install then you did asterisk make clean, make and make install |
| 23:56.28 | ta[i]nted | Juxt why don't u just purchase a license from digium |
| 23:56.40 | ariel_ | Juxt, what is available on your sip phones? |
| 23:56.50 | captrb | ariel_: hrm. guess I didn't recompile asterisk when I switched kernels. |
| 23:56.55 | Juxt | well i'm still in the testing mode, i haven't purchased any phones yet |
| 23:56.58 | TechDawg | Okay, had to do a make clean but now I get: termcap support not found |
| 23:57.06 | Juxt | g.729 seems promissing |
| 23:57.08 | captrb | but asterisk isn't even running |
| 23:57.41 | ariel_ | Juxt, ok get your self xlite use gsm or ulaw for the testing. they work great. |
| 23:57.51 | ta[i]nted | Juxt then use the 'open source' implementation of g729 |
| 23:58.13 | *** join/#asterisk Arthemys (~arthemys@dsl39.barrvtel.sover.net) |
| 23:58.25 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 23:58.26 | Juxt | i already have all the codecs working well, except for g729 |
| 23:58.26 | ta[i]nted | Juxt and when u like it, buy licenses from digium |
| 23:59.19 | Juxt | well my thinking is... if i want to use asterisk just to pass thru g.729 why would i need to install the codec on it? |
