| 00:00.15 | *** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-56.hsd1.ma.comcast.net) |
| 00:01.16 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 00:02.50 | Dr-Linux | anybody is using digium 24 ports FXS card? |
| 00:13.04 | surfdue | Strom_C, what do you mean? |
| 00:15.30 | *** join/#asterisk nain (i=nain@137.101.145.50) |
| 00:15.43 | nain | Hi |
| 00:16.36 | nain | How can i set different sip port for 2 users, for example, 8060 for 1st user and 5060 for rest of all users by default ? |
| 00:29.44 | SplasPood | What are the laws regarding LNP? Can a company refuse to let you port your numbers, presuming you have no outstanding balance with them? |
| 00:30.03 | *** join/#asterisk domingues_br (n=hotbuttl@200.99.149.139) |
| 00:30.10 | domingues_br | Hello ALl |
| 00:30.16 | knarfly | :( - I can't seem to get the conference room going...can anyone offer advice? |
| 00:32.11 | domingues_br | Hi I am trying to use Hylafax to send fax over IAXMODEM, but every fax that I send I get timeout after asterisk dial and connect to the receive fax machine, any idea? |
| 00:34.26 | nain | How can i set different sip port for 2 users, for example, 8060 for 1st user and 5060 for rest of all users by default ? |
| 00:35.27 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:36.55 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 00:37.33 | knarfly | Okay, I guess this is where the ztdummy comes in...but I run FreeBSD with no zaptel hardware. How can I run the ztdummy driver? |
| 00:40.04 | domingues_br | does anyaboy know a solution to send fax in asterisk? |
| 00:43.32 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:46.03 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 00:46.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.226.157) |
| 01:11.34 | *** join/#asterisk hfb (n=hfb@adsl-75-4-220-175.dsl.irvnca.sbcglobal.net) |
| 01:15.09 | xheliox | Sure is quiet in here. |
| 01:15.31 | *** join/#asterisk quadrata (n=jhuntwor@ool-44c61ecb.dyn.optonline.net) |
| 01:19.04 | surfdue | hi |
| 01:19.15 | surfdue | anyone here? |
| 01:19.19 | surfdue | i cant get my pap2 to ring |
| 01:19.27 | surfdue | even on notify |
| 01:20.28 | tenlet | hey |
| 01:20.30 | tenlet | sorry im a newb lol |
| 01:20.41 | tenlet | do you happen to know anything about digital cable though? |
| 01:20.58 | *** join/#asterisk kcio (n=cassio@c9117290.rjo.virtua.com.br) |
| 01:22.25 | *** part/#asterisk kcio (n=cassio@c9117290.rjo.virtua.com.br) |
| 01:22.38 | surfdue | yes |
| 01:23.05 | tenlet | ah really.. |
| 01:23.30 | tenlet | i always wondered about each step from the provider-customer |
| 01:24.09 | tenlet | lets say cable company A, sends a signal to subscriber A, it goes from the cable company-? (a box outside the house? regional box?)-receiver (Digital television) to the TV? |
| 01:28.55 | *** part/#asterisk domingues_br (n=hotbuttl@200.99.149.139) |
| 01:32.07 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 01:36.03 | *** join/#asterisk drbrown (n=chatzill@host-216-153-226-30.day.choiceone.net) |
| 01:37.01 | *** join/#asterisk [LiFE] (n=LiFE@Toronto-HSE-ppp3711342.sympatico.ca) |
| 01:37.12 | *** part/#asterisk [LiFE] (n=LiFE@Toronto-HSE-ppp3711342.sympatico.ca) |
| 01:37.20 | drbrown | I am having probs understanding the syntax for the GotoIf command |
| 01:37.42 | drbrown | I am trying to evaluate a variable to determine if it is equal to a number |
| 01:39.05 | drbrown | here is what I have tried exten => s,5,GotoIf($[${TESTVAR} = 2]?cfwd,s,1) |
| 01:39.40 | drbrown | as well as exten = s,5,GotoIf($["${TESTVAR}" = "2"]?cfwd,s,1) |
| 01:40.14 | drbrown | and exten = s,5,GotoIf(["${TESTVAR}" = "2"]?cfwd,s,1) |
| 01:40.26 | drbrown | any help would be appreciated |
| 01:40.34 | *** join/#asterisk XircUser (n=zerst@66.172.109.250.nwinternet.com) |
| 01:41.20 | XircUser | anyone have experience with the ztclock application? |
| 01:42.14 | zerst | I run ztclock against my T100P card and receive 8 frame slips every 3.12 seconds. |
| 01:42.21 | zerst | This can't be a good thing. |
| 01:42.47 | FuriousGeorge | so i wsa starting to ask in here last night. im noticing that if someone calls and quickly disconnects on my analog DiDs, asterisk will sometimes fail to terminate that channel |
| 01:43.07 | FuriousGeorge | as a result my fxo eventually become terminally in use with no one on the line until finally we dont get any incomming calls, and its time to restart the service |
| 01:43.41 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 01:43.42 | FuriousGeorge | its always sad when that happens |
| 01:43.59 | Qwell[laptop] | FuriousGeorge: sad when I join? |
| 01:44.12 | nain | Hi |
| 01:44.16 | nain | How can i set different sip port for 2 users, for example, 8060 for 1st user and 5060 for rest of all users by default ? |
| 01:44.19 | surfdue | im having problems |
| 01:44.23 | surfdue | Qwell[laptop], you know my situation |
| 01:44.27 | surfdue | my pap2 wont ring |
| 01:44.44 | FuriousGeorge | Qwell[laptop]: well, that too, but i was asking about why it is that if someone calls and disconnects very quickly * sometimes doesnt terminate that channel, so it stays open indefinately until all my fxo are in use |
| 01:44.49 | FuriousGeorge | and we cant get incomming calls |
| 01:45.25 | FuriousGeorge | i get this with all of my servers, someone said they had the same problem when verizon was their telco as well |
| 01:45.58 | FuriousGeorge | someone else suggested i tirun up the gain, but then there is noticeable echo, so i had to turn it back down |
| 01:46.58 | nain | Is it possible to set different SIP port for each peer/user in sip.conf ? |
| 01:48.43 | surfdue | anyone |
| 01:48.48 | surfdue | im having problems with my pap2 |
| 01:50.38 | Bullseye_Network | i've suddenly started gettting this deadlock how can I find out what channel it is: : channel.c:787 channel_find_locked: Avoided deadlock for '0x9d056b8', 10 retries |
| 01:56.00 | Bullseye_Network | nm I got it |
| 01:56.44 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
| 01:57.53 | mindwarp | quick, stupid question (but i couldn't find the answer in the obvious places): how do i disable music on hold completely, just turn it off for the time being? |
| 02:00.54 | danp | rename your musiconhold.conf file or possibly rename the directory where your MOH files are |
| 02:01.35 | Qwell[laptop] | ~timriker |
| 02:01.43 | jbot | i heard timriker is my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 02:04.43 | Wi_Fi | hey guys |
| 02:05.13 | Wi_Fi | will the linksys pap2 work with asterisk as the main pots line |
| 02:05.29 | SarahEmm | pap2 or pap2-na? |
| 02:05.35 | SarahEmm | the unlocked one will work fine with asterisk |
| 02:05.41 | SarahEmm | what do you mean by 'main pots line'? |
| 02:06.04 | Wi_Fi | right now i have the x100p |
| 02:06.38 | Wi_Fi | zaptel |
| 02:06.43 | SarahEmm | okay... |
| 02:06.57 | SarahEmm | and? |
| 02:07.03 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 02:07.08 | Wi_Fi | will it work |
| 02:07.22 | Wi_Fi | i want to get rid of x100p |
| 02:08.40 | SarahEmm | oh.. no, the PAP2 is for plugging a phone into |
| 02:08.44 | SarahEmm | it's an FXS, you want an FXO |
| 02:09.48 | *** join/#asterisk h3x0r (n=hex@CPE-24-208-84-116.neb.res.rr.com) |
| 02:15.51 | drbrown | is there any special syntax in the gotoif command if you are using a global variable???? |
| 02:16.53 | Wi_Fi | damn SarahEmm was afraid you say that |
| 02:17.01 | Wi_Fi | so i need sipura 3000 |
| 02:18.37 | SarahEmm | yep |
| 02:18.41 | SarahEmm | that'll give you 1FXS 1FXO |
| 02:18.52 | SarahEmm | why are you switching, if i may as? |
| 02:18.53 | SarahEmm | k |
| 02:20.33 | mindwarp | danp: thanks |
| 02:23.50 | Corydon76-home | drbrown: not currently, no |
| 02:24.12 | Corydon76-home | drbrown: just be aware that if you have a channel variable of the same name, it currently will mask out the global variable |
| 02:24.35 | Corydon76-home | If you have such a channel variable, there is currently no way of accessing the underlying global variable |
| 02:25.27 | Corydon76-home | In the upcoming version of Asterisk due to be released this summer, there will be a way to retrieve that global variable, exclusive of any channel variable |
| 02:25.32 | drbrown | Corydon76-home: I am not sure what my prob is, but I am having a hell of a time using the GotoIf command to evaluate a simple variable |
| 02:25.46 | Corydon76-home | Are you using the evaluation syntax? |
| 02:26.02 | Corydon76-home | What is the full expression that you're using? |
| 02:26.38 | drbrown | ,GotoIf([${TESTVAR} = 2]?cfwd,s,1) |
| 02:26.50 | drbrown | and I have tried |
| 02:26.52 | Corydon76-home | You're missing a $ before the [ |
| 02:26.55 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 02:27.00 | mbit | hey guys |
| 02:27.02 | drbrown | and I have tried ,GotoIf($[${TESTVAR} = 2]?cfwd,s,1) |
| 02:27.17 | mbit | has anyone integrated many existing pbxes with asterisk? |
| 02:27.23 | mbit | via a e1 link? |
| 02:27.42 | Corydon76-home | drbrown: is it possible for TESTVAR not to be defined? |
| 02:27.52 | mitcheloc | hey mbit how is your stuff coming along? |
| 02:28.14 | mbit | good i am helping out the poundkey project now |
| 02:28.20 | mbit | and have it integrated with freepbx |
| 02:28.23 | drbrown | I have set TESTVAR manually, and still have probs |
| 02:28.27 | mitcheloc | you are a developer on it? |
| 02:28.43 | mbit | on the freepbx part |
| 02:28.54 | Corydon76-home | drbrown: because I would additionally prefix the variable with a 0, to ensure it still works if the variable value is blank: $[0${TESTVAR} = 2]? |
| 02:29.06 | mitcheloc | last week i got to meet most of the people working on it in person :) |
| 02:29.23 | mbit | nice where at |
| 02:29.24 | drbrown | ok |
| 02:29.29 | mitcheloc | huntsville |
| 02:30.26 | mitcheloc | i have to run, catch ya later |
| 02:30.51 | mbit | cya |
| 02:31.31 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 02:31.46 | drfreeze | hi, I need parkext help |
| 02:32.54 | IOscanner | anyone remeber how to set the DID number in asterisk. I need to set the DID number if sip alert is present |
| 02:33.10 | IOscanner | I have the sip alert done I just can't remember how to set DID number |
| 02:33.42 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 02:34.24 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 02:34.24 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:35.36 | *** part/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 02:36.06 | drfreeze | Are there any known problems with call parking on Asterisk 1.2.6?? |
| 02:36.39 | drbrown | Corydon76-home: does the following syntax look correct ???? ,Set(${CFWDEXTEN}CFWD=2|g) |
| 02:38.08 | Corydon76-home | drbrown: what variable name are you trying to set? |
| 02:39.51 | drbrown | I am trying to name the variable by combining the extension and the letters cfwd |
| 02:40.09 | drbrown | is the syntax incorrect???? |
| 02:40.23 | Corydon76-home | Is there something in CFWDEXTEN ? |
| 02:40.28 | drbrown | yes |
| 02:40.59 | Corydon76-home | Are there any special characters in it, like spaces or :'s? |
| 02:41.28 | drbrown | no it is numerical |
| 02:41.35 | drbrown | it is a phone extension |
| 02:42.03 | drbrown | when running verbose it gives the impression that it is setting the variable correctly |
| 02:43.03 | drbrown | for example Setting global variable '4003CFWD' to '2 |
| 02:45.03 | Corydon76-home | I can't see anything obviously wrong |
| 02:45.17 | Corydon76-home | There's one thing that smells wrong, but it might be fine |
| 02:46.11 | Corydon76-home | I can't find anything in the source that would prevent you from starting a variable name with a digit, but it's not a wonderful idea |
| 02:46.45 | Corydon76-home | I'd swap around that the Set(CFWD${CFWDEXTEN}=2|g) |
| 02:46.46 | drbrown | I didn't think of that |
| 02:50.34 | *** mode/#asterisk [+o Corydon-w] by Corydon76-home |
| 02:52.34 | mdiehl | Can someone tell me why my mailbox can't accept any more voicemail messages after I pulled a zap card out of the server? |
| 02:54.31 | drbrown | Corydon76-home: That worked I appreciate your help |
| 02:55.53 | *** join/#asterisk evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 02:56.15 | *** part/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
| 02:57.47 | IOscanner | I have a call comming in and I want to set the DID number hwo do I do that? I thought set(DID=1234567) would work, but it doesn't |
| 02:59.23 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 03:00.40 | [TK]D-Fender | IOscanner : thats jsut a var name... what id your intention? The term DID is supposed to refer the the number the CALLER called, not something YOU set. |
| 03:01.15 | IOscanner | is it FROM_DID? |
| 03:03.00 | [TK]D-Fender | IOscanner : You aren't being any clearer. YOU don't set a DID. the DID is the number the caller called. |
| 03:03.12 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 03:03.15 | IOscanner | correct |
| 03:03.27 | IOscanner | I want to change that number when it come into asterisk |
| 03:03.36 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 03:04.16 | [TK]D-Fender | IOscanner : as in I have a PRI w/ 100 DID's (#'s pointing to it). incoming calls are ON the DID the caller called and dials an extenmatching it in my incoming context. |
| 03:04.16 | IOscanner | I know what it is set to when it come in I want to change it to be another number in the PBX. so I can route it. |
| 03:04.16 | [TK]D-Fender | IOscanner : What is your call coming in on? |
| 03:04.44 | IOscanner | broadvoice and I want to map the alter number using the alert info to change the from_DID number |
| 03:04.54 | [TK]D-Fender | IOscanner : You don't need to change it to route it. You just use whatever they send if you don't care. Then again if you have multiple DID's, whats the point of treating them all the same? |
| 03:05.16 | [TK]D-Fender | IOscanner : For outbound? |
| 03:06.17 | IOscanner | I have 2 number, but the alt looks like the primary, other then the alert info field. |
| 03:06.20 | IOscanner | no inbound |
| 03:06.31 | IOscanner | I don't care for outbound yet that is next |
| 03:07.00 | IOscanner | I have done this about a year ago, but I don't remember how |
| 03:07.05 | nain | Is it possible to set different SIP port for each peer/user in sip.conf ? |
| 03:07.13 | [TK]D-Fender | IOscanner : Ok, no idea how to break that off the header. Normally you might use 2 registrations or they might dial a different exten off your single reg. |
| 03:07.31 | [TK]D-Fender | nain : I'm betting its a single user/peer |
| 03:08.02 | tlow | IOscanner, would be much easier if you know the number |
| 03:08.12 | tlow | then you could implement something like, exten => s,1,GotoIf($["${CALLERIDNUM}" = "3138853352"]?5:2) |
| 03:08.12 | IOscanner | I have the line that finds it and goes on if it has the alert header set. I now need to set from_did, but I don't remember how |
| 03:09.26 | IOscanner | Set(FROM_DID=()) |
| 03:09.27 | IOscanner | got it |
| 03:10.04 | type0 | [TK]D-Fender.. what was the company called that uses * for a dateline implementation? |
| 03:10.07 | [TK]D-Fender | tlow : thats quite off |
| 03:10.20 | [TK]D-Fender | type0 : Can't recall at this point, been too long. |
| 03:10.26 | [TK]D-Fender | Was a local Montreal place |
| 03:10.48 | nain | <[TK]D-Fender>: Actually i am talking about 2 different user who can registered to same box with different ports, e.g: 1st user A can registered on 5060 port and B user can registered on 8060 port ? |
| 03:11.07 | [TK]D-Fender | nain : Sure |
| 03:11.31 | [TK]D-Fender | nain : Thats exactly how Sipura's do it. |
| 03:12.29 | nain | <[TK]D-Fender>: Well, How can i defined custom port for each sip user? i have tried setting bindport in each user context but unable to registered except port defined in [general] section with bindport... |
| 03:13.40 | tlow | port= |
| 03:13.49 | [TK]D-Fender | nain : bindport is only for [general] and should be left at 0.0.0.0 as most cases |
| 03:14.03 | [TK]D-Fender | nain : use "port=[theport]" in the peer entry |
| 03:14.10 | nain | tlow: what's the difference between bindport and port? |
| 03:14.28 | tlow | bindport is where you actually bind 'asterisk' to an ip or multiple ips. |
| 03:14.33 | tlow | port is for the poor. |
| 03:14.40 | [TK]D-Fender | nain : bindport is only for [general] and is used to restrict * to a specifi interface if there are more than 1 IP allocated to it. |
| 03:15.13 | *** part/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 03:16.18 | nain | well, what i understood, i should use [general] section bindport=0.0.0.0 or 5060 or even comment(might be default port will use), and for each peer or user should use port=8060 for A user, port=9060 for B user. if i am not wrong ??? |
| 03:16.51 | tlow | nods |
| 03:17.16 | [TK]D-Fender | nain : Sorry, bindport should be 5060, bindADDR should be 0.0.0.0 |
| 03:17.28 | [TK]D-Fender | nain : you new sample looks right |
| 03:17.29 | tlow | missread |
| 03:18.34 | nain | Thanks, Let me try it |
| 03:20.19 | *** part/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 03:22.11 | nain | <[TK]D-Fender>: :(, Sorry, Login Failed, Request Time out, with new configrations |
| 03:22.46 | [TK]D-Fender | nain : pastebin. |
| 03:22.52 | nain | ok |
| 03:27.00 | nain | <[TK]D-Fender>: http://pastebin.ca/84605, check the configration |
| 03:27.39 | nain | <[TK]D-Fender>: user 420 is unable to register with sip port 8060 but 421 successfully log into with 5060 port .. |
| 03:31.39 | *** join/#asterisk smackus (n=smackus@63.149.122.93) |
| 03:32.32 | nain | Any Clue ? |
| 03:32.45 | smackus | I am looking for the command to be able to silent monitor telephone calls. I think I am looking for the wrong terminology. I am used to the term monitor, and unless I am misreading the command that is used just for recording. Please direct me to the correct term. |
| 03:32.47 | [TK]D-Fender | nain : What kind of device, and where is it located relate to *? |
| 03:33.07 | [TK]D-Fender | smackus : ChanSpy |
| 03:33.11 | smackus | thank you |
| 03:35.02 | nain | <[TK]D-Fender>: end device is a Softphone Eyebeam and it is located in PK |
| 03:35.11 | IOscanner | How do I print a message to the CLI when I get to a certain part of a context |
| 03:35.58 | nain | It Only register on port defined in General section, what ever be the port is, 5060, 8060 9060 but must be general section will register successfully, |
| 03:36.04 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 03:36.08 | IOscanner | I need to do some debuging |
| 03:36.11 | [TK]D-Fender | nain : Describe the networking on both sides |
| 03:36.22 | [TK]D-Fender | IOscanner : NoOp |
| 03:37.01 | IOscanner | thanks |
| 03:38.29 | nain | <[TK]D-Fender>: Client end is connected with direct DSL Connection(No Nating) in Pakistan and Remote asterisk is colocated in Spain. |
| 03:40.00 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 03:40.20 | [TK]D-Fender | nain : * on public IP? |
| 03:41.07 | smackus | is there a way to do a show channels and make it just update instead of running show channels over and over? |
| 03:41.17 | nain | <[TK]D-Fender>: Yes * is on public IP and client is on Public IP as well. |
| 03:41.44 | [TK]D-Fender | nain : So Clien is intending on running 2 eyebeam clients on his box? |
| 03:44.04 | nain | <[TK]D-Fender>: no, client has only 1 eyebeam running, but when it try to register on port 5060 it's fine with User A, but when try to register with User B with 8060 port, "TIME OUT" |
| 03:44.53 | nain | <[TK]D-Fender>: It doesn't matter on which port client is being register, if i set the general section on 8060 port, client will register to 8060 port successfully but with port define on user context , user time out |
| 03:45.24 | nain | <[TK]D-Fender>: wait let me pastbin the debug log of * |
| 03:45.36 | [TK]D-Fender | nain : Ah, 2 reg on 1 client. Not sure how that works on it. |
| 03:45.47 | [TK]D-Fender | nain : try using closer ports. 5060/5061 |
| 03:45.59 | [TK]D-Fender | nain : should have no issue reging. |
| 03:47.09 | nain | http://pastebin.ca/84613 |
| 03:48.12 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:48.29 | [TK]D-Fender | nain : That little bit doesn't say much. |
| 03:48.57 | nain | <[TK]D-Fender>: I tried using closer ports with 5060 and 5061 but still same. |
| 03:49.10 | Wi_Fi | SarahEmm sorry |
| 03:49.17 | Wi_Fi | got distracted |
| 03:49.30 | [TK]D-Fender | nain : ok, well we'd need to see a real regist attempt fom both connections to get a better idea of whats going on. |
| 03:49.48 | ph|ber | hey.. im getting this in my logs.. is this from me or my provider.. |
| 03:49.51 | SarahEmm | Wi_Fi: it's okay |
| 03:49.52 | ph|ber | connect attempt from 72.5.55.200, request '1@from-pstn' does not exist |
| 03:50.27 | nain | <[TK]D-Fender>: wait let me provide with the real debug log from both ends. |
| 03:50.32 | Wi_Fi | SarahEmm i want to switch cause im tryin to get trixbox to run in VM |
| 03:50.45 | Qwell[laptop] | Wi_Fi: It won't work very well at all |
| 03:50.53 | SarahEmm | Wi_Fi: ahh.... |
| 03:50.55 | SarahEmm | what's trixbox? |
| 03:51.03 | SarahEmm | ahh i see |
| 03:51.06 | SarahEmm | why not Qwell[laptop]? |
| 03:51.11 | SarahEmm | i run * in VMWare Server |
| 03:51.13 | SarahEmm | no issues here |
| 03:51.19 | Qwell[laptop] | SarahEmm: not very many channels, I should hope.. |
| 03:51.20 | Wi_Fi | it should with sipura3000 |
| 03:51.29 | SarahEmm | Qwell[laptop]: it's just my home system, yeah. |
| 03:51.33 | SarahEmm | but so was Wi_Fi's |
| 03:51.59 | [TK]D-Fender | *cough* AMP |
| 03:52.20 | SarahEmm | [TK]D-Fender: what about amp? |
| 03:52.47 | pdtmobile | i run asterisk in vmware for dev and it works ok, I wouldn't do it in production |
| 03:52.55 | *** join/#asterisk bmg505 (n=leon@c1-116-9.rndf.isadsl.co.za) |
| 03:53.00 | Qwell[laptop] | I wouldn't do it for dev |
| 03:53.24 | pdtmobile | Qwell[laptop]: why? |
| 03:53.40 | Qwell[laptop] | pdtmobile: it wouldn't surprise me if odd bugs showed up/disappeared |
| 03:54.01 | Wi_Fi | Qwell[laptop] have you experienced VM |
| 03:54.22 | Wi_Fi | ESX to be exact |
| 03:54.29 | Qwell[laptop] | sure |
| 03:54.31 | pdtmobile | well the only ones I could think of would be related to hardware cards... which you can't test in VMWare anyway |
| 03:54.51 | pdtmobile | if you are using ztdummy for timing it should be pretty generic |
| 03:55.18 | pdtmobile | definitely if you are doing dev that related to any tdm cards you would have to have a real box |
| 03:55.28 | ph|ber | anyone?? |
| 03:55.50 | Qwell[laptop] | pdtmobile: ask some of the devs about the (heavily) failed attempt at running a conf call in vmware |
| 03:56.08 | pdtmobile | really? |
| 03:56.20 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 03:56.33 | Qwell[laptop] | TrixVox: /topic |
| 03:56.42 | Qwell[laptop] | or, V..nm |
| 03:57.35 | [TK]D-Fender | Qwell[laptop] : Let them incriminate themselve, THEN decapitate, ok? ;) |
| 03:57.44 | Qwell[laptop] | [TK]D-Fender: pfft |
| 03:58.00 | pdtmobile | lol |
| 03:59.48 | CunningPike | ~lart CPAN |
| 03:59.52 | pdtmobile | well, i don't have a billion boxes laying around so it'll have to do |
| 04:00.02 | pdtmobile | CunningPike: you've been fighting with cpan allthis time? |
| 04:00.12 | Corydon76-home | clue gun? |
| 04:00.21 | pdtmobile | i do test everything on real hardware before it goes anywhere though... |
| 04:00.35 | CunningPike | pdtmobile: All I want is to install perl-Curses and a few other libraries. It's like pulling teeth..... |
| 04:00.51 | CunningPike | pdtmobile: I even had to force install Bundle::CPAN, ffs |
| 04:00.52 | pdtmobile | are you running a recent version of cpan? |
| 04:00.56 | file | moo |
| 04:00.59 | nain | <[TK]D-Fender>: http://pastebin.ca/84619 (8060 port from * registered) |
| 04:01.00 | nain | http://pastebin.ca/84621 (8060 port from eyebeam registerd) |
| 04:01.14 | pdtmobile | whats it bitching about? |
| 04:02.09 | CunningPike | pdtmobile: Some gpgring problem for Bundle::CPAN, I forced that - now perl-Curses spews out a list of functions it can't find |
| 04:02.23 | pdtmobile | oh |
| 04:02.27 | pdtmobile | mkdir ~/.gnupg |
| 04:02.28 | *** join/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
| 04:02.46 | pdtmobile | it tries to check the signature of the package and you don't have a gnupg directory and it freaks out |
| 04:03.13 | *** part/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
| 04:03.43 | pdtmobile | functions or modules? |
| 04:04.20 | *** join/#asterisk techie (n=gus@voipops.net) |
| 04:04.39 | *** join/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
| 04:05.53 | livinded | running on a 1 ghz celeron with 128 megs of ram about how many channels would i be able to use with ulaw before the calls start breaking up? |
| 04:05.53 | nain | <PROTECTED> |
| 04:07.32 | CunningPike | pdtmobile: functions |
| 04:07.36 | *** join/#asterisk widge (n=widge@69-12-146-88.dsl.static.sonic.net) |
| 04:07.51 | pdtmobile | pastebin it |
| 04:07.55 | CunningPike | pdtmobile: It's a pretty bare bones CentOS, so I'm probably missing libraries somewhere (sigh) |
| 04:08.08 | widge | hi |
| 04:08.18 | CunningPike | pdtmobile: OK - one sec - I'm going to reinstall Bundle::CPAN first |
| 04:08.27 | pdtmobile | k |
| 04:08.56 | widge | when I get a call on my Zap channel with no caller ID, my sip phone shows "asterisk" as the CID |
| 04:09.11 | widge | is there an easy way to change this? |
| 04:09.45 | type0 | change it to like |
| 04:09.50 | type0 | ISUCKANDHAVENOCALLERID |
| 04:10.33 | widge | I don't see this in zapata.conf or sip.conf |
| 04:10.45 | CunningPike | pdtmobile: http://pastebin.ca/84629 |
| 04:10.47 | smackus | is anyone here using hp proliant servers? |
| 04:10.57 | CunningPike | smackus: Yup |
| 04:11.33 | smackus | ok, here is my question... I am trying to set my digium card so that it has its own irq |
| 04:11.56 | smackus | but in the bios when i try to set the irq, there is no way I can see to do such. it wants to share with aprox 5 other devices. |
| 04:11.56 | CunningPike | pdtmobile: Looks like it's not picking up my ncurses library...... |
| 04:12.18 | CunningPike | smackus: What we had to do was disable USB - that was the only way to get the PCI slot on its own IRQ |
| 04:12.59 | smackus | ok, so you are talking in the system config where you can disable the usb 2.0 controller and usb? |
| 04:12.59 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 04:13.10 | smackus | then save and reboot, and there should be no more usb irqs? |
| 04:13.15 | pdtmobile | yum -y install curses-devel && cpan -i Curses |
| 04:13.20 | pdtmobile | give that a shot |
| 04:14.05 | CunningPike | pdtmobile: OK - thanks - I appreciate the help - this _is_ actually * related - I'm trying to install HUDLite :D |
| 04:14.20 | pdtmobile | CunningPike: let me know how that goes, I haven't messed with it |
| 04:14.27 | pdtmobile | looks neat |
| 04:14.38 | CunningPike | pdtmobile: Sure will - we're trying to evaluate HUDLite against FOP |
| 04:15.22 | pdtmobile | hell I don't care if it's * related or not, you have helped one of my guys a bunch, so it's the least I can do |
| 04:16.18 | pdtmobile | I sent Waverly360 to queue hell and you saved him :) |
| 04:16.59 | smackus | CunningPike: so I did all that. It still wants to share an irq with the "HP NC7782 Gigabit Server Adapter port 2" |
| 04:17.06 | smackus | I dont know what that is, any help? |
| 04:17.29 | CunningPike | smackus: That's your NIC - bad to share with that. What model do you have? |
| 04:18.02 | smackus | dl380 |
| 04:18.08 | smackus | i disabled the nic port 2 |
| 04:18.17 | smackus | should i save and restart again? would that make it go away? |
| 04:18.21 | CunningPike | smackus: We have DL360s |
| 04:18.35 | CunningPike | smackus: Have you rebooted since you disabled USB?? |
| 04:18.37 | smackus | should be close i would think, right? |
| 04:18.40 | smackus | yes |
| 04:18.44 | smackus | not since i disabled the nic |
| 04:18.51 | smackus | lemme do that and see what happens. |
| 04:18.57 | CunningPike | smackus: OK |
| 04:20.06 | pdtmobile | CunningPike: have you had lots of problems with FOP getting confused? |
| 04:20.33 | CunningPike | pdtmobile: Haven't gotten that far yet - HUDLite is first on our list |
| 04:20.36 | pdtmobile | oh |
| 04:20.45 | smackus | that did the trick |
| 04:21.22 | CunningPike | smackus: OK- good- but try juggling the IRQs around - we have both NICs working - we need them for VLANs |
| 04:21.36 | smackus | now we will see how stable/unstable this bitch is. The netadmin built the machine and had the digium card shared with the gig nic, the raid controller and a handful of other things |
| 04:21.44 | smackus | cool, will play with the irqs a bit more |
| 04:21.54 | smackus | I may just put a pci gigabit nic in. |
| 04:23.21 | CunningPike | Who da man? pdtmobile da man, dats who |
| 04:23.41 | pdtmobile | pretty much |
| 04:27.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:28.30 | smackus | CunningPike: would you mind just double checking me on my lspci output to make sure I haven't done anything stupid? http://pastebin.ca/84637 |
| 04:28.43 | CunningPike | smackus: Sure - hang on |
| 04:29.39 | CunningPike | smackus: Looks OK - want to see mine? |
| 04:29.45 | smackus | sure, thanks |
| 04:29.48 | type0 | I am going on strike tommorow, for 24 hours, I will refuse to acknowledge the existance of IRQ's |
| 04:30.20 | smackus | sounds like a plan |
| 04:30.37 | smackus | I have not had to mess with IRQs for years |
| 04:30.42 | type0 | either have I |
| 04:30.44 | smackus | why now? |
| 04:30.54 | type0 | I just hate the fact they even are around |
| 04:31.15 | smackus | is it just the digium cards? i know a lot of people have tried to get me to by the sanoma card. is that an issue with those cards as well? |
| 04:31.50 | CunningPike | smackus: http://pastebin.ca/84640 |
| 04:32.17 | CunningPike | smackus: I don't believe so, but I think Sangomas have other installation quirks |
| 04:32.32 | pdtmobile | the sangomas do have their own issues, but irq's doesn't seem to be one of them |
| 04:32.48 | pdtmobile | and they are a hell of a lot less picky on what board you put them in, you might even call them whore cards |
| 04:33.03 | *** part/#asterisk widge (n=widge@69-12-146-88.dsl.static.sonic.net) |
| 04:33.20 | type0 | Will analog dialogic equipment work with asterisk? |
| 04:33.34 | SarahEmm | yep |
| 04:33.40 | SarahEmm | err waiat |
| 04:33.48 | SarahEmm | i misread dialogic as digium sorry :) |
| 04:33.53 | SarahEmm | ignore me. :) |
| 04:33.53 | pdtmobile | doh |
| 04:34.08 | pdtmobile | get his hopes up and then smash his head on the rocks |
| 04:34.16 | SarahEmm | sorry! |
| 04:34.20 | pdtmobile | heheh |
| 04:34.31 | SarahEmm | they may, i don't know. they support some dialogic with an unFree driver iirc |
| 04:35.04 | *** join/#asterisk setti (n=setti@68-79-112.adsl.terra.cl) |
| 04:35.16 | type0 | my head hurts. |
| 04:35.25 | type0 | don't do that again. |
| 04:35.36 | smackus | ok, so now I cannot get calls in or out now that I have changed my irqs and rebooted. |
| 04:36.34 | CunningPike | pdtmobile: You don't happen to know what to install to get perl/Tk loaded do you? It needs X11R6/include etc..... |
| 04:36.50 | pdtmobile | xorg-crap-i-forgot |
| 04:36.56 | CunningPike | pdtmobile: :D |
| 04:37.53 | pdtmobile | xorg-x11-devel ? |
| 04:38.15 | CunningPike | Worked last time lol |
| 04:39.33 | *** join/#asterisk DCGrendel (n=dcgrende@ip68-1-157-197.mc.at.cox.net) |
| 04:40.33 | CunningPike | Once again pdtmobile saves the day - keep shooting! |
| 04:51.35 | bugz | i landed an interview for 85K!!!!!!!!!! |
| 04:52.02 | pdtmobile | crap ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 04:52.36 | bugz | pdtmobile: paste /etc/zaptel.conf and /etc/asterisk/zapata.conf on pastebin.ca |
| 04:52.47 | bugz | what kernel? |
| 04:53.34 | pdtmobile | 2.6.9-34.EL |
| 04:53.58 | bugz | lets see the config files |
| 04:55.02 | pdtmobile | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 04:55.09 | pdtmobile | different now, i pulled ztdummy out |
| 04:55.24 | pdtmobile | leme pastebin zaptel |
| 04:55.46 | type0 | my interweb is slow |
| 04:56.01 | pdtmobile | http://pastebin.ca/84654 |
| 04:56.08 | DCGrendel | how do you dial from the console? |
| 04:56.29 | type0 | with /bin/finger |
| 04:56.36 | type0 | at least that's what I heard.. |
| 04:56.38 | pdtmobile | i wish somebody would pay me 85k for an interview, i'd do it allthe tie |
| 04:56.40 | pdtmobile | time |
| 04:56.43 | bugz | DCGrendel: dail <context>,<tech>/<exten> |
| 04:56.44 | pdtmobile | :) |
| 04:56.57 | DCGrendel | bugz: i get "No such command 'dial'" |
| 04:57.01 | bugz | eww |
| 04:57.05 | type0 | DCGrendel |
| 04:57.07 | pdtmobile | nice |
| 04:57.08 | Qwell[laptop] | You need chan_oss or chan_alsa loaded |
| 04:57.12 | type0 | .. /bin/finger |
| 04:57.13 | DCGrendel | i have chan_oss loaded |
| 04:57.37 | pdtmobile | ~sex |
| 04:57.39 | 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, or super extractor, http://sf.net/projects/sex/ |
| 04:57.39 | bugz | pdtmobile: this is what i do to fix this.. |
| 04:57.48 | *** join/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 04:57.56 | bugz | pdtmobile: i have the PRI card located in the last slot, located on the edge of the mobo |
| 04:58.03 | bugz | and i put the fxs card in the next |
| 04:58.12 | pdtmobile | what is the problem? |
| 04:58.21 | type0 | I wish I had a pri card |
| 04:58.30 | type0 | I'm going to install asterisk with no card at all |
| 04:58.32 | type0 | HOW COOL AM I? |
| 04:58.34 | bugz | fix your config to have the pri span bcha on 1-12 |
| 04:59.02 | bugz | make sure your cards are detected and reboot |
| 04:59.13 | dudes | <type0> - We can conclude not any cooler than anyone else who doesn't have a PRI |
| 04:59.26 | bugz | there is a condition in some cases where the zap driver will send on request for service packet only to the incoming PRI |
| 04:59.48 | bugz | if the incoming PRI end is not ready zap wont send another request |
| 04:59.58 | bugz | and it sits there waiting |
| 05:00.11 | pdtmobile | fix my config, I can't change where the b-chans are |
| 05:00.16 | bugz | make sure your cards are in order, put the config file in order too |
| 05:00.17 | bugz | ok |
| 05:00.20 | pdtmobile | they are 13-23 |
| 05:00.46 | type0 | I have a PRI |
| 05:00.50 | type0 | .. |
| 05:00.54 | type0 | I dont have a PRI card.. |
| 05:01.01 | type0 | btw, who's a really good iax provider? |
| 05:01.19 | dudes | you mean you have two groups. g1 1-12 and g2 13-23 with a dchan on 24 |
| 05:01.37 | pdtmobile | no, i mean i don't have a 1-12 |
| 05:01.45 | pdtmobile | the channels are delivered on 13-23 |
| 05:01.50 | dudes | txlink is good with iax |
| 05:02.00 | bugz | http://pastebin.ca/84658 |
| 05:02.03 | pdtmobile | now, why my provider did it that way, i duno |
| 05:02.12 | bugz | then just do that. its basically the same thing |
| 05:02.15 | pdtmobile | i think at one point this was an integrated circuit and they split out the PRI |
| 05:02.25 | bugz | make sure your cards are powered up when you load the modules, make sure the lights are on |
| 05:02.35 | bugz | make sure they are in lspci |
| 05:02.56 | bugz | my experience is to reboot |
| 05:03.06 | bugz | which sucks when making changes to a pri |
| 05:03.21 | bugz | or 16 of them hehe |
| 05:03.25 | pdtmobile | well the box isn't in production yet, bout to swap the production one with this one |
| 05:03.27 | pdtmobile | so not a biggie |
| 05:03.33 | bugz | kick it |
| 05:03.39 | dudes | are you using a crossover cable |
| 05:03.51 | bugz | yeah, theres the cable issue to |
| 05:03.59 | bugz | what does 'pri show span 1' say? |
| 05:04.15 | bugz | 'provisioned, down, active' is where im going with this |
| 05:04.18 | dudes | do you have the right signal? |
| 05:04.20 | pdtmobile | it should be in alarm state if it comes up, it's got air gap right now |
| 05:05.23 | bugz | 1,1,0 says you are using master signalling |
| 05:05.38 | bugz | 1,0,0 would let the PRI handle that |
| 05:05.59 | bugz | ive rarely used 1,1,0 |
| 05:06.11 | bugz | that has negligble effects on starting the span though |
| 05:06.32 | bugz | theyll just fight over it for a while and do all kinds of HDLC errs |
| 05:06.39 | pdtmobile | will it bitch like that when it's alarm? |
| 05:06.43 | pdtmobile | i don't ever remember seeing that |
| 05:07.06 | bugz | being in alarm means there is something wrong physically if the configs are right |
| 05:07.13 | type0 | pdt.. you need a tberd |
| 05:07.31 | pdtmobile | i have a tberd in the closet |
| 05:07.37 | type0 | lies. |
| 05:07.39 | pdtmobile | bugz, it's not connected to the telco yet |
| 05:07.47 | type0 | how did you fit a car into your closet? |
| 05:08.19 | pdtmobile | i am just trying to get the box configed before I swap the production pbx |
| 05:09.19 | pdtmobile | well hell |
| 05:09.35 | pdtmobile | i think it's time for bed |
| 05:09.45 | pdtmobile | i borked my asterisk rpm |
| 05:09.46 | type0 | I think its time for ummm |
| 05:09.54 | pdtmobile | it doesn't even know what a pri is |
| 05:10.01 | type0 | its an rpm |
| 05:10.04 | type0 | you have to introduce them |
| 05:10.09 | type0 | they get to know eachother |
| 05:10.20 | pdtmobile | ya, I forgot BuildRequires: libpri-devel |
| 05:10.27 | pdtmobile | along with the 50 others |
| 05:10.33 | pdtmobile | well i have the others, just neglected that one |
| 05:10.38 | type0 | exchange memory addresses |
| 05:10.45 | type0 | after that.. they'll know eachother.. |
| 05:11.09 | type0 | why cant I pull porn torrents at 500k/sec |
| 05:11.20 | type0 | Life would be a lot less stressful.. I can tell you that |
| 05:11.25 | type0 | EH COMRADE??! |
| 05:11.26 | type0 | EH?! |
| 05:11.39 | pdtmobile | get a fatter pipe |
| 05:11.44 | type0 | OHHHHHH |
| 05:11.52 | type0 | too bad Im pulling gentoo at 500k/sec |
| 05:11.58 | type0 | porn torrents are FAR less |
| 05:12.00 | type0 | 20k/sec |
| 05:12.14 | smackus | ok... now here is a new one. |
| 05:12.14 | pdtmobile | you need to get more people interested in porn I guess |
| 05:12.18 | pdtmobile | at least ones with bigger pipes |
| 05:12.25 | smackus | I have made a bunch of hardware changes, but no changes to the configs. |
| 05:12.33 | type0 | I think the ones with bigger pipes, are IN the porn |
| 05:12.38 | type0 | Just what I heard.. |
| 05:12.38 | smackus | I can only make about 3 calls into one of my numbers before I get a busy signal. |
| 05:12.48 | smackus | the output on the cli really is no help. |
| 05:13.05 | smackus | where should I start looking? |
| 05:13.37 | SarahEmm | nini! |
| 05:13.38 | *** part/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 05:13.42 | type0 | hmmm |
| 05:13.54 | type0 | why even say good night.. if you bail right after |
| 05:13.56 | smackus | well, actually the number of calls varies. that time it was one call |
| 05:14.16 | pdtmobile | smackus: how is that number delievered? pri, voip, pots? |
| 05:14.22 | smackus | pri |
| 05:14.34 | pdtmobile | so inbound is what fails? |
| 05:14.43 | smackus | correct |
| 05:14.53 | pdtmobile | what do you get, something about the line being in use? |
| 05:15.54 | smackus | == Spawn extension (progrexion, 3972700, 1) exited non-zero on 'SIP/122-00fb' |
| 05:15.54 | smackus | <PROTECTED> |
| 05:16.36 | pdtmobile | what do you get if you try pri debug span 1 |
| 05:16.47 | CunningPike | I have a theory that installation of _any_ Perl module requires the installation of the whole darn CPAN library...... :) |
| 05:17.04 | pdtmobile | nah |
| 05:17.15 | pdtmobile | like 20 or 30 |
| 05:17.19 | CunningPike | Heh heh |
| 05:17.26 | CunningPike | Just feels that way........ |
| 05:17.37 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 05:17.42 | mdiehl | Hi all. |
| 05:18.12 | type0 | are there any free iax providers? |
| 05:18.14 | setti | hi |
| 05:18.31 | mdiehl | Is there any guidelines for how wide to make the port range in rtp.conf? |
| 05:18.48 | pdtmobile | it's one of the biggest benefits of cpan and the biggest problems as well |
| 05:18.54 | pdtmobile | you don't have to re-invent the wheel |
| 05:19.06 | pdtmobile | but... you have to have 35 components to build the wheel |
| 05:19.25 | CunningPike | pdtmobile: Ya - I remember installing netdisco - took forever........ |
| 05:20.15 | smackus | pdtmobile: http://pastebin.ca/84665 |
| 05:20.16 | Juggie | mdiehl, two ports for every client. |
| 05:20.45 | smackus | 122 is the one that rang busy |
| 05:22.08 | mdiehl | Juggie, seems like each client can only use one at a time. What am I missing? |
| 05:22.32 | smackus | I am also getting :http://pastebin.ca/84667 |
| 05:23.12 | pdtmobile | so it's outbound calls? |
| 05:24.08 | Juggie | mdiehl, source/destination i believe, i thought it was 2. |
| 05:24.32 | pdtmobile | what did you change about the hardware? |
| 05:24.36 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 05:24.43 | mdiehl | So both the source and destination are allocated from the configured range? |
| 05:24.51 | mdiehl | Usually the source is allocated by the OS. |
| 05:25.00 | mdiehl | Well, if this is true, it makes sense. |
| 05:25.01 | smackus | well, i am using internal phones to dial, but I get it with my cell phone also. it just does not show anything in the cli if i call from my cell phone |
| 05:25.01 | Juggie | depends on the style of RTP. |
| 05:25.08 | mdiehl | Ok. |
| 05:25.17 | Juggie | symetric rtp matches ports. |
| 05:25.22 | mdiehl | Thanx for the advise. I can narrow things quite a bit. |
| 05:25.24 | Juggie | its 1 or 2 ports per client, i dont remember :) |
| 05:25.39 | Juggie | it should be 1 you would think but there may be a situation when it can be 2. |
| 05:26.07 | mdiehl | But, as long as you only support one call/line per IP address, you should only need one? |
| 05:26.26 | Juggie | destination ports cant be shared for RTP. |
| 05:26.58 | mdiehl | So, if I only had a total of 5 ports and 50 customers on different IP's, I'd be ok. |
| 05:27.19 | Juggie | no, not 'realistically' each customer needs a destination RTP port |
| 05:27.50 | mdiehl | But they can share the dest port, no? |
| 05:28.28 | Juggie | no |
| 05:28.32 | *** join/#asterisk smackus (n=smackus@63.149.122.93) |
| 05:28.46 | pdtmobile | i should go, i'm about to fall asleep and I still need to drive home |
| 05:29.10 | Juggie | http://www.voip-info.org/wiki-RTP |
| 05:29.15 | pdtmobile | caffeeeeeeeeein |
| 05:29.18 | mdiehl | Reading. |
| 05:29.21 | Juggie | in most cases that RTP range is never used anyways |
| 05:29.24 | Juggie | but thats another story. |
| 05:29.34 | Juggie | mostly what gets used is symetric RTP |
| 05:29.44 | Juggie | assuming one point is behind nat, and usually they are. |
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| 05:37.45 | CunningPike | Which module provides AMI? |
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| 05:48.32 | FuriousGeorge | why does mpg123 behave so damn poorly |
| 05:48.43 | FuriousGeorge | every now and then i catch it at like 99% |
| 05:48.58 | FuriousGeorge | cpu usage |
| 05:49.55 | dudes | bad timing |
| 05:50.06 | FuriousGeorge | huh? |
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| 05:54.25 | CunningPike | Netsplit, anyone? |
| 05:54.25 | FuriousGeorge | no thanks, just had one |
| 05:54.25 | CunningPike | :D |
| 05:54.25 | Assid | :P |
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| 06:17.43 | spooky7 | Good morning to everyone |
| 06:17.56 | spooky7 | Here in Greece is morning |
| 06:18.16 | Kernel_core | spooky7: in asia too |
| 06:18.20 | Kernel_core | good morning |
| 06:18.23 | spooky7 | Because i am new to asterisk I want some help with asterisk |
| 06:18.42 | spooky7 | Can anyone help me ? |
| 06:19.38 | kaldemar | http://www.voip-info.org is a good place to start. |
| 06:21.56 | spooky7 | I have looked there but Icouldn't find help to that I want |
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| 06:25.14 | _omer | http://pastebin.ca/84683 .... need expert suggestions please... |
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| 06:30.10 | smackus | what causes these: Jul 11 00:27:21 WARNING[5317] interface.c: Junk at the beginning of frame 00000000 |
| 06:30.30 | dudes | bad network packets maybe |
| 06:31.21 | dudes | and it's only a warning, so you can normally just ignore it. Same with notices. |
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| 06:32.51 | DCGrendel | can you use a macro inside another macro? |
| 06:32.58 | DCGrendel | (in the dialplan) |
| 06:33.12 | dudes | try it and see |
| 06:33.25 | dudes | or maybe try ael |
| 06:33.59 | DCGrendel | i actually probably don't need to call a macro from another come to think of it |
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| 06:42.15 | DCGrendel | nope, was able to do it all with AGI instead |
| 06:43.51 | _omer | DCGrendel : Do you use AGI ? |
| 06:44.05 | _omer | DCGrendel : Do you use AGI with Java, C , Perl or what ? |
| 06:44.05 | DCGrendel | i wrote some simple agi |
| 06:44.27 | DCGrendel | bash |
| 06:44.39 | DCGrendel | i needed to setup an AVR system so my customers could call and check the status of the server that hosts their sites |
| 06:44.50 | DCGrendel | and possibly kickstart it if its donw |
| 06:44.52 | DCGrendel | *down |
| 06:45.27 | _omer | alright... |
| 06:45.55 | _omer | I need expert suggestions regarding AGI ..can you please have a look for a moment ? |
| 06:45.56 | _omer | <PROTECTED> |
| 06:46.26 | DCGrendel | so i have a simple bash script that pulls the status of my unixshell server and plays a message based on what it gets back. |
| 06:47.54 | DCGrendel | wow |
| 06:48.05 | DCGrendel | i dunno how many calls you can handle |
| 06:48.13 | *** part/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 06:48.29 | DCGrendel | i'm running on a lowend system, and don't expect many calls really |
| 06:49.03 | DCGrendel | i've got < 10 customers who would potentially be calling my AVR |
| 06:49.25 | _omer | I see! |
| 06:49.34 | _omer | thats gonna be no problem then.. |
| 06:53.47 | DCGrendel | http://pastebin.ca/84693 |
| 06:53.52 | DCGrendel | thats what i've got working |
| 06:54.23 | DCGrendel | eventually i need to make it able to check multiple servers |
| 06:54.25 | *** join/#asterisk orlock (n=jwr@202.44.174.4.static.nexnet.net.au) |
| 06:54.47 | orlock | Can somebody tell me the UDP port ranges used for the voice data on SIP phones? |
| 06:55.00 | _omer | great! |
| 06:55.29 | orlock | 16382-20382? |
| 07:02.20 | *** join/#asterisk xblurone1 (n=me@203.125.98.22) |
| 07:02.38 | xblurone1 | Hi. |
| 07:03.37 | xblurone1 | I would like to log the IP address of the phones in the CDR, is this possible on MySQL and if so, where do I change what? |
| 07:08.12 | DCGrendel | is there a way to make voicemail return to a context instead of hanging up on exit? |
| 07:08.51 | CunningPike | DCGrendel: You could try an 'h' extension in the context from which Voicemail() was called |
| 07:10.31 | DCGrendel | will calling hangup from the h extension cause a loop? |
| 07:11.35 | DCGrendel | and no, adding h didnt help |
| 07:12.00 | DCGrendel | there really needs to be a small delay after vm-goodbye is played before hanging up the call |
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| 07:12.26 | DCGrendel | otherwise the call ends part way thru the word |
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| 07:14.14 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 07:14.15 | parag_ast | If i want that my all requests on every port should go threw localhost:888 then how can i write the iptables rule.... |
| 07:14.32 | parag_ast | tzafrir, do u hv idea of iptables |
| 07:14.41 | parag_ast | I need that my asterisk server |
| 07:15.00 | parag_ast | server's all request should go threw localhost:888 |
| 07:15.06 | _omer | parag_ast : I think thats the question related to ##Linux |
| 07:15.28 | parag_ast | Its mixed question |
| 07:15.39 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 07:15.42 | _omer | :) ... then shoot in both channels ;) |
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| 07:16.16 | tzafrir | parag_ast, who listens on localhost port 888? (udp or tcp)? |
| 07:16.26 | parag_ast | tcp and udp both |
| 07:16.39 | parag_ast | its acutally external proxy |
| 07:16.44 | tzafrir | udp and tcp are two different things |
| 07:16.48 | parag_ast | tcp |
| 07:16.50 | parag_ast | i need |
| 07:17.09 | parag_ast | actually in our area our telco is blocking all the sip and iax |
| 07:17.14 | parag_ast | so i m using external proxy |
| 07:17.17 | DCGrendel | wow. |
| 07:17.22 | parag_ast | which is running on localhost:888 |
| 07:17.31 | DCGrendel | they can get in trouble for that if you are in the right area |
| 07:17.34 | parag_ast | i need that my alll trafic should |
| 07:17.47 | parag_ast | pass threw this port |
| 07:26.18 | tzafrir | parag_ast, sip can be redirected. rtp can't easily be redirected. It inherently uses multiple ports |
| 07:26.31 | tzafrir | IAX is more of a single port protocol |
| 07:26.45 | parag_ast | I know that |
| 07:26.46 | parag_ast | i m using |
| 07:26.48 | parag_ast | IAX |
| 07:26.50 | parag_ast | only |
| 07:27.18 | tzafrir | If you only have a single UDP port at your disposal, consider using a VPN tunnel to some remote server |
| 07:28.55 | tzafrir | Also consider using a port number >= 1024 and make asterisk listen on it explicitly |
| 07:29.26 | parag_ast | so how can i right the rule |
| 07:29.58 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.145) |
| 07:31.00 | tzafrir | sending multiple "streams" to one port just doesn't make sense to me. Are you usre that's what you're supposed to do? |
| 07:31.48 | _omer | tzafrir: I need experts suggestions at http://pastebin.ca/84683 |
| 07:31.59 | _omer | I think you should check it once :) |
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| 07:39.39 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-179-48.red.bezeqint.net) |
| 07:41.29 | *** join/#asterisk erwinism (n=erwinpog@61.9.118.37) |
| 07:41.53 | erwinism | hello, what is a good sip or iax client that can be embeded on the web? |
| 07:49.18 | smackus | where is this being called from? |
| 07:49.19 | smackus | res_musiconhold.c:896 local_ast_moh_start: No class: default |
| 07:49.37 | smackus | I have verified that all of my setmusiconhold() are set to native, not default |
| 07:50.39 | *** join/#asterisk tparcina (n=tparcina@lns02-1215.dsl.iskon.hr) |
| 07:50.43 | tparcina | hi channel |
| 07:53.11 | smackus | http://pastebin.ca/84712 |
| 07:53.26 | smackus | this was working, but I reinstalled asterisk and now it is not . |
| 07:53.39 | smackus | any help would be appreciated |
| 07:53.49 | smackus | music on hold is not playing. |
| 07:54.09 | DCGrendel | heh |
| 07:54.13 | DCGrendel | my music on hold is playing |
| 07:54.17 | DCGrendel | but at the wrong speed |
| 07:54.23 | smackus | nice |
| 07:54.34 | smackus | the file exists but asterisk says it does not |
| 07:54.40 | DCGrendel | its not converting it to 8khz sample rate |
| 07:54.43 | erwinism | DCGrendel: like chipmunks? |
| 07:54.54 | DCGrendel | erwinism: other way. |
| 07:55.02 | DCGrendel | its playing a 44100 mp3 at 8000 |
| 07:55.11 | DCGrendel | smackus: check permissions and paths. |
| 07:55.30 | DCGrendel | /var/lib/asterisk/mohmp3 vs /usr/share/asterisk/mohmp3 |
| 07:55.43 | smackus | 'tis correct. |
| 07:55.56 | DCGrendel | maybe mpg123 is missing ? |
| 07:56.01 | smackus | using native |
| 07:56.06 | DCGrendel | dunno |
| 07:56.10 | smackus | k |
| 07:56.23 | DCGrendel | i am too and i get that rate conversion problem |
| 07:56.23 | DCGrendel | :P |
| 07:57.24 | tparcina | grandstream GPX 2000 - how to transfer all incoming calls to some extension? on Cisoc 7940 it can be done with one bottun, with gxp 2000 I don't can it be done at all... |
| 08:04.25 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:05.59 | smackus | ok... so why does music on hold sound like crap when someone else gets put on hold. |
| 08:06.21 | smackus | one caller on hold it sounds fine, two, it sounds like shit |
| 08:06.27 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-179-48.red.bezeqint.net) |
| 08:07.03 | DCGrendel | heh |
| 08:07.11 | DCGrendel | now my moh isnt even working period |
| 08:07.17 | evisu | ok, i give up. Anyone selling support ? |
| 08:08.02 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
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| 08:08.06 | DCGrendel | evisu: i'll let you speak to my helpdesk in india for $8/min, you'll have to excuse the zapping sounds, we have to use monkeys with tazers for supervisors. |
| 08:08.10 | smackus | speak with [TK]defender |
| 08:08.22 | smackus | about support |
| 08:08.37 | DCGrendel | (if you don't get that one, you are not having enough fun) |
| 08:08.49 | evisu | cheers i'll give him a try. regarding the monkeys i would settle for that right about now |
| 08:08.54 | DCGrendel | heh |
| 08:09.21 | DCGrendel | nice auto response. |
| 08:09.27 | evisu | whooops sorry, yeah auto |
| 08:10.21 | DCGrendel | <- tempted to use annoying kylie minogue songs for musiconhold |
| 08:10.47 | evisu | [tk]defender doesnt seem to be here, or maybe i just cant find him in this list |
| 08:10.51 | DCGrendel | the idea is to keep the queues empty, yes? |
| 08:11.27 | tparcina | grandstream gxp2000 - anybody use them? |
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| 08:14.12 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
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| 08:18.34 | Eimann | re |
| 08:21.22 | smackus | [TK] is most likely gone for the night. |
| 08:21.29 | evisu | ah |
| 08:21.46 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:21.54 | DCGrendel | there, i got my music on hold going :) |
| 08:23.37 | evisu | any other recomendations for support perhaps? |
| 08:24.33 | glLoadIdentity | last time i tried, moh needed specific version of mpg123 ... am i getting old? |
| 08:24.46 | DCGrendel | no, i hacked my own replacement :) |
| 08:24.46 | Assid | evisu: whats your problem anyhow |
| 08:25.12 | evisu | mainly setting up my te110p... |
| 08:25.26 | evisu | should be quite streight forward.. |
| 08:25.31 | Assid | glLoadIdentity: you could use native.. thats part of asterisk-addons package |
| 08:25.53 | Assid | evisu: i dont have one.. but have you tried the install/readme file and installed as it is |
| 08:26.12 | Assid | and what does ztcfg -v show ? |
| 08:26.17 | evisu | i've been glued to the screen for the past two days, it isnt working out well. |
| 08:26.29 | Assid | you need antistick |
| 08:26.50 | DCGrendel | http://pastebin.ca/84734 |
| 08:27.00 | DCGrendel | thats my replacement for mpg123, using sox |
| 08:27.15 | evisu | SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 08:27.15 | evisu | 31 channels configured. |
| 08:28.18 | Assid | okay so you have the channels configured |
| 08:28.22 | Assid | whats the problem? |
| 08:28.35 | evisu | just keep on getting a busy signal when i call the line |
| 08:28.36 | Assid | DCGrendel: so long as you get the end result :P |
| 08:28.47 | DCGrendel | yea |
| 08:28.54 | DCGrendel | i'm probably doing it wrong |
| 08:28.54 | tparcina | glLoadIdentity: yes, you are getting old - use native MOH |
| 08:28.56 | Assid | evisu: you tried outgoing call? |
| 08:29.05 | DCGrendel | i think its only going to play the first file in my mohmp3 dir |
| 08:29.34 | DCGrendel | i just can't remeber if its $* or $@ to include all paramaters |
| 08:29.37 | Assid | native moh.. the only way to go |
| 08:29.41 | evisu | assid, I really would, but im now stuck without my port 5060 working for some strange reason so i cant log into sip. ... This was going to wait on this question but you got it out of me |
| 08:29.52 | DCGrendel | assid: native plays *really* slow for me |
| 08:30.02 | DCGrendel | i'd have to re-encode my mp3s to mono 8000khz to use it |
| 08:30.08 | DCGrendel | which i should probably do anyway :P |
| 08:30.45 | Assid | why cant you login? whats the CLI saying? authentication issues? or just cant connect? |
| 08:30.48 | evisu | i go out for dinner, come back and i cant log into sip anymore. netstat -l shows port 5060 but not as 'Listening' |
| 08:31.10 | Assid | shut down asterisk.. start it up again |
| 08:31.12 | Assid | see if that helps |
| 08:31.16 | evisu | rebooted several times, |
| 08:31.23 | tzafrir | evisu, netstat -lu , right? |
| 08:31.38 | tzafrir | evisu, netstat -lup , even better |
| 08:31.40 | Assid | where is the sip phone as compared to the box? |
| 08:31.42 | evisu | will try that a sec |
| 08:31.50 | evisu | box is at the ISP |
| 08:31.51 | Assid | same network ? connecting from outside?.. ? |
| 08:31.52 | evisu | sip phone is at home |
| 08:32.17 | evisu | netstat -lu does show 5060 |
| 08:32.20 | Assid | reboot your router at home |
| 08:32.28 | Assid | and your phone |
| 08:32.53 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 08:33.06 | evisu | hmmm. ok .. |
| 08:33.12 | CMike | morning all |
| 08:33.14 | evisu | will be back in a min ! |
| 08:33.52 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:34.20 | Assid | musta been a softphone |
| 08:36.26 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 08:37.37 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-106-157.red.bezeqint.net) |
| 08:37.39 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 08:37.52 | evisu | restart didnt help unfortunatly |
| 08:38.08 | evisu | still getting a 'login timeout' on the softphone |
| 08:38.19 | DCGrendel | http://pastebin.ca/84739 <- if you have sox installed, that will play all mp3s in your music on hold directory in order (when used as a custom handler) |
| 08:40.37 | DCGrendel | i used to have one that used mikmod |
| 08:50.23 | *** join/#asterisk tsetane (n=tsetane@87.252.71.180) |
| 08:53.43 | *** join/#asterisk Vec (n=Vector@dsl-146-83-167.telkomadsl.co.za) |
| 08:53.49 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:59.59 | tparcina | does anybody use sip video - videosuppert=yes in sip.conf? |
| 09:00.12 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:00.34 | tparcina | i have tried to use it once but it didn't work. now i'm planing to try it again and i would like to hear some experiences |
| 09:00.45 | *** join/#asterisk RoyK[nl] (n=roy@194.178.111.19) |
| 09:03.19 | erwinism | hello, what is a good sip or iax client that can be embeded on the web? |
| 09:05.00 | Assid | http://www.microappliances.com/site/html/index.php?section=Products&page=sipphone.php |
| 09:07.33 | erwinism | Assid: yeah i saw that, but its for sale :) |
| 09:11.19 | Assid | well |
| 09:11.28 | Assid | if you do manage to find one.. let me know |
| 09:12.59 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 09:15.51 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 09:16.28 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-102-242.solcon.nl) |
| 09:18.16 | QuAtRo[NL] | When i have a agent logged on to a queue, and there is only one agent.. And the second call comes in.. Is there any posibility to make his phone ring again of make some LED burn? |
| 09:18.29 | DCGrendel | what? |
| 09:18.29 | QuAtRo[NL] | I have a Linksys SPA942 |
| 09:18.41 | DCGrendel | QuAtRo[NL]: you mean make all the leds flash? |
| 09:18.51 | DCGrendel | or something? |
| 09:19.04 | QuAtRo[NL] | Yes, it should be visible for the agent that someone is in the queue |
| 09:19.06 | DCGrendel | oh, when a second call comes into the queue you want him to know |
| 09:19.26 | QuAtRo[NL] | Yes, that's exactly what i want |
| 09:19.40 | *** join/#asterisk kyborg (n=kyborg@201.200.65.230) |
| 09:20.16 | DCGrendel | can probably flash an led or send a text message, i'm not familiar with that phone though |
| 09:21.07 | DCGrendel | or flash the message waiting light. |
| 09:21.25 | QuAtRo[NL] | Is there a feature in Asterisk which can do that? |
| 09:22.22 | DCGrendel | trying to figure that out now for you |
| 09:24.38 | CMike | anyone in here that uses Sangoma A104 ? |
| 09:25.19 | RoyK[nl] | CMike: i do |
| 09:25.50 | CMike | Ah Hi roy.. hm .. You in .nl now ? .) |
| 09:25.59 | RoyK[nl] | yep |
| 09:26.13 | CMike | thought you were in .au ? |
| 09:26.20 | CMike | .at even |
| 09:26.41 | CMike | I still got this really strange problem .. |
| 09:28.16 | RoyK[nl] | i was in .at |
| 09:28.19 | RoyK[nl] | two days ago :) |
| 09:28.40 | RoyK[nl] | 300km/h through germany by train |
| 09:28.56 | CMike | oh :) vacation ? |
| 09:29.06 | RoyK[nl] | yep |
| 09:29.12 | RoyK[nl] | but what's wrong with the card? |
| 09:29.23 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 09:29.39 | TeePOG | hi |
| 09:30.19 | CMike | try to call a number., and you can here for your self ? |
| 09:30.23 | CMike | Very strange sound |
| 09:30.24 | DCGrendel | QuAtRo[NL]: you might be able to have the queue call your phone even if you are in a call, and you'll get call waiting tone / flash |
| 09:31.01 | DCGrendel | QuAtRo[NL]: but i'm not experienced enough to know any other options |
| 09:32.02 | QuAtRo[NL] | My idea was to put a 'call agent' before i put the the caller into the queue |
| 09:32.08 | *** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net) |
| 09:32.39 | DCGrendel | yea |
| 09:36.56 | tparcina | RoyK: you are on vacation? why don't you chace some chiks, why do you read this channel? |
| 09:37.13 | tparcina | RoyK: pardon my question, but how old are you? |
| 09:37.28 | dlynes_laptop | tparcina: nosey, aren't you? |
| 09:37.49 | tparcina | dlynes, yes i'm |
| 09:38.23 | tparcina | dlynes, and you with laptop, are you also on vacation? :))) |
| 09:38.33 | dlynes_laptop | nah...just at a coffee shop |
| 09:38.40 | dlynes_laptop | easier to work here than in the office |
| 09:38.49 | tparcina | how come? |
| 09:38.52 | dlynes_laptop | the office is kinda stifling...no air conditioning |
| 09:39.06 | tparcina | that sucks |
| 09:39.47 | tparcina | dlynes, give a quit to oyur boss - or give him deadline until he has to get you air conditioner |
| 09:39.57 | dlynes_laptop | nah |
| 09:40.05 | dlynes_laptop | i don't want to work in the office anyways |
| 09:40.14 | dlynes_laptop | I'd rather work from the coffee shop, or from home |
| 09:40.24 | TeePOG | over a shell, everywhere is your office |
| 09:40.27 | tparcina | ok, is there any nice lady in coffe shop? |
| 09:40.32 | dlynes_laptop | TeePOG: exactly |
| 09:40.42 | dlynes_laptop | tparcina: not atm, no |
| 09:40.50 | dlynes_laptop | tparcina: but then again, it's 2:30am |
| 09:40.59 | JamesDotCom | that's when the finest ladies are out |
| 09:41.05 | JamesDotCom | ;) |
| 09:41.07 | tparcina | ok, then for you it's allowed to read this channel ;)) |
| 09:41.53 | dlynes_laptop | JamesDotCom: if you consider hte ones you have to pay for being the finest ones |
| 09:41.55 | TeePOG | can anyone tell me, is there a problem with the CentOS mirrors? i'm trying to run the install-pdf script, but it keeps trying mirror after mirror and getting "[Errno -1] Header not complete" for the package ghostscript |
| 09:42.03 | tparcina | no, finest ladies if they are not in bed till 1am, they go home ;)) |
| 09:42.12 | JamesDotCom | dlynes_laptop: haha, that was the joke ;> |
| 09:42.28 | dlynes_laptop | JamesDotCom: you mean all the crack and heroin addicts? |
| 09:43.03 | dlynes_laptop | TeePOG: time to dump centos, and install slackware instead |
| 09:43.40 | TeePOG | oh dlynes_laptop: sorry, it's a@h [#freepbx ignored my question there] |
| 09:44.15 | dlynes_laptop | TeePOG: why can't you run freepbx on slackware instead of centos? |
| 09:44.37 | TeePOG | dlynes_laptop: asterisk@home [the distro] is centos-based |
| 09:44.45 | dlynes_laptop | TeePOG: anyways, try asking in #centos |
| 09:44.58 | dlynes_laptop | TeePOG: there's a lot more peeps in there than freepbx, probably |
| 09:45.06 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
| 09:45.34 | TeePOG | i got the install CD from my digium-card supplier, don't have the b/w to download any other distro this month |
| 09:45.38 | dlynes_laptop | actually...nvm |
| 09:45.44 | TeePOG | thanks dlynes_laptop, will try that |
| 09:45.46 | dlynes_laptop | they've only got 156 peeps in their channel atm |
| 09:45.57 | dlynes_laptop | you probably won't get an answer there |
| 09:46.04 | dlynes_laptop | :P |
| 09:46.48 | tparcina | so, does anybody use video calls with asterisk? |
| 09:47.13 | tparcina | should I expect any problems? |
| 09:47.24 | TeePOG | hmm, will try again later |
| 09:48.24 | JamesDotCom | tparcina: use a sip proxy if you dont want any problems |
| 09:49.00 | TeePOG | wish there were a debian- slackware-based asterisk livecd... then i could use apt instead of bloody yum |
| 09:49.40 | tparcina | JamesDotCom: i din't plan to use one. is it necesery? |
| 09:50.51 | dlynes_laptop | TeePOG: livecd? |
| 09:50.54 | JamesDotCom | tparcina: might not be, asterisk might handle it fine... but what you're asking to do is for asterisk to handle the media, with a video call you're much better off being UA <-> UA |
| 09:51.02 | dlynes_laptop | TeePOG: a@h is a livecd? it's not one that gets installed? |
| 09:52.27 | dlynes_laptop | TeePOG: btw...i was kidding about #centos...sheesh...thought it was obvious :) |
| 09:52.28 | tzafrir | TeePOG, Asterisk@Home is not a liveCD |
| 09:52.54 | dlynes_laptop | tzafrir: he never explicitly said it was, but he seems to have implied it is |
| 09:53.56 | DCGrendel | _X will match any single-digit, yes? |
| 09:54.03 | dlynes_laptop | DCGrendel: correct |
| 09:54.39 | DCGrendel | what about */# ? |
| 09:56.20 | dlynes_laptop | DCGrendel: no idea, myself...never used those type of expressions in the dial plan |
| 09:56.27 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 09:57.03 | dlynes_laptop | I dub thee TeePOG the Centosian! |
| 09:57.50 | TeePOG | noooooooo! anything but that! I'll admit to kissing my cousin! anything! |
| 09:58.30 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 09:59.33 | *** join/#asterisk glLoadIdentity (n=tyn@195.175.21.46) |
| 10:01.32 | mosty | i have a zap card which doesn't detect when the remote end has hung up. what could be wrong? |
| 10:02.00 | DCGrendel | anyone know what version VMAuthenticate was introduced? |
| 10:02.36 | mosty | ie if i dial in, leave voicemail, then hang up, asterisk leaves the channel open and writes silence to the voicemail wav file, it's still writing now |
| 10:03.58 | *** join/#asterisk clive- (n=pirch@dsl-146-124-61.telkomadsl.co.za) |
| 10:06.31 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 10:10.22 | erwinism | in the type, what is the difference between peer and friend? |
| 10:11.38 | QuAtRo[NL] | A peer is a provider |
| 10:11.46 | QuAtRo[NL] | A friend is more a user |
| 10:11.51 | QuAtRo[NL] | In my vision :P |
| 10:11.52 | drray | a peer is a trunk |
| 10:12.00 | erwinism | ohh i see, thank you |
| 10:12.07 | drray | a friend is a subordinate/user |
| 10:12.18 | drray | in my vision |
| 10:13.19 | xblurone1 | I would like to log the IP address of the phones in the CDR, is this possible on MySQL and if so, where do I change what? |
| 10:15.28 | Splat | actually isn;t friend a combination of peer and user? |
| 10:15.45 | JunK-Y | xblurone1: use the userfield |
| 10:18.56 | *** join/#asterisk agallo (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 10:20.15 | agallo | Hello, my NVBackGRoundDetect function is now broken after upgrading asterisk, anyone has an updated sources code for it? It returns immediately without waiting for DTMF |
| 10:20.58 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 10:21.42 | clive- | teepog howzit |
| 10:22.22 | TeePOG | hi clive-, not bad, just on an unstable gprs connection |
| 10:22.48 | clive- | who you using for your gprs |
| 10:23.00 | TeePOG | vodacom |
| 10:23.05 | TeePOG | [South Africa] |
| 10:23.19 | clive- | i hear mtn is better because of "edge" |
| 10:23.47 | TeePOG | my phone is too old for edge :-( |
| 10:24.05 | DCGrendel | argh, i can dial out from my freedigits account, but now i can't accept incoming calls |
| 10:24.56 | clive- | Teepog I wonder if you could get voip to work on that |
| 10:25.19 | TeePOG | tzafrir: continue pvt convo? |
| 10:25.38 | TeePOG | clive- : my server is on adsl, i'm on holiday in the bushveld |
| 10:26.19 | clive- | Teepog lekker, and your still online?......??:) |
| 10:27.16 | TeePOG | lol clive- ... this is why I use both mtn and vodacom. failover reliability in a way: if cell-c pulls vodacom down, i put mtn sim card in |
| 10:27.21 | erwinism | i added stana for my trunk, should it be a friend or a peer in the incomming call settings |
| 10:27.58 | erwinism | seems that callers from stana cant hear anything but the call is established |
| 10:28.48 | DCGrendel | which is incoming from a provider, peer or user? |
| 10:29.07 | DCGrendel | peer is coming from them, user going to? |
| 10:29.10 | DCGrendel | or the other waY? |
| 10:29.34 | erwinism | its from stana |
| 10:29.54 | erwinism | incming calls from stanaphone |
| 10:30.20 | erwinism | i add one stana trunk in my asterisk |
| 10:30.59 | DCGrendel | i'm trying to get freedigits working both ways :P |
| 10:33.27 | erwinism | ohhh it might be because of my bandwidth problem |
| 10:34.55 | *** join/#asterisk TeePOG (n=1234@mtngprs4.mtn.co.za) |
| 10:35.00 | *** join/#asterisk evisu (i=hIRC@213.8.30.106) |
| 10:39.43 | TeePOG | hi hi |
| 10:41.01 | *** join/#asterisk Nettux (n=Quaaaack@adsl-195.137.136.17.tzm.net) |
| 10:42.40 | Nettux | Hello everybody. I am working on asterisk 1.2.9.1 and am experiencing a problem with background command. What happens is that if i try to background a sound it won't go on in the dialplan but it waits for the sound to be played completely (as if I had used playback instead). Whay is this happening? |
| 10:42.46 | Nettux | why* |
| 10:46.28 | Assid | background isnt for jumping to the next directive in the dialplan |
| 10:46.35 | Assid | its to accept incoming dtmf signals |
| 10:46.50 | Assid | like what you use in IVR systems |
| 10:46.52 | Nettux | well the doc says it should |
| 10:47.27 | Assid | well imagine someone running 3 background functions one after the other in order of priority |
| 10:47.28 | Nettux | Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next commands (if any) execute. |
| 10:47.32 | Assid | any clue what would happen ??? |
| 10:47.46 | *** part/#asterisk tparcina (n=tparcina@lns02-1215.dsl.iskon.hr) |
| 10:48.02 | Assid | i guress that needs to be updated... |
| 10:48.17 | agallo | Hello, my NVBackGRoundDetect function is now broken after upgrading asterisk, anyone has an updated sources code for it? It returns immediately without waiting for DTMF |
| 10:48.20 | Assid | as far as i know. background is only to play the sounds and wait for dtmf |
| 10:48.29 | Nettux | Assid then i'll have to find a workaround... what I need is a sound that plays while asterisk finds an available exten that can answer the call |
| 10:48.50 | Assid | Nettux: you want a queue then ? |
| 10:49.00 | Nettux | I tried with queues but they won't let me do it in the order i want |
| 10:49.21 | Assid | well.. you could try MOH then |
| 10:49.46 | Nettux | Assid how do I get out of MOH when the call is answered? |
| 10:49.55 | Nettux | what i mean is |
| 10:50.06 | Nettux | I tried that but if the call is put on hold |
| 10:50.14 | Nettux | it won't go on with the dialplan |
| 10:50.17 | Assid | or use MOH/music instead of ringing |
| 10:50.32 | Nettux | Assid how do i do that? |
| 10:50.38 | Assid | read up on the dial function |
| 10:50.43 | Assid | option m |
| 10:50.47 | Nettux | k thanks i will |
| 10:52.25 | agallo | Uhm... what's wrong with Background also? it doesnt not wait for the timeout, it quits ASAP the file is played? O.o |
| 10:53.54 | agallo | i've the exact reverse problem of Nettux, background and nvbackground are not waiting me for DTMF O.o |
| 10:54.14 | Assid | asetyou timeout |
| 10:54.25 | Assid | digit timeout |
| 10:54.28 | Assid | and response timeout |
| 10:54.35 | Assid | check the wiki |
| 10:54.48 | agallo | i did |
| 10:54.57 | agallo | exten => s,1,Set(TIMEOUT(digit)=30000) |
| 10:54.57 | agallo | exten => s,1,Set(TIMEOUT(absolute)=30000) |
| 10:54.57 | agallo | exten => s,1,Set(TIMEOUT(response)=30000) |
| 10:55.13 | Assid | how the hell do you expect that to work? |
| 10:55.48 | agallo | values are too bigs? |
| 10:56.00 | Assid | besides that.. check your priorities |
| 10:56.04 | agallo | ouych rofl, "1" :-) |
| 10:56.10 | drray | n |
| 10:56.52 | agallo | priorities are ok, i just maded a mistake with cut and copy |
| 10:57.01 | Assid | thats why they arent okay |
| 10:57.10 | Assid | needs to be correct |
| 10:57.27 | Assid | and your timeouts are ridiculously high |
| 10:57.42 | agallo | i can put any value they just don't work anymore |
| 10:57.52 | agallo | the function return asap the background file ends |
| 10:58.07 | Assid | pastebin your dialplan |
| 10:58.30 | knarfly | anyone know about zaptel & FreeBSD issues? |
| 10:59.59 | agallo | exten => s,1,Macro(ringing) |
| 10:59.59 | agallo | exten => s,n,Set(TIMEOUT(digit)=5) |
| 10:59.59 | agallo | exten => s,n,Set(TIMEOUT(response)=5) |
| 10:59.59 | agallo | exten => s,n,Background(vm-enter-num-to-call) |
| 10:59.59 | agallo | exten => s,n,NoOp(error) |
| 11:00.15 | Assid | PASTEBIN!!!!!!!! |
| 11:00.19 | Assid | anwyays |
| 11:00.22 | Assid | whats the problem |
| 11:00.49 | Assid | does your cli show Background(vm-enter-num-to-call) being executed? |
| 11:00.50 | mut | man fog is BAD today, 2 of our radio links are down because it's so thick |
| 11:01.04 | Assid | radio links are affected by fog? |
| 11:01.17 | mut | 2.4ghz and 5.8ghz |
| 11:01.31 | agallo | Assid, yes it show it executed, and the NoOp after the background file ends to play |
| 11:01.43 | Assid | so whats wrong |
| 11:01.49 | Assid | doesnt accept the digits? |
| 11:01.49 | agallo | Assid, if i dial a tone before the voice ends it works |
| 11:02.05 | agallo | Assid, when background file ends it does not wait me the extra seconds |
| 11:02.17 | Assid | increase your response to 40 |
| 11:02.26 | Assid | and make sure to reload extensions |
| 11:03.31 | agallo | Assid, i already did that with 30, 45, 120 and 30000 |
| 11:03.53 | Assid | have you checked whether the problem exists on your sip client which is calling in? |
| 11:03.54 | agallo | Assid, i tought NVBackGround detect was broken but then testing with Background i discovered the same wrong behavior |
| 11:04.08 | Assid | or however else your calling in |
| 11:04.16 | agallo | Assid, i tested with sjphone, thomson phone and IDefisk IAX softphomne |
| 11:04.29 | Assid | weird |
| 11:04.59 | agallo | Assid, is there a way i can debug/strace that Backgound function? |
| 11:05.10 | Assid | hold |
| 11:05.10 | Assid | whats the file type? |
| 11:05.50 | Assid | sorry mate.. my knowwledge in that is prettry limited |
| 11:05.50 | Assid | also what version of asterisk are you using |
| 11:05.56 | agallo | latest |
| 11:06.06 | agallo | 1.2.9.1 |
| 11:07.12 | Assid | you know what |
| 11:07.18 | Assid | remove the digit timeout |
| 11:07.29 | Assid | and set only the response timeout to 20 |
| 11:07.41 | Assid | stop asterisk.. and then start it again.. and then see if it works |
| 11:07.56 | agallo | Assid, i start to suspect that Wiki is broken since "show application Background" gives me a different man page.... O.o |
| 11:08.01 | Assid | also.. add exten => s,n,WaitExten after the Background |
| 11:08.32 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:09.07 | Assid | that should take care of your headache |
| 11:10.35 | agallo | ofc with WaitExten will work... but the Wiki is not reporting that asterisk is now behaving differently O.io |
| 11:11.00 | agallo | on my old setup Background was waiting ResponseTimeout |
| 11:11.07 | Assid | well.. so long as you get what you need |
| 11:11.33 | Assid | i just try and make sure everything is compliant.. so i dont have problems in future |
| 11:11.58 | Assid | be it a bug or not |
| 11:12.43 | agallo | who i should report that wiki is incoherent with 1.2.9.1 ? |
| 11:13.04 | Assid | i guess the mailing list? |
| 11:14.45 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:14.47 | PakiPenguin | hello |
| 11:15.08 | agallo | Assid, thank you very much |
| 11:15.17 | parag_ast | r u from pakistan |
| 11:15.20 | Assid | np agallo |
| 11:15.28 | agallo | Assid, off for lunch, if you ever get in Italy you have 1 big beer payed |
| 11:15.32 | PakiPenguin | parag_ast, yes |
| 11:15.39 | parag_ast | okk i m from India |
| 11:15.46 | PakiPenguin | good to know that |
| 11:15.54 | Assid | hahahaa |
| 11:16.00 | parag_ast | right now in Dubai |
| 11:16.00 | Assid | that was funny |
| 11:16.15 | Assid | "are you from pakistan" "yes" "ok im from india" |
| 11:16.20 | Assid | damn |
| 11:16.33 | parag_ast | sorry |
| 11:16.35 | parag_ast | Assid |
| 11:16.42 | Assid | nah dont be |
| 11:16.46 | Assid | it was funny |
| 11:17.11 | PakiPenguin | cool :) |
| 11:17.18 | Assid | okay.. i need a good way to relax.. cartoons here i come! |
| 11:17.28 | PakiPenguin | Assid, which ones? |
| 11:17.33 | Assid | who knws |
| 11:17.39 | Assid | i just put on the channel |
| 11:18.10 | Assid | besides.. hallmark showing some nice series later |
| 11:18.22 | Assid | im kinda wasting some time for 10 mins.. |
| 11:19.11 | PakiPenguin | :) |
| 11:19.19 | PakiPenguin | Assid, asterisk 1.0.11 can do eagi? |
| 11:19.53 | Assid | not sure.. why arent you using latest stable ? |
| 11:20.02 | parag_ast | Zaptel is sucking here in UAE...Not hanging the line |
| 11:20.19 | parag_ast | hanging up |
| 11:20.32 | Assid | parag_ast: try adding a hangup() after the dial() |
| 11:20.59 | Assid | if you hangup.. it will send a signal back to hangup |
| 11:21.13 | Assid | isnt voip illegal in uae |
| 11:21.47 | parag_ast | Yes it is |
| 11:22.43 | PakiPenguin | Assid, i cannot , this is someone's machine |
| 11:23.10 | parag_ast | Assid, I was talking about the incomming one |
| 11:23.12 | parag_ast | not the |
| 11:23.14 | Assid | if your handling it.. its yourcall to upgrade |
| 11:23.14 | parag_ast | outgoing |
| 11:23.36 | Assid | parag_ast: you have a dial(SIP/whaeteer) ? |
| 11:23.45 | parag_ast | yes |
| 11:23.52 | Assid | then add it after dial |
| 11:24.48 | *** join/#asterisk glLoadIdentity (n=tyn@81.214.255.57) |
| 11:24.49 | parag_ast | but zaptel hardware not able to detect hangup man |
| 11:26.55 | Assid | okay 10 mins over... tv time |
| 11:27.36 | Assid | parag_ast: no clue.. i barely use any zaptel devices |
| 11:52.45 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-106-157.red.bezeqint.net) |
| 11:53.47 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 11:54.31 | evisu | trying this again: Anybody interested in providing paid support ? need help with a few things (ie install te110p..) |
| 11:54.42 | drray | where? |
| 11:55.05 | evisu | where is the server? or |
| 12:02.40 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:04.35 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 12:05.12 | brif8 | anyone using Teliax, I can ping but I get unreachable in sip show peers and registration keeps timing out ? |
| 12:06.06 | mut | man i could go for a big mac right now |
| 12:07.01 | PakiPenguin | iw ant one too |
| 12:07.48 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.158) |
| 12:10.30 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 12:25.36 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 12:27.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:27.32 | brif8 | anyone using Teliax, I can ping but I get unreachable in sip show peers and registration keeps timing out ? |
| 12:27.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 12:28.13 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 12:28.25 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:28.53 | *** join/#asterisk daysmen3 (n=primus@host86-139-53-168.range86-139.btcentralplus.com) |
| 12:30.58 | brif8 | anyone know of VoIP to PSTN providers in the Tampa FL area ? |
| 12:31.11 | ph|ber | sellvoip has an office in miami |
| 12:34.20 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 12:37.15 | xheliox | Teliax down, again. |
| 12:39.10 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 12:41.27 | brif8 | looking for a GOOD VoIP to PSTN provider in the Tamp FL area ? |
| 12:43.07 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 12:47.28 | *** join/#asterisk itssimon (n=itssimon@60-240-115-18-nsw-pppoe.tpgi.com.au) |
| 12:48.18 | itssimon | Hello to all, I have an absolute emergency, can someone please help me with an Asterisk configuration file question. We have system down! :( |
| 12:50.04 | itssimon | ? |
| 12:50.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:50.32 | itssimon | anyone? |
| 12:50.37 | tzafrir | itssimon, not if you don't ask it |
| 12:50.42 | itssimon | ok |
| 12:51.39 | itssimon | The problem is this.......I use WinSCP as my SSH client to edit my sip.conf.....Something went wrong and it locked up and now it has deleted my sip.conf, It is 0 bytes!!!! So I don't have a backup and I'm in deep trouble. However, it is still in memory on the dialplan....Is there any way to make asterisk output to file the sip.conf that is in memory? |
| 12:52.36 | tzafrir | for starters, don't 'reload' |
| 12:52.48 | itssimon | yes!! I am definitely not going to do that |
| 12:52.58 | [TK]D-Fender | itssimon : "sip show peers". This will give you the list. Then "sip show peer [peer]" to get the details. |
| 12:53.05 | [TK]D-Fender | itssimon : How big a system? |
| 12:53.54 | itssimon | hi, it is has about 100 calls a day |
| 12:53.59 | itssimon | the day is just about to begin |
| 12:54.09 | itssimon | TK....I tried that but it just shows me basics |
| 12:54.22 | itssimon | is there any way to just get the sip.conf dumped to file from within memory? |
| 12:55.36 | mut | no |
| 12:56.59 | itssimon | dammit! |
| 12:58.05 | mut | just don't stop the server |
| 12:58.10 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 12:58.11 | mut | and spend the day rebuilding it |
| 12:58.22 | [TK]D-Fender | itssimon : No, how many PHONES do you have configured? |
| 12:58.29 | mut | learned the hard way how to make a backup eh |
| 12:58.37 | [TK]D-Fender | itssimon : Any VoIP accounts? |
| 12:58.45 | file | or check your configs into something like subversion |
| 12:58.48 | itssimon | yeah...stupid me! Just 1 local VOIP account, but connected to a tollfree number |
| 12:59.20 | mut | file? |
| 12:59.27 | file | [TK]D-Fender: this timezone change has my body confused, I'm waking up at almost 6AM on the dot |
| 12:59.42 | [TK]D-Fender | file : I'm not awake *NOW* |
| 12:59.53 | [TK]D-Fender | file : I feel like shit on a stick today. |
| 13:00.19 | file | mut: if you put your configs into SVN you can then just commit when you make changes... and grab new ones to deploy servers, revert to an older version... all that fun stuff |
| 13:00.32 | mut | oh |
| 13:00.33 | mut | yea |
| 13:01.03 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:01.06 | file | [TK]D-Fender: the answer is... potato |
| 13:01.36 | Katty | hi file (= |
| 13:01.58 | [TK]D-Fender | Katty : mew. |
| 13:02.06 | iCEBrkr | Bitches... |
| 13:02.08 | Katty | mister fender (= |
| 13:02.08 | iCEBrkr | Hoes... |
| 13:03.18 | ariel_ | Katty, morning |
| 13:03.30 | Katty | hi ariel_ (= |
| 13:03.37 | ariel_ | hope everyone is doing fine today. |
| 13:04.15 | Katty | still waking up |
| 13:04.43 | ariel_ | k |
| 13:04.55 | file | I am... doing the same |
| 13:05.30 | ariel_ | I am just getting ready to drive up 68 miles north to a job. 3 days a week for now. part time work with asterisk setups. Strange don't really want to work for a company. |
| 13:05.44 | file | come to the dark side... |
| 13:06.03 | ariel_ | hello file, |
| 13:06.10 | file | hola |
| 13:06.11 | ariel_ | but I need the income... |
| 13:06.45 | ariel_ | my 18 year old starts Collage in 3 weeks... argh is it expensive.... |
| 13:07.37 | Katty | wish my parents would have paid for my college |
| 13:07.44 | jbalcomb | ditto |
| 13:08.02 | jbalcomb | or my first car, house, or wedding. ;) |
| 13:08.02 | [TK]D-Fender | ariel_ : Come to Canada! And when we're done he'll just get exported back like the rest! |
| 13:08.04 | Katty | would have been nice if they even offered to help a little. |
| 13:08.05 | ariel_ | well she has some loans, grants and I have to cover the rest. still expensive |
| 13:08.05 | mut | i don't |
| 13:08.11 | mut | otherwise i'de still be in school |
| 13:08.54 | ariel_ | well she is going to be an (Industrial Engineer) first time I have heard of that one. |
| 13:09.43 | [TK]D-Fender | ariel_ : Better than a Bac in philisophy.... so she can have deep thoughts about her unemployment ;) |
| 13:09.50 | ariel_ | OK go to go now.. need to take that long drive in the stupid road system down here.... Argh |
| 13:10.32 | ariel_ | [TK]D-Fender, she was going for the Mechanical one but they told her this one includes it and a 2nd in Elec too. |
| 13:11.13 | ariel_ | just strange in my view... at least it's something she can use. Not like mine I have a BS in Chemistry..... Never used it. |
| 13:11.26 | PakiPenguin | ariel_, BSC in chem. is awesome |
| 13:11.33 | mut | how can you not use a bs in chem? |
| 13:11.42 | PakiPenguin | <-- aeronautics and communications engineer |
| 13:12.00 | ariel_ | I instead was drawn by the US Air Force and started flying instead... |
| 13:12.07 | [TK]D-Fender | <- 2 bit hack (1 is for PARITY) |
| 13:12.33 | PakiPenguin | :p |
| 13:12.35 | ariel_ | The Dark Side took me.... |
| 13:12.56 | PakiPenguin | ariel_, which plane? |
| 13:13.19 | ariel_ | I ended up not flying them for them But as a Flight Engineer for the C-130E/H |
| 13:13.26 | [TK]D-Fender | PakiPenguin : He'd tell you, but then he'd have to bomb you ;) |
| 13:13.34 | ariel_ | haha |
| 13:13.53 | ariel_ | got to go... see you all much later... |
| 13:13.58 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 13:14.10 | PakiPenguin | :) |
| 13:14.11 | PakiPenguin | lol |
| 13:14.20 | PakiPenguin | [TK]D-Fender, if he doesnt , your govt. will |
| 13:14.24 | PakiPenguin | sometime soon :p |
| 13:14.32 | file | I am damned... I am the master of hundreds of workers... they all look to me... |
| 13:15.07 | [TK]D-Fender | PakiPenguin : MINE?! We don't HAVE bombs! We send over peace-keeping troops to help manage the messes they make... |
| 13:15.18 | PakiPenguin | lol :) |
| 13:15.48 | [TK]D-Fender | file : Boy are they screw, eh? ;) |
| 13:15.56 | file | [TK]D-Fender: :D |
| 13:17.29 | *** join/#asterisk MikeJ__ (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 13:19.15 | file | not even good attempts |
| 13:19.17 | file | darn |
| 13:19.35 | file | I should go... get ready... |
| 13:19.46 | *** join/#asterisk Sudith (n=none@adsl-70-128-231-105.dsl.austtx.swbell.net) |
| 13:21.01 | *** part/#asterisk Sudith (n=none@adsl-70-128-231-105.dsl.austtx.swbell.net) |
| 13:23.53 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 13:24.05 | unixgeek | Good Morning..... |
| 13:24.24 | QuAtRo[NL] | Morning |
| 13:24.39 | unixgeek | Working on a PRI right now that will accept a 5 digit extention and pass it to a Nortel Meridian switch. |
| 13:24.58 | unixgeek | But it will not accept a 7 digit number for a local exchange. |
| 13:25.01 | unixgeek | Any thoughts? |
| 13:25.21 | unixgeek | The dialplan passes both calls over the T1. |
| 13:27.46 | Hmmhesays | morning geeks |
| 13:29.28 | file | Hmmhesays: mornings are evil |
| 13:29.34 | Katty | Hmmhesays: hey hun |
| 13:29.35 | Hmmhesays | so true |
| 13:29.42 | Hmmhesays | Katty |
| 13:29.54 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.186.Dial1.SanJose1.Level3.net) |
| 13:30.08 | *** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 13:30.09 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.186.Dial1.SanJose1.Level3.net) |
| 13:30.29 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:31.02 | unixgeek | Is there any way to see what asterisk is sending over the D channel of a PRI? |
| 13:31.14 | Katty | Hmmhesays: what's this asterisk business edition thing? |
| 13:31.40 | Hmmhesays | Um, isn't it supposed to be the more 'stable' asterisk with digum support ...etc |
| 13:31.45 | Hmmhesays | *digium |
| 13:31.56 | Katty | i don't know. my boss wants me to check into it tho |
| 13:32.20 | unixgeek | Nevermind. found the pri command. |
| 13:32.22 | Hmmhesays | why? your asterisk not working? |
| 13:32.29 | *** join/#asterisk alexns (n=ibtek04@static-acs-24-154-114-15.zoominternet.net) |
| 13:32.35 | Katty | it's working fine. |
| 13:32.42 | alexns | damn teliax down again |
| 13:33.07 | brif8 | alexns: yes since before 7:21 am EDT and they say they might be up in 15 minutes +/- |
| 13:33.08 | Katty | Hmmhesays: he seems to think that asterisk business edition would be Better(tm) |
| 13:33.16 | Hmmhesays | For what reason? |
| 13:33.20 | Hmmhesays | Are you having problems? |
| 13:33.38 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 13:33.43 | Hmmhesays | I wish people would throw money at me |
| 13:33.44 | Katty | Hmmhesays: not a single problem. |
| 13:33.51 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 13:33.57 | Hmmhesays | well then no, it would not be 'better' |
| 13:34.07 | Katty | well unfortunately he won't listen to me |
| 13:34.14 | Katty | feel free to call my boss and argue with him |
| 13:34.16 | xheliox | He'd be paying for a warranty, basically. |
| 13:34.18 | file | Katty: it's a tested version of Asterisk, and you get support from Digium if you have issues... |
| 13:34.29 | alexns | brif8:back up now |
| 13:34.48 | Hmmhesays | Katty: I don't want to argue with your boss, tell him we said your setup is fine |
| 13:34.54 | xheliox | alexns: Yeah. But like brif8 said, it's been down since early morning. |
| 13:35.20 | Katty | Hmmhesays: i've told him the setup is fine. |
| 13:35.24 | alexns | xheliux: not a good thing |
| 13:35.25 | Katty | Hmmhesays: he doesn't care. |
| 13:35.29 | Hmmhesays | then tell him to stfu |
| 13:35.32 | Katty | Hmmhesays: haha |
| 13:35.35 | Katty | Hmmhesays: that'd be the day |
| 13:35.38 | xheliox | alexns: Nope. Sorta discouraging. |
| 13:35.43 | Hmmhesays | i need coffee now |
| 13:35.54 | xheliox | Katty: I'll argue with him, but I charge $95/hr for phone support. :) |
| 13:36.03 | brif8 | alexns: yes just came up |
| 13:36.47 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-63-77.nwrknj.east.verizon.net) |
| 13:37.03 | Katty | xheliox: i can get support for free ;) |
| 13:37.15 | Katty | xheliox: it's call you politely ask. |
| 13:37.19 | Katty | s/call/called |
| 13:37.26 | Hmmhesays | I'm cheaper $75/hour |
| 13:37.40 | [TK]D-Fender | Katty : Do you have actual issues? |
| 13:37.46 | Katty | Hmmhesays: i'd buy you a drink and we'd be square. |
| 13:38.05 | Katty | [TK]D-Fender: i've got so many issues, you can afford to subscribe. |
| 13:38.20 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 13:38.25 | [TK]D-Fender | Katty : I meant with * :) |
| 13:38.28 | Katty | [TK]D-Fender: so, no. the bossman just seems to think that asterisk business edition would be Better(tm) and won't listen to me, etc. but that's pretty much normal now. |
| 13:38.34 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F1352.dip0.t-ipconnect.de) |
| 13:38.47 | [TK]D-Fender | Katty : Its an OLDER release and doesn't offer anything you don't have already. |
| 13:39.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:39.24 | Katty | [TK]D-Fender: i'm well aware. |
| 13:39.50 | [TK]D-Fender | Katty : You can tell him he's a schmuck. ABE is for people who need hand-holding or the illusion of corporate backing to validate CHOOSING *. You already have it, so whats the point. </twocents> |
| 13:39.58 | [ProB]CrazyMan | hello I tried to use hint in my dialplan, so when I do "show hints" i see the state from my phones, but on my Snom phone the lights are still off |
| 13:39.59 | Katty | for goodness sake, fender. |
| 13:40.28 | Dovid | i am cheaper |
| 13:40.29 | [TK]D-Fender | Katty : I already know I'm good, but this isn't the channel to discuss my prowess ;) |
| 13:40.35 | Dovid | $45.00 and hour |
| 13:40.37 | Katty | [TK]D-Fender: you hush. |
| 13:40.43 | Katty | [TK]D-Fender: there are probably children in here! |
| 13:40.53 | [TK]D-Fender | Katty : Regardless of age ;) |
| 13:42.21 | [ProB]CrazyMan | does I need to do any special to force the snom phone to subscribe |
| 13:42.42 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 13:44.48 | oscarh | hi, when i originate a call to a Zap FXO channel / phonenumber it gets "Answered" streight away. Is there any way to change this behaviour? |
| 13:46.00 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 13:46.11 | tzafrir | oscarh, don't Answer()? |
| 13:46.31 | oscarh | tzafrir, i dont |
| 13:46.43 | oscarh | maybee i'm doing it the wrong way |
| 13:46.47 | file | nah unless you have callprogress turned on it'll be answered right away, and if you have callprogress turned on... well it only works for the US and it may cause your calls to randomly hang up |
| 13:47.01 | [TK]D-Fender | oscarh : No, thats analog tech for you... doesn't know any better. |
| 13:47.24 | [TK]D-Fender | oscarh : Get a PRI. |
| 13:47.47 | oscarh | oh, so is it inposible to make an outgoing call and connect it to an AGI channel when it is connected? |
| 13:47.56 | oscarh | sorry, when it has ben answered |
| 13:51.19 | jbalcomb | anyone using APC PowerChute on their Asterisk server to tell it to shutdown when the UPS gets too low during an outage? |
| 13:52.36 | Katty | [TK]D-Fender: i sent my boss a nice email. |
| 13:53.02 | [TK]D-Fender | :O |
| 13:53.40 | Assid | oh man.. |
| 13:53.41 | *** part/#asterisk evilmnky (n=evilmnky@69-163-156-13.atlsfl.adelphia.net) |
| 13:53.42 | Assid | this sucks |
| 13:53.49 | Assid | blasts in local train here in bombay |
| 13:54.50 | drray | local crazies or imported jerks |
| 13:54.51 | drray | ? |
| 13:55.10 | Katty | does anyone else have anything to say about ABE? |
| 13:55.24 | Katty | opinion, experience, etc. |
| 13:57.20 | clive- | abe? |
| 13:57.40 | Katty | asterisk business edition |
| 14:01.41 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 14:03.10 | DrkShdw | Katty: as mentioned earlier, if your astrerisk is working, ABE is gonna be no different. ABE just gives you the ability to pickup the phone and call for support |
| 14:03.47 | gandhijee | Assid: what happened? |
| 14:04.03 | [TK]D-Fender | Katty : And the ability to integrate with a proprietary solution without GPL "complications". |
| 14:04.14 | DrkShdw | If you want it, to help support digium, go for it. if you want it, cuz you think it's gonna be different than the Free asterisk? not much difference |
| 14:04.57 | Katty | [TK]D-Fender: i'm going to interview for addition emailing porpoises. |
| 14:04.58 | DrkShdw | if you want it, cuz your boss thinks it's better (as mentioned earlier) he's an idiot, unless he likes the idea of the company support |
| 14:05.28 | Katty | DrkShdw: well unfortunately my boss wants Facts and not your opinion. |
| 14:05.49 | *** join/#asterisk ahsan (n=ahsan@host-83-146-53-46.bulldogdsl.com) |
| 14:05.59 | DrkShdw | 09:55 <Katty> opinion, experience, etc. <-- sorry. I thought you said opinions. Sorry. I'll learn some english one of these days. |
| 14:06.07 | Qwell[laptop] | Katty: facts shmacts |
| 14:06.07 | Katty | DrkShdw: good job (= |
| 14:06.19 | Katty | DrkShdw: I don't care what you think, if you're just going to insult it. |
| 14:06.26 | Katty | DrkShdw: when i ask for opinions, i ask for reasons. |
| 14:06.44 | DrkShdw | I'm not insulting it. I gave you my opinion. and additionally, I gave you reasons. |
| 14:06.53 | Katty | Qwell[laptop]: if you have a few minutes, i wouldn't mind getting your opinion too |
| 14:06.53 | DrkShdw | piss off. |
| 14:06.59 | *** join/#asterisk Drew99 (n=top@ppp83-237-189-211.pppoe.mtu-net.ru) |
| 14:07.00 | Katty | DrkShdw: awww, poor baby. |
| 14:07.05 | Qwell[laptop] | Katty: on? |
| 14:07.09 | Katty | Qwell[laptop]: ABE |
| 14:07.19 | Qwell[laptop] | what about it? |
| 14:07.21 | Katty | Qwell[laptop]: but not in here, people like throwing insults in randomly (= |
| 14:08.04 | Drew99 | hi! on the incoming call I have: http://phonix.ru/iax.txt what does it mean? anybody know? |
| 14:08.24 | *** part/#asterisk BeanDip (n=benattar@ip-d88bfed0.hostwaydcs.com) |
| 14:10.13 | Assid | gandhijee: 5 blasts |
| 14:10.19 | Assid | they are confimring the 6th one |
| 14:12.04 | gandhijee | holy shit |
| 14:12.49 | drray | is it the 11th in india or is it the 12th? |
| 14:13.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 14:13.46 | gandhijee | still the 11th |
| 14:13.51 | gandhijee | 10.5 hours ahead |
| 14:13.55 | _problem_ | Assid: where happens the blasts in india? |
| 14:14.36 | Assid | mumbai |
| 14:14.51 | _problem_ | ohh ok |
| 14:15.15 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 14:15.57 | Winkie | sup guys, anyone got much experience with inbound zaptel calls to agents, i just want to clear something up regarding why two Local/ channels are spawned with two different uniqueid |
| 14:15.57 | Winkie | s |
| 14:16.21 | *** part/#asterisk Drew99 (n=top@ppp83-237-189-211.pppoe.mtu-net.ru) |
| 14:16.25 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:18.48 | Vec | How can I check if a SIP phone is registered to asterisk ? |
| 14:19.15 | jbalcomb | Vec: sip show peers |
| 14:19.16 | Vec | sip show registry, shows nothing but I am pretty sure the phone is registered. |
| 14:19.29 | jbalcomb | Vec: What phone? |
| 14:19.30 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:19.42 | jbalcomb | Vec: Most of them have on screen status of registration |
| 14:19.46 | Vec | jbalcomb : cool sip show peers worked!, thanks |
| 14:20.01 | jbalcomb | Vec: Also, if you have guest access turned off, unregistered phones don't work at all |
| 14:20.08 | Vec | jbalcomb : it is a Sipura 3000 |
| 14:20.23 | jbalcomb | Vec: ah.. well, glad to help |
| 14:20.42 | Vec | jbalcomb : the phone part works perfectly, but the PSTN part is giving me sh1t |
| 14:21.18 | Winkie | Vec: what's up? |
| 14:21.53 | Vec | jbalcomb : when I phone the PSTN line, it does not Playback(tt-monkeys), but goes beebeebeebeebebe... |
| 14:22.07 | Vec | my extensions.conf file looks right |
| 14:22.43 | Winkie | Vec: what sort of PSTN line? ISDN (bri/pri?) etc |
| 14:23.08 | Vec | Its a normal analogue line, no ISDN. |
| 14:23.14 | Vec | POTS lien |
| 14:23.16 | Vec | line |
| 14:24.12 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:24.21 | Vec | exten => homephoneline,1,Answer --- is the "homephoneline" in that line arbitary ? |
| 14:24.29 | Vec | or is it supposed to refer to something |
| 14:25.34 | Winkie | Vec: it should be the extension, i would have s there |
| 14:25.39 | Winkie | but i've never tried it |
| 14:25.53 | Winkie | with my E1 we use our DDI pattern, IE _XXXXXX |
| 14:29.56 | gandhijee | anyone know what exactly is the DAA is? |
| 14:31.47 | Winkie | gandhijee: context? |
| 14:31.57 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 14:32.11 | Hmmhesays | suddenly asterisk stopped sending calls to the right context by domain |
| 14:32.26 | gandhijee | happens when i try to load the wcfxo module |
| 14:35.09 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:35.45 | jbalcomb | Ah, GrandStream.. when you turn off config via keypad it also disables the ability to factory default the phone... |
| 14:38.20 | *** part/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-102-242.solcon.nl) |
| 14:38.32 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 14:39.07 | [TK]D-Fender | jbalcomb : Grandsuck strikes again! |
| 14:40.30 | jbalcomb | indeed |
| 14:41.44 | [TK]D-Fender | jbalcomb : Ordered my IP 430, should have it for friday :) |
| 14:42.26 | jbalcomb | [TK]D-Fender Excellent. You /must/ let me know how that feelsw. |
| 14:42.30 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 14:42.34 | jbalcomb | s/w// |
| 14:42.41 | jbalcomb | damn you jbot |
| 14:43.08 | jbalcomb | I am so bad with SED and limiting the scope of my matches |
| 14:43.30 | [TK]D-Fender | Can promise you I'm WORSE :) |
| 14:43.41 | jbalcomb | :) |
| 14:43.43 | DrkShdw | lol |
| 14:44.08 | jbalcomb | [TK]D-Fender: I finally got my polycom to autoprovision |
| 14:44.47 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 14:44.48 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 14:44.52 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 14:45.03 | [TK]D-Fender | jbalcomb : as in out of the box based on DHCP configurable stuff only? |
| 14:45.56 | jbalcomb | [TK]D-Fender: Yeah, its fetches DHCP with option 66, contacts the HTTP/FTP/TFTP server and grabs the firmware, sounds, and default configs. |
| 14:46.23 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:46.43 | *** join/#asterisk Tordah (n=Ross@213-152-55-237.dsl.eclipse.net.uk) |
| 14:46.46 | Winkie | jbalcomb: lol next-server |
| 14:47.06 | Tordah | Hey m8rs |
| 14:48.23 | Tordah | quick question, how do you add a prefix to the start of a call automatically? For example exten => _<add a 9 here>XXXXXXX,1,Dial(Zap/1/%{EXTEN}) |
| 14:48.59 | Tordah | tried searching for it and can't find any info on it=/ |
| 14:49.17 | [TK]D-Fender | Tordah : exten => XXXXXXX,1,Dial(Zap/1/9${EXTEN}) |
| 14:49.23 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
| 14:49.26 | kindor | helo! |
| 14:49.52 | Tordah | ahhhh i see |
| 14:49.56 | Tordah | you add it to the end. |
| 14:49.58 | Tordah | Thanks |
| 14:49.59 | Tordah | :D |
| 14:50.11 | [TK]D-Fender | np |
| 14:50.31 | Winkie | Tordah: more exactly, the first XXXX bit defines what extension runs the command listed later, and [TK]D-Fender is just adding a 9 before you include the matching string :) |
| 14:51.20 | Tordah | yeah, i understood that part, kind of logical really |
| 14:51.32 | Vec | Would I need to tell a PSTN adapter (sipura 3000), to forward incoming calls to asterisk or does it just have to be registered with asterisk ? |
| 14:51.42 | [TK]D-Fender | Winkie : He asked to ADD a prefix, not STRIP one. |
| 14:51.59 | [TK]D-Fender | Vec : You have to tell it where to send calls. |
| 14:52.04 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 14:52.31 | [TK]D-Fender | Vec : You could have it selectively send specific calls elsewhwew. Its a powerful little box actually. |
| 14:53.04 | Tordah | I only just got into Asterisk this week so pretty new to me still ;). I've been using example scripts and now I'm starting to make my own. Still working on accepting incoming calls from voip ;p |
| 14:54.29 | Vec | [TK]D-Fender : I am going to play around with the "Cfwd Sel1 Dest: " section on the web interface of the sipura, and see if I can figure it out. Unless you know off hand where to specify that ? |
| 14:55.37 | Tordah | now I have to wait for someone to stop using the phone so I can reload the configs=/ |
| 14:56.03 | [TK]D-Fender | in your dialplans you'll need something like <s0:> or something to forward ALL incoming calls on the SPA to *. |
| 14:56.33 | [TK]D-Fender | Vec : Go to www.voxilla.com and read their forums. There are several excellent guides on how to set this unit up for * |
| 14:56.50 | [TK]D-Fender | Tordah : Which configs? |
| 14:56.52 | Vec | [TK]D-Fender : thanks on my way. |
| 14:57.33 | Tordah | all of zem! |
| 14:58.36 | Tordah | I'm such a noob:< |
| 14:58.36 | Qb3rt | i looking to find a good queue_log reader.. it could be in php mysql or not.... can someone help me with this??? |
| 14:59.05 | _problem_ | Qb3rt: queue_stats by asteriskguru.com |
| 15:01.20 | Qb3rt | _problem_: its in PostgreSQL and i never did that... |
| 15:02.52 | _problem_ | Qb3rt: ya it is in Postgresql..well i believe that is not a problem...but few months ago they mentioned that future release will work with mysql also... |
| 15:05.40 | kindor | Hi, when compiling the addon package i get this error: app_addon_sql_mysql.c:15:22: asterisk.h: No such file or directory, when i place the asterisk.h in /usr/include it gives another error on lock.h. What can this be? |
| 15:05.55 | kindor | i use a debian system |
| 15:07.51 | *** join/#asterisk Precion (n=crhodes@adsl-75-7-75-29.dsl.milwwi.sbcglobal.net) |
| 15:10.06 | setti | kindor, wich version of mysql you have install? |
| 15:10.33 | kindor | setti: 4.0.24 |
| 15:10.47 | Tordah | hypothetically should this work: |
| 15:10.48 | Tordah | [bt_pstn] ; allow incoming analog calls to be recieved |
| 15:10.49 | Tordah | exten => s,1,Dial(SIP/dg,25,t,r) |
| 15:12.14 | *** join/#asterisk Chai_Sangeen (n=Chai-San@82.194.38.2) |
| 15:12.20 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 15:12.23 | Chai_Sangeen | hi guyz |
| 15:12.42 | Tordah | lo |
| 15:12.50 | Chai_Sangeen | anyone one can help with eyebeam ports? |
| 15:15.05 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 15:15.56 | AlexCTI | Hi, Any digium guy here? |
| 15:16.06 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:16.09 | Chai_Sangeen | what ports other than the 5060 and 10000 20000 do i have to open? for eyebeam? |
| 15:16.45 | file | AlexCTI: sure, what's up? |
| 15:16.47 | Chai_Sangeen | when it used to work, i think i had port 8000 open but not sure about that.. tried it but no luck. |
| 15:18.20 | Hmmhesays | how long as domain been part of sip.conf? |
| 15:18.27 | jbalcomb | Anyone using the DHCP next-server <ip-address> option? |
| 15:18.39 | *** join/#asterisk Tier_1 (n=tier@c-67-176-28-65.hsd1.co.comcast.net) |
| 15:19.05 | Winkie | aye but not for phones, what's up jbalcomb |
| 15:21.35 | jbalcomb | Winkie: just don't see much documentation for it. The tftp-server-name says that some stuff doesn |
| 15:22.06 | jbalcomb | some stuff will treat it the same as the Server Name option but I don't see any option server-name option |
| 15:22.13 | jbalcomb | wondering what next-server does. |
| 15:22.15 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 15:22.20 | _problem_ | gandhijee: 100 dead in mumbai blasts. |
| 15:22.25 | AlexCTI | File, I have a question, I have a borad with 4 PRIs and it was working perfect but now one of the ports is not working, so I swicth cables to test but definitly is the board which do not provide de D-channel, but if you see on zttool loosk like ok |
| 15:22.46 | Winkie | jbalcomb: as far as i understand next-server simply tells whatever the DHCP agent is where to contact next, usually to download a file |
| 15:22.57 | file | AlexCTI: call support :) |
| 15:23.03 | Winkie | indeed |
| 15:23.13 | Winkie | are you getting alarms on the interface or anything obvious? |
| 15:23.34 | AlexCTI | nothing, not alarms |
| 15:23.35 | Winkie | jbalcomb: what exactly are you trying to do? I had a hell of a time getting PXE booting working flawlesslyu |
| 15:23.55 | Winkie | AlexCTI: what does zap show state (or whatever it is) give you? |
| 15:24.09 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 15:24.11 | Winkie | zap show status :) |
| 15:24.27 | AlexCTI | Winkie, yes it show status fine, it is a zaptel board |
| 15:24.54 | AlexCTI | The link on the board is OK, solid green |
| 15:24.56 | Winkie | AlexCTI: it shows "OK"? is your ISP getting any alarms from it / can you make any calls out? |
| 15:25.35 | jbalcomb | Winkie: What is PXE? |
| 15:26.22 | AlexCTI | Winkkie, on the ISP site it doesn't send the D, and I can not make outbound calls |
| 15:26.23 | jbalcomb | Winkie: I'm just trying to get my phone to fetch via DHCP to get to my HTTP/FTP/TFTP server and grab default configs |
| 15:27.11 | Winkie | jbalcomb: PXE is a type of network boot, is this phone not connecting at all? |
| 15:27.21 | Winkie | AlexCTI: are you sure your B / D channel configuration is correct on that span? |
| 15:27.26 | eKo1 | Anybody know what the cheapest motherboard with PCI 3.3V is for my TE410P? |
| 15:28.08 | AlexCTI | winkie, I'm positive, I was working before perfect and all othere ports are working with the same config. |
| 15:28.16 | jbalcomb | Winkie: it is now actually and i have not implemented next-server so its not needed. i'm just trying to flesh out my knowledge and understanding of DHCP. |
| 15:29.02 | Winkie | AlexCTI: it's an interesting problem, i'm assuming you've restarted, checked /proc/zaptel and reinserted the card? Did you mention you'd swapped T/E1s round and the same port caused a problem? |
| 15:29.08 | Tordah | is there a certain way you can check if the number is greater than a certain length when doing extensions? Greater than 8> etc |
| 15:29.33 | Winkie | jbalcomb: fair enough, well i'll PM you my config using next-server |
| 15:30.57 | jbalcomb | cool |
| 15:31.25 | Winkie | Tordah: in what context? |
| 15:31.32 | AlexCTI | Winkie, I restart zaptel, modules and asterisk, I reboot the server, I swapped the T1s and you right it only happen in that port, but I didn't swapped teh board to other slot. |
| 15:31.32 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 15:31.36 | Tordah | when dialing a number |
| 15:32.00 | Tordah | exten => _(if>8length) |
| 15:32.11 | Winkie | AlexCTI: other than that i'd imagine you have a hardware issue and would contact support |
| 15:34.29 | AlexCTI | Ok, i'll contact support, thanks anyway.. |
| 15:34.35 | *** join/#asterisk Galeras (n=Galeras@litigaractivos1.att.net.co) |
| 15:35.30 | Galeras | Hi, Please give me some advice about how to detect a hangup on a Grandstream HT488 which FXO port is connected to an old panasonic KX-TEM824. This Panasonic doesn't provide polarity reversal. It just plays a busy tone.. |
| 15:35.32 | *** join/#asterisk saftsack (n=saftsack@p54A7EF6D.dip.t-dialin.net) |
| 15:38.42 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:39.04 | Tordah | I guess that is a bit too much to ask of asterisk ^^ |
| 15:40.08 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 15:40.12 | Bert- | helllooooo there :) |
| 15:40.35 | Bert- | aterisk more and more each day :d |
| 15:40.55 | Bert- | (forgot 'rox') |
| 15:41.42 | Bert- | anyway I've a little question about voicemailmain : Is it possible to restrict the IVR ?? There is too much options, and users are lost |
| 15:42.18 | Bert- | I can't see any tip about that so I ask here ... I can modify the code, but I don't know I it's legal |
| 15:42.48 | [TK]D-Fender | Bert- : you have the sorce... get coding! |
| 15:42.55 | Bert- | hehe :) |
| 15:42.59 | kindor | anyone knows what goes wrong here; http://paste.ubuntu-nl.org/17768? |
| 15:43.01 | [TK]D-Fender | Bert- : Short answer = NO. |
| 15:43.18 | Bert- | okay so no other way |
| 15:43.19 | Bert- | the I'll modify code |
| 15:43.29 | Bert- | should I post any thing I'll do somewhere ? |
| 15:43.36 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 15:44.08 | [TK]D-Fender | Bert- : Depends how you do it and how much work it is to redo it when a new release of that app comes along. |
| 15:44.39 | [TK]D-Fender | Bert- : if you make your patch an easy thing to configure they may integrate it into * |
| 15:44.39 | Bert- | hmm |
| 15:44.58 | Winkie | i need to SVN up asterisk and check if any of my patches need modifying |
| 15:45.08 | Winkie | i sent off my disclaimer the other day but have been way too busy to repatch everything |
| 15:45.19 | Bert- | the fact is I'll modify code to match what my boss wants, so it may be not interresting for community |
| 15:45.30 | Bert- | as I'll only remove some functionnalities |
| 15:51.06 | Tili | how can i tell if digium card is setup for T1 or E1 |
| 15:51.27 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 15:58.10 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:01.36 | *** join/#asterisk glLoadIdentity (n=tyn@195.175.21.46) |
| 16:04.59 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:05.42 | jbalcomb | ok, so the GXP-2000 defaults to fm.granstream.com for the config server and override via DCHP option 66 is off... |
| 16:06.40 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 16:06.41 | jbalcomb | Has anyone worked out a good way to get the GXP-2000 to autoprovision? |
| 16:06.57 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 16:07.00 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 16:13.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.170) |
| 16:15.06 | *** join/#asterisk Nix (n=Nix@195.175.21.46) |
| 16:17.49 | jbalcomb | Maybe a firewall rule to redirect fm.grandstream.com back to my tftp server? |
| 16:19.13 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 16:19.48 | Bert- | tftp sucks because UDP don't have state |
| 16:19.50 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 16:20.00 | *** join/#asterisk hi365 (n=FreePBX9@bzq-88-155-37-190.red.bezeqint.net) |
| 16:20.00 | Bert- | you have to open a range ports |
| 16:20.17 | Bert- | the source port used by you grandstream should be into this range |
| 16:20.34 | Bert- | it is the way i used to update granstream through TFTP |
| 16:20.57 | hi365 | Hi! I get the following erros when runing wancfg on a songoma a200 |
| 16:20.58 | hi365 | Failed to get Card Type from selected card line!! line: AFT-A200-SH SLOT=14 BUS=2 IRQ=5 CPU=A PORT=PRI |
| 16:21.03 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:21.31 | hi365 | what does it mean? why can i start my zaptel drivers properly? |
| 16:21.57 | hi365 | I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
| 16:23.29 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:23.33 | [TK]D-Fender | jbalcomb : Local DNS <- |
| 16:24.11 | jbalcomb | [TK]D-Fender does it for sure use my option name-server declaration? |
| 16:24.57 | hi365 | Can anyone help troubleshot a Singoa card? |
| 16:24.59 | [TK]D-Fender | jbalcomb : No clue, just suggesting the easy fix for redirecting the failover. |
| 16:26.02 | jbalcomb | [TK]D-Fender: ok, I added an SOA for grandstream.com to my internal dns server. testing... |
| 16:26.07 | rob0 | If they use DNS to resolve fm.granstream.com ... yes |
| 16:26.27 | rob0 | I would ONLY hijack the specific RR you need tho. |
| 16:26.46 | rob0 | fm.granstream.com, not grandstream.com |
| 16:27.03 | rob0 | (I copied your typo above :) ) |
| 16:27.24 | rob0 | fm.grandstream.com, not grandstream.com |
| 16:28.14 | syzygybsd | but that takes all the fun out of trying to figure out why it still doesnt' work |
| 16:29.09 | kay2 | when I do an Originate, how can I know what's the channel ID to it ? |
| 16:29.57 | jbalcomb | [TK]D-Fender fm.grandstream.com is an alias for tftp.voip.imtco.com |
| 16:30.05 | jbalcomb | so far so good |
| 16:30.16 | jbalcomb | need to set up HTTP on the TFTP server |
| 16:31.22 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 16:32.45 | jbalcomb | I'm sad to have to install web server app just to get my gxp-2000 to autoprovision. |
| 16:33.04 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 16:33.06 | *** join/#asterisk laurentc92320 (n=l@84-72-235-191.dclient.hispeed.ch) |
| 16:33.14 | [TK]D-Fender | I have a solution.... actually SEVERAL... and all of them corrosive ;) |
| 16:33.18 | drray | :) |
| 16:33.31 | laurentc92320 | hi all |
| 16:33.38 | rowter | anyone know if a digium card te05P could handle E1 and T1 simultanously? |
| 16:34.00 | rowter | anyone know if a digium card te405P could handle E1 and T1 simultanously? |
| 16:34.00 | drray | on different spans? |
| 16:34.12 | laurentc92320 | someone know how to setup a streaming musiconhold? |
| 16:34.17 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 16:34.23 | jbalcomb | [TK]D-Fender: I don't think that sort of work is covered by the PO... |
| 16:34.37 | [TK]D-Fender | jbalcomb : Its complimentary :D |
| 16:35.12 | jbalcomb | [TK]D-Fender oh, well right on then! ;) |
| 16:35.36 | *** join/#asterisk Chai_Sangeen (n=alialjis@82.194.38.2) |
| 16:37.07 | topping | has anyone dealt with nufone and this number move they have been trying to do? |
| 16:37.19 | topping | i have an 866 with them and it's been down for months |
| 16:37.35 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 16:37.53 | topping | tres frustrating |
| 16:38.17 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 16:40.07 | Nix | sangoma cards can.. dont know about digium |
| 16:40.59 | Strom_C | I believe the TE405P can handle T1/E1 on separate spans of the same card |
| 16:41.18 | Strom_C | but don't quote me on that one; I have no firsthand experience attempting to set that up |
| 16:41.34 | eKo1 | Yes it can. |
| 16:41.39 | eKo1 | I've done it. |
| 16:41.44 | Strom_C | there you go then |
| 16:41.45 | CunningPike | Has anyone had any success with installing HUDLite? |
| 16:41.58 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:41.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:42.25 | pdtmobile | CunningPike: still at it eh? |
| 16:42.53 | Chai_Sangeen | same here tried installing HUDLite on my asterisk@home 2.8 no luck.... |
| 16:43.11 | CunningPike | pdtmobile: :D I have the server side running, and when I connect with a regular IRC client, all the right things seem to happen. But I can't get the HUDLite client itself to connect..... |
| 16:43.15 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 16:43.26 | pdtmobile | hmm |
| 16:44.05 | CunningPike | It connects, and the server issues the invites to the chatrooms, but the client never responds. A 'real' IRC client works just fone |
| 16:44.09 | CunningPike | s/fone/fine/ |
| 16:44.30 | jbalcomb | [TK]D-Fender: IT WORKED!! [Tue Jul 11 12:44:01 2006] [error] [client 10.0.101.151] File does not exist: /var/lib/tftproot/gs/ring1.bin |
| 16:44.52 | CunningPike | Time to try FOP :D |
| 16:46.01 | [TK]D-Fender | jbalcomb : Nothing says thank-you quite like PayPal ;) |
| 16:47.19 | jbalcomb | [TK]D-Fender: HA! you didn't tell me anything I didn't already think of but I did assume they were using thier own name servers as well. |
| 16:47.22 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 16:48.15 | Winkie | jbalcomb: oh you got it working then? :) |
| 16:48.37 | Winkie | also how the hell do you guys have T1 and E1 in the same place, damn telcos :) |
| 16:48.49 | Luke-Jr | yay ulaw =p |
| 16:48.53 | Luke-Jr | w/ ulaw and voip |
| 16:48.57 | Luke-Jr | (just telling everyone since he hears so often that voip can't do fax =p) |
| 16:49.11 | mut | can't do fax reliably* |
| 16:49.19 | Luke-Jr | mut: works for me =p |
| 16:49.21 | mut | sure |
| 16:49.24 | mut | works |
| 16:49.25 | jbalcomb | Winkie: yeah, my polycoms and grandstreams now can pull thier configs straight out of the box. |
| 16:49.34 | mut | but may not for every situation |
| 16:49.45 | Luke-Jr | pfft, only if you have phone issues in general |
| 16:49.47 | Luke-Jr | like huge latency |
| 16:49.50 | syzygybsd | Luke-Jr: a lot of it depends on the fax machine sending or recieving |
| 16:49.52 | [TK]D-Fender | Luke-Jr : I dunno.... you sent 4 pages and only got 2! ;) |
| 16:50.03 | Luke-Jr | [TK]D-Fender: no, I got 2 and faxed them back twice |
| 16:50.15 | mut | if the fax machine is retarded then it won't work |
| 16:50.17 | Luke-Jr | [TK]D-Fender: I'm at one end |
| 16:50.19 | Hmmhesays | what is euroisdn, is that etsi? |
| 16:50.31 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:50.33 | Luke-Jr | mut: isn't there some standard fax machines need to follow or else they aren't really 'fax' =p |
| 16:50.34 | Luke-Jr | ? |
| 16:50.45 | syzygybsd | well, the fax machines have a higher tolerance for errors then Voip i think, or the other way around, not sure |
| 16:50.45 | Luke-Jr | or is fax older than standards? |
| 16:50.46 | mut | yea, but all fax machines aren't created equally |
| 16:50.58 | topping | are there better codecs to use for fax than others? |
| 16:51.06 | Luke-Jr | lol |
| 16:51.10 | Luke-Jr | don't try fax w/ any compression =p |
| 16:51.17 | topping | :) |
| 16:51.19 | syzygybsd | ^^^ |
| 16:51.26 | Winkie | i use fax over ZAP but that's about all i'd ever stretch to |
| 16:51.28 | Hmmhesays | can someone give me a pointer on a cisco as5300? |
| 16:51.29 | Luke-Jr | *then* you'll fail miserably, I suspect |
| 16:51.33 | Winkie | maybe fax over IAX but only if i could dedicate some bandwidth to it |
| 16:51.39 | mut | i've had a fax work via gsm |
| 16:51.44 | Luke-Jr | mut: really? O.o |
| 16:51.47 | mut | it just sucks a lot |
| 16:51.51 | mut | and hardly ever works |
| 16:51.53 | Winkie | and by a lot you mean A LOT |
| 16:51.54 | syzygybsd | lol.. that silence suppression must be great... |
| 16:51.55 | Luke-Jr | I'd expect all compression to cut out the frequencies fax uses |
| 16:51.57 | jbalcomb | Can someone write a BASH script for me? It just needs to copy 000000000000.cfg to <000B82XXXXXX>.cfg where XXXXXX is every conceivable combination of legal mac address characters. |
| 16:52.08 | Winkie | you're joking me right? |
| 16:52.14 | jbalcomb | =) |
| 16:52.17 | Luke-Jr | jbalcomb: ...why? |
| 16:52.28 | [TK]D-Fender | jbalcomb : SLOPPY! ;) |
| 16:52.33 | mut | heh |
| 16:52.33 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 16:52.44 | topping | jbalcomb: http://www.mturk.com/mturk/welcome |
| 16:52.45 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 16:52.50 | jbalcomb | because the GXP-2000 doesn't look for a generic default config file anymore. it only looks for its <mac address>.cfg. |
| 16:52.51 | Winkie | topping: you wouldn't need mturk for it |
| 16:53.03 | Winkie | jbalcomb: so add in the mac addresses manually, or for the love of god symlink them |
| 16:53.04 | syzygybsd | jbalcomb: you do know that is 16777216 files right? |
| 16:53.11 | Hmmhesays | anyone anyone? |
| 16:53.12 | Winkie | syzygybsd: you know it's more than that right? |
| 16:53.13 | Luke-Jr | jbalcomb: why not hack the server to send a generic if the file isn't found? |
| 16:53.13 | topping | Winkie: why not? you could get thousands of replies, one would be right! |
| 16:53.22 | SplasPood | hack the gibson?! |
| 16:53.24 | syzygybsd | Winkie: nope |
| 16:53.31 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:53.33 | Winkie | syzygybsd: it's in the reigon of several billion iirc |
| 16:53.48 | Luke-Jr | Winkie: that's IPv6 |
| 16:53.50 | tanvalmg | Has anyone installed Asterisk in a Compaq EVO workstation? |
| 16:53.51 | syzygybsd | Winkie: he only wants the last 6 characters of the mac address created |
| 16:53.54 | Winkie | Luke-Jr: this is MAC not ipv4 |
| 16:54.05 | Luke-Jr | syzygybsd: did he say that? |
| 16:54.07 | Winkie | syzygybsd: i suppose so really if it's already got a company identifier |
| 16:54.10 | mut | if he can;t make a bash script |
| 16:54.14 | mut | he can't hack the server |
| 16:54.17 | syzygybsd | Luke-Jr: yes |
| 16:54.17 | oscarh | is the manager command Queues not supposed to send back a package? |
| 16:54.20 | Winkie | Luke-Jr: mac addresses are assigned by blocks so yeah he sorta would |
| 16:54.22 | jbalcomb | syzygybsd: not exactly but i knew it would be a whole hell of a lot |
| 16:54.33 | Luke-Jr | ah, I see |
| 16:54.35 | oscarh | but rather just the thext No Queues if there are none? |
| 16:54.39 | Luke-Jr | 000B82 is the MAC prefix |
| 16:54.45 | Winkie | jbalcomb: just `ln -s lol.cfg macaddress.cfg` when you add a new phone? |
| 16:54.51 | syzygybsd | :) reding is phun! |
| 16:55.05 | jbalcomb | Winkie: i dont want to have to add a new phone. :(.. |
| 16:55.17 | Winkie | jbalcomb: too bad |
| 16:55.26 | mut | it's job security man |
| 16:55.39 | mut | when you get fired you can charge them $100/hr to add phones |
| 16:55.43 | mut | o_O |
| 16:55.43 | Winkie | confirming that |
| 16:55.54 | Luke-Jr | lol |
| 16:56.08 | jbalcomb | Winkie: I'm thinking a perl script that tails the HTTP/FTP/TFTP log for failed requests for <mac address>.cfg and then automatically does the `ln -s 000000000000 <mac address>.cfg` |
| 16:56.11 | mut | really |
| 16:56.25 | mut | think about it.. how many other people could that company find in your area in a timely manner to do the job? |
| 16:56.41 | mut | they can't sit on a phone for a new person and/or dead phone for a week waiting |
| 16:56.45 | mut | instead they phone up you |
| 16:56.46 | mut | :P |
| 16:56.49 | jbalcomb | mut: the guy that handled the phone system before i took over still works here. |
| 16:57.05 | jbalcomb | mut: and besides, i really hate "job security" |
| 16:57.27 | jbalcomb | mut: if i could fix everything, automate the entire IT dept., and get the hell out of here I would be a happy man. |
| 16:57.27 | mut | heh job security oozes out my ass |
| 16:57.31 | mut | it's too bad i get paid shit |
| 16:57.36 | syzygybsd | jbalcomb: just have a cron that checks the arp table every minute and adds a cfg for all entries |
| 16:57.44 | [TK]D-Fender | jbalcomb : Thats what I'd think too... Then again I think you should just use that fancy web server you're now mounting to let the user input 1 friggen field into it to create the config :) |
| 16:57.59 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.170) |
| 16:58.24 | [TK]D-Fender | mut : You in manure processing? ;) |
| 16:58.25 | jbalcomb | syzygybsd: hrmm.. that isn't a bad idea. I already have arpwatch running and a perl script that grabs the MAC and IP and loads them into a MySQL DB. I could easily add a function to do that. |
| 16:58.45 | mut | [TK]D-Fender: no but sadly it'd probly pay better |
| 16:58.59 | Luke-Jr | mut: maybe that's *why* you get paid like that? O.o |
| 16:59.09 | [TK]D-Fender | jbalcomb : You auto-provisioner is starting to ack more and more like a VIRUS! |
| 16:59.19 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:59.23 | qseek | hello all |
| 16:59.30 | mut | why would they pay me crap |
| 16:59.30 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 16:59.43 | qseek | hey jbalcomb |
| 16:59.44 | mut | if i leave to go find a job somewhere else that pays better.. |
| 16:59.46 | syzygybsd | manure processing is a shitty job |
| 16:59.52 | jbalcomb | [TK]D-Fender: well, I already have the means in place to automate this system. It's just that by the time I'm done it's going to need its own $2000 server to run on. ;) |
| 16:59.53 | Luke-Jr | mut: because you ensure 'job security' ? =p |
| 16:59.57 | mut | they're screwed on all my 'job security' |
| 17:00.02 | jbalcomb | hayo!~ qseek |
| 17:00.12 | mut | heh |
| 17:00.18 | qseek | so how is the grandstream autoconfig coming along jbal |
| 17:00.24 | mut | i ensure MY job security |
| 17:00.26 | [TK]D-Fender | jbalcomb : All so you don't have to configure 5 more phones. Would you like your elephant gun sir? ;) |
| 17:00.31 | Luke-Jr | mut: so why not quit and let that job security kick in? |
| 17:00.32 | mut | i get rid of everyone elses in the company by automating their jobs |
| 17:01.00 | mut | that only lasts until they learn it all |
| 17:01.01 | [TK]D-Fender | jbalcomb : SHOOT IT! SHOOT IT! |
| 17:01.08 | Luke-Jr | mut: but it's under your terms until then =p |
| 17:01.13 | jbalcomb | [TK]D-Fender: I'd prefer my flame thrower! |
| 17:01.18 | mut | true |
| 17:01.26 | mut | but then i lose my free internet |
| 17:01.28 | mut | phoneline |
| 17:01.48 | mut | can't make car/house/utility/other payments |
| 17:02.03 | mut | and i can't lose my internet! |
| 17:02.08 | mut | i'de DIE |
| 17:03.38 | *** part/#asterisk Chai_Sangeen (n=alialjis@82.194.38.2) |
| 17:03.38 | drray | you understand the path to power, is adding layers of complexity |
| 17:03.45 | drray | not simplifying things |
| 17:04.02 | Hmmhesays | anyone familiar with as5300's I could use a hand |
| 17:04.14 | drray | quit thinking like a nerd, and spend more time being a greedy asshole. |
| 17:04.25 | Luke-Jr | mut: why? your terms can easily be more income than before |
| 17:04.42 | Luke-Jr | and who needs a phone line? ;) |
| 17:04.52 | *** join/#asterisk kiong (i=BekokBau@bb58-185-165-38.singnet.com.sg) |
| 17:05.00 | Luke-Jr | this is #asterisk after all |
| 17:05.06 | mut | phoneline = voi |
| 17:05.07 | mut | p |
| 17:05.34 | kiong | any recommendation for a to z termination, which offer fair price and good quality? |
| 17:05.52 | Luke-Jr | mut: then you need what? $5/mo for that? =p |
| 17:06.02 | Luke-Jr | kiong: VoipJet |
| 17:06.31 | Luke-Jr | I think VoipJet's the only VoIP company I've dealt with that hasn't had any problems |
| 17:06.32 | kiong | luke, are you using it ? |
| 17:06.34 | Luke-Jr | yes |
| 17:06.42 | TrixVox | voipjet? haha |
| 17:06.42 | Luke-Jr | have been for 1-2 years |
| 17:06.47 | kiong | ow... |
| 17:06.49 | Luke-Jr | ... |
| 17:06.56 | kiong | i must put deposit 1st right? |
| 17:06.59 | Luke-Jr | of course |
| 17:07.19 | kiong | oke... going to their site |
| 17:07.21 | Luke-Jr | something wrong with that? =p |
| 17:07.40 | Luke-Jr | I just wish they offered origination |
| 17:07.44 | TrixVox | How many minutes will you be using per month, kiong? |
| 17:08.21 | TrixVox | And is it for home or business use? |
| 17:09.13 | laurentc92320 | same question for home TrixVox |
| 17:09.26 | Luke-Jr | TrixVox: do many business-with-lots-of-minutes use VoIP? ;p |
| 17:09.53 | mut | i have lots-of-customers-with-minutes using voip |
| 17:09.57 | mut | ;P |
| 17:10.00 | TrixVox | Luke-Jr: Are you sure you're in the right channel? |
| 17:10.26 | Luke-Jr | TrixVox: I'd suspect a PRI might be cheaper with a lot of use |
| 17:10.45 | Luke-Jr | Asterisk isn't VoIP-only... |
| 17:10.53 | mut | i'm still suprised |
| 17:10.59 | mut | it takes ~350 people to fill a single pri |
| 17:11.19 | Luke-Jr | mut: a T1? |
| 17:11.26 | mut | yeh |
| 17:11.35 | Luke-Jr | O.o |
| 17:11.59 | Hmmhesays | as5300 anyone? |
| 17:12.00 | ahsan | exit |
| 17:12.08 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:12.19 | mut | i have a 5350 if ya wanna buy it |
| 17:12.20 | mut | :P |
| 17:12.23 | Luke-Jr | lol |
| 17:12.57 | *** part/#asterisk ahsan (n=ahsan@host-83-146-53-46.bulldogdsl.com) |
| 17:16.08 | [TK]D-Fender | 5150 "rocks"! ;) |
| 17:16.37 | laurentc92320 | hi nobody know how to setup correctly a streaming musiconhold? :) |
| 17:16.56 | syzygybsd | oh.. lots of people know, and if you read online you could too! |
| 17:16.57 | qseek | guys anyone use snom 300 |
| 17:17.11 | qseek | what is the consensus..good phone or bad? |
| 17:17.15 | laurentc92320 | I have try but not working |
| 17:17.37 | syzygybsd | what isn't working about it... |
| 17:17.58 | [TK]D-Fender | qseek : Where are you located? |
| 17:18.09 | qseek | dallas |
| 17:18.10 | mut | squeek! |
| 17:18.19 | mut | ftw |
| 17:18.33 | qseek | neat mut |
| 17:18.34 | laurentc92320 | syzygybsd I have try to setup it but I hear nothink on the phone |
| 17:18.49 | [TK]D-Fender | qseek : Then forget sno, go Polycom. Superior Quality, and nearly the same price. |
| 17:18.51 | syzygybsd | well is it running? |
| 17:19.08 | qseek | yeah but i am having configuring issues with polycom |
| 17:19.21 | qseek | and the 301 does not have full duplex speaker |
| 17:19.48 | [TK]D-Fender | qseek : IP 430 does. onlya few $ more expensive than the Snom 300 |
| 17:19.54 | [TK]D-Fender | qseek : As for config problems, whats up? |
| 17:19.57 | qseek | how much is the ip430 |
| 17:20.03 | *** join/#asterisk saftsack (n=oliver@p54A7E0CC.dip.t-dialin.net) |
| 17:20.22 | laurentc92320 | this extension doesn't work too I see i connect to the website but nothink in the phone exten => 466,3,MP3Player(http://sip-server.dnsalias.net/QuajiroPromo.mp3) |
| 17:20.52 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 17:20.52 | surfdue | ello |
| 17:21.00 | Zeeek | ood orning |
| 17:21.07 | syzygybsd | ol |
| 17:21.10 | surfdue | I have some errors http://pastebin.ca/85079 < there is the error, extensions.conf and sip.conf in that pastebin. Thanks |
| 17:21.25 | Zeeek | as of this minute, all vowels are deprecated |
| 17:21.49 | syzygybsd | wll thts n fn |
| 17:21.55 | surfdue | whats a vowel |
| 17:22.01 | surfdue | lol |
| 17:22.05 | qseek | TKD any ideas? |
| 17:22.42 | syzygybsd | surfdue: what are you trying to do |
| 17:22.53 | surfdue | simply trying to get it to call my pap2 ext 200 |
| 17:23.02 | surfdue | it seems to be getting 404 and sip show peers shows em |
| 17:23.04 | Strom_C | what is "it" |
| 17:23.10 | Strom_C | what are you calling from |
| 17:23.15 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.94) |
| 17:23.21 | *** join/#asterisk stoffell_h (n=stoffell@d51A5839B.access.telenet.be) |
| 17:23.26 | [TK]D-Fender | qseek : IP 430 = $150 USD and has full duplex, built in PoE, and lighted indicators |
| 17:23.33 | surfdue | http://pastebin.ca/85083 |
| 17:23.43 | syzygybsd | what is streaming musiconhold |
| 17:23.46 | Strom_C | surfdue: stop pasting things at me and answer my questions |
| 17:23.48 | syzygybsd | what is 72.40.10.150 |
| 17:23.49 | surfdue | Strom_C, calling the 1-800 number and trying to get it to go to my sip |
| 17:23.54 | syzygybsd | sorry, stupid copy and paste ... |
| 17:23.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:24.00 | surfdue | syzygybsd, thats my sip pap2 |
| 17:24.12 | mocker | Anyone recommend cheap places for asterisk schwag? |
| 17:24.12 | syzygybsd | k, that doesn't have extension 200 on it |
| 17:24.14 | Strom_C | so you're coming in via an outside IAX trunk and trying to call a phone attached to your PAP2? |
| 17:24.40 | syzygybsd | and since we dont' have any configs from there, I can't tell you why |
| 17:24.45 | surfdue | Strom_C, im coming in from asterlink my provider via sip i think, and im trying to call out to my pap2 |
| 17:25.36 | kiong | huuuh i try to register with voipjet yet the email never arrive |
| 17:25.42 | Strom_C | surfdue: how is your PAP2 configured? |
| 17:25.58 | surfdue | it says online, and it can call out |
| 17:25.59 | syzygybsd | ya, we need your pap2 configuration |
| 17:26.12 | surfdue | just ip then username and display is 200, and the password |
| 17:26.39 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 17:27.37 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
| 17:28.08 | syzygybsd | surfdue: can you get the config for your pap2 to us somehow? |
| 17:28.10 | Egonis | I have alsa loaded into asterisk, it plays fine with noxmms -- when I try to Dial(Console/dsp) I get -- Unable to create channel of type 'Console' (cause 66 - Channel not implemented) |
| 17:28.51 | syzygybsd | you can dial console? |
| 17:28.53 | surfdue | http://img461.imageshack.us/img461/1774/untitled0bx.jpg |
| 17:29.18 | syzygybsd | surfdue: add ans call without reg to yes |
| 17:30.18 | surfdue | k |
| 17:30.24 | rene- | hey, i have a realtime sip setup, my grandstream never lose registration but my PAP2 lose and gain registration all the time, i added |
| 17:30.24 | rene- | rtcachefriends=yes which helped a bit and also rtautoclear=no, the grandstream works like a champ, has anyone seen something like this before? |
| 17:30.26 | surfdue | done |
| 17:31.07 | Egonis | Should I just use OSS? because when I use oss it complains about not being able to open /dev/dsp |
| 17:31.24 | syzygybsd | rene-: shorten you register expires in your pap2 configuration |
| 17:31.31 | surfdue | still dosnt work syzygybsd |
| 17:32.09 | rene- | syzygybsd: they are at 60secs |
| 17:32.17 | rene- | what do you suggest |
| 17:32.25 | syzygybsd | that should be plenty low... |
| 17:32.27 | surfdue | syzygybsd, http://pastebin.ca/85097 |
| 17:33.41 | syzygybsd | can you post the sip tab of your pap2 configuration? |
| 17:34.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:34.30 | [TK]D-Fender | Hmmhesays : Hey, remember that MIDI master controller I showed you that I figiured I could get maybe a bit lower than the $460 I mentioned? Well they updated their price and I'm getting it for $375! A friggen STEAL at that price :) |
| 17:34.44 | surfdue | yes it already was |
| 17:34.49 | Hmmhesays | awesome |
| 17:34.57 | eKo1 | What is a MIDI master controller? |
| 17:34.57 | syzygybsd | where? |
| 17:35.03 | syzygybsd | I just see line2 |
| 17:35.33 | surfdue | http://pastebin.ca/85079 |
| 17:35.36 | surfdue | at the bottom |
| 17:35.48 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:35.58 | syzygybsd | it would be an image.... |
| 17:36.06 | surfdue | what ya mean? |
| 17:36.07 | surfdue | OH |
| 17:36.09 | surfdue | 1 sec |
| 17:36.11 | syzygybsd | http://img461.imageshack.us/img461/1774/untitled0bx.jpg but the sip tab |
| 17:36.24 | Tier_1 | <Tier_1> ok who here knows how to do the mail command from a command line |
| 17:36.25 | Tier_1 | <Tier_1> on linux |
| 17:36.29 | [TK]D-Fender | eKo1 : http://www.m-audio.ca/products/en_ca/KeystationPro88-main.html |
| 17:36.29 | Tier_1 | please never had to do it |
| 17:36.38 | syzygybsd | lol.. copy and paste for the win! |
| 17:36.51 | surfdue | advanced or basic? |
| 17:37.08 | syzygybsd | Tier_1: I do... depends on how technical you want to get, you watn to telnet to the server, or not |
| 17:37.08 | Nugget | telnet is eeeeeeevil! |
| 17:37.24 | syzygybsd | is that an auto response to telnet? |
| 17:37.31 | Tier_1 | syz can you pm me |
| 17:37.44 | syzygybsd | nope, keep it in the channel |
| 17:38.02 | syzygybsd | pms are annoying |
| 17:38.16 | Tier_1 | i need to test a system and cc 22 users from the command line |
| 17:38.20 | [TK]D-Fender | syzygybsd : Its that time of the month ;) |
| 17:38.54 | jbalcomb | ~seen SoloFlyer |
| 17:39.07 | jbot | soloflyer <n=soloflye@eth13515.sa.adsl.internode.on.net> was last seen on IRC in channel #debian, 63d 15h 12m 4s ago, saying: 'wine_0.9.11-1'. |
| 17:39.07 | syzygybsd | lol, with this many people it could always be THAT time of the month |
| 17:39.38 | *** join/#asterisk glLoadIdentity (n=tyn@81.214.255.57) |
| 17:39.45 | syzygybsd | Tier_1: it depends on what mail command you have, what mail server you have etc |
| 17:40.16 | surfdue | syzygybsd, sorry here it is :) http://img111.imageshack.us/img111/7646/untitled0jp.jpg |
| 17:40.40 | syzygybsd | any chance you could do the user2 one as well... |
| 17:40.56 | surfdue | k |
| 17:42.34 | syzygybsd | Tier_1: start with 'man mail' |
| 17:43.07 | surfdue | http://img472.imageshack.us/img472/9919/untitled7pu.jpg |
| 17:43.41 | syzygybsd | it will likely end up as something like 'mail -s [subject] [to] < filetosend' |
| 17:45.29 | syzygybsd | why sorry... anyway, can you get calls on line 1? |
| 17:45.31 | [TK]D-Fender | Quick and silly php/file standards type question for EOL sequence. Isn't its supposed to be \l\n" for an MS CR/LF combo? Doesn't seem to be interpreting properly in Firefox. |
| 17:46.09 | syzygybsd | \r\n |
| 17:46.29 | jbalcomb | Can someone give me the ip for www.grandstream.com? |
| 17:46.47 | syzygybsd | 66.94.237.70 |
| 17:47.00 | [TK]D-Fender | syzygybsd : Ah, thanks... been so long I couldn't remember :) |
| 17:47.28 | syzygybsd | or if you really want to get into the nitty gritty chr(13)chr(10) |
| 17:47.50 | syzygybsd | or whatever php uses for that.. maybe it is char() |
| 17:48.07 | *** join/#asterisk ManuMendoza (n=ManuMend@200.69.182.135) |
| 17:48.25 | [TK]D-Fender | syzygybsd : Thanks, I would do taht but I want to work the "normal" way first. Chr brings me back to coding in gw-basic ;) |
| 17:48.29 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 17:48.59 | benjamin7062 | Is there a way to get voicemail to 'forward' emails with the attachment and not keep a local copy in a users mailbox? |
| 17:49.00 | syzygybsd | most basic I did was on my calculator in HS.. been a while |
| 17:49.22 | syzygybsd | benjamin7062: yes, in voicemail.conf add the email address, then delete=1 |
| 17:49.36 | benjamin7062 | delete=1 is what I was missing.. thaks |
| 17:49.37 | benjamin7062 | thanks |
| 17:49.55 | jbalcomb | thanks syzygybsd |
| 17:49.58 | syzygybsd | it has to be after a | though |
| 17:50.21 | benjamin7062 | syzygybsd, Is there a more up-to-date wiki for voicemail config than voip-wiki? |
| 17:50.31 | syzygybsd | http://www.voip-info.org//tiki-index.php?page=asterisk+config+voicemail.conf |
| 17:51.02 | syzygybsd | has all the info you need |
| 17:51.12 | benjamin7062 | sigh, mkay.. so I can't read.. that's a different problem. Thank you! |
| 17:51.24 | Egonis | I can play through /dev/dsp from noxmms, but not asterisk -- I have tried chan_oss and chan_alsa |
| 17:51.31 | syzygybsd | look at one of the examples, or just find delete on the page |
| 17:51.44 | benjamin7062 | Yeah, found 'er... |
| 17:51.48 | benjamin7062 | You got me on the right path |
| 17:52.33 | syzygybsd | surfdue: can you get calls on line1? |
| 17:53.15 | surfdue | um |
| 17:53.16 | surfdue | no |
| 17:53.36 | surfdue | line 1 says Registration State:Offline |
| 17:53.37 | syzygybsd | is it enabled at all |
| 17:53.39 | surfdue | it wo nt connect |
| 17:53.40 | surfdue | yes |
| 17:53.44 | syzygybsd | disable it |
| 17:53.53 | surfdue | k |
| 17:54.09 | syzygybsd | if it works I will let you know why |
| 17:54.16 | *** part/#asterisk ManuMendoza (n=ManuMend@200.69.182.135) |
| 17:54.46 | rene- | i have one PAp on SIP realtime and the other on SIP.conf lets see how it goes |
| 17:55.30 | surfdue | nope |
| 17:55.36 | surfdue | still get 404 erros |
| 17:55.37 | surfdue | wtf :( |
| 17:56.01 | syzygybsd | hmm.. well the reason I had you turn that off was because port 5060 was likely being used by both of them... |
| 17:56.23 | syzygybsd | my next question is going to be... do you have a phone plugged in to line2? |
| 17:56.58 | syzygybsd | well, both lines |
| 17:58.19 | surfdue | tes |
| 17:58.23 | surfdue | yes |
| 18:00.01 | surfdue | i keep getting SIP/2.0 404 Not Found |
| 18:00.17 | surfdue | To: <sip:200@72.40.10.150:5060>;tag=e7e72a58deb44c8ci0 |
| 18:00.18 | surfdue | From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as797f5fb9 |
| 18:00.27 | surfdue | Server: Linksys/PAP2-3.1.3(LS) |
| 18:00.30 | Tier_1 | ok the man page does not tell you how to send the mail it tells you how to write it |
| 18:00.31 | surfdue | so i know it sees it |
| 18:00.44 | Tier_1 | how do you end the line and get it to send |
| 18:00.55 | syzygybsd | ya, it is a configuration with your pap2.. |
| 18:01.49 | *** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 18:01.51 | *** join/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 18:02.12 | _-Jon-_ | Hey |
| 18:02.14 | surfdue | syzygybsd, should i reset it? |
| 18:02.18 | syzygybsd | na |
| 18:02.20 | _-Jon-_ | Anyone know if shido still comes around here? |
| 18:02.23 | surfdue | syzygybsd, do you want remote access? |
| 18:02.34 | syzygybsd | na... |
| 18:02.34 | surfdue | _-Jon-_, i know jon does :P |
| 18:02.49 | _-Jon-_ | Heh, that would be me :P |
| 18:03.31 | surfdue | syzygybsd, i can give you access to my vnc |
| 18:04.02 | _-Jon-_ | Hmm, alright, well is anyone from nufone around? :P |
| 18:04.15 | Trionnis | not likely |
| 18:04.22 | Trionnis | he seems to be hiding lately |
| 18:04.36 | Trionnis | probably tired of all the customers he screwed over bugging him |
| 18:04.45 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 18:05.02 | _-Jon-_ | How'd be screw them over? |
| 18:05.13 | Trionnis | the number fiasco from april/may |
| 18:05.22 | _-Jon-_ | Ohhh that |
| 18:05.25 | Trionnis | that's still ongoing, I might add |
| 18:05.33 | surfdue | hello? |
| 18:05.39 | surfdue | if anyone can help me i will give you vdt |
| 18:05.43 | surfdue | virtual desktop access |
| 18:05.45 | _-Jon-_ | Oh, I wouldn't know. I rarely use them, and only for outgoing |
| 18:06.17 | _-Jon-_ | Someone seems to have jacked my account though. TONS of calls places to NC from it |
| 18:06.44 | Trionnis | that sucks :( |
| 18:06.49 | _-Jon-_ | Not sure if it's a glitch on their end or what |
| 18:06.54 | _-Jon-_ | I know |
| 18:06.58 | surfdue | hmm |
| 18:07.02 | Trionnis | well, I wish you luck in resolving it, but I wouldn't hold your breath |
| 18:07.02 | surfdue | i have alot of calls to alaska |
| 18:07.03 | surfdue | :| |
| 18:07.17 | *** join/#asterisk So3kris (n=jan-will@soekris.xs4all.nl) |
| 18:07.24 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:07.27 | So3kris | hello |
| 18:07.29 | surfdue | seems those pesky eskimos like to use my minutes. |
| 18:07.30 | surfdue | hi |
| 18:07.33 | surfdue | lol |
| 18:07.36 | Trionnis | hi So3kris |
| 18:07.43 | surfdue | syzygybsd, ? |
| 18:07.49 | syzygybsd | reading, sorry |
| 18:07.52 | surfdue | :P |
| 18:08.12 | _-Jon-_ | Yeah I'm not expecting them to do anything about it. Though I am debating calling this Sharon E Campen person to find out who the hell she's been talking to for 40 minutes at a time |
| 18:08.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-126-111.red.bezeqint.net) |
| 18:08.33 | Trionnis | it's entirely possible it got brute forced |
| 18:08.40 | Trionnis | there's been some of that going around recently |
| 18:08.44 | _-Jon-_ | Oh really? |
| 18:08.56 | Trionnis | change your password and see if it stops |
| 18:09.07 | So3kris | I have a asterisk server. But i can't get the dialing rights good. Whats the place to say person 1 may call out-side and person 2 not. But the may call eathother |
| 18:09.07 | *** join/#asterisk kiong (i=BekokBau@bb58-185-165-38.singnet.com.sg) |
| 18:09.08 | surfdue | jon call her |
| 18:09.16 | surfdue | be like WTF you doin on my system man.. |
| 18:09.16 | surfdue | lol |
| 18:09.39 | _-Jon-_ | lol. "btw your husband is having an affair *click*" |
| 18:09.41 | surfdue | it cost like $1 for an hour of talk time so i wouldnt really mind but hey :) |
| 18:09.41 | [TK]D-Fender | So3kris : Set each user in a different context |
| 18:09.43 | Trionnis | haha |
| 18:09.44 | Trionnis | ouch |
| 18:09.45 | surfdue | lol.!! |
| 18:10.00 | surfdue | [TK]D-Fender, member me :) |
| 18:10.04 | _-Jon-_ | surfdue, they used $12 so far :P |
| 18:10.06 | surfdue | [TK]D-Fender, im the guy that dosnt like to read :P |
| 18:10.26 | [TK]D-Fender | surfdue : How could I not without years of therapy? ;) |
| 18:10.28 | Trionnis | I dunno, I'd hazard a guess that it's part of the whole "brute force then resell minutes" ordeal |
| 18:10.30 | surfdue | oh no, they passed the $10, oh man this is call for suewage (i like to make words ;)) |
| 18:10.34 | So3kris | [TK]D-Fender: i have that and then i can manage it at the extentions.conf ? |
| 18:10.36 | surfdue | [TK]D-Fender, LOL |
| 18:10.40 | surfdue | [TK]D-Fender, your so mean! |
| 18:10.48 | _-Jon-_ | hah |
| 18:11.10 | [TK]D-Fender | surfdue : thats ok.... I'm getting volume rebates on it now :) |
| 18:11.29 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:11.42 | surfdue | [TK]D-Fender, anyways im having a problem with my pap2 I keep getting 404 errors, syzygybsd thinks its my pap2 settings but its connected and online, here is a pastebin: http://pastebin.ca/85079 http://pastebin.ca/85083 |
| 18:14.11 | *** join/#asterisk trumee (n=trumee@cpc4-cmbg1-0-0-cust330.cmbg.cable.ntl.com) |
| 18:15.21 | trumee | guys, i am trying to setup asterisk to talk with voiptalk. They have a page https://www.voiptalk.org/products/iaxconfig.html which tells how to talk to asterisk. But what is puzzling me what softphone should i use? |
| 18:15.35 | [TK]D-Fender | surfdue : 2nd one shows it registered. What is the the problem with it exactly? |
| 18:15.41 | Strom_C | trumee: whatever softphone you want |
| 18:15.57 | trumee | i have installed kiax and can connect to asterisk but how do i call out |
| 18:16.22 | trumee | Strom_C:can i use sip phone to talk 2 asterisk which then call voiptalk over iax? |
| 18:16.36 | surfdue | it wont ring my pap2 |
| 18:17.01 | Qb3rt | when a extension is already in used... the queue system is not seending call on the second line of the extension! is there a manner the force that? |
| 18:17.11 | [TK]D-Fender | surfdue : got a screen shot of the web-coinfig pages for it? |
| 18:17.29 | Strom_C | trumee: yes |
| 18:17.35 | surfdue | [TK]D-Fender, ill brb yes |
| 18:17.37 | Egonis | in extensions.conf I have CONSOLE=Console/alsa -- is this correct? |
| 18:17.55 | syzygybsd | he posted them earlier... bout 100 lines up |
| 18:18.01 | trumee | Strom_C:excellent, so i dont need kiax which is OSS based. i will use twinkle which is alsa based :) |
| 18:19.12 | Egonis | I get 'no channel type registered for 'Console' when I Dial(Console/alsa) |
| 18:19.33 | rene- | my PAP2 are also being picky, they are loosing and gaining registration randomly and it doesnt matter weather they are sip.conf or mysql realtime based since the problem shows in both cases, i restart the boxes they register but registration is lost after 5 -15 minutes, registration comes back at random |
| 18:19.48 | rene- | my setup is LAN |
| 18:20.21 | rene- | would qualify help me here? |
| 18:20.42 | rene- | i am not using it |
| 18:20.52 | So3kris | rene-: i got the 2100 but if i call with a analog phone a register wil be forced |
| 18:21.17 | *** part/#asterisk Egonis (n=chultay@207.245.14.10) |
| 18:21.43 | *** join/#asterisk hi365 (n=hi365@bzq-88-155-37-190.red.bezeqint.net) |
| 18:21.43 | rene- | So3kris: what if you are calling to the 2100 |
| 18:21.55 | rene- | right now all ports are online again |
| 18:22.18 | jbalcomb | ok, why don't the GXP-2000s ask for the cfg.txt default config anymore? |
| 18:22.41 | So3kris | rene-: than not asterisk must first hava a registerd account or a full static config form a device |
| 18:24.23 | *** join/#asterisk Eggplant (n=none@dsl-72-19-47-026.cascadeaccess.com) |
| 18:27.00 | rene- | this seems related: http://bugs.digium.com/view.php?id=4343 |
| 18:30.31 | *** join/#asterisk Felix2 (n=felix@p5493F823.dip.t-dialin.net) |
| 18:30.37 | Felix2 | Hello |
| 18:30.52 | kindor | should dtmf tones work with voipbuster as incoming trunk ? |
| 18:30.57 | Felix2 | I need help |
| 18:32.33 | gandhijee | anyone know if libtonezone.so is required to load the wcfxo module? |
| 18:32.55 | Felix2 | Can i install asterisk on my linux root server? The root server is standing in the Data Center. Works asterisk on my root server? |
| 18:33.24 | Trionnis | er? |
| 18:33.30 | So3kris | Felix2: if you have a normal server you can do that. |
| 18:34.51 | Felix2 | I have an linux root server but the server stand not by me at home. Is it workin when i install the asterisk on my root server and configure the system that all calls come to me at home? Is that working? |
| 18:35.24 | Trionnis | as long as it's a "real" server, yes |
| 18:35.34 | hi365 | does anyone have experience with sangoma cards?? im getting this error: http://pastebin.ca/85147 |
| 18:35.37 | Trionnis | VPS'es tend not to work due to needing kernel modules |
| 18:35.39 | Felix2 | Its a real Server |
| 18:36.46 | Felix2 | Its a Hight-End Server of strato.de |
| 18:37.13 | *** join/#asterisk MXC4 (n=mark@dsl-146-66-33.telkomadsl.co.za) |
| 18:37.22 | eKo1 | Felix2: Deutscher? |
| 18:37.30 | Felix2 | yes |
| 18:37.33 | Felix2 | Ja |
| 18:37.39 | MXC4 | hi all |
| 18:37.43 | kindor | sup |
| 18:37.45 | Felix2 | Mein Englisch ist grauenhaft |
| 18:37.50 | gandhijee | hi365: do you have the modules for the sangoma loaded? |
| 18:38.01 | Felix2 | Sprichste auch Deutsch? |
| 18:38.05 | gandhijee | thier wiki is pretty stright foward on how to get the hardware going |
| 18:38.05 | kpettit | I've got some g729 licenses that I just installed. "show channels" only shows 12 or so connections but I get a constant "Need mroe licenses" message flooding the screent. "show g729" shows I have 20 licenses |
| 18:38.22 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:38.38 | MXC4 | has anyone had problems with the digium cards not hanging up properly for incoming cals |
| 18:38.53 | MXC4 | sorry calls |
| 18:39.17 | trumee | guys, on an incoming call i am getting the error Rejected connect attempt from 217.14.138.48, request '01223858818@default' does not exist |
| 18:39.18 | eKo1 | Felix2: Ya aber bitte frag hier. Kein PM. |
| 18:39.21 | MXC4 | in the flash operator panel it just shows 2 of the 4 trunk in red and they never seem to drop |
| 18:39.23 | MXC4 | the lines |
| 18:39.32 | Felix2 | ok |
| 18:39.39 | trumee | but i do have the line exten => 01223858818,1,Dial(SIP/1234) |
| 18:40.04 | MXC4 | i have search the web but found nothing about this |
| 18:40.07 | MXC4 | any ideas? |
| 18:40.31 | trumee | Strom_C:any idea? |
| 18:40.31 | Felix2 | Also ich hab nen Root Server, will darauf ne TK SOftware drauf machen, da dachte ich an asterisk, aber das dann geht, weißte wenn das proggie aufm server läauft, das der dann die anrufe an die verschiedenen telefonnummern weiterleitet... |
| 18:40.38 | trumee | Strom_C: on an incoming call i am getting the error Rejected connect attempt from 217.14.138.48, request '01223858818@default' does not exist |
| 18:40.45 | trumee | Strom_C:but i do have the line exten => 01223858818,1,Dial(SIP/1234) |
| 18:40.52 | Felix2 | Würde auch gehen das ich per VoIP auf den Server connecte |
| 18:41.16 | Felix2 | Funktioniert das? |
| 18:43.05 | eKo1 | Felix2: Yes, Asterisk will do that for you. |
| 18:43.32 | Assid | oay if its showing native bridge.. AND sip show channels show 1 on 729 but the other end as ulaw |
| 18:43.39 | Assid | how is it connecting eventually? |
| 18:44.07 | Felix2 | Kann ich die Anrufe auf verschiedene Telefonnummern weiterleiten oder muss ich per VoIP auf den server connecten? |
| 18:45.47 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 18:45.50 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 18:46.49 | eKo1 | Felix2: Beide. |
| 18:47.33 | Felix2 | gut |
| 18:47.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.98) |
| 18:48.40 | fgwaller | is anyone working on the deadlock issues in manager.c? |
| 18:48.45 | Felix2 | Meine Server Daten: 2,4 GHz, 1 GB Ram, wie viel % wird der server ausgelastet bei ca. 3 Gesprächen aufeinmal und wie viel Traffic verbraucht das so im Monat? |
| 18:49.00 | rene- | fgwaller: link please? |
| 18:49.16 | fgwaller | bug base link? |
| 18:49.21 | rene- | fgwaller: i am not working on them |
| 18:49.30 | rene- | fgwaller: i just want to be informed |
| 18:49.45 | rene- | i ll chek the tracker |
| 18:49.54 | eKo1 | Felix2: I suggest you find out for yourself. |
| 18:51.08 | fgwaller | rene: thanks, I have been following all the svn posts but did not see any activity in that section lately |
| 18:51.31 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 18:53.34 | *** join/#asterisk babyju (n=babyju@151.202.195.132) |
| 18:55.40 | [TK]D-Fender | trumee : Pastebin your extensiosn.conf and the CLI output of that failed incoming call. |
| 18:55.58 | trumee | [TK]D-Fender:it is working now :) |
| 18:56.06 | trumee | [TK]D-Fender:thanks |
| 18:56.15 | dlynes_laptop | [TK]D-Fender: btw |
| 18:56.33 | dlynes_laptop | [TK]D-Fender: if you're interested, sangoma will probably be coming out with some bri cards around the end of this year |
| 18:56.45 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 18:56.59 | [TK]D-Fender | dlynes_laptop : Not for my use, but as "suggestion fodder", sure |
| 18:57.11 | [TK]D-Fender | dlynes_laptop : Just not a "reality" for us you know.... |
| 18:57.17 | dlynes_laptop | yeah...same here |
| 18:57.25 | benjk | any BRIstuff users here? |
| 18:57.28 | dlynes_laptop | but i figured for suggesting to all the euros on here |
| 18:57.35 | dlynes_laptop | like probably benjk |
| 18:57.35 | [TK]D-Fender | dlynes_laptop : I meant "us" inclusively |
| 18:57.41 | [TK]D-Fender | dlynes_laptop : Exactly |
| 18:57.57 | dlynes_laptop | well, japan |
| 18:58.03 | dlynes_laptop | nippon |
| 18:58.33 | benjk | Japan's NTT apparently uses DSS1 with ulaw instead of alaw |
| 18:58.49 | dlynes_laptop | benjk: but it's still BRI |
| 18:58.51 | [TK]D-Fender | Japan is a really wierd mix of standards |
| 18:58.53 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:59.13 | dlynes_laptop | qseek: oooh....another norteller |
| 18:59.31 | [TK]D-Fender | *shudder* |
| 18:59.34 | benjk | well, I don't really think that such a minor adjustment as using a different codec actually qualifies to be called a different standard or even having its own name |
| 18:59.40 | dlynes_laptop | [TK]D-Fender: nortellers are a good thing in here |
| 18:59.41 | qseek | dang |
| 18:59.48 | qseek | the spoofing is still on |
| 18:59.53 | [TK]D-Fender | benjk : I mean for analog as well... |
| 18:59.56 | trumee | [TK]D-Fender:i have a stupid question |
| 19:00.03 | benjk | analog is totally screwed up here |
| 19:00.05 | dlynes_laptop | [TK]D-Fender: they're good for the grassroots campaign from within nortel to get nortel handsets working with asterisk :) |
| 19:00.11 | [TK]D-Fender | trumee : Wouldn't be the first I've heard ;) |
| 19:00.37 | kpettit | I've got a group of phones behind NAT. And they all register with the same port/ip |
| 19:00.44 | benjk | so, is there anybody here who knows how to test/troubleshoot a BRIstuff setup? |
| 19:00.47 | [TK]D-Fender | dlynes_laptop : No, they're good for creating a flash-mob to storm NT's headquarters and lynch everyone! |
| 19:01.35 | kpettit | I've got nat=yes which dosen't seem to help. I've got a couple doze other phones in different locations that don't have this issues. not sure what I can do to get these phones on different ports. |
| 19:01.37 | kpettit | I've even tried setting bindport on the sip peers which didn't seem to help |
| 19:01.37 | dlynes_laptop | [TK]D-Fender: not that it'll happen, but it's a good though :) |
| 19:01.37 | dlynes_laptop | s/though/thought/ |
| 19:01.37 | gandhijee | dlynes_laptop: will they work for the bri lines here in america? |
| 19:01.38 | trumee | [TK]D-Fender:i have setup an incoming call setups for iax and sip which work separately. for iax i have exten => 01223858818,1,Dial(SIP/1234,20,tr) and for sip i have exten => s,1,Dial(SIP/1234,20,tr). But both of them dont work if i put them in the same file |
| 19:01.38 | [TK]D-Fender | kpettit : set each to a seperate SIP port in their peer entry and you should be OK |
| 19:01.43 | kpettit | [TK]D-Fender, I've done that, but they all stay same |
| 19:01.50 | [TK]D-Fender | trumee : Pastebin your extensions.conf |
| 19:02.04 | dlynes_laptop | gandhijee: bri is still used somewhere in north america? |
| 19:02.12 | gandhijee | yea |
| 19:02.15 | gandhijee | ISDN lines |
| 19:02.17 | [TK]D-Fender | kpettit "port=5061" for example |
| 19:02.21 | dlynes_laptop | gandhijee: you ask the telco around here, and they won't even admit bri still exists |
| 19:02.47 | gandhijee | where would that be? |
| 19:02.49 | kpettit | [TK]D-Fender, yeah I've done that. I gave each sip peer a different port, but it uses the one they all previously had |
| 19:03.05 | dlynes_laptop | gandhijee: when you corner them and get them to admit that it still exists, they want to charge you five times the cost of an analog line |
| 19:03.11 | gandhijee | really? |
| 19:03.21 | dlynes_laptop | yeah |
| 19:03.23 | gandhijee | i gotta pretty good quote for a BRI line when i asked for one |
| 19:03.28 | dlynes_laptop | thought it was that way in all of north america |
| 19:03.47 | gandhijee | only thing that stopped me was the fact i couldn't get a BRI card for asterisk and North America |
| 19:03.49 | [TK]D-Fender | kpettit : Make sure its configured on the phone as well. |
| 19:03.52 | *** join/#asterisk doolph (n=doolph@200.124.16.143) |
| 19:03.52 | dlynes_laptop | gandhijee: how much a month? |
| 19:03.54 | doolph | hi there |
| 19:04.01 | doolph | anyone suggest me a good qos solution in the same asterisk box? |
| 19:04.02 | trumee | [TK]D-Fender:here it is http://pastebin.ca/85181 |
| 19:04.06 | gandhijee | the analog is running us like 140 or so |
| 19:04.12 | gandhijee | the BRI would have been like 170 |
| 19:04.22 | benjk | you don't use a special BRI card for the US |
| 19:04.22 | dlynes_laptop | gandhijee: eh? 140 for an analog? |
| 19:04.30 | gandhijee | and that was for a business, 4 lines |
| 19:04.31 | dlynes_laptop | gandhijee: your phone company really seen you coming |
| 19:04.35 | benjk | don't need, I meant to sa |
| 19:04.36 | benjk | say |
| 19:04.37 | dlynes_laptop | gandhijee: oh |
| 19:04.43 | gandhijee | 35 a piece |
| 19:04.48 | trumee | [TK]D-Fender:is the incoming setup ok? |
| 19:04.56 | dlynes_laptop | gandhijee: yeah...different story...that's pretty cheap for isdn then |
| 19:04.56 | gandhijee | that was including the taxes |
| 19:05.00 | gandhijee | yeah |
| 19:05.08 | dlynes_laptop | gandhijee: $170 for 2 ISDN lines then? |
| 19:05.10 | gandhijee | it was a really good price |
| 19:05.15 | dlynes_laptop | gandhijee: or $170 for a 128K ISDN? |
| 19:05.28 | gandhijee | 170 for 2 ISDN's |
| 19:05.31 | benjk | 170 USD for 2xBRI? |
| 19:05.33 | gandhijee | 2 chan a piece |
| 19:05.39 | gandhijee | yeapo |
| 19:05.40 | benjk | that's outragrous |
| 19:05.51 | kpettit | [TK]D-Fender, I have polycom phones. Do I need to set the reg.1.server.1.port="5060" to the same port I define in sip.conf? |
| 19:05.51 | gandhijee | i gotta quote for like 180 for 4 PRI lines |
| 19:05.53 | benjk | 22 USD per BRI in Japan |
| 19:06.07 | dlynes_laptop | benjk: NA telcos don't want you to use BRI |
| 19:06.08 | benjk | if you own the rights to a phone line that is |
| 19:06.09 | kpettit | [TK]D-Fender, I'm alittle confused if it's taking media stream or registration there |
| 19:06.13 | drray | who is your provider? |
| 19:06.14 | gandhijee | we are not all fortunate enough to live in japan |
| 19:06.23 | benjk | if you have to rent the phone rights, then its 28 USD |
| 19:06.33 | [TK]D-Fender | kpettit : AH... not sure about the intricacies of NAT settings on them, but yes that is where you would set it I believe. |
| 19:06.42 | benjk | they sell you an elusive thing here called a phone right |
| 19:06.42 | dlynes_laptop | gandhijee: 4 pri lines? $180? wtf? It costs us like $1200 for one |
| 19:06.51 | benjk | its about 700 USD |
| 19:06.52 | drray | 4 did's |
| 19:06.55 | benjk | per line |
| 19:06.56 | drray | on one PRI |
| 19:07.03 | gandhijee | lemme change taht |
| 19:07.10 | gandhijee | a 4 channel PRI |
| 19:07.10 | benjk | its like a phone company membership |
| 19:07.20 | benjk | you can sell it |
| 19:07.24 | gandhijee | and you are gettin raped on your PRI |
| 19:07.32 | benjk | some companies trade in "second hand phone rights" |
| 19:07.39 | gandhijee | i can get a 24 chan for like 700 + taxes |
| 19:07.54 | benjk | no that 700 is a one time membership fee |
| 19:08.00 | dlynes_laptop | gandhijee: $1200 Cdn, and that includes colo facilities, and locker rental |
| 19:08.01 | gandhijee | no its not |
| 19:08.11 | benjk | for each phone circuit you have to buy a membership here |
| 19:08.15 | *** part/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 19:08.17 | gandhijee | i see |
| 19:08.24 | dlynes_laptop | gandhijee: he's talking about the $700USD right for a phone line in Japan |
| 19:08.29 | dlynes_laptop | gandhijee: not a pri |
| 19:08.32 | benjk | that's the reason why mobile phones took off with young folks in Japan |
| 19:08.45 | benjk | a mobile phone doesn't have these gimmicks |
| 19:08.51 | gandhijee | it would with me too if it cost that much |
| 19:08.58 | *** join/#asterisk Jimerson (n=Jim@c-71-200-9-249.hsd1.md.comcast.net) |
| 19:09.00 | benjk | so they youngsters simply didn't bother with a fixed line anymore |
| 19:09.01 | trumee | [TK]D-Fender:is the file ok? |
| 19:09.11 | dlynes_laptop | benjk: that sounds similar to the racket they've got here in Vancouver with taxi cab licenses |
| 19:09.19 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 19:09.20 | benjk | yeah, something like that |
| 19:09.22 | Jimerson | Is anyone around that can help with an error? |
| 19:09.23 | SarahEmm | hihi |
| 19:09.24 | dlynes_laptop | benjk: there's a finite number of licenses to operate a taxi cab |
| 19:09.47 | nortex | Jimerson, What kind of error? |
| 19:09.53 | dlynes_laptop | benjk: and demand outstrips supply, so when you can buy a license, you're usually paying about $500K USD for a license |
| 19:09.55 | [TK]D-Fender | trumee : Not at all. You have NO proper contexts, let alone the [default] it was looking for. |
| 19:10.08 | [TK]D-Fender | trumee : You seriously need to read THE BOOK |
| 19:10.09 | [TK]D-Fender | ~book |
| 19:10.11 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:10.11 | trumee | [TK]D-Fender:that sounds shit. |
| 19:10.11 | dlynes_laptop | benjk: and that's $500K USD per taxi cab |
| 19:10.15 | Jimerson | nortex when I try and start asterisk using the /usr/sbin/amportal start |
| 19:10.15 | Jimerson | <PROTECTED> |
| 19:10.24 | Jimerson | I get STARTING ASTERISK |
| 19:10.24 | Jimerson | Cannot find your TTY (9) |
| 19:10.24 | Jimerson | safe_asterisk: no process killed |
| 19:10.25 | Jimerson | mpg123: no process killed |
| 19:10.26 | doolph | anyone suggest me a good qos solution in the same asterisk box? |
| 19:10.47 | trumee | [TK]D-Fender:i get this iax2_read: I should never be called! on CLI |
| 19:10.47 | dlynes_laptop | Jimerson: type /topic |
| 19:10.49 | benjk | that's a bargain compared to 700 USD per phone line membership |
| 19:10.51 | [TK]D-Fender | Jimerson : Please read the channel topic... |
| 19:11.02 | benjk | now according to http://www.cia.gov/cia/publications/factbook/geos/ja.html |
| 19:11.04 | trumee | [TK]D-Fender:the iax setting is what i got from voiptalk.org |
| 19:11.05 | rene- | 500K for a taxi cab license? i guess taxi service is VERY expensive there |
| 19:11.11 | benjk | Japan had 58 million fixed lines in 2004 |
| 19:11.12 | nortex | Jimerson, You might check with the guys in #freepbx since that is one of their scripts. |
| 19:11.14 | [TK]D-Fender | trumee : Well fix your extensions.conf then we'll review iax.conf after |
| 19:11.18 | benjk | multiply that with 700 USD |
| 19:11.19 | Jimerson | Sorry - typed in wrong channel again. |
| 19:11.20 | Jimerson | lol |
| 19:11.21 | dlynes_laptop | rene-: nope...average |
| 19:11.23 | Jimerson | too many. |
| 19:11.25 | Jimerson | My appologies. |
| 19:11.26 | Hmmhesays | anyone in here using as5300's? |
| 19:11.42 | trumee | [TK]D-Fender:it wasnt the full file, i had got rid of global setting while pasting |
| 19:11.46 | benjk | that's a pile of cash for NTT |
| 19:12.13 | benjk | anyway, nowadays they have a "lite" plan |
| 19:12.18 | rene- | A Mexico city taxi license is like $15K no wonder there are tens of thousands cabs here (half of them with no legal licenses) |
| 19:12.28 | [TK]D-Fender | Hmmhesays : Just ordered that MIDI controller for $375! Cheaper in CAD than I could even find it in USD and no shipping :) |
| 19:12.31 | gandhijee | anyone know if libtonezone.so is required for the wc* modules to load |
| 19:12.34 | benjk | you don't need the membership but pay a few dollars per month more on the line rental |
| 19:12.37 | Hmmhesays | sweet [TK]D-Fender |
| 19:12.50 | dlynes_laptop | rene-: $2.30 Cdn(?) to get in, $1/mi for idling, $1/km or something like that for driving |
| 19:12.53 | [TK]D-Fender | trumee : Pastebin the WHOLE thing then and sip/iax confs |
| 19:13.18 | dlynes_laptop | s/\/mi/\/min/ |
| 19:13.22 | benjk | so idling is per mile and driving is per km |
| 19:13.27 | benjk | that's funny |
| 19:13.31 | SarahEmm | err |
| 19:13.34 | SarahEmm | idling is per minute |
| 19:13.35 | dlynes_laptop | benjk: mi == minute |
| 19:13.38 | SarahEmm | if you're idling, you're not moving |
| 19:13.39 | benjk | ah ok |
| 19:13.40 | m4rkl4r | is this a good place for STUN questions? |
| 19:13.43 | SarahEmm | it'd be hard to charge for idling per mile ;) |
| 19:13.50 | benjk | haha |
| 19:13.54 | dlynes_laptop | SarahEmm: forgive him...he's from Japan...they don't know these kinda things |
| 19:14.14 | benjk | maybe if the driver switches off the engine on the top of a hill |
| 19:14.18 | SarahEmm | lol! |
| 19:14.22 | benjk | then they charge you for idling by the mile |
| 19:14.27 | [TK]D-Fender | m4rkl4r : No.... Asterisk is set to KILL |
| 19:15.14 | Katty | allo. |
| 19:15.25 | m4rkl4r | It is a bad, bad program that doesn't follow asimov's three laws |
| 19:15.38 | dlynes_laptop | meow. |
| 19:15.46 | SarahEmm | meow! |
| 19:15.48 | SarahEmm | it'd be hard to charge for idling per mile ;) |
| 19:15.49 | drray | why does a robot/ai have to follow these laws? |
| 19:15.51 | SarahEmm | oops |
| 19:16.00 | benjk | anybody any idea why ztcfg would report this ... |
| 19:16.01 | benjk | Notice: Configuration file is /etc/zaptel.conf |
| 19:16.01 | benjk | line 8: Unable to open master device '/dev/zap/ctl' |
| 19:16.14 | benjk | there is no /dev/zap/ctl only /dev/zapctl |
| 19:16.26 | m4rkl4r | well, this is maybe a better question for the stun server mailing list at vovida.org |
| 19:16.28 | [TK]D-Fender | Katty : Salut ma cher ;D |
| 19:16.46 | dlynes_laptop | SarahEmm: write an agi script to automatically playback the soundtrack to some 80's porn everytime someone dials *69? |
| 19:16.55 | SarahEmm | umm.... |
| 19:16.58 | SarahEmm | not so much.. |
| 19:17.02 | SarahEmm | i'm the only one that uses the system ;) |
| 19:17.08 | dlynes_laptop | ah |
| 19:17.13 | dlynes_laptop | then that wouldn't be any fun |
| 19:17.14 | [TK]D-Fender | dlynes_laptop : No need for AGI, can can set her up in straigh dial-plan ;) |
| 19:17.17 | trumee | [TK]D-Fender:Here it is, all confs in full http://pastebin.ca/85198 |
| 19:17.59 | benjk | I wonder if there is a different IRC channel for BRI folks |
| 19:18.08 | dlynes_laptop | [TK]D-Fender: she's ignoring you :) |
| 19:18.21 | [TK]D-Fender | trumee : And CLI output of the failed call please. |
| 19:18.52 | trumee | [TK]D-Fender:well sometimes it fails, sometime it doesnt. a bit erratic |
| 19:19.04 | [TK]D-Fender | dlynes_laptop : s'ok the voices still keep me company... |
| 19:19.11 | trumee | [TK]D-Fender:but i was getting this han_iax2.c:2447 iax2_read: I should never be called! |
| 19:19.16 | [TK]D-Fender | trumee : Find a failure..... |
| 19:19.18 | tzafrir_laptop | benjk, is there a file with that name? /dev/zap/ctl ? |
| 19:19.32 | benjk | no, that's the odd thing, its /dev/zapctl |
| 19:19.33 | trumee | [TK]D-Fender:what do u think about the conf? |
| 19:19.53 | benjk | wonder is that is a typo in the error message (Asterisk has tons of those) or a genuine bug |
| 19:20.07 | tzafrir_laptop | benjk, looks like a missing / in the udev rules |
| 19:20.11 | *** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be) |
| 19:20.20 | tzafrir_laptop | you can always manually symlink... |
| 19:20.32 | tzafrir_laptop | or grep for zaptel under /etc/udev |
| 19:20.38 | tzafrir_laptop | or manually mknod |
| 19:20.56 | trumee | [TK]D-Fender:no failure now :( |
| 19:21.10 | tzafrir_laptop | benjk, what distro? |
| 19:21.18 | benjk | SUSE 10.1 |
| 19:21.25 | trumee | [TK]D-Fender:i am thinking of getting rid of voipstunt incoming. dont trust that configuration |
| 19:21.46 | benjk | rules.d or udev.conf ? |
| 19:22.10 | SarahEmm | hrm... i could dig into code again. |
| 19:22.12 | SarahEmm | i might do that... |
| 19:22.12 | tzafrir_laptop | grep -r zaptel /etc/udev ... |
| 19:22.23 | tzafrir_laptop | maybe one is a symlink? |
| 19:22.55 | benjk | nothing at all |
| 19:23.00 | tzafrir_laptop | you should see CLASS=="zaptel" as a condition to that rule |
| 19:23.35 | tzafrir_laptop | so maybe this is not udev, but rather a manually-created file. Wen was it created? |
| 19:24.07 | benjk | all today |
| 19:24.28 | tzafrir_laptop | anyway: crw-rw---- 1 root root 196, 0 2006-07-10 23:37 /dev/zap/ctl |
| 19:24.31 | benjk | zap1, zap2, zap3, zapchannel ... |
| 19:24.42 | benjk | yeah kind of strange |
| 19:24.55 | tzafrir_laptop | mkdir /dev/zap; mknod /dev/zap/ctl c 196 0 |
| 19:25.05 | benjk | Kapejod's scripts screwed this up it seems |
| 19:25.42 | trumee | [TK]D-Fender:u around? |
| 19:25.55 | tzafrir_laptop | I actually tend to suspect the zaptel Makefile |
| 19:25.56 | benjk | tzafrir this is a vanilla install of BRIstuff |
| 19:26.25 | tzafrir_laptop | ps aux | grep udevd |
| 19:26.28 | tzafrir_laptop | anything? |
| 19:26.50 | [TK]D-Fender | trumee : Well it may work, but you mix incoming and outgoing stuff all in the same context. an unhealthy thing to do for sure. |
| 19:27.02 | benjk | root 734 0.0 0.3 1844 624 ? S<s 01:04 0:01 /sbin/udevd --daemon |
| 19:27.49 | trumee | [TK]D-Fender:oh, should i have separate contexts then. i suppose you are talking about extensions,conf. How can i have separate contexts? |
| 19:28.04 | benjk | [incoming] |
| 19:28.06 | benjk | ... |
| 19:28.10 | benjk | [outgoing] |
| 19:28.11 | benjk | ... |
| 19:28.16 | benjk | very simple |
| 19:28.20 | tzafrir_laptop | those names are basically the names of the files under /sys/class/zaptel |
| 19:28.48 | trumee | benjk:uh ok. and is that the only change i need to do, or do i need to change something in sip.conf and iax.con too? |
| 19:28.51 | [TK]D-Fender | trumee : The "book" awaits you.... |
| 19:29.32 | benjk | trumee, if you have a SIP phone, you'll have an entry for that phone in sip.conf and in that entry there should be a line "context=blah" |
| 19:29.51 | trumee | benjk:uh ok, i got the idea. let me set it up |
| 19:29.55 | benjk | that's the context the phone will automatically drop into |
| 19:30.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:30.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:30.40 | benjk | tzafrir_laptop: what do you have in /sys/class/zaptel ? |
| 19:30.54 | *** join/#asterisk type0 (i=type0@205-34-178-69.gci.net) |
| 19:31.23 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
| 19:31.30 | trumee | benjk:do i need to have separate outgoings contexts for different sip providers as well? |
| 19:31.40 | benjk | sure |
| 19:32.18 | alexns | anyone using multiple teliax accounts on one machine ? |
| 19:32.32 | tzafrir_laptop | zap1 zap11 zap13 zap15 zap17 zap19 zap20 zap22 zap24 zap26 zap28 zap3 zap31 zap33 zap35 zap37 zap4 zap6 zap8 zapchannel zappseudo |
| 19:32.33 | tzafrir_laptop | zap10 zap12 zap14 zap16 zap18 zap2 zap21 zap23 zap25 zap27 zap29 zap30 zap32 zap34 zap36 zap38 zap5 zap7 zap9 zapctl zaptimer |
| 19:32.41 | xheliox | alexns: Accounts or DIDs? |
| 19:32.48 | alexns | accounts |
| 19:32.51 | benjk | ok, that looks like what I have |
| 19:33.00 | xheliox | Not I. |
| 19:33.20 | benjk | so those are ok, but in /dev they should be separated with a slash |
| 19:33.42 | [TK]D-Fender | trumee : Depends on your setup. You should different INCOMING contexts for them for sure to avoid any cross-talk, etc. |
| 19:33.51 | alexns | xheliox: dids no problem but i have some people interested in tenant service... teliax said it wouldnt work before, but it may now |
| 19:34.08 | surfdue | i have some bad echo cancelation or something my pap2 is getting really chopped but its even in and out and in and out |
| 19:34.26 | xheliox | alexns: I use IAX with Teliax, I imagine if you used the "with auth" option, it would work just fine. |
| 19:34.33 | trumee | [TK]D-Fender:what should be in the default in extensions.conf, i am getting this error now on dialout, pbx.c:1332 pbx_extension_helper: Cannot find extension context 'default' |
| 19:34.36 | tzafrir_laptop | benjk, they were either created by a udev rule (in that case, grep for zaptel under /etc/udev or whatever) |
| 19:34.53 | benjk | already done that but returned nothing |
| 19:34.57 | tzafrir_laptop | or they are manually-generated. In that case: mv or re-mknod |
| 19:35.18 | tzafrir_laptop | It's easy to test: do the files disappear when you rmmod? |
| 19:35.18 | benjk | well, not really manually because I would have been the only one to have done that |
| 19:35.25 | benjk | and I am not that retarded yet |
| 19:35.56 | alexns | xheliox: tried that last time, they said to try without auth as well as doing one account sip the other iax |
| 19:36.02 | tzafrir_laptop | manually:= not by udev |
| 19:36.07 | surfdue | anyone? |
| 19:36.29 | benjk | rmmod zaphfc => zap1, zap2, zap3 are gone |
| 19:36.32 | alexns | xheliox: cant remember what happened though aside from not working... just curious if anyone was doing it |
| 19:36.55 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 19:36.56 | tzafrir_laptop | so it's udev . |
| 19:37.09 | benjk | but where are the rules then? |
| 19:37.33 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:37.36 | *** join/#asterisk damned (n=vpol@17.15.leased.lanck.net) |
| 19:37.41 | dlynes_laptop | catch ya all later |
| 19:37.42 | dlynes_laptop | gotta run |
| 19:37.54 | damned | hi all. can anybody suggest where can i discuss commercial support? |
| 19:38.17 | benjk | dial 500 |
| 19:38.33 | benjk | then press 2 for sales (or whatever the voice menu says) |
| 19:38.34 | xheliox | Damin: Providing or receiving? |
| 19:38.44 | damned | xheliox: recieving |
| 19:38.50 | xheliox | Step into my office? :) |
| 19:39.05 | dlynes_laptop | damned: Just ask on here....there's plenty of people on here that offer it |
| 19:39.19 | dlynes_laptop | damned: you can also try the asterisk-biz mailing list |
| 19:39.25 | dlynes_laptop | ~mailinglist |
| 19:39.27 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 19:39.36 | Cresl1n | ~book |
| 19:39.37 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:39.42 | Cresl1n | ~jsmith |
| 19:39.43 | jbot | well, jsmith is perpetually hungry, or the author of Asterisk: The Future of Telephony |
| 19:39.43 | Assid | aargh |
| 19:39.48 | surfdue | anyone anyone anyone? |
| 19:39.49 | tzafrir_laptop | benjk, maybe as on #suse ? |
| 19:39.50 | Assid | why isnt my caller id working correctly |
| 19:39.51 | surfdue | i have some bad echo cancelation or something my pap2 is getting really chopped but its even in and out and in and out |
| 19:39.59 | dlynes_laptop | Cresl1n: You're jsmith? |
| 19:40.07 | Cresl1n | ~Cresl1n |
| 19:40.14 | Cresl1n | ooooohhhhhh |
| 19:40.14 | *** join/#asterisk ys1234 (i=dontknow@61.247.229.200) |
| 19:40.15 | Cresl1n | nothing |
| 19:40.18 | benjk | its weird, I have never seen this before on SUSE |
| 19:40.18 | dlynes_laptop | You're not terribly famous |
| 19:40.21 | damned | xheliox: if you r able to provide support in russia? :) |
| 19:40.30 | Cresl1n | ~file |
| 19:40.36 | benjk | where in Russia? |
| 19:40.44 | dlynes_laptop | ~nickometer Cresl1n |
| 19:40.50 | Cresl1n | heh |
| 19:40.54 | ys1234 | help |
| 19:40.56 | xheliox | damned: If you provide an ssh connection or air fare, sure, why not? :) |
| 19:40.57 | damned | benjk: Saint-Petersburg |
| 19:41.01 | Cresl1n | ~nickometer dlynes_laptop |
| 19:41.04 | file | jbot: file? |
| 19:41.06 | jbot | well, file is Joshua Colp, jcolp@digium.com, and is the best looking software developer at Digium |
| 19:41.06 | Cresl1n | oooh |
| 19:41.15 | Cresl1n | jbot: Cresl1n |
| 19:41.25 | ys1234 | any one have experience on asterisk behind isa 2004. plz |
| 19:41.25 | SarahEmm | jbot: kitrich |
| 19:41.26 | jbot | i heard kitrich is a cross between and ostrich and a kitten |
| 19:41.26 | tzafrir_laptop | benjk, if nodes suddnly appear when a module is loaded, it is probably udev |
| 19:41.27 | eKo1 | hehehe |
| 19:41.27 | file | jbot: Cresl1n? |
| 19:41.29 | SarahEmm | woot! |
| 19:41.32 | SarahEmm | jbot: sarahemm |
| 19:41.33 | jbot | i guess sarahemm is kitrichernesses. |
| 19:41.36 | file | poor Cresl1n |
| 19:41.43 | tzafrir_laptop | benjk, is can be hotplug or something similar, but why? |
| 19:41.45 | surfdue | hey file can you help me :) |
| 19:42.02 | file | surfdue: what's up now? |
| 19:42.04 | kindor | what dtmfmode to use for voipbuster and like systems? |
| 19:42.09 | surfdue | i have some bad echo cancelation or something my pap2 is getting really chopped but its even in and out and in and out |
| 19:42.12 | surfdue | file ^ |
| 19:42.13 | ys1234 | any one have experience on asterisk behind isa 2004. plz |
| 19:42.34 | dlynes_laptop | kindor: usually rfc2833 is a safe bet for most providers, but I would still check with them to be sure, if it's not working for you |
| 19:42.35 | surfdue | ys1234, that dosnt make any sence? |
| 19:42.45 | file | surfdue: nope |
| 19:42.51 | kindor | thanks dlynes |
| 19:42.56 | trumee | [TK]D-Fender:ok, i have separated out now. but now i can only receive calls from voiptalk but not from voipstunt. What context do i specify in sip,conf for my softphone? |
| 19:43.01 | file | ys1234: I think Juggie tried it... and had problems with IAX2 because it was being stupid (ISA) |
| 19:43.04 | surfdue | dlynes probly can |
| 19:43.15 | ys1234 | i m trying to install asterisk behind isa 2004 |
| 19:43.21 | benjk | I have 45-isdn.rules |
| 19:43.30 | trumee | benjk:What context do i specify in sip,conf for my softphone? Currently, i have context=outgoing on it. but it is not taking calls :( |
| 19:43.30 | benjk | but that looks like it has nothing to do with zaptel |
| 19:43.55 | trumee | benjk:i mean it is not receiving any sip incoming calls |
| 19:43.56 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:43.57 | benjk | trumee, perhaps you want to have a context in extensions.conf for all your phones |
| 19:44.01 | benjk | [home] |
| 19:44.03 | benjk | ... |
| 19:44.04 | doolph | anyone suggest me a good qos solution in the same asterisk box? |
| 19:44.04 | benjk | then |
| 19:44.13 | benjk | include => outgoing-PSTN |
| 19:44.25 | benjk | incluce => outgoing-Blah |
| 19:44.28 | benjk | etc |
| 19:44.55 | trumee | benjk:i have 3 contexts now in extension.comf, [incoming-voiptalk], [incoming-voipstunt] and [outgoing] |
| 19:45.19 | Assid | incomign voipstunt ? |
| 19:45.20 | trumee | benjk:but i dont have any default in extensions.conf now. |
| 19:45.27 | benjk | make a context [home] or [office] or whatever in your extensions.conf |
| 19:45.42 | Assid | stupid voipstunt doesnt wanna give me a number cause im not in US |
| 19:45.46 | benjk | then assign that to all your sip phones in sip.conf with context=home or whatever |
| 19:46.00 | benjk | then do include => outgoing-THIS |
| 19:46.01 | trumee | benjk:so, what do i specify in sip.conf for my softphone. [incoming] or [outgoing] ? |
| 19:46.08 | benjk | include => outgoing-THAT |
| 19:46.12 | benjk | under [home] |
| 19:46.19 | Cresl1n | jbot: vim |
| 19:46.20 | jbot | vim is probably better than emacs |
| 19:46.22 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:46.25 | ys1234 | yes file |
| 19:46.28 | benjk | no, you to context=home |
| 19:46.31 | Cresl1n | jbot: emacs |
| 19:46.32 | jbot | hmm... emacs is as useless as vim if you don't wanna learn it |
| 19:46.36 | benjk | and in extensions.conf you do |
| 19:46.38 | benjk | [home] |
| 19:46.38 | file | ys1234: SIP? |
| 19:46.40 | trumee | benjk:oh, so instead of context , i have put include? |
| 19:46.45 | ys1234 | yes |
| 19:46.51 | benjk | include => outgoing-blahblah |
| 19:46.51 | Cresl1n | jbot: benjk |
| 19:46.52 | jbot | [benjk] abusing the bot, or he is l33t with the bot, or but in reality benjk is teaching the bot, or he is truely trying to crash the bot |
| 19:46.55 | ys1234 | i m getting one way audo |
| 19:47.00 | ys1234 | i m getting one way audio |
| 19:47.08 | ys1234 | xlite as clients |
| 19:47.10 | trumee | benjk:include in sip.conf , you meam |
| 19:47.19 | file | ys1234: have you setup sip.conf with your external IP address, internal network information, and forwarded the ports? |
| 19:47.25 | Cresl1n | jbot: jbot |
| 19:47.27 | jbot | rumour has it, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt |
| 19:47.44 | file | Cresl1n: it's sad without you over here >.< |
| 19:47.46 | ys1234 | yes |
| 19:47.46 | trumee | benjk:shall i post my conf for you to have a look? |
| 19:48.03 | ys1234 | i hv set externip an local host |
| 19:48.03 | trumee | benjk:it is a bit confusing to me |
| 19:48.12 | file | ys1234: which direction is not working? |
| 19:48.39 | benjk | just do what I said |
| 19:48.47 | ys1234 | no audio coming inside from external to internal. the other way works fine |
| 19:48.57 | file | ys1234: did you set localnet? |
| 19:49.04 | ys1234 | yes |
| 19:49.10 | benjk | use IAX or LTP instead of SIP |
| 19:49.27 | benjk | those are designed to traverse NAT |
| 19:49.34 | file | have you done a sip debug to look at the messages? |
| 19:50.18 | Hmmhesays | Chuck Norris owns the greatest Poker Face of all-time. It helped him win the 1983 World Series of Poker despite him holding just a Joker, a Get out of Jail Free Monopoly card, a 2 of clubs, 7 of spades and a green #4 card from the game Uno. |
| 19:50.29 | gandhijee | anyone here familiar with the workings of the zaptel drivers? |
| 19:50.35 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-2-14.nwrknj.east.verizon.net) |
| 19:50.36 | ys1234 | i used etherreal on my eternal machine. the packets being sent to the isa has the internal address 192.x.x.x. and not the public ip |
| 19:50.41 | gandhijee | i wanna know if they are considered memory hungry |
| 19:50.55 | file | ys1234: pastebin a sip debug, and your sip.conf |
| 19:52.16 | benjk | tzafrif, that isdn.rules file has nothing to do with zaptel nor zaphfc |
| 19:52.37 | benjk | I guess the system installer must have put it there after seeing the HFC card |
| 19:53.01 | trumee | benjk:ok, i understood what you meant by include now :) |
| 19:53.04 | ys1234 | this chat machine is on another network. Can i send the file to your email id? |
| 19:53.15 | trumee | benjk:i have put include in extensiosn.conf |
| 19:53.28 | benjk | ok |
| 19:53.45 | benjk | think of include as a tool to grant access rights to services |
| 19:54.45 | So3kris | is there some info about dail plan rights. I counld not find info about the right way to permit some users to use some extentions |
| 19:54.51 | benjk | if you have multiple user groups, such as adults and teenagers in a household or marketing and support in a company, you can grant each group different access by including (or not including) various contexts in the context that is their home |
| 19:55.09 | [TK]D-Fender | So3kris : ... |
| 19:55.10 | trumee | benjk:yes it is working fine now. i have put incoming-voiptalk in default include though, because iax.conf has context=incoming-voiptalk |
| 19:55.12 | [TK]D-Fender | ~book |
| 19:55.13 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:55.24 | benjk | www.voip-info.org has lots of stuff on dialplan scripting |
| 19:55.34 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:55.55 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:56.54 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 19:57.18 | So3kris | benjk: if you include a extention with a include inside include asterisk both extentions ? |
| 19:57.31 | benjk | yes |
| 19:58.13 | ManxPower | EXCEPT for the special extensions i, h, etc |
| 20:00.13 | Assid | maybe its better to use sip - sip (* box )- sip (* box) - sip instead of iax to interconnect the 2 boxes |
| 20:00.33 | Assid | you might be able to benefit from canreinvite |
| 20:00.37 | So3kris | If i have 3 extentions A,B and C. A must can use B and C but B may only use A. At A i must include B and C and if I include A at B. B gets default C |
| 20:01.13 | benjk | IAX also has a mechanism that is similar in nature to reinvite |
| 20:01.27 | benjk | and LTP has a forward |
| 20:01.48 | Assid | do i need to enable anything special for the end sip clients to directly talk ? |
| 20:02.23 | pdtmobile | does anybody know of a SIP/IAX to RCA/Mini-plug device |
| 20:02.39 | pdtmobile | want to use something like that for one way paging |
| 20:02.42 | SarahEmm | pdtmobile: it's on my list of things to build. |
| 20:02.46 | pdtmobile | lol |
| 20:02.51 | SarahEmm | that along with a SIP/IAX visual ringer. |
| 20:02.59 | SarahEmm | neither are things i could find commercially |
| 20:03.07 | pdtmobile | thats what i was afraid of |
| 20:03.10 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 20:03.11 | SarahEmm | yeah. |
| 20:03.23 | pdtmobile | i have thought of just using a grandstream hacked up with auto answer and using the headphone jack |
| 20:03.25 | SarahEmm | an ATA with a flasher on it seems like a hacky solution, so i'm going to (eventually) build one.. |
| 20:03.51 | MikeJ[Laptop] | pdthome, there are devices like thatt for analog phones |
| 20:03.57 | MikeJ[Laptop] | you could use that and an ata |
| 20:04.04 | SarahEmm | yeah... |
| 20:04.26 | pdtmobile | ya i found fxo to rca |
| 20:04.34 | pdtmobile | but i was trying to skip the middle man ;) |
| 20:04.38 | MikeJ[Laptop] | heh |
| 20:04.42 | [Airwolf] | Can anyone tell me if it's possible to block the text that comes with the authenticate command ? |
| 20:04.45 | pdtmobile | less moving parts, less to break... |
| 20:04.46 | ManxPower | What is wrong with a sound card in the server and chan_alsa? |
| 20:04.57 | pdtmobile | number of channels |
| 20:05.01 | pdtmobile | multiple paging zones |
| 20:05.02 | [Airwolf] | Like I can block the standard greeting text with voicemail |
| 20:05.35 | ManxPower | Ahrimanes, OK |
| 20:06.05 | pdtmobile | also just a distance issue, it's alot easier to move a packet between buildings, than it is to get the audio out of a sound card half a block away |
| 20:06.16 | benjk | tzafrir, no luck in #suse |
| 20:06.49 | *** join/#asterisk ManuMendoza (n=ManuMend@200.69.182.135) |
| 20:07.13 | benjk | anybody here has got any idea which piece of code creates the /dev/zap entries when running modprobe? |
| 20:07.25 | tzafrir_laptop | benjk, is there /etc/udev* ? |
| 20:07.36 | benjk | yes |
| 20:07.47 | benjk | but I didn't find anything specific to zaptel there |
| 20:08.11 | tzafrir_laptop | grep, not just a specific file, right? |
| 20:08.20 | benjk | in all the files |
| 20:08.43 | benjk | I am now reading them all to see if there is any hint |
| 20:11.46 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:12.38 | tzafrir_laptop | 'grep -r zaptel /etc/udev' produced nothing . 'grep -r zap /etc/udev' did produce those rules. pastebin them? |
| 20:14.57 | kpettit | [TK]D-Fender, this port thing is killing me. It's like asterisk has it in memory and I can't get rid of it |
| 20:15.16 | [TK]D-Fender | kpettit : You mean the port? |
| 20:15.34 | kpettit | [TK]D-Fender, I've tried going from a public IP to a vpn NAT address but the IP address and port stays the same |
| 20:15.51 | tanvalmg | Hi...has anyone here ever installed a Digium TDM400P in a Compaq Evo? |
| 20:16.04 | kpettit | [TK]D-Fender, I've got 5 phones behind a nat that all have the same port. I've tried specifying individual ports in sip.conf and on thep hones, but they don't take |
| 20:16.33 | [TK]D-Fender | kpettit : And you reset the phones & reloaded sip.conf and still nothing? |
| 20:16.34 | kpettit | So I thought I'd switch it back to a VPN and go nat 192.XXXX address, but they still show the public address and the same damn port for all those phones |
| 20:16.34 | ManxPower | kpettit, It is the job of your NAT router to change the ports. |
| 20:16.46 | ManxPower | If you REALLY want to change the source ports, you do that in the actual phone |
| 20:17.14 | kpettit | ManxPower, I've done it on the phone. they are polycom and i've changed it in the phone configs to match what I put in sip.conf |
| 20:17.31 | benjk | tzafrir, grep -r zap /etc/udev turns up nothing |
| 20:18.11 | benjk | there are some files in the /etc/udev/rules.d directory |
| 20:18.33 | benjk | but they are not related to zaptel and none of them contains the sequence "zap" |
| 20:19.36 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 20:20.03 | Assid | <PROTECTED> |
| 20:20.14 | Assid | can someone tell me why this isnt picking up the extension? |
| 20:20.44 | file | Assid: so you have a phone that dialed an extension, and it's Dial is in the context ila at the extension 207? |
| 20:21.05 | Assid | yes.. i want to pickup the extension 207 of ila |
| 20:21.23 | file | show me. |
| 20:21.26 | Assid | i can see it run in CLI nothing happens in reality |
| 20:21.35 | Assid | err.. what do i show? |
| 20:21.45 | *** join/#asterisk JonParkins (i=Jon@ns1.pseudointeractive.com) |
| 20:21.47 | Assid | do i need the actual sip account or the extension? |
| 20:21.48 | file | I need to see a full CLI output containing both calls, plus dialplan logic |
| 20:22.30 | ys1234 | file: sent mail with sip conf and sip debug. Thanks |
| 20:22.42 | Assid | pm okay? |
| 20:22.44 | Assid | 4 lines |
| 20:22.50 | Assid | 5 actually |
| 20:22.53 | file | ys1234: you MUST have localnet set |
| 20:23.34 | JonParkins | I have a really simple question, and am wondering if anyone has a quick answer for it. When in the asterisk command line, is there a way to scroll back up to view text or pause it in some way? please forgive my ignorance |
| 20:24.08 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-106-157.red.bezeqint.net) |
| 20:24.17 | ys1234 | ok thanx will try and get back to u |
| 20:26.16 | pdtmobile | what can cause this: |
| 20:26.17 | pdtmobile | Jul 11 15:25:57 WARNING[2710]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/13 already in use on span 1. Hanging up owner. |
| 20:26.18 | doolph | how can I record in g729 format or convert to g729 format |
| 20:27.56 | terrapen | jon, that's a terminal emulator issue |
| 20:27.59 | terrapen | not an asterisk issue. |
| 20:28.06 | JonParkins | k thanks |
| 20:32.11 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 20:33.46 | Idle | my asterisk server seems to keep calling me... when I pick up, all I get it my own echo... |
| 20:33.52 | Idle | completely randomly... |
| 20:34.00 | Strom_C | it likes you |
| 20:34.05 | Idle | aparently |
| 20:34.14 | Strom_C | it wants to ask you out on a dat but it keeps getting cold feet |
| 20:34.17 | Strom_C | s/dat/date/ |
| 20:34.21 | Idle | I've got it in debug, so when it happens again I can see wtf it thinks its doing |
| 20:34.41 | Idle | haha, funny little bot |
| 20:35.24 | jbalcomb | HAHA!! "It is recommended to do firmware upgrade through TFTP..." then why does the damn phone default to HTTP!?!? |
| 20:35.49 | *** join/#asterisk Uhh (n=esmith@office.xpressamerica.net) |
| 20:36.25 | Alric | Is that Polycom? |
| 20:36.43 | jbalcomb | gxp-2000 1.1.0.16 release notes |
| 20:36.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 20:37.07 | tzafrir_laptop | benjk, http://pastebin.ca/85278 |
| 20:37.18 | benjk | The only thing I can think of is that BRIstuff futzed up zaptel |
| 20:37.24 | pdtmobile | can anybody help me with this error: Jul 11 15:36:06 WARNING[2710]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/13 already in use on span 1. Hanging up owner. |
| 20:37.41 | pdtmobile | inbound calls don't work |
| 20:37.45 | pdtmobile | outbound make it through |
| 20:38.12 | benjk | tzafrir, nice one |
| 20:38.17 | benjk | which file is that in? |
| 20:38.18 | tzafrir_laptop | benjk, the bristuff patch doesn't change much the kernel-space zaptel |
| 20:38.34 | tzafrir_laptop | somewhere under udev.d ... |
| 20:38.35 | benjk | well, its a vanilla install |
| 20:38.42 | *** part/#asterisk ManuMendoza (n=ManuMend@200.69.182.135) |
| 20:38.45 | benjk | and it is totally not working |
| 20:39.36 | *** join/#asterisk saftsack (n=saftsack@p54A7EF6D.dip.t-dialin.net) |
| 20:39.56 | tzafrir_laptop | should udevd be notified of changes t its scripts? doesn't make sense |
| 20:40.27 | benjk | I can create a rules file in /etc/udev/rules.d |
| 20:40.41 | benjk | but I guess it would help to know which file you found those in |
| 20:41.06 | tzafrir_laptop | /etc/udev/rules.d/udev.rules |
| 20:41.15 | tzafrir_laptop | that's the output of a grep |
| 20:41.22 | tzafrir_laptop | delete anything before the : |
| 20:41.44 | benjk | hmmm, I don't have udev.rules |
| 20:42.03 | benjk | i have early, default and late and a whole bunch of other rules |
| 20:42.12 | saftsack | ManxPower, hi |
| 20:42.14 | doolph | anyone know how do I use those g729 sound files correctly? |
| 20:42.39 | benjk | maybe udev-late.rules? what do you reckon? |
| 20:43.20 | benjk | I guess ISDN cards are late items relative to filesystem stuff and networking etc |
| 20:43.29 | tzafrir_laptop | I'm no udev expert. I actually don't use it in production. only on my laptop... |
| 20:43.56 | saftsack | tzafrir_laptop, hi did i speak with you because of sangomas cards? |
| 20:44.19 | tzafrir_laptop | on pkg-voip? |
| 20:44.25 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:44.28 | benjk | well, how did zaptel get those devices created without udev then? |
| 20:45.02 | tzafrir_laptop | benjk, no idea? it can't be devfs, as there is udev |
| 20:45.28 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 20:45.51 | tzafrir_laptop | benjk, anything in the system logs ? |
| 20:45.53 | benjk | what the heck, I will stick them into udev-default.rules |
| 20:46.06 | benjk | nothing that gives any clues |
| 20:46.23 | benjk | it says that zaptel is not supported by Novell, set module tainted |
| 20:46.28 | benjk | stull like that |
| 20:47.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:50.39 | benjk | ok, I put those rules into udev-default.rules, modprobed zaptel and zaphfc and now the /dev/zap structure looks fine |
| 20:50.57 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 20:52.56 | benjk | and now more crap |
| 20:53.09 | rob0 | Off-topic: anyone here know what it takes to compile a Linux i386 kernel on a Linux x86_64 machine? I'm chroot'ing into a 32-bit OS and using ARCH=i386, but no luck so far. |
| 20:53.28 | benjk | Jul 12 06:21:15 ERROR[3717]: chan_zap.c:6585 mkintf: Unable to open channel 1: No such device or address |
| 20:53.29 | benjk | here = 0, tmp->channel = 1, channel = 1 |
| 20:53.54 | benjk | I really hate this craptel stuff |
| 20:54.10 | Cresl1n | sounds like you forgot to run ztcfg |
| 20:54.12 | Strom_C | what user is asterisk running as? |
| 20:54.14 | Cresl1n | lame...... |
| 20:54.16 | rob0 | benjk: "dmesg | tail", did the driver load? "ztcfg -vv" |
| 20:54.18 | saftsack | tzafrir_laptop, no here in #asterisk |
| 20:54.25 | benjk | I did run ztcfg |
| 20:54.34 | Strom_C | benjk: what user is asterisk running as? |
| 20:54.39 | Cresl1n | file!!! |
| 20:54.52 | tzafrir_laptop | saftsack, not sure. what was it about? |
| 20:54.53 | benjk | maybe all the experimenting borked the stuff |
| 20:54.56 | benjk | lemme reboot |
| 20:56.12 | saftsack | about BRI and sangoma cards |
| 20:56.13 | benjk | root, as I did sudo asterisk -vvvgnc |
| 20:56.19 | tzafrir_laptop | benjk, this could be a handful of things. First off: is there a configured zaptel channel? did ztcfg run successfully? |
| 20:56.48 | SarahEmm | file: you were working on a bunch of * text processing code weren't you? |
| 20:56.55 | benjk | I think my experimenting screwed something up, I am rebooting now |
| 20:57.18 | Cresl1n | SarahEmm: file says yes |
| 20:58.06 | SarahEmm | uhh.. okay... does text get bridged across channels with the voice? |
| 20:58.18 | Cresl1n | yeah, it should |
| 20:58.26 | Cresl1n | it's just in an ast_frame |
| 20:58.28 | SarahEmm | cool. |
| 20:58.37 | SarahEmm | hmmm. |
| 20:59.19 | *** join/#asterisk damned (n=vpol@17.15.leased.lanck.net) |
| 20:59.39 | tzafrir_laptop | saftsack, I don't remember this, but maybe I just forgot... |
| 21:00.51 | *** join/#asterisk sekuru (n=niklas@h183n1fls31o831.telia.com) |
| 21:00.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 21:01.41 | benjk | Ok, now we're getting somewhere here |
| 21:02.16 | benjk | *CLI> pri show span 1 |
| 21:02.16 | benjk | Primary D-channel: 3 |
| 21:02.16 | benjk | Status: Provisioned, Up, Active |
| 21:02.18 | benjk | Yay |
| 21:02.30 | Strom_C | your d-channel is on THREE?! |
| 21:02.33 | Strom_C | how odd |
| 21:02.34 | benjk | yea |
| 21:02.42 | benjk | now this is because its BRI |
| 21:02.49 | Strom_C | oh ok |
| 21:02.51 | tzafrir_laptop | Strom_C, bristuff: zap bri |
| 21:02.55 | benjk | and zaphfc uses libpri |
| 21:03.02 | Strom_C | I was wondering which bonkers telco provisioned a PRI D-channel on 3 |
| 21:03.03 | benjk | so it shows up as pri but its not |
| 21:03.08 | Galeras | Seems GrandStream HT488 ATA, is unable to detect busy tone to hangup a call. I'm wondering if there is something to do at asterisk level with that sip calls. I mean something like busydetect on zapata.conf. Thanks. |
| 21:03.29 | Strom_C | Galeras: busy on an analog line? |
| 21:03.32 | benjk | hmmm |
| 21:03.54 | benjk | <PROTECTED> |
| 21:04.04 | benjk | what is Extension '' ? |
| 21:04.34 | Galeras | Strom_C yes, from panasonic pbx |
| 21:04.43 | tzafrir_laptop | Strom_C, the Junghanns's gsm card's driver provide spans of a single b channel. There you'd have: Primary D-channel: 2 |
| 21:04.56 | benjk | hehe |
| 21:05.11 | benjk | I wish they had a UMTS card |
| 21:05.19 | benjk | we don't have GSM over here |
| 21:05.35 | tzafrir_laptop | What do you have? cdma? |
| 21:05.47 | benjk | we have a bunch of things |
| 21:06.07 | SarahEmm | hmmm. |
| 21:06.09 | benjk | PDC, PHS, CDMA-J, WCDMA |
| 21:06.20 | tzafrir_laptop | Here we have two major cellular carriers using gsm, and one using cdma |
| 21:06.41 | SarahEmm | here we're one GSM, a bunch CDMA |
| 21:06.47 | SarahEmm | unfortunately. GSM++ |
| 21:06.59 | benjk | the Japanese thought they were being smart by raising one proprietary standard after another and keeping GSM out |
| 21:07.31 | tzafrir_laptop | SarahEmm, unfortunetly? why? |
| 21:07.37 | benjk | but all this did was keeping Japanese companies from getting into the global GSM market |
| 21:07.43 | benjk | so they shot themselves in the foot |
| 21:07.58 | SarahEmm | tzafrir: unfortunately only one GSM provider here i mean, and they're.. not that good |
| 21:08.12 | benjk | well, we've got 0 GSM providers here |
| 21:08.16 | type0 | what about that japan only protocol? |
| 21:08.18 | type0 | DoCoMo? |
| 21:08.25 | type0 | I heard that shit is nutcase fast |
| 21:08.25 | eKo1 | docomo is a company |
| 21:08.30 | Strom_C | Galeras: can your PBX do a polarity reversal or a battery drop upon disconnect? |
| 21:08.34 | type0 | well.. what does docomo use? |
| 21:08.35 | benjk | DoCoMo is the name of the ex-state-monopoly mobile phone company |
| 21:08.58 | type0 | I was looking at their throughput speeds the other day to mobile phones |
| 21:09.15 | benjk | DoCoMo have PDC (their proprietary standard) and PHS (another proprietary standard opened up when things didn't work out) and WCDMA |
| 21:09.36 | benjk | PDC throughput is 9.6 Kbps |
| 21:09.39 | benjk | that's iMode |
| 21:09.56 | *** join/#asterisk evisu (n=hIRC@bzq-88-155-173-175.red.bezeqint.net) |
| 21:09.57 | benjk | PHS througput is 128Kbps |
| 21:10.05 | type0 | HSPDA will offer data speeds up to 8-10 Mbps (and 20 Mbps for MIMO systems) |
| 21:10.10 | *** join/#asterisk h3x (n=hex@CPE-24-208-84-116.neb.res.rr.com) |
| 21:10.16 | benjk | WCDMA is 384 kbps |
| 21:10.29 | ManxPower | type0, And I will become the richest person in the world! |
| 21:10.42 | type0 | you dont think its possible? |
| 21:10.44 | benjk | they're also testing Advanced PHS now with some Megabits bandwidth |
| 21:10.53 | *** join/#asterisk Eggplant (i=No@dsl-72-19-47-026.cascadeaccess.com) |
| 21:11.08 | type0 | I'm still waiting for a wimax mobile handset |
| 21:11.12 | benjk | anyway, anybody any idea what context '' is? |
| 21:11.14 | ManxPower | type0, I don't know, but I know it's not available in the near future so why even waste time thinking about it? |
| 21:11.26 | benjk | er extension |
| 21:11.30 | type0 | we have a huge wimax infrastructure here.. would be rad to do voip through wimax |
| 21:11.32 | ManxPower | ben_d, usually it means "empty context" |
| 21:11.37 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 21:11.55 | benjk | sure but why would extension s not match that? |
| 21:12.02 | Galeras | Srtom_C: It seems that none of these 2, it is an old panasonic pbx KX-TEM824 |
| 21:12.12 | ManxPower | type0, Ah. OK. Since WiMax has not even been finalized yet, I find it interesting that you have a huge WiMax deployment |
| 21:12.34 | ManxPower | benjk, I don't know, perhaps the destination extension is empty, rather than unknown. |
| 21:12.51 | type0 | www.clearwire.com |
| 21:12.54 | benjk | its an incoming call from the PSTN on my ISDN line |
| 21:12.57 | ManxPower | type0, What specific make/models of WiMax APs do you have? |
| 21:13.11 | benjk | so the destination is my primary phone number |
| 21:13.30 | type0 | I live in the largest city by square mileage in the country.. and every inch is covered by wimax data |
| 21:13.35 | ManxPower | benjk, not if it's saying "extension ''" |
| 21:13.47 | benjk | well, the call came in on that line |
| 21:13.50 | type0 | additionally, there is a CDMA EVDO network here that covers the same area |
| 21:13.55 | benjk | maybe the signalling is borked then |
| 21:14.43 | Galeras | Strom_C: is an old panasonic pbx KX-TEM824, seems can't do a polarity reversal or battery drop upon disconnect. Just play a hangup tone. |
| 21:15.02 | ManxPower | type0, What specific make/model of WiMax CPE do you use? |
| 21:15.10 | type0 | i dont use wimax |
| 21:15.15 | type0 | im simply saying, its here.. and selling |
| 21:15.20 | type0 | www.clearwire.com is the company |
| 21:15.22 | terrapen | hahahah i am so evil |
| 21:15.25 | ManxPower | type0 we have a huge wimax infrastructure here.. would be rad to do voip through wimax |
| 21:15.38 | type0 | the city that I live in (we) |
| 21:15.39 | terrapen | i keep an archive of netblocks by-country on my web server |
| 21:15.48 | terrapen | and some asshole set up a cron job to request them every hour |
| 21:16.00 | terrapen | i'm sure he's using them for his rbldnsd or other mail filter |
| 21:16.13 | terrapen | well, I just added every class A network to the files :) |
| 21:16.21 | ManxPower | terrapen, 8-) |
| 21:16.22 | terrapen | his mail server will break very shortly :) |
| 21:16.29 | eKo1 | hehehe |
| 21:16.43 | terrapen | 24.0.0.0/8 yeah baby |
| 21:17.33 | Qwell[laptop] | that's just mean |
| 21:17.51 | terrapen | next time, ask before cron'ing requests to my servers :) |
| 21:18.25 | file | Qwell[laptop]: moo! |
| 21:18.48 | ManxPower | type0, How do you know. There is not a single mention of WiMax on their web site. |
| 21:18.59 | type0 | heh .. hold on |
| 21:19.13 | file | it's pre-wimax |
| 21:19.34 | ManxPower | Anyway, it doesn't do me much good since you can't actually buy the APs and clients without service as far as I can tell. |
| 21:19.43 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 21:20.00 | *** join/#asterisk sekuru (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 21:20.05 | ManxPower | file, so it's not licensed MMDS type of stuff? |
| 21:20.15 | Egonis | When I try to Dial(Console/alsa) I get 'No channel type registered for 'Console'' although asterisk.conf has CONSOLE=Console/alsa configured |
| 21:20.24 | file | that's pretty much all I know about it, haven't researched much |
| 21:20.35 | type0 | http://www.redherring.com/Article.aspx?a=16166&hed=Startup+Casts+%241B+into+WiMAX |
| 21:21.15 | ManxPower | Egonis, That means there is no channel type of Console registered. You prolly did not have the ALSA development packages installed |
| 21:22.09 | Egonis | Ah, so I need alsa-lib, alsa-driver, alsa-firmware, alsa-headers before compiling asterisk? |
| 21:22.15 | Egonis | ManxPower: I use Gentoo btw |
| 21:22.28 | ManxPower | <PROTECTED> |
| 21:22.28 | ManxPower | <PROTECTED> |
| 21:22.31 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 21:22.57 | ManxPower | sounds to me there is no actuall indication they are doing ANY WiMax at this point. They currently operate in the licensed bands in the USA. |
| 21:23.16 | type0 | alaska they're doing wimax |
| 21:23.32 | type0 | they put the wimax repeaters on top of all the cellular towers in anchorage |
| 21:23.36 | ManxPower | As usual there is no actual products available for sale. |
| 21:23.44 | Egonis | ManxPower: So should I recompile Asterisk with all alsa packages present and pray? :) |
| 21:23.53 | benjk | ok, found the culprit ... only works with immediate=yes |
| 21:24.15 | file | they are using their own pre-wimax equipment... |
| 21:24.19 | type0 | Currently the company has limited access to the 2.5 GHz licensed spectrum because the mobile carrier Sprint owns or leases licenses covering 85 percent of the U.S. population, .. Sprint owns ZERO licenses in alaska |
| 21:24.19 | ManxPower | benjk, that is WEIRD. immediate=yes normally BREAKS stuff. |
| 21:24.42 | benjk | <PROTECTED> |
| 21:24.42 | benjk | <PROTECTED> |
| 21:24.42 | type0 | GCI owns most of the PCS spectrum.. and guess who their big i provider is.. they have a rack in the south anchorage network center ;) |
| 21:24.52 | ManxPower | file, I'm having trouble figuring out why I should CARE. |
| 21:25.03 | file | type0: so am I |
| 21:25.22 | ManxPower | file, sounds to me like just another carrier lock in, like the Sprint wireless fixed broadband stuff. |
| 21:25.53 | file | I would most likely say it is |
| 21:26.07 | type0 | why cant phone companies work together? |
| 21:26.14 | Qwell[laptop] | What are you, new? |
| 21:26.18 | type0 | haha |
| 21:26.18 | Qwell[laptop] | $$$ |
| 21:26.21 | ManxPower | Wake me up when you can actually BUY WiMax (or pre standard) APs and clients. |
| 21:26.42 | type0 | what is clearwire giving out then? |
| 21:26.48 | type0 | they actually give you a clearwire "router" |
| 21:27.04 | ManxPower | type0, I don't want their service. I want to do my own WiMax stuff |
| 21:27.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.98) |
| 21:27.36 | ManxPower | Can you buy their equipment without signing up for their service? |
| 21:27.41 | type0 | http://www.clearwire.com/wireless-broadband/overview.php |
| 21:27.46 | ManxPower | nortex, you can't since they only use licensed bands right now. |
| 21:27.51 | type0 | I'm sure I could figure something out |
| 21:28.21 | Assid | anprovider like voipstunt/sipdiscount.. but allows you to set your own callerid? |
| 21:28.26 | file | type0: I highly doubt it |
| 21:28.30 | ManxPower | I don't see anything that will remove the need for me to spend $500/month on a T-1 for internet. |
| 21:29.34 | Egonis | ManxPower: I recompiled asterisk with full alsa support, but still get the same message (cause 66 - Channel not implemented) |
| 21:30.02 | ManxPower | Eggplant, "show modules" is chan_alsa loaded? |
| 21:30.21 | *** join/#asterisk kc5cqm (n=michael@2002:a55f:d1d:0:0:0:0:1) |
| 21:30.45 | ManxPower | Eggplant, in /etc/asterisk/modules.conf is there a noload for alsa? |
| 21:31.04 | Uhh | ManxPower: Since he's Egonis and not Eggplant, he might not know you're talking to him :) |
| 21:31.18 | kc5cqm | howdy |
| 21:31.24 | Egonis | ManxPower: noload => chan_oss.so and load => chan_alsa.so |
| 21:31.28 | ManxPower | Uhh, sill autocomplere |
| 21:31.42 | ManxPower | Egonis, and does "show modules" show it? |
| 21:32.06 | saftsack | ManxPower, hi |
| 21:32.06 | Egonis | good question |
| 21:32.32 | saftsack | ManxPower, is it right, that you have echo cancellor hardware in use? |
| 21:32.33 | Egonis | MansPower: Yes, but it has a use count of 0 |
| 21:34.54 | ManxPower | saftsack, I have a tellabs EC I got from ebay in use |
| 21:35.05 | ManxPower | Egonis, hold on |
| 21:35.36 | saftsack | ManxPower, did you use software echo cancelling before? |
| 21:35.55 | ManxPower | saftsack, YES |
| 21:36.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:36.26 | saftsack | did you experiene any differents? |
| 21:37.34 | ManxPower | saftsack, yes |
| 21:38.34 | So3kris | Good night |
| 21:39.03 | ManxPower | Egonis, http://pastebin.ca/85349 |
| 21:39.04 | saftsack | ManxPower, my problem here is, that i can't call from my voip telephon into the telephon net here without getting echo |
| 21:39.17 | *** join/#asterisk ajohnson (i=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 21:39.23 | saftsack | is your sound quality as good as with a classic TK? |
| 21:39.33 | ManxPower | saftsack, TK? |
| 21:39.38 | *** part/#asterisk ajohnson (i=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 21:39.44 | ManxPower | saftsack, Tellabs EC is what the TELCOS use for EC |
| 21:40.58 | saftsack | TK = is what asterisk is just in hardware like ELMEG systems etc. |
| 21:41.18 | saftsack | is the tellabs EC bound to a card? |
| 21:41.23 | ManxPower | I have never used a Digium EC module |
| 21:41.45 | ManxPower | The Tellabx EC takes a T-1, does echocancel, then spits out a T-1 |
| 21:41.47 | saftsack | me too ;) |
| 21:42.08 | ManxPower | you place it between the telco and Asterisk |
| 21:42.12 | saftsack | ok, do you know if there are BRI isdn ECs too? |
| 21:42.24 | ManxPower | saftsack, there are none that I know of |
| 21:42.28 | saftsack | kk |
| 21:42.35 | saftsack | i hate germany with its BRI lines ^^ |
| 21:42.48 | Egonis | ManxPower: I still get the same message, very interesting |
| 21:42.51 | saftsack | TELCO = telephone provider? |
| 21:42.53 | eKo1 | It is better than regular POTS though |
| 21:42.58 | Egonis | ManxPower: I can play to the device via noxmms with no issue |
| 21:43.16 | ManxPower | Eggplant, that pastebin is COPIED from a working configuration |
| 21:43.50 | Egonis | ManxPower: I agree... that's why it's so strange to me |
| 21:44.04 | Egonis | ManxPower: In your extensions.conf, what is CONSOLE=? |
| 21:44.20 | ManxPower | Egonis, I do not define it. |
| 21:44.27 | Egonis | ManxPower: I will try that |
| 21:44.28 | saftsack | TELCO -> tellabs EC -> Asterisk Server -> tons of IP PHONES |
| 21:44.29 | ManxPower | Console is like Zap or SIP. It's a technology |
| 21:44.39 | saftsack | would this built up work properly without echos? |
| 21:44.54 | ManxPower | saftsack, It should |
| 21:45.28 | Egonis | ManxPower: Didn't work, I'm trying a recompile |
| 21:45.51 | ManxPower | After I put in the Tellabs I set my txgain and rxgain to 0 in Asterisk, then disabled Asterisk's EC. Faxes started working well and no complaints about volume or echo |
| 21:46.08 | saftsack | hmm the problem is, that a our TELCO here in germany wants 280EURO / month for an E1 |
| 21:46.29 | ManxPower | saftsack, that is very inexpensive |
| 21:46.57 | saftsack | but too high just for testing issues |
| 21:47.00 | ManxPower | I wish I could find a T-1 for 280EURO |
| 21:47.19 | saftsack | where do you live? |
| 21:47.41 | ManxPower | saftsack, Near Birmingham Alabama, USA, 11 miles from the CO |
| 21:48.24 | saftsack | is telephony in USA more expensive than in germany? OO |
| 21:48.50 | ManxPower | saftsack, Yes. No. |
| 21:48.57 | saftsack | ^^ |
| 21:49.06 | ManxPower | BRI is much much much more expensive |
| 21:49.18 | ManxPower | PRI can be less expensive or more expensive depending on where you live. |
| 21:49.36 | saftsack | 20eu per month here + telephone bills |
| 21:49.44 | saftsack | for 2 B-Channels (BRI) |
| 21:49.55 | Strom_C | thats cheap |
| 21:50.02 | ManxPower | saftsack, most of the USA BRI costs $100/month + taxes |
| 21:50.06 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 21:50.18 | saftsack | what? ^^ |
| 21:50.26 | ManxPower | Which is why nobody uses BRI in the USA |
| 21:50.31 | saftsack | hrhr |
| 21:50.43 | saftsack | but a normal analog one is very cheap i think |
| 21:50.56 | saftsack | thats the reason why there are just crap cards which are supporting BRI ^^ |
| 21:51.43 | ariel_ | afternoon folks |
| 21:52.05 | saftsack | hi |
| 21:54.33 | saftsack | are there any converters or extractors so that i can do TELCO FXO -> PRI -> TELLABS EC -> PRI -> FXO -> Asterisk ..... ? |
| 21:54.42 | saftsack | or do i need a channelbank though? |
| 21:55.24 | ManxPower | saftsack, why not Telco FXO -> T-1 -> Tellabs EC -> T-1 -> Asterisk |
| 21:55.39 | ManxPower | Then you don't even have to buy an E-1 channel bank and do not have to find an E-1 tellabs |
| 21:55.49 | ManxPower | ..er.. |
| 21:56.03 | ManxPower | saftsack, why not Telco FXO -> Channel Bank -> T-1 -> Tellabs EC -> T-1 -> Asterisk |
| 21:56.09 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 21:56.45 | denon | ManxPower: convert a PRI to analog, just to use the EC? |
| 21:56.58 | ManxPower | denon, not PRI. |
| 21:57.04 | denon | oh |
| 21:57.33 | [TK]D-Fender | I prefer TELCO -> OTASIC -> Asterisk myself ;) |
| 21:57.46 | [TK]D-Fender | Nice straight line :) |
| 21:57.52 | saftsack | ManxPower, because im not rich ;) |
| 21:58.00 | ManxPower | denon, E-1 or PRI is "too expensive" |
| 21:58.02 | saftsack | i just want to pay maybe 200EU for this testing environemnt |
| 21:58.07 | denon | oic |
| 21:58.12 | ManxPower | saftsack, Have a nice life. |
| 21:58.16 | saftsack | hrhr |
| 21:58.27 | saftsack | i have no problem with buying expensive hardware |
| 21:58.42 | saftsack | but with for a test environment i dont want to pay much money |
| 21:59.04 | kindor | anyone got dtmf working with voipbuster? |
| 21:59.11 | saftsack | the onliest thing what i want is to have a hardware EC and nothing more |
| 21:59.12 | *** join/#asterisk AsteriskNoob (i=Asterisk@24-117-117-211.cpe.cableone.net) |
| 21:59.28 | AsteriskNoob | hey everyone! |
| 21:59.31 | ManxPower | Tellabs EC can be as cheap as $30 on eBay if you are willing to do some work. Channel Bank can be as low as $350 on ebay |
| 21:59.42 | ManxPower | The T-1 card for Asterisk is the expesnive part, but a 1 port is under $500 |
| 21:59.49 | drray | low as $100 |
| 21:59.53 | saftsack | ManxPower, yes I know that |
| 21:59.54 | drray | zhone |
| 22:00.11 | ManxPower | drray, I'm an Adtran guy. |
| 22:00.19 | saftsack | but there would be the possibilty too then to buy eicons SINGLE port BRI card for 400eu with hardware EC |
| 22:00.26 | ariel_ | does anyone remember the problem of no sound when you had the te4XX board installed? |
| 22:00.34 | drray | I'm an adit guy now, but I cut my teeth on a zhone |
| 22:00.49 | saftsack | it is the expensivest card ive ever seen for just having 2 channels available *G* |
| 22:00.52 | Alric | Ah, Zhones. |
| 22:01.08 | eKo1 | expensivest? |
| 22:01.08 | ManxPower | drray, I thought everyone that uzed zhone committed suicide. |
| 22:01.10 | AsteriskNoob | who here would like to help me test something on a PRI? |
| 22:01.11 | drray | zhones are cruddy, but they are good enough for learning |
| 22:02.02 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:02.19 | xheliox | ~centosbug |
| 22:02.25 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:02.25 | drray | other than caller ID, and forward disconnect detection.. and the losing its marbles for no reason |
| 22:02.53 | xheliox | thank you jbot, always helpful :) |
| 22:03.30 | AsteriskNoob | who here runs their asterisk on an NI2 PRI??? |
| 22:04.59 | ManxPower | AsteriskNoob, EVERYONE THAT CAN |
| 22:05.33 | topping | anyone know any peeps at NuFone? |
| 22:05.54 | AsteriskNoob | lol @ Manx, I wanted to know who does right now, cause I have a patch I want to test somewhere else before doing the whole disclaimer thing etc... cause I have a hard time with legal mumbo jumbo |
| 22:05.55 | terrapen | topping: hahaha |
| 22:05.57 | topping | still have a dead 866 number, months after first reporting it |
| 22:06.00 | topping | yah |
| 22:06.01 | ariel_ | topping, you can always go to the #nufone section |
| 22:06.07 | topping | haha really? cool |
| 22:06.11 | terrapen | where you can yell at nobody. |
| 22:06.22 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:06.51 | ariel_ | topping, I ended up transfering the number from them to another provider. |
| 22:07.04 | ManxPower | AsteriskNoob, My servers do NOT get patches |
| 22:07.23 | topping | so i guess i'm not alone then? |
| 22:07.25 | AsteriskNoob | Manx, running the business edition? |
| 22:07.35 | topping | that's a relief |
| 22:07.35 | ariel_ | topping, no your not |
| 22:07.40 | topping | thanks :) |
| 22:08.02 | terrapen | ugh...gnarly rain storms heading my way |
| 22:08.04 | ManxPower | AsteriskNoob, no |
| 22:08.07 | terrapen | no mountain biking for me tonight :( |
| 22:08.10 | topping | the reorder message changed yesterday, so I think there may be some progress with them, but... |
| 22:08.12 | AsteriskNoob | *shrug* It's just a patch everyone has been asking for, 2BCT over NI2 :) |
| 22:08.31 | AsteriskNoob | I figured it out a couple months ago, tried to claim a bounty but that person never responded |
| 22:08.48 | terrapen | what is NI2? |
| 22:08.55 | ariel_ | hell there are no mountains here to see or climb much less ride. |
| 22:09.11 | ManxPower | AsteriskNoob, We don't have much need for 2BCT at the moment. |
| 22:09.18 | AsteriskNoob | terrapen, NI2 is a protocol that runs over an ISDN PRI circuit |
| 22:09.22 | ManxPower | terrapen, USA ISDN PRI protocol |
| 22:09.36 | terrapen | and 2BCT? |
| 22:09.41 | AsteriskNoob | Manx, thats cool, somebody out there wants to try it I'm sure |
| 22:10.04 | ManxPower | terrapen, allows you to transfer the call to a PSTN number and not tie up 2 channels |
| 22:10.13 | terrapen | ahh, neat |
| 22:10.14 | ariel_ | AsteriskNoob, send an email out to the -dev mailing list and see if someone there would like to test it out for you. |
| 22:10.30 | AsteriskNoob | ariel_ i'm not sure where that list is |
| 22:10.40 | ariel_ | digium |
| 22:10.45 | ariel_ | asterisk-dev |
| 22:10.58 | drray | I unsubscribed to all the lists yesterday :( |
| 22:11.01 | ariel_ | it's a channel here as well #asterisk-dev |
| 22:11.14 | AsteriskNoob | I would just post it somewhere but I dont understand the "disclaimer" thing you have to sign |
| 22:11.36 | drray | maybe offer it to freepbx? |
| 22:11.48 | ariel_ | AsteriskNoob, the disclaimer is you sign all your rigths to the code away to them... |
| 22:12.21 | ariel_ | drray, it's a driver patch not a gui item. |
| 22:12.27 | ManxPower | AsteriskNoob, there are 2 disclaimers. You pick which one. the first you give ALL rights to digium. The second one you just give Digium a license to use the code in any way they want to, but you keep the copyright. |
| 22:12.36 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 22:12.53 | AsteriskNoob | but if its a mod to existing code I have to put it back to the open source community anyway.... |
| 22:13.04 | AsteriskNoob | its like 10 lines of code in 1 of the files |
| 22:13.06 | AsteriskNoob | hmm |
| 22:13.12 | AsteriskNoob | i'll figure it out, i'm looking for the list |
| 22:14.19 | Docelm0 | OI! |
| 22:14.41 | Galeras | Someone achieved to detect hangup on the fxo port of a grandstream handytone 488 connected to a panasonic pbx KX---? |
| 22:15.09 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:18.10 | *** join/#asterisk dreamless (n=dreamles@port63.ds1-arsy.adsl.cybercity.dk) |
| 22:23.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.98) |
| 22:26.21 | saftsack | tell me a good reason why big companies should use asterisk and not any other systems (are there any other computer based systems?) |
| 22:28.23 | ariel_ | cost, flexable, able to bridge multi-type of systems |
| 22:28.38 | *** join/#asterisk veepster_ (n=veepster@67.130.38.2) |
| 22:29.36 | saftsack | ok but i had many problems with my hfc based asterisk in a production environment |
| 22:29.52 | saftsack | for example providing an internal dialtone was quiet impossible |
| 22:30.04 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 22:30.45 | denon | saftsack: hmm? asterisk can be very robust, when done properly |
| 22:31.01 | denon | and extremely flexible, in complex ivr routing or applications |
| 22:31.26 | AsteriskNoob | saftsack, theres lots of other computer based systems, but they're more expensive and we wont mention them... except for cisco to give you an idea, the cisco callmanager can run you about 20000 |
| 22:31.47 | saftsack | wow thats hard |
| 22:31.56 | AsteriskNoob | and it cant do half of what asterisk can |
| 22:32.06 | saftsack | okay thats an argument. |
| 22:32.09 | saftsack | http://www.digiumcards.com/digium_b410p.html |
| 22:32.22 | saftsack | WHATS THAT? digium said about 400 or 500eu on cebit |
| 22:32.40 | Strom_C | saftsack: digiumcards.com is not digium |
| 22:32.47 | AsteriskNoob | i run a cisco call manager based system with lots of phones, but i also run about 20 phones off of asterisk for friends family and my smaller business, the cisco one was in place before i was hired |
| 22:32.50 | Strom_C | saftsack: they're not even an authorized reseller |
| 22:32.57 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 22:33.33 | saftsack | AsteriskNoob, ok but one thing i cant understand. why do all systems use normal x86 computers? |
| 22:33.48 | ariel_ | saftsack, I have no issues with giving dial tones with asterisk. |
| 22:33.53 | saftsack | would be a powerpc with a flashdisk not safer? |
| 22:33.56 | SarahEmm | saftsack: they're cheap and industry standard |
| 22:34.04 | SarahEmm | how would powerpc be safer? |
| 22:34.05 | AsteriskNoob | saftsack, cause they're cheap, and i forgot to mention 20k for the cisco, thats just the call manager, the call control, you need to buy interfaces (gateways) phones, switches etc... |
| 22:34.32 | saftsack | hmm .... is it possible to built a redundant telephony system? |
| 22:34.34 | saftsack | with asterisk? |
| 22:34.44 | ariel_ | Got get a nice TNT 50,000 gateway... |
| 22:34.46 | AsteriskNoob | saftsack, my system hasnt been rebooted in 8 months |
| 22:35.14 | AsteriskNoob | 8 months! I recompiled asterisk, added features and such without ever rebooting. it hasnt crashed ever |
| 22:35.36 | saftsack | AsteriskNoob, yes thats cool but there is one thing. we have just BRI here in germany. and the bri drivers which are available ATM crashes about every 4 weeks |
| 22:35.44 | AsteriskNoob | to date i've passed a total of 14587 calls over 9 PRI channels and 20 SIP phones |
| 22:36.04 | AsteriskNoob | saft, single BRI? |
| 22:36.09 | AsteriskNoob | or multiple? |
| 22:36.39 | saftsack | the leading company which provides hfc chipset drivers offered a stable driver which had a KNOWN bug that it crashes after couple of hours |
| 22:36.42 | saftsack | 4 port BRI |
| 22:36.48 | saftsack | 2 b-Channels per port |
| 22:36.56 | AsteriskNoob | are you utilizing them all? |
| 22:37.10 | syzygybsd | AsteriskNoob: I did that in a couple days.. well not 9 PRIs.. but the number of calls |
| 22:37.14 | saftsack | yes i use all four s/t ports |
| 22:37.24 | saftsack | 2 in TE and 2 in NT mode |
| 22:37.35 | AsteriskNoob | so you have 4 channels coming in from the telco then... |
| 22:37.44 | saftsack | yes |
| 22:37.54 | AsteriskNoob | uh....... i dont know about there but over here it's a whole lot cheaper to go for a PRI after having either 6 POTS or 2 BRI's |
| 22:38.24 | hads | AsteriskNoob: Probably not in EU |
| 22:38.45 | AsteriskNoob | if i were you i'd look into a fractional PRI, just with 4 to 6 channels and see if the cost is comparible, you'll get alot more out of PRI service and you can use a Digium card, very reliable |
| 22:38.48 | saftsack | Oo ManxPower told that he pays about 1000¤ for a PRI |
| 22:38.56 | saftsack | and he lives in the usa |
| 22:39.01 | saftsack | AsteriskNoob, where do you come from? |
| 22:39.01 | AsteriskNoob | 1000? ouch |
| 22:39.07 | AsteriskNoob | I'm in Idaho (USA) |
| 22:39.11 | saftsack | i heard 280¤ one time |
| 22:39.18 | saftsack | 40¤ for 4 channels here |
| 22:39.22 | saftsack | per month |
| 22:39.26 | kc5cqm | how much for a fractional T1? |
| 22:39.30 | kc5cqm | or is that PRI? |
| 22:39.39 | ariel_ | depends on location on how much you pay for a pri where there about 400 dollars |
| 22:39.45 | AsteriskNoob | i pay $28 a channel for a mix of voice and data, i have 8 channels of data right now and 6 channels of voice |
| 22:39.46 | saftsack | PRI = E1 |
| 22:39.49 | AsteriskNoob | plus the D chan |
| 22:39.53 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 22:40.12 | AsteriskNoob | 420 bucks for 15 out of 24 possible channels |
| 22:40.16 | saftsack | AsteriskNoob, is it really possible to get a T1 with not all channels? |
| 22:40.19 | AsteriskNoob | i got exactly the service i wanted |
| 22:40.20 | saftsack | didnt know that |
| 22:40.28 | syzygybsd | AsteriskNoob: did I set that up for you? |
| 22:40.32 | AsteriskNoob | saft, here it is |
| 22:40.36 | saftsack | thought it is only possible to get the whole package |
| 22:40.50 | AsteriskNoob | syzy, i dont know... did you... i have no idea who you are... lol |
| 22:40.55 | saftsack | one second i ll visit the telcos site mom |
| 22:41.21 | AsteriskNoob | syzy: i set up the asterisk system myself... do you work for a telco? |
| 22:42.23 | kc5cqm | for typical (non-call-center) type businesses, how many lines per deskphones is the norm? |
| 22:42.24 | syzygybsd | nope, just sounds like a system I setup for someone in ID |
| 22:42.30 | AsteriskNoob | so what I have is called a "Custom T" its a T1 that goes into a box on my wall, they split off a PRI for my voice and an ethernet cable for my net. |
| 22:42.43 | ariel_ | kc5cqm 1 |
| 22:42.48 | syzygybsd | I haven't heard of many hybrid systems, and definently not in ID |
| 22:43.10 | kc5cqm | what's the use of having a PBX if you could afford 1:1? |
| 22:43.16 | AsteriskNoob | theres also time warner's "VersiPak" they do the same but let you use the unused voice bandwidth, causes trouble with data calls |
| 22:43.23 | syzygybsd | AsteriskNoob: you can do that with just your asterisk box if you make it your router too |
| 22:43.24 | ariel_ | kc5cqm, sorry I miss understood, about 10 to 1 |
| 22:43.27 | kc5cqm | gotcha |
| 22:43.33 | kc5cqm | that's what I was thinking ballpark |
| 22:43.45 | SarahEmm | that heavilly depends on the kind of office and size tho :) |
| 22:43.51 | SarahEmm | an office of 10 will need more than 1 ;) |
| 22:44.01 | ariel_ | but that is when you have allot of users. the less users you have the more lines you need |
| 22:44.09 | kc5cqm | linear above a certain number...makes sense |
| 22:44.15 | AsteriskNoob | yeah syzy i know.... but the telco likes to have their box do it, for one reason they can get in it and run tests from my side |
| 22:44.22 | ariel_ | SarahEmm, your correct |
| 22:44.44 | kc5cqm | one of these days I'd like to pitch small businesses doing this |
| 22:44.50 | ariel_ | I have a few offices of 15 people with 5 lines, that is where I start. |
| 22:44.58 | benjk | AsteriskNoob, what you have is called a fractional T1 |
| 22:45.07 | AsteriskNoob | yes, i know |
| 22:45.13 | AsteriskNoob | read above :P |
| 22:45.19 | saftsack | http://www.t-com.de/is-bin/INTERSHOP.enfinity/WFS/EKI-GK-Site/de_DE/-/EUR/ViewDisplayProductInformation-Start;sid=e5OyJqAJKWKyJuS9Uoy4voEBrXvLZ804QzpX2eO-Q0P4GQ==?PreView=true&ProductRef=0100001000091%40EKI-GK |
| 22:45.22 | kc5cqm | pretty new to * at the time. Have it at home with 1 digium card in it, 5 ata's to the phones, a few other sip and voip based providers |
| 22:45.24 | saftsack | for those who can german ;) |
| 22:45.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:45.38 | saftsack | 287,78 ¤ / MONTH |
| 22:45.50 | AsteriskNoob | ariel_: wanna partner? i was gonna do a virtual PBX sell a single internet line to people and set up their numbers, extensions, menus all on my system :) |
| 22:45.50 | ariel_ | in fact I have setup a few locations with a line from XO or FDN that has 5 phone lines rest data. Goes for 399 works great. |
| 22:46.17 | syzygybsd | kanst du nicht deutsch sprechen |
| 22:46.23 | syzygybsd | ouch... I can't spell |
| 22:46.26 | benjk | gosh these germanwordsaresoincrediblylonglonglong |
| 22:46.29 | syzygybsd | been too long... |
| 22:46.41 | saftsack | benjk, LOL |
| 22:46.44 | ariel_ | AsteriskNoob, I don't want to be a telco/VSP |
| 22:46.49 | AsteriskNoob | ja |
| 22:46.56 | saftsack | you mean primärmultiplexeranschluss? |
| 22:46.57 | saftsack | *G* |
| 22:47.48 | AsteriskNoob | saft, is that comparable to your BRI circuits? |
| 22:47.50 | benjk | primary multiplex connection or something like that |
| 22:47.58 | benjk | no its an E1 PRI |
| 22:48.05 | benjk | with 30 channels |
| 22:48.16 | AsteriskNoob | ben, i wask asking him if its comparable in price.... |
| 22:48.25 | AsteriskNoob | he's having trouble running asterisk with BRI's |
| 22:48.31 | benjk | that would be rather amazing |
| 22:48.43 | benjk | if they did sell BRI for a similar cost as a PRI |
| 22:48.56 | AsteriskNoob | he's got 2 BRI's |
| 22:49.01 | saftsack | BRI is a little bit more expensive |
| 22:49.06 | AsteriskNoob | and he's looking for a fractional E1 |
| 22:49.19 | saftsack | but if you have a little buero you need just 2 BRIs |
| 22:49.21 | benjk | over here BRI is cheaper than analog |
| 22:49.30 | saftsack | but the BRI telephony hardware is crap |
| 22:49.36 | saftsack | benjk, where do you live? |
| 22:49.39 | saftsack | here too |
| 22:49.40 | benjk | Tokyo |
| 22:50.00 | benjk | and we've got EuroISDN in Japan |
| 22:50.20 | benjk | bamboozlingamazingly |
| 22:50.41 | benjk | nobody who knows this will tell you that |
| 22:50.46 | benjk | they call it INS64 |
| 22:50.54 | benjk | to put you off |
| 22:51.10 | benjk | but its just EuroISDN with ulaw |
| 22:51.17 | saftsack | k |
| 22:51.24 | saftsack | benjk, do you use hfc cards? |
| 22:51.28 | benjk | yep |
| 22:51.43 | saftsack | they are crap imho ^^ |
| 22:51.46 | benjk | I just got it working last night (its 7:50 am) |
| 22:51.51 | AsteriskNoob | so saft, that line you posted, was that the price for the full 30? |
| 22:51.59 | saftsack | yes :) |
| 22:52.13 | AsteriskNoob | and all 30 were cheaper than your 4? |
| 22:52.15 | AsteriskNoob | dammnnnnn |
| 22:52.33 | saftsack | NO |
| 22:52.43 | AsteriskNoob | ohok, i thought you said that |
| 22:52.46 | saftsack | my 4 are about 40eu |
| 22:52.59 | AsteriskNoob | total? wow |
| 22:53.02 | saftsack | yes |
| 22:53.05 | benjk | is that 4 BRI or 2 BRI with 4 channels? |
| 22:53.08 | saftsack | + bills |
| 22:53.13 | saftsack | 2 BRI with 4 channels |
| 22:53.28 | saftsack | but "anlagenanschluss" |
| 22:53.36 | saftsack | 2 ISDN merged into one |
| 22:53.49 | AsteriskNoob | ok |
| 22:53.57 | saftsack | so that i have blockdialing :) |
| 22:54.24 | saftsack | but i search @ the telcos atm for a fractal PRI |
| 22:54.27 | benjk | hm, that is 2900 yen per BRI |
| 22:54.43 | benjk | about the same as here if you don't "own" a line |
| 22:55.09 | saftsack | benjk, with which driver did you set up your hfc card? |
| 22:55.18 | benjk | zaphfc |
| 22:55.42 | benjk | I am in the process of testing a bunch of different cards now |
| 22:55.52 | saftsack | i use this too in production environment with a 4 port hfc card |
| 22:55.56 | *** part/#asterisk smackus (n=smackus@63.149.122.93) |
| 22:56.21 | saftsack | after terminating the bus it sounds good in most situations but there are a few moments where it sounds not as well as with a normal TK |
| 22:56.31 | benjk | well, I got an HFC-S PCI card for starters to see if it works on the Japanese BRI |
| 22:56.36 | *** join/#asterisk h3x (n=hex@CPE-24-208-84-116.neb.res.rr.com) |
| 22:56.37 | saftsack | and furthermore i have 2 cheap unterminated hfc cards and the qualitiy sucks |
| 22:56.51 | saftsack | same cards as you i see |
| 22:56.52 | benjk | because until a few hours ago I wasn't sure it would work |
| 22:56.58 | saftsack | did you terminated your cards |
| 22:56.59 | AsteriskNoob | ack, i have to go |
| 22:56.59 | saftsack | ;) |
| 22:57.04 | saftsack | AsteriskNoob, k gn8 |
| 22:57.05 | AsteriskNoob | saft, good luck man |
| 22:57.07 | saftsack | see you tomorrow |
| 22:57.09 | saftsack | thanks :) |
| 22:57.32 | benjk | no, just a straight ISDN cable from the DSU to the card |
| 22:58.36 | benjk | I had to patch zaptel though to adjust for some post 2.6.15 kernel changes |
| 22:58.58 | benjk | and the udev stuff was also borked |
| 22:59.07 | saftsack | ok. did you give it a try? |
| 22:59.15 | saftsack | i mean did you phone someone? |
| 22:59.17 | benjk | yeah, it works nicely now |
| 22:59.19 | benjk | sure |
| 22:59.41 | benjk | callerid inbound and outbound is working fine and voice quality is suberp |
| 22:59.44 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:59.46 | saftsack | but with mass echo ;) |
| 22:59.47 | benjk | superb |
| 22:59.47 | saftsack | WHAT? |
| 22:59.51 | benjk | no echo |
| 23:00.02 | saftsack | voice quality suberp? doesnt believe that :( i get tons of echo here |
| 23:00.09 | saftsack | do you use an ip phone on the another side? |
| 23:01.21 | benjk | on the other side I called a few cell phones, an ordinary analog phone on an ordinary POTS line, a proprietary phone on a proprietary digital PBX and another IP phone behind another Asterisk box on what I believe is a PRI |
| 23:02.02 | benjk | the voice quality on the Docomo cell phone (using PDC) was of course crap |
| 23:02.05 | SarahEmm | cl |
| 23:02.06 | SarahEmm | oops |
| 23:02.06 | benjk | PDC is crap |
| 23:02.35 | benjk | but calling a PHS or a WCDMA cell phone is superb |
| 23:03.04 | benjk | PHS being wireless BRI anyway |
| 23:03.30 | saftsack | hmm |
| 23:03.35 | saftsack | whats your setup? |
| 23:03.43 | saftsack | telco -> BRI -> asterisk -> FXS phone?? |
| 23:04.27 | benjk | NTT ---BRI---> asterisk ---SIP---> SIP office phone |
| 23:04.40 | benjk | also asterisk ---IAXy---> analog phone |
| 23:04.47 | saftsack | k |
| 23:05.00 | saftsack | hmm thats spurious that you dont experience any echo |
| 23:05.03 | saftsack | Primär-Multiplexanschluss mit 15-ISDN-Anschlüssen gleichzeitig. |
| 23:05.49 | saftsack | :) thats a little bit smaller one |
| 23:05.49 | twisted[asteria] | T1 --PRI--> my nutsack -> te410 -> asterisk --sip--> ser --SIP--> cisco 5350 --BRI--> desk set |
| 23:05.49 | twisted[asteria] | i can't get calls |
| 23:05.49 | benjk | how much for that 15 channel E1? |
| 23:05.50 | Qwell[laptop] | twisted[asteria]: It's because you're using SER |
| 23:05.52 | twisted[asteria] | lol |
| 23:05.57 | saftsack | expensive 194eu |
| 23:06.15 | saftsack | it is a local company here in my town and i hate this company ^^ |
| 23:06.21 | surfdue | hey |
| 23:06.31 | benjk | thats as much as KVH charge for a J1 with 23 channels |
| 23:06.47 | surfdue | if anyone can call my phone +18006170141 with skype you will notice its chopping in and out how can i fix this |
| 23:06.54 | benjk | NTT charge almost 100 USD more for a J1 |
| 23:07.06 | twisted[asteria] | surfdue, more bandwidth :P |
| 23:07.13 | surfdue | twisted[asteria], it worked normallyy |
| 23:07.17 | surfdue | on a normal psnt |
| 23:07.18 | ariel_ | or just get a real service |
| 23:07.38 | twisted[asteria] | surfdue, is your number a pstn number or a skype number? |
| 23:07.39 | *** join/#asterisk lucaferr (n=lucaferr@81-232-63-166-no42.business.telia.com) |
| 23:07.45 | benjk | surfdue, Skype don't have any infrastructure, they hijack their customers' (your) PCs and bandwidth to make the service work |
| 23:07.47 | surfdue | pstn |
| 23:07.56 | twisted[asteria] | surfdue, so you're calling pstn to pstn and it's fine |
| 23:07.58 | saftsack | i need a PRI with about 6 channels |
| 23:08.02 | saftsack | thats all what i need |
| 23:08.02 | twisted[asteria] | if you call over skype, you're using BANDWIDTH, not the pstn |
| 23:08.08 | twisted[asteria] | (at least for most of the call) |
| 23:08.08 | surfdue | i know but my sip line also does it |
| 23:08.14 | benjk | sometimes that means bandwidth and CPU horsepower can be in short supply and then you get lousy call quality |
| 23:08.16 | surfdue | i think asterisk has like ehco cancelation |
| 23:08.16 | twisted[asteria] | that should point to your bandwidth then |
| 23:08.18 | surfdue | how do i turn it off? |
| 23:08.27 | surfdue | 20mb/s premium |
| 23:08.29 | surfdue | :"| |
| 23:08.30 | surfdue | lol |
| 23:08.43 | twisted[asteria] | that doesn't mean you're not still running into issues |
| 23:08.54 | Qwell[laptop] | surfdue: skype requires 800km of bandwidth |
| 23:08.59 | surfdue | FORGET SKYPE |
| 23:09.00 | surfdue | please |
| 23:09.58 | Strom_C | Skype eats the bandwidth out of your muffins! |
| 23:10.25 | benjk | Skype sucks the blood out of your ballz |
| 23:10.36 | surfdue | FORGET skype |
| 23:10.37 | surfdue | please |
| 23:10.41 | surfdue | its asterisk to pap2 |
| 23:10.52 | benjk | doesn't matter |
| 23:11.13 | benjk | if Skype is in use on your internet connection, that's bad news |
| 23:11.31 | Strom_C | If Skype is anywhere on your hard drive, it's bad news ;) |
| 23:11.32 | surfdue | its not |
| 23:11.33 | benjk | it will saturate your available capacity |
| 23:11.38 | surfdue | i just unistalled it |
| 23:11.40 | surfdue | same problem |
| 23:11.53 | benjk | if its on the hard disk but not running, that's not going to be a problem |
| 23:12.00 | twisted[asteria] | benjk, that sounded something like out of johnny mnemonic |
| 23:12.07 | surfdue | its gone |
| 23:12.09 | surfdue | i erased it |
| 23:12.15 | surfdue | and overrode the sectors it was on |
| 23:12.16 | surfdue | jeez |
| 23:12.18 | surfdue | now please help me |
| 23:12.19 | benjk | but if it is launched, it will use up all your bandwidth and CPU power in case your machine has become a supernode |
| 23:12.39 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 23:12.40 | h3x | wtf |
| 23:12.50 | h3x | why does skype have to act like ptp |
| 23:14.06 | benjk | surfdue, VoIP service over the internet is generally a no-guaranteed-service-levels affair |
| 23:15.10 | benjk | lets say you are on CATV internet and everybody in your neighbourhood is downloading pr0n, then you are likely going to experience bandwidth shortages and your calls will be choppy |
| 23:15.24 | surfdue | omg |
| 23:15.26 | surfdue | stop it! |
| 23:15.34 | surfdue | i just want to turn echo cancelation off. |
| 23:15.56 | file | echo cancellation only happens on zaptel, move on to the next window |
| 23:16.02 | benjk | hehe |
| 23:16.17 | file | (at least in Asterisk, not to say the ATA or phone doesn't echo cancel) |
| 23:16.43 | surfdue | file whats another reason |
| 23:16.45 | surfdue | becides bw and proc |
| 23:16.48 | benjk | yeah, this ATA has such an echo canceller I can hardly hear myself while I am in the house |
| 23:16.48 | surfdue | they are both not aproblem |
| 23:17.06 | file | provider? |
| 23:17.15 | benjk | CPU |
| 23:17.38 | surfdue | asterlink |
| 23:17.42 | file | provider, CPU, running processes, bandwidth... internet connection... |
| 23:17.43 | surfdue | duel athlons |
| 23:17.45 | benjk | somebody standing on the ethernet cable |
| 23:17.47 | surfdue | 20mb/s |
| 23:18.07 | benjk | that's your problem |
| 23:18.09 | file | I'm not asking questions, I'm saying what the problems could be |
| 23:18.16 | benjk | the Athlons dueling each other |
| 23:18.31 | Strom_C | what about fencing athlons |
| 23:20.08 | benjk | no athlons use pistols |
| 23:20.26 | benjk | pentium IIIs use swords |
| 23:21.09 | Strom_C | ahhh ok |
| 23:21.11 | benjk | Xeons use Ronald Reagan Star Wars laser cannons |
| 23:21.18 | Strom_C | is that where fencepost errors come from? |
| 23:21.24 | benjk | hehe |
| 23:22.02 | benjk | ARM use boxing gloves |
| 23:22.14 | file | you are very sill |
| 23:22.15 | file | er silly |
| 23:22.49 | benjk | that's why I am in this channel I suppose |
| 23:22.57 | Strom_C | asterisk is silly |
| 23:23.07 | file | it's developers are very silly |
| 23:23.22 | h3x | p3 uses useless north korean warheads |
| 23:23.35 | benjk | no that's Samsung |
| 23:23.46 | h3x | thats south korea :P |
| 23:24.01 | benjk | yeah but they don't make any chips in the North |
| 23:24.11 | benjk | some potato chips perhaps |
| 23:24.23 | Strom_C | linksys phones are lovingly handcrafted by Mongolian gnomes |
| 23:24.32 | benjk | lol |
| 23:24.58 | twisted[asteria] | asdfasdfhahahahahah |
| 23:25.08 | file | we now return you to your regular programming (hardy har har) |
| 23:25.10 | benjk | Cisco phones are designed by a million monkeys on typewriters |
| 23:25.20 | Strom_C | no, that's "Hamlet" |
| 23:25.26 | benjk | ah ok |
| 23:25.54 | file | out of this world! |
| 23:25.54 | Strom_C | ow |
| 23:25.59 | benjk | did you read the RFC for that |
| 23:26.04 | surfdue | lol |
| 23:26.11 | Strom_C | wait, wasnt that the show about the girl who could freeze time? |
| 23:26.17 | Strom_C | or something |
| 23:26.22 | benjk | IMPS |
| 23:26.26 | saftsack | has someone of you good contacts to digium? |
| 23:26.30 | surfdue | i can change time |
| 23:26.31 | benjk | Infinite Monkey Protocol Suite |
| 23:26.37 | surfdue | look at your clock its now a time |
| 23:26.39 | saftsack | because i wrote an email to them and they dont reply |
| 23:26.43 | benjk | RFC 2795 |
| 23:26.57 | saftsack | would it be better to phone digium? |
| 23:27.05 | benjk | http://rfc.net/rfc2795.html |
| 23:27.09 | file | saftsack: how long ago did you write? |
| 23:27.10 | Qwell[laptop] | saftsack: digium is mostly closed |
| 23:27.20 | file | "mostly closed" is relative |
| 23:27.20 | saftsack | i wrote on saturday |
| 23:27.31 | saftsack | Qwell[laptop], why? fair? |
| 23:27.45 | benjk | isn't "mostly closed" an oxymoron? |
| 23:27.50 | Qwell[laptop] | saftsack: Something like that |
| 23:27.54 | benjk | like "a little pregnant" |
| 23:27.57 | saftsack | what means that? |
| 23:28.07 | saftsack | lacks their money? |
| 23:28.10 | file | phones are good |
| 23:28.11 | saftsack | do they have to close? |
| 23:28.32 | Strom_C | if digium were open 24/7 it would be like a frat party |
| 23:28.50 | h3x | lots of echos ? :) |
| 23:28.58 | Qwell[laptop] | saftsack: for the evening... |
| 23:29.05 | Qwell[laptop] | just ignore me |
| 23:29.06 | file | support might still be open... |
| 23:29.09 | saftsack | kk |
| 23:29.10 | file | I don't know the hours |
| 23:29.19 | Strom_C | support is open through 7 pm central |
| 23:29.23 | saftsack | so had someone of you ever called digium? |
| 23:29.27 | Strom_C | so for another 30 minutes |
| 23:29.32 | file | you might say I have |
| 23:29.38 | saftsack | k |
| 23:29.39 | benjk | sure, just dial 500 |
| 23:29.53 | saftsack | im living in germany here so dial might be expensive ;) |
| 23:30.01 | benjk | nope |
| 23:30.07 | benjk | 500 |
| 23:30.10 | benjk | its free |
| 23:30.18 | Strom_C | FREEEEEEEEEEEEEEEE |
| 23:30.27 | saftsack | and theres another thing ^^ my english isnt as well as i think i can communicate very well with digiums support |
| 23:30.35 | saftsack | yes maybe free if you dial from the USA |
| 23:30.40 | benjk | no |
| 23:30.43 | file | you can call over VoIP |
| 23:30.49 | Strom_C | saftsack: free via IAX |
| 23:30.49 | saftsack | sounds great :) |
| 23:30.56 | saftsack | GREAT :) |
| 23:31.08 | benjk | its free wherever you have an asterisk box connected to an internet connection capable of carring an IAX call |
| 23:31.13 | saftsack | so if i have connect to the internet i can call them without paying anything? |
| 23:31.15 | benjk | its set up by defaiult |
| 23:31.21 | benjk | just dial 500 |
| 23:31.24 | saftsack | fat :> |
| 23:31.35 | saftsack | but they arent in their company now, or? |
| 23:32.06 | Strom_C | saftsack: you have another 30 minutes |
| 23:32.10 | Strom_C | call now |
| 23:32.22 | type0 | they're PST arent they? |
| 23:32.23 | benjk | first you get the prerecorded "this is the demo" stuff, then it will transfer to Digium's Asterisk box then you get a menu, 1 for sales, 2 for support etc |
| 23:32.25 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:32.38 | benjk | its just like if you called them on their PSTN line |
| 23:33.06 | benjk | of course if you call at an odd time, then you will get to voicemail |
| 23:33.21 | tzafrir_laptop | saftsack, are you in Germany? if so: why use PSTN rather than ISDN? |
| 23:33.36 | saftsack | thought about that |
| 23:33.44 | saftsack | my father ordered today one PSTN for testing issues |
| 23:33.44 | benjk | PSTN includes ISDN |
| 23:33.57 | benjk | do you mean POTS? |
| 23:34.21 | saftsack | digium will laugh about my bad voice quality of my grandstream craptel 101 |
| 23:34.26 | saftsack | ^^ |
| 23:34.52 | CunningPike | Anyone here know anything about FOP? |
| 23:35.14 | tzafrir_laptop | a bit |
| 23:35.16 | saftsack | 30 minutes to set up asterisk |
| 23:35.17 | saftsack | lol |
| 23:35.23 | benjk | FOP? |
| 23:36.26 | tzafrir_laptop | http://xmlgraphics.apache.org/fop/ |
| 23:36.27 | saftsack | aah :) i have openpbx running *duck* |
| 23:36.37 | CunningPike | ~FOP |
| 23:36.43 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 23:36.48 | benjk | running Duck? |
| 23:36.52 | benjk | DuckOS? |
| 23:36.56 | saftsack | lol |
| 23:37.22 | saftsack | duck is getting away for not being knocked out |
| 23:37.36 | *** join/#asterisk MiniDevil (n=MiniDevi@71.224.106.188) |
| 23:37.45 | MiniDevil | Hey guys |
| 23:37.56 | tzafrir_laptop | CunningPike, actually, I only know of the second. The first was just the first hit fom google. Consider asking a question some time in the near future |
| 23:39.42 | MiniDevil | Is this channel for Asterisk tech support/information, or just asterisk in general? |
| 23:39.43 | CunningPike | tzafrir_laptop: Did I do something to offend you? |
| 23:39.43 | Qwell[laptop] | MiniDevil: both |
| 23:39.43 | Strom_C | MiniDevil: both |
| 23:39.43 | SarahEmm | MiniDevil: general asterisk stuff, support, ostrichness, etc. |
| 23:39.43 | MiniDevil | Perhaps someone here can assist me. |
| 23:39.43 | Strom_C | also muffins |
| 23:39.43 | saftsack | OpenPBX Ready. |
| 23:39.43 | Qwell[laptop] | MiniDevil: perhaps, but we'll never know until you ask... |
| 23:39.43 | Strom_C | MiniDevil: just ask your question |
| 23:39.48 | saftsack | so what now. i need a SIP provider to call digium, or? |
| 23:39.53 | Qwell[laptop] | saftsack: #openpbx |
| 23:40.10 | saftsack | its similiar to asterisk :-P in iax2 things |
| 23:40.38 | tzafrir_laptop | CunningPike, no. I just assumed that there will be a followup to "Anyone here know anything about FOP?" |
| 23:40.50 | MiniDevil | I am looking at setting up a PBX for a near-future broadvoice setup. I am running on a P3 box.. 750mhz 512mb sdram, 20gb drive. I was given an Asterisk@Home 1.5CD and loaded that on there. However browsing around it appears Asterisk@Home is not Asterisk, is that true? |
| 23:41.01 | benjk | saftsack you can call them over IAX2 guest@misery.digium.com |
| 23:41.06 | Strom_C | MiniDevil: asterisk@home is icky |
| 23:41.11 | MiniDevil | I agree, Strom. |
| 23:41.17 | Strom_C | MiniDevil: wipe it clean and start with linux and asterisk :) |
| 23:41.22 | tzafrir_laptop | MiniDevil, Asterisk@Home is a system based on Asterisk |
| 23:41.41 | tzafrir_laptop | (and has been renamed recently to trixbox) |
| 23:41.44 | MiniDevil | I am trying to locate somewhere where I can get a few ISOs to make my Asterisk PBX |
| 23:41.52 | Strom_C | MiniDevil: asterisk.org |
| 23:41.55 | Strom_C | no isos necessary |
| 23:41.58 | saftsack | <PROTECTED> |
| 23:42.00 | Strom_C | just download teh tarballs |
| 23:42.05 | saftsack | benjk, this way? |
| 23:42.13 | tzafrir_laptop | No easy way to upgrade it, I believe |
| 23:42.20 | Strom_C | and install them on top of whatever linux distro you like |
| 23:42.22 | MiniDevil | You see.. that is my problem. I am not familiar with Linux. All I know is that Asterisk@Home runs off of CentOS linux. |
| 23:42.30 | benjk | saftsack, if you have an Asterisk box you can do Goto(demo,500,1) |
| 23:42.49 | *** join/#asterisk lucaferr (n=lucaferr@81-232-63-166-no42.business.telia.com) |
| 23:42.59 | benjk | if you have OpenPBX then you would need to do something like Dial(IAX2/guest@misery.digium.com) |
| 23:43.20 | MiniDevil | Which Linux would you suggest I get? I do have Mandrake ISOs but it seems bloated for a PBX. |
| 23:43.21 | saftsack | <PROTECTED> |
| 23:43.26 | saftsack | so i will try this now |
| 23:43.34 | benjk | that's kinda bad |
| 23:43.51 | benjk | because it means no matter what you dial after 9 it will always go there |
| 23:43.55 | saftsack | yes |
| 23:44.03 | saftsack | i wrote this line just for this call now :) |
| 23:44.15 | saftsack | so it wont be there after this call ;) |
| 23:44.20 | benjk | a specific number might be better |
| 23:44.24 | tzafrir_laptop | MiniDevil, basically any Linux will do. Unless you're terribly short of disk space |
| 23:44.56 | saftsack | <PROTECTED> |
| 23:44.56 | saftsack | <PROTECTED> |
| 23:45.02 | saftsack | just for you benjk :) |
| 23:45.02 | tzafrir_laptop | You should set the system not to run an X server, though. |
| 23:45.03 | benjk | yeah |
| 23:45.07 | benjk | actually it is exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) |
| 23:45.42 | MiniDevil | CentOS good to work off of? Or should I grab something like Redhat? |
| 23:45.42 | tzafrir_laptop | What version of Mandrake is it? |
| 23:45.42 | MiniDevil | The latest |
| 23:45.44 | Strom_C | MiniDevil: CentOS has bugs |
| 23:45.54 | Strom_C | MiniDevil: I like Debian, others like Gentoo or Slack |
| 23:46.04 | MiniDevil | I head alot about slackware. |
| 23:46.16 | MiniDevil | Gentoo I havent heard much of, once or twice. |
| 23:46.24 | benjk | you mean Debian, Gentoo and Slackware don't have bugs? |
| 23:46.33 | Qwell[laptop] | gentoo doesn't |
| 23:46.40 | benjk | ah yeah |
| 23:46.44 | saftsack | the voice menu doesnt works :( |
| 23:46.46 | Strom_C | well, CentOS has at least one showstoppingly disastrous bug |
| 23:46.50 | saftsack | Jul 12 01:46:45 WARNING[3054349232]: timer.c:84 _set_interval: Requested a timer with 1000000 nanosecond interval, but system timer reports a resolution of 4000250 nanosec. Timing may be unreliable! |
| 23:46.52 | Qwell[laptop] | ~centosbug |
| 23:46.58 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:46.58 | tzafrir_laptop | I see no reason why Mandrake would not work. But I don't know it wel enough |
| 23:46.58 | saftsack | :/ so i have to take asterisk |
| 23:46.58 | Qwell[laptop] | saftsack: #openpbx |
| 23:47.10 | saftsack | vdr:~# nano /etc/asterisk/extensions.conf |
| 23:47.13 | tzafrir_laptop | mutt doesn't have bugs. It has fleas (according to mutt(1) ) |
| 23:47.19 | benjk | saftsack, I think you have some issue with your box |
| 23:47.38 | benjk | LOL |
| 23:48.36 | MiniDevil | Gentoo 2006.0 Universal install CD ---- ppc = pentium processor core? |
| 23:48.43 | trelane | Power PC |
| 23:48.44 | SarahEmm | no |
| 23:48.45 | Strom_C | power PC |
| 23:48.46 | trelane | try #gentoo |
| 23:48.46 | tzafrir_laptop | benjk, there is a FLEAS section in the man page immedietly after that. And it comes with a utility called "flea" to report those fleas. |
| 23:48.47 | SarahEmm | ppc = PowerPC |
| 23:48.53 | MiniDevil | ah ok #gentoo |
| 23:48.55 | benjk | HAHA |
| 23:49.40 | saftsack | benjk, no error with * now :) |
| 23:49.46 | trelane | MiniDevil, gentoo is not a good distribution for those who know not what they're doing, read the manuals before you start, they're friendly and well written, whereas I'm not :) |
| 23:49.57 | saftsack | i cant go through the voice menu :( |
| 23:50.08 | saftsack | if i press a digit nothing happens :( |
| 23:50.19 | Strom_C | saftsack: oh bloody hell, just use a regular phone |
| 23:50.33 | benjk | saftsack, are you using a SIP phone between you and your Asterisk box by any chance? |
| 23:50.59 | saftsack | yes |
| 23:51.20 | benjk | looks like your SIP phone's DTMF setting is out of whack |
| 23:51.44 | saftsack | grandstream 101 ^^ |
| 23:51.48 | benjk | make sure it is the same on the phone and in your sip.conf entry for that phone |
| 23:52.12 | benjk | for example dtmfmode=info in sip.conf and SIP INFO in the admin menu of the Barbitone |
| 23:52.16 | benjk | Barbietone |
| 23:52.31 | Strom_C | Barbietone? :) |
| 23:52.34 | benjk | yeah |
| 23:52.50 | Strom_C | I would have thought "sneezeanditfallsaparttone" |
| 23:53.02 | benjk | hence barbietone |
| 23:53.08 | Strom_C | heh |
| 23:53.11 | benjk | as in toy |
| 23:53.18 | Strom_C | yup :) |
| 23:53.23 | saftsack | whats barbietones standard pw? |
| 23:53.29 | saftsack | admin :) |
| 23:54.27 | saftsack | works :) |
| 23:54.59 | saftsack | FUCK :( no one is there anymore :( |
| 23:55.05 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 23:55.07 | benjk | they did that on purpose |
| 23:55.10 | SarahEmm | eep! |
| 23:55.14 | SarahEmm | hihi trelane, ltns |
| 23:55.14 | benjk | just to upset you |
| 23:55.20 | saftsack | benjk, ^^ |
| 23:55.35 | Egonis | When I try to load asterisk I get -- /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_osp_terminate |
| 23:56.10 | saftsack | i go to bed now |
| 23:56.12 | saftsack | it is 2:00 here |
| 23:56.27 | saftsack | thanks for your help :) ill be back later |
| 23:56.28 | benjk | good night then |
| 23:56.40 | saftsack | thanks |
