| 00:00.05 | file | CDFAssociates: you have to give me the link... |
| 00:00.06 | dlynes_home | damn |
| 00:00.07 | Qwell | ~lart himself |
| 00:00.09 | TripleFFFF | he check out what the bleep to we know.. ifs quantum phys coolio |
| 00:00.10 | TripleFFFF | ;) |
| 00:00.17 | dlynes_home | didn't know jbot had one of those appendages |
| 00:00.19 | CDFAssociates | file: http://pastebin.com/743855 |
| 00:00.31 | TripleFFFF | so anyone have eyebeam solution ? |
| 00:01.53 | file | looks fine... except for the retransmits... |
| 00:03.41 | CDFAssociates | ??? |
| 00:05.55 | *** join/#asterisk moua (i=david@free.hd.free.fr) |
| 00:05.57 | moua | hi |
| 00:06.08 | cj | howdy moua |
| 00:07.39 | *** join/#asterisk Malthus (n=admin@uslec-66-255-41-2.cust.uslec.net) |
| 00:08.12 | moua | is there any how-to to run my own easypabx.com/pbxes.com service ? it's for personnal use only |
| 00:09.13 | moua | I wish to trigger a callback, by calling a DID number from a PSTN line to SIP account #1 |
| 00:09.13 | moua | then another Sip account call me back and give me the tone, |
| 00:09.13 | moua | to use the best SIP account for my desired destination. |
| 00:10.01 | moua | i have a dedicated server to run that |
| 00:12.13 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:12.23 | CDFAssociates | dlynes_home: do you have any more ideas? |
| 00:12.27 | cj | so... asterisk seems to be responding to the request for extension #25 |
| 00:12.31 | cj | but this is all I get... |
| 00:12.37 | cj | May 28 17:11:45 WARNING[15008]: chan_sip.c:3490 process_sdp: Unknown SDP media type in offer: video 5014 RTP/AVP 31 |
| 00:12.42 | cj | and I get a tone in my softphone |
| 00:12.53 | *** join/#asterisk orlock (n=jwr@202.44.174.4) |
| 00:13.12 | orlock | Do i need my phones firmware to be able to use chan_sccp? |
| 00:15.06 | dlynes_home | CDFAssociates: your phone is set to ulaw,alaw,gsm? |
| 00:16.18 | dlynes_home | cj: turn off video on your softphone |
| 00:17.14 | CDFAssociates | Yes |
| 00:17.50 | dlynes_home | CDFAssociates: turn on g729...i'm guessing your provider requires it for outbound calls |
| 00:18.18 | dlynes_home | CDFAssociates: i..e the call you're trying to place only has terminators that can handle g729 (they don't handle ulaw, alaw, or gsm) |
| 00:18.53 | CDFAssociates | According to there documentation we use g711u |
| 00:19.04 | Malthus | anyone tried messing with the firmware on the artdios? |
| 00:19.12 | dlynes_home | CDFAssociates: you're not getting audio, right? |
| 00:19.23 | Malthus | for some odd reason they don't support callerid name |
| 00:19.28 | CDFAssociates | We get a busy signal. |
| 00:19.32 | cj | dlynes_home: heh, now I just get the tone and no errors on the console :) |
| 00:19.39 | dlynes_home | CDFAssociates: yeah, so do as i say |
| 00:19.44 | CDFAssociates | k |
| 00:19.44 | dlynes_home | CDFAssociates: enable g729 on your phone |
| 00:19.58 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a222.nwlnnh.tds.net) |
| 00:19.59 | dlynes_home | cj: any errors in your log file? |
| 00:20.05 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 00:20.52 | cj | http://rafb.net/paste/results/7rECaB68.html |
| 00:20.55 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a222.nwlnnh.tds.net) |
| 00:23.07 | *** join/#asterisk FaithX (n=FaithX@mail.familyfirst.org.au) |
| 00:23.17 | CDFAssociates | Tried that and it did not work. |
| 00:23.29 | dlynes_home | CDFAssociates: did you adjust it in your sip.conf file as well? |
| 00:24.39 | CDFAssociates | yes |
| 00:24.58 | dlynes_home | CDFAssociates: set your preferred codec to g729 on both side and on the phone |
| 00:26.06 | cj | dlynes_home: any suggestions? :) |
| 00:26.25 | dlynes_home | cj: didn't see taht |
| 00:27.17 | *** join/#asterisk lately (n=doug@ppp167-252-31.static.internode.on.net) |
| 00:27.48 | dlynes_home | cj: that's not the whole log file |
| 00:27.55 | dlynes_home | cj: could you paste the whole call's log? |
| 00:28.38 | cj | dlynes_home: the whole call's log... I'll grab the contents of the full log and paste them in a pastebin... is that what you mean? |
| 00:29.08 | dlynes_home | cj: just the part of tghe log where the call starts and the call ends |
| 00:29.17 | cj | http://rafb.net/paste/results/RfKj4X72.html |
| 00:29.22 | cj | same thing |
| 00:29.54 | [TK]D-Fender | cj : Ok, what exactly are you calling from, and to? |
| 00:30.29 | dlynes_home | cj: where's the rest of the call? |
| 00:30.44 | dlynes_home | cj: i only see the debug portion, not the rest of it |
| 00:31.04 | dlynes_home | cj: no errors, no warnings, no verbose, no notices |
| 00:32.34 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 00:39.41 | *** join/#asterisk wilane_ (n=user@196.207.218.107) |
| 00:40.55 | CDFAssociates | dlynes_home: I tried setting everything to g729 and still no luck. |
| 00:41.58 | CDFAssociates | dlynes_home: The provider is using a Tekelec 9000 switch if that helps any. |
| 00:43.40 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 00:44.58 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 00:45.06 | jhiver | hi all |
| 00:45.21 | jhiver | any idea how I can retrieve the caller number under AGI? |
| 00:46.38 | jhiver | oh boy, maybe this is a bad time, the channel looks pretty dead :) |
| 00:46.43 | dlynes_home | CDFAssociates: no idea...never heard of it |
| 00:47.08 | CDFAssociates | k |
| 00:47.45 | dlynes_home | CDFAssociates: i'm out of ideas, but i'm far from being a sip expert, too |
| 00:49.11 | nextime | jhiver : you can use GET VARIABLE agi command |
| 00:49.43 | jhiver | so... GET CALLERID ? |
| 00:49.52 | nextime | jhiver : no, GET VARIABLE CALLERID |
| 00:49.57 | jhiver | ok |
| 00:50.01 | jhiver | I'll give that a try |
| 00:50.09 | jhiver | thw |
| 00:50.11 | jhiver | thx |
| 00:53.06 | P-NuT | Hi all, Is anybody using a X100P under ubuntu 5.10? Have they got the drivers working? |
| 00:53.26 | jhiver | mhhh actually I have another question |
| 00:53.37 | dlynes_home | dood |
| 00:53.41 | dlynes_home | it's not winter anymore |
| 00:53.58 | jhiver | is it possible to set a variable which will be different depending on the peer / user which the call comes from? |
| 00:54.12 | jhiver | any ideas? |
| 00:54.14 | dlynes_home | jhiver: there's already one set |
| 00:54.20 | jhiver | ? |
| 00:54.24 | jhiver | sounds good :) |
| 00:54.25 | dlynes_home | jhiver: CallerID(num) |
| 00:54.43 | jhiver | sorry, no that's the number of the caller |
| 00:54.51 | *** join/#asterisk bigmac4444 (n=mtur2848@CPE-58-170-42-34.qld.bigpond.net.au) |
| 00:54.55 | dlynes_home | jhiver: which is the peer/user the call came from |
| 00:54.56 | jhiver | but not the provider which carries the call |
| 00:55.09 | dlynes_home | jhiver: you said the call comes from, not the call goes to |
| 00:55.15 | jhiver | ? |
| 00:55.17 | bigmac4444 | g'day all =) |
| 00:55.19 | file | dlynes_home: give up now, just give up |
| 00:55.24 | dlynes_home | file: lol |
| 00:55.38 | jhiver | dlynes_home, I'm confused :) |
| 00:55.45 | dlynes_home | obviously |
| 00:55.49 | dlynes_home | if you weren't confused |
| 00:55.50 | file | jhiver: there's a setvar option that allows you to set whatever variable you want with whatever information you want... on channels that get authenticated from a user/peer - at least in SIP |
| 00:55.55 | dlynes_home | you wouldn't be asking that question |
| 00:56.04 | jhiver | :-) |
| 00:56.09 | jhiver | so right |
| 00:56.15 | file | setvar=VARNAME=contents |
| 00:56.34 | jhiver | You can do setvar in a [user] definition in sip.conf or iax.conf? |
| 00:56.44 | file | I know you can do it in sip |
| 00:56.57 | file | I can check iax in a sec |
| 00:57.02 | jhiver | oh well that sounds good |
| 00:57.12 | dlynes_home | jhiver: or in sip.conf use callerid=666 |
| 00:57.13 | file | you can do it in iax too |
| 00:57.25 | dlynes_home | jhiver: same for h323.conf and iax.conf |
| 00:57.28 | jhiver | sounds cool |
| 00:57.31 | jhiver | thanks lads |
| 00:57.36 | litage | how can i play gsm-encoded audio files within programs such as xmms, noatun, etc? |
| 00:57.58 | jhiver | so setvar=FROM_NETWORK=<user_name> will do it |
| 00:57.59 | jhiver | cool |
| 00:58.52 | jhiver | Gosh I'm so weak with Asterisk :) |
| 00:59.06 | *** join/#asterisk psi_force (n=mark@c220-237-128-179.mckinn1.vic.optusnet.com.au) |
| 00:59.25 | psi_force | hi all I'm having a problem with voice mail |
| 00:59.52 | psi_force | I get the following message from the console "app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '007007'" |
| 01:00.04 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 01:00.18 | file | I think it's rather explanitory |
| 01:00.33 | rainkid | can anyone point me to a music-on-hold install doc? the config file is not very helpful. |
| 01:00.56 | psi_force | but "show voicemail users" states |
| 01:00.59 | bigmac4444 | anyone: where would i go to configure the voicemail operator options? Some options the operator gives us we dont want. Thx |
| 01:01.07 | psi_force | local 007007 Mark 0 |
| 01:01.28 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 01:01.42 | file | psi_force: did you specify the voicemail context? if not it searches in default, and if your entry 007007 is not in default... it won't find it |
| 01:02.07 | dlynes_home | file: it says right above that 007007 is in the local context, not the default context :) |
| 01:02.18 | mitcheloc | does anyonek now the best speech to text translator around? windows/linux, i don't care? (command line interface though, or sdk) |
| 01:02.18 | file | dlynes_home: I meant in his dialplan |
| 01:02.38 | droops | hey im trying to send asterisk a restart gracefully from bash. i do an: asterisk -rx restart gracefully and it doesnt work, what is teh correct syntax for that |
| 01:02.59 | Qwell | mitcheloc: The one from lumenvox (file? is that right?) seemed really good |
| 01:03.28 | file | yes Lumenvox's is cool... works well |
| 01:03.52 | mitcheloc | i'm looking it up |
| 01:03.59 | psi_force | file: doh! should have 007007@local |
| 01:04.01 | psi_force | file: thanks |
| 01:06.21 | bigmac4444 | droops: asterisk -rx "restart now" |
| 01:06.28 | bigmac4444 | include quotes |
| 01:06.37 | droops | thank you sir |
| 01:06.40 | bigmac4444 | yw |
| 01:06.54 | mitcheloc | lumenvox looks good, i'm wondering about their licensing though, i.e. redistribution with another product |
| 01:07.17 | bigmac4444 | rainkid: have you installed mpg123 ? |
| 01:07.28 | file | hell if I know anything about that... I just write code :D |
| 01:07.39 | rainkid | no, using 1.2 and the addon package |
| 01:07.43 | rainkid | format_mp3 |
| 01:08.07 | rainkid | hmmm, do you need to have kernel sound support and a sound card? |
| 01:09.34 | dlynes_home | rainkid: only if you |
| 01:09.41 | bigmac4444 | as far as i know you dont. We didnt. |
| 01:09.42 | dlynes_home | rainkid: only if you're planning to play it out your speakers |
| 01:09.59 | file | mitcheloc: nuance is another one, but I have no experience with them or their stuff |
| 01:10.00 | rainkid | right.. all i want to do is send it down the wire to the caller |
| 01:10.08 | dlynes_home | rainkid: for moh, it just sends the output of the mp3 file to stdout |
| 01:10.20 | bigmac4444 | rainkid: what lines have you got in your musiconhold.conf ? |
| 01:10.24 | dlynes_home | rainkid: then asterisk grabs the stdout, and plays it out to the channel |
| 01:10.51 | *** join/#asterisk cybergyp1y (n=mark@APoitiers-156-1-42-86.w86-213.abo.wanadoo.fr) |
| 01:11.18 | mitcheloc | file: i'm looking them up, i think that licensing on these types of products is going to kill my plan for world domination |
| 01:12.02 | dlynes_home | oh yeah...did anyone see the newsblast? |
| 01:12.06 | dlynes_home | microsoft is doing iptv now |
| 01:12.14 | rainkid | ahh, it's working. i was using mode=quietmp3 instead of mode=files |
| 01:12.25 | bigmac4444 | as in tv network streaming? |
| 01:12.32 | bigmac4444 | =) |
| 01:12.52 | dlynes_home | rainkid: if mode=files isn't working it's because you never installed format_mp3 from asterisk-addons and you never loaded it in modules.conf, or some combination thereof |
| 01:12.54 | mitcheloc | dlnes_home: i think they've been at it for over a year now...? |
| 01:12.57 | dlynes_home | bigmac4444: yes |
| 01:13.13 | dlynes_home | mitcheloc: dunno...they just announced it in this month's telephony magazine |
| 01:14.15 | bigmac4444 | gees, there'd be a massive load on bandwidth wouldnt there? |
| 01:14.29 | bigmac4444 | depending on connections, lol |
| 01:14.44 | mitcheloc | *predicts the future*....the lines are blurring between phone calls and tv now, they'll be one and the same soon, next stop for iptv is interactive services and simple voip with webcams in the tv units |
| 01:14.45 | dlynes_home | bigmac4444: if you're the isp, who cares? |
| 01:14.50 | dlynes_home | bigmac4444: it's all edge cached, then |
| 01:15.12 | mitcheloc | it's not like verizon hasn't been rolling out fiber directly to homes around here anyway... |
| 01:15.15 | bigmac4444 | in one way, yes |
| 01:15.46 | bigmac4444 | might look into that |
| 01:15.49 | mitcheloc | i bet we see iptv with interactive gambling first ;) |
| 01:15.55 | bigmac4444 | lol |
| 01:16.03 | bigmac4444 | and the adult channels |
| 01:16.07 | jhiver | aaaargh, my AGI script doesn't do anything (not even print the debug commands) and I don't know what's going on |
| 01:16.12 | jhiver | works fine on the command line |
| 01:16.17 | jhiver | it's executable too |
| 01:17.01 | mitcheloc | jhiver: post it up somewhere |
| 01:17.13 | jhiver | sure I can do that |
| 01:17.17 | jhiver | w8 a sec :) |
| 01:17.58 | tainted_ | mitcheloc i have one reason why tv will never merge with phone |
| 01:18.26 | jhiver | http://pastebin.ca/index.php |
| 01:18.28 | jhiver | oops |
| 01:18.29 | jhiver | :) |
| 01:18.37 | jhiver | http://pastebin.ca/59484 |
| 01:19.50 | nextime | jhiver : agi debug on the * CLI? |
| 01:20.10 | jhiver | ok let me try that :) |
| 01:21.12 | nextime | jhiver : in which timezone you are? |
| 01:21.20 | jhiver | GMT+4 |
| 01:21.24 | jhiver | now that's stange |
| 01:21.27 | jhiver | I have this: |
| 01:21.36 | mitcheloc | tainted: ? |
| 01:21.43 | jhiver | AGI Rx << GET VARIABLE CALLERID |
| 01:21.43 | jhiver | AGI Tx >> 200 result=0 |
| 01:21.50 | jhiver | and AGI Rx << GET VARIABLE SYNAPSE_CALLEE |
| 01:21.50 | jhiver | AGI Tx >> 200 result=0 |
| 01:22.00 | jhiver | but in the dialplan I set this last varialbe |
| 01:22.22 | jhiver | exten => _0262XXXXXX,1,Macro(dial-lcr,262${EXTEN:1}) |
| 01:22.29 | jhiver | and then: |
| 01:22.30 | jhiver | [macro-dial-lcr] |
| 01:22.31 | jhiver | exten => s,1,SetVar(SYNAPSE_CALLEE,${ARG1}) |
| 01:22.31 | jhiver | exten => s,2,DeadAGI(synapse_lcr.agi) |
| 01:22.46 | dlynes_home | edmonchuck's gonna kick so much ass!!!!!!!!!!!!! |
| 01:23.13 | Qwell | jhiver: Set(SYNAPSE_CALLEE=${ARG1}) |
| 01:24.16 | jhiver | Arrgh |
| 01:24.29 | jhiver | and I get _no_ error message? What a drag :) |
| 01:24.54 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 01:25.54 | jhiver | May 29 05:25:44 WARNING[31601]: pbx.c:1294 pbx_extension_helper: No application 'Set' for extension (macro-dial-lcr, s, 1) |
| 01:26.04 | jhiver | I think it's SetVar with 1.0.9 |
| 01:26.07 | Qwell | What are you using, 1.0? |
| 01:26.08 | Qwell | bah |
| 01:26.15 | marl | hi, can someone tell me if i have the following wrong? .call files can be setup to call an internal extesnsion and when its answered then transfer the call to an outgoing line (eg. only make the external call if the internal extesnion has been asnwered)? |
| 01:26.27 | Qwell | 1.4 beta is happening in like 3 days. you REALLY need to upgrade |
| 01:27.03 | jhiver | yeah I know I know :) |
| 01:27.04 | trelane | 1.x? |
| 01:27.15 | trelane | err 1.0.x... are we even supporting THAT? |
| 01:27.22 | jhiver | I just have so much stuff to do :) |
| 01:27.33 | trelane | jhiver, I did the migration in several hours per machine |
| 01:28.08 | jhiver | ok looks like it works :) |
| 01:28.12 | jhiver | thanks |
| 01:28.39 | trelane | jhiver, I strongly recommend migrating ASAP |
| 01:29.18 | jhiver | I know I will but I prolly need to get a second TDM400 first |
| 01:29.42 | jhiver | so that I have another timing device so that I can set up the next upgraded box with it |
| 01:29.57 | jhiver | don't want to suffer too much downtime obviously |
| 01:30.00 | trelane | jhiver, digium.com is your friend |
| 01:30.03 | trelane | by direct from mark |
| 01:30.05 | litage | hey guys, i have calls going like this: user device --> ser --> asterisk --> callee . within asterisk, if i set a password for an extension, the user device can't make calls because asterisk keeps sending 407 (proxy authenticate) messages. how can i fix this? |
| 01:30.08 | trelane | as far as I'm concerned he 100% rocks |
| 01:30.08 | jhiver | I only use the TDM400 as a timing device though :) |
| 01:30.38 | trelane | get the one fxo module |
| 01:30.40 | trelane | it's cheapeast |
| 01:30.43 | trelane | cheapest |
| 01:30.45 | jhiver | oh now I have some strange behavior |
| 01:30.53 | trelane | I've got several I use at work simply for timing in a distributed confrence bridge |
| 01:30.59 | trelane | ooh? |
| 01:31.30 | jhiver | Well I have set up the script to direct the call to a gateway which is alive but not configured yet to accept calls from this asterisk box |
| 01:31.40 | jhiver | so it does this: |
| 01:31.43 | jhiver | <PROTECTED> |
| 01:31.43 | jhiver | <PROTECTED> |
| 01:31.43 | jhiver | M |
| 01:31.50 | jhiver | and then obviously I have this: |
| 01:31.55 | jhiver | May 29 05:27:43 NOTICE[31634]: chan_sip.c:6877 handle_response: Failed to authenticate on INVITE to '"jhiver" <sip:asterisk@83.206.114.91>;tag=as38d7d4ce' |
| 01:32.00 | jhiver | which is fine |
| 01:32.16 | jhiver | but then it hangs for like 2 minutes before dropping the attempt? how come? |
| 01:33.16 | jhiver | I would expect it to fail immediately on a failed INVITE |
| 01:33.38 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 01:34.17 | *** join/#asterisk asteriskmonkey (n=phil@bas4-toronto12-1128731519.dsl.bell.ca) |
| 01:34.26 | mitcheloc | so um heres a random question, is there a reason why sip was created instead of using xmpp??? |
| 01:34.38 | trelane | jhiver, timeout |
| 01:34.44 | asteriskmonkey | hey anyone evey got the error "broken pipe" when tryign to connect back to what was a running asterisk service |
| 01:35.24 | trelane | asteriskmonkey, I find it more reliable to run asterisk in screen |
| 01:35.43 | [TK]D-Fender | asteriskmonkey : PM |
| 01:36.38 | asteriskmonkey | its an anoying eror i know asteirsk is running but it wont let me reconnecct anyone had any bugs like htat before/ |
| 01:37.35 | jhiver | ok it's a timeout but... does asterisk try to INVITE the same gateway for 2 minutes? |
| 01:39.40 | orlock | Hmm, has anybody here used a 7910 with asterisk and chan_sccp? |
| 01:40.37 | Qwell | orlock: no, but you should test it with chan_skinny |
| 01:41.56 | orlock | I'm actually having issues specifying the tftp server for it |
| 01:42.16 | litage | my users register with ser, and ser forwards calls to asterisk. how can i prevent my users from registering directly with my asterisk server? |
| 01:42.20 | orlock | it seems to be ignoring dhcp |
| 01:42.34 | Qwell | orlock: option 66? |
| 01:42.58 | jhiver | another strange thing is that the DIALSTATUS for a failed auth on INVITE is a... CANCEL? |
| 01:43.03 | jhiver | how straaaange |
| 01:47.43 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 01:49.33 | orlock | Qwell: any ide ahow how to unlock/erase the config? |
| 01:49.40 | Qwell | orlock: on a 7910, no |
| 01:50.03 | orlock | hmm |
| 01:50.15 | orlock | it get the ip via dhcp ok, but not the tftp server address |
| 01:50.52 | Qwell | option 66? |
| 01:51.00 | bdunn | Can anyone recommend someone that could do a bit of Asterisk settings for a small company? We have it all up and working very well, but we would like to do a couple of interesting things with the extensions.conf file. |
| 01:51.18 | Qwell | bdunn: there is a large list of consultants on the wiki |
| 01:51.20 | Qwell | ~wikis |
| 01:51.22 | jbot | it has been said that wikis is http://www.voip-info.org |
| 01:51.58 | jhiver | I thought CANCEL would be used only if the caller hangs up before the call goes through |
| 01:52.19 | jhiver | is there any way to reduce the time asterisk "gives up" on unsuccessful INVITEs? |
| 01:53.36 | file | are you the one using 1.0? |
| 01:53.57 | jhiver | yeah my bad, you think it's a bug with this version? |
| 01:54.12 | file | well, chan_sip has drastically changed since the 1.0 days |
| 01:54.27 | file | and by drastically I mean, OMFG IT'S TOTALLY DIFFERENT!!! |
| 01:54.50 | jhiver | oh you mean so it works now? :) |
| 01:55.08 | file | chan_sip is under appreciated for what it does |
| 01:55.28 | file | :P |
| 01:58.31 | dlynes_home | wtf? |
| 01:58.38 | dlynes_home | there's still people using asterisk 1.0? |
| 01:59.03 | TripleFFFF | lol |
| 01:59.05 | file | yes, yes there are... |
| 01:59.08 | TripleFFFF | cvsup ports |
| 01:59.13 | jhiver | yeah yeah, I do but that's only because I had a million other things to do and I haven't used it so much |
| 01:59.13 | dlynes_home | isn't that like people that are still using Bind4? |
| 01:59.16 | TripleFFFF | and install 1.2.7.1 if you got them |
| 01:59.28 | TripleFFFF | like people using win3.1 |
| 01:59.29 | TripleFFFF | ;) |
| 01:59.31 | TripleFFFF | i knwo some |
| 01:59.41 | dlynes_home | TripleFFFF: no flipping way! |
| 01:59.46 | TripleFFFF | actually hte most stable windows out htere |
| 01:59.54 | dlynes_home | TripleFFFF: wrong! |
| 02:00.07 | TripleFFFF | dos was ? |
| 02:00.08 | file | my bank's ATMs run Windows actually... it's disturbing |
| 02:00.12 | TripleFFFF | never saw a blue screen in dos |
| 02:00.13 | dlynes_home | TripleFFFF: the most stable windows out there is the Windows emulation under OS/2 :) |
| 02:00.19 | TripleFFFF | and xppro sp2 i get some all the time |
| 02:00.23 | TripleFFFF | cauz of usb |
| 02:00.38 | file | and the bus ticket dispensers in Pisa, Italy run DOS |
| 02:00.47 | TripleFFFF | file neat |
| 02:01.01 | dlynes_home | file: lots of stuff still runs DOS |
| 02:01.19 | dlynes_home | but DOS is completely different from winblows |
| 02:01.23 | file | if it works don't touch it... but it didn't work :( |
| 02:01.38 | file | I know, I was just making reference to it... as it's something you would not expect |
| 02:01.49 | mitcheloc | hmm, does anyone here actually pay attention to the mailing list? |
| 02:01.58 | dlynes_home | mitcheloc: of course |
| 02:02.02 | file | mitcheloc: depends |
| 02:02.14 | dlynes_home | mitcheloc: i only have 3GB's of asterisk mailing list mail on my hard drive just for the hell of it |
| 02:03.15 | dlynes_home | ah man |
| 02:03.27 | dlynes_home | some of those chinese chick medics from singapore in indonesia are pretty cute |
| 02:04.19 | mitcheloc | dlynes_office: my family just got back from there, they were talking it up saying we should move |
| 02:04.32 | dlynes_home | mitcheloc: move to indonesia? |
| 02:04.51 | mitcheloc | dlynes_home: singapore |
| 02:04.58 | dlynes_home | mitcheloc: ah |
| 02:05.07 | dlynes_home | yeah...lotsa chinese pussy there :) |
| 02:05.28 | mitcheloc | heh, well... |
| 02:07.06 | dlynes_home | dood |
| 02:07.07 | TripleFFFF | !dlynes_home show us |
| 02:07.25 | dlynes_home | singapore would rock |
| 02:08.12 | dlynes_home | http://www.sggirls.com/ |
| 02:08.39 | dlynes_home | don't worry about your eyes...these girls are pretty easy on the eyes :) |
| 02:09.32 | dlynes_home | TripleFFFF: if you manage to pick up a singapore girl, you're going to have to understand wtf she's saying |
| 02:09.44 | dlynes_home | TripleFFFF: so you're going to need this site, too: http://www.talkingcock.com/ |
| 02:10.36 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-211.cybersurf.com) |
| 02:11.27 | mitcheloc | hmm...not really my type.... |
| 02:11.35 | dlynes_home | mitcheloc: don't like asian chicks? |
| 02:11.48 | dlynes_home | mitcheloc: or are they just too young? |
| 02:11.58 | TripleFFFF | actually learning manadarin |
| 02:12.30 | dlynes_home | ni hao zhi dao na ghe putonghua? |
| 02:12.31 | mitcheloc | nahh, i'm just attracted to like less then maybe 2% of them... some are hot..but it's not really my thing |
| 02:12.37 | *** part/#asterisk CDFAssociates (n=CDFAssoc@doc-24-32-55-141.we.ok.cebridge.net) |
| 02:12.53 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 02:13.03 | dlynes_home | mitcheloc: ah...i don't like caucasian chicks myself |
| 02:13.04 | TripleFFFF | hehee |
| 02:13.10 | dlynes_home | mitcheloc: mostly just filipinas and chinese :) |
| 02:13.29 | mitcheloc | well thats fine for me, more for you more for me ;) |
| 02:13.59 | dlynes_home | TripleFFFF: Ni duo qian zhi dao putonghua ma? |
| 02:14.39 | TripleFFFF | hehe |
| 02:14.44 | TripleFFFF | i said is tarted learning |
| 02:14.47 | TripleFFFF | started |
| 02:14.53 | dlynes_home | TripleFFFF: How much Mandarin do you know? |
| 02:15.30 | dlynes_home | TripleFFFF: has anyone said 'yang gui zi' to you yet? |
| 02:16.26 | TripleFFFF | none |
| 02:16.34 | TripleFFFF | but now i know why you on ggoirls |
| 02:16.35 | TripleFFFF | ;0 |
| 02:16.49 | TripleFFFF | im still in nouns |
| 02:16.49 | dlynes_home | you mean on sggirls.com? :) |
| 02:16.57 | dlynes_home | ah |
| 02:16.57 | TripleFFFF | eyah i meant that |
| 02:17.08 | dlynes_home | TripleFFFF: ni shi piaoliang xiaojie :) |
| 02:17.09 | TripleFFFF | as in plane, girl , woemen , man , boat, |
| 02:17.10 | TripleFFFF | balls |
| 02:17.11 | TripleFFFF | ;) |
| 02:17.28 | TripleFFFF | actually using rosetta stone for what its worth |
| 02:17.48 | dlynes_home | girl = xiaojie |
| 02:17.51 | dlynes_home | well |
| 02:17.55 | dlynes_home | xiao1jie2 |
| 02:18.04 | bigmac4444 | how can i change the voicemail menu selections? some i dont want. plz |
| 02:18.09 | TripleFFFF | http://sggirlsmirror3.j37.com/sggirlsphotos/9/8/8/sgGirls.com_-_00133370.jpg |
| 02:18.10 | mitcheloc | *closes sggirls.com* |
| 02:18.13 | TripleFFFF | thhe middle one ;) |
| 02:18.29 | mitcheloc | damn, and i just closed the window, thanks triple |
| 02:18.33 | dlynes_home | yeah..looks kinda slutty :) |
| 02:18.34 | TripleFFFF | yeah sounds like htat |
| 02:18.38 | TripleFFFF | xiao |
| 02:18.44 | TripleFFFF | nue |
| 02:18.47 | mitcheloc | i'm just not attracted to them... |
| 02:18.49 | dlynes_home | xiao = means small |
| 02:18.52 | mitcheloc | somthings wrong with me |
| 02:18.59 | TripleFFFF | yeah |
| 02:19.03 | TripleFFFF | hhe mitch |
| 02:19.07 | TripleFFFF | no worries you keep the russians |
| 02:19.08 | TripleFFFF | ;0 |
| 02:19.09 | mitcheloc | nice bodies, but it's the faces... |
| 02:19.25 | dlynes_home | dood...the faces are the best part about chinese girls :) |
| 02:19.25 | TripleFFFF | <PROTECTED> |
| 02:19.29 | mitcheloc | shush, i'll stick to south american....french...oooh or italian... |
| 02:19.29 | TripleFFFF | these look adorable |
| 02:19.37 | mitcheloc | same link |
| 02:19.48 | dlynes_home | TripleFFFF: dork...same chicks |
| 02:19.52 | TripleFFFF | only reason i didnt move to thailand was coz of darn he shes.. im sure to end up in jail for murder of one of those things |
| 02:20.03 | TripleFFFF | yeah brazilian |
| 02:20.03 | dlynes_home | TripleFFFF: lol |
| 02:20.04 | TripleFFFF | man |
| 02:20.09 | TripleFFFF | we need cash to travel |
| 02:20.09 | TripleFFFF | ;) |
| 02:20.36 | dlynes_home | Or convince coppice to do an exchange |
| 02:20.40 | TripleFFFF | actualy lived 6 monts in peurto vallarta MX.. |
| 02:20.43 | dlynes_home | He lives in Hong Kong :0 |
| 02:20.45 | TripleFFFF | puerto |
| 02:20.46 | TripleFFFF | ;) |
| 02:20.50 | *** part/#asterisk asteriskmonkey (n=phil@bas4-toronto12-1128731519.dsl.bell.ca) |
| 02:20.53 | mitcheloc | nice |
| 02:21.05 | mitcheloc | well if you all have empty couches, i could use some travelin |
| 02:21.05 | dlynes_home | you monkey! |
| 02:21.22 | dlynes_home | man |
| 02:21.28 | mitcheloc | i'll hit the road and install asterisk from city to city, yay |
| 02:21.36 | dlynes_home | we gotta get all you monkeys in toronto to move to vancouver :0 |
| 02:22.01 | dlynes_home | dooed |
| 02:22.07 | dlynes_home | hurricane heading for mexico already |
| 02:22.46 | Strom_C | pffffft, it's all about scenic downtown Regina |
| 02:22.54 | dlynes_home | scenic? |
| 02:22.55 | dlynes_home | lol |
| 02:23.11 | TripleFFFF | hehe |
| 02:23.12 | dlynes_home | Yeah...like the dancers at the Prince Albert |
| 02:23.12 | [TK]D-Fender | SK : Where the land is so flat you can your dog running away from you for DAYS |
| 02:23.19 | Strom_C | hah |
| 02:23.21 | TripleFFFF | come to canada, im 3 hours from mtl |
| 02:23.26 | Strom_C | I've only been to Vancouver |
| 02:23.38 | dlynes_home | Strom_C: or is the prince albert still there? |
| 02:23.49 | dlynes_home | Strom_C: or am i getting regina mixed up with saskatoon? |
| 02:23.58 | dlynes_home | Strom_C: yeah...nvm...that was saskatoon |
| 02:23.59 | TripleFFFF | what up with hurrican what ? |
| 02:24.01 | TripleFFFF | where when how |
| 02:24.31 | TripleFFFF | this frenchie should be messed up by now ..savelivesinmay.com said end of lives on 25th may from acomet..lol |
| 02:25.24 | dlynes_home | ummm |
| 02:25.26 | dlynes_home | wtf? |
| 02:25.57 | TripleFFFF | http://passion.com/search/p97162c?max_age=29&country=Canada&override=1&city=Valcartier+Station&ip=auto&show_city=1&min_age=18&picid=1Aq1RNOQtujVsoLcXuSJrCCvH4oqKXMXEVVK0qKmc&models=0 |
| 02:26.16 | TripleFFFF | #3 is boo.. #11 is my style |
| 02:26.27 | TripleFFFF | all nice chicks here |
| 02:26.30 | TripleFFFF | 95% |
| 02:27.13 | dlynes_home | ummmm |
| 02:27.15 | dlynes_home | wtf??? |
| 02:27.18 | dlynes_home | she's a kid |
| 02:27.37 | mitcheloc | 15 is more me |
| 02:27.44 | dlynes_home | TripleFFFF wants to be a child pornographer when he grows up |
| 02:28.03 | TripleFFFF | heheeh |
| 02:28.12 | mitcheloc | dlynes_home: actually, i think it showed results local to our ip addresses.... |
| 02:28.19 | TripleFFFF | nah we are like chineese we look young |
| 02:28.25 | TripleFFFF | but in fact are all over 75 |
| 02:28.33 | TripleFFFF | oh |
| 02:28.34 | TripleFFFF | lol |
| 02:28.36 | TripleFFFF | your right |
| 02:28.55 | dlynes_home | mitcheloc: well, yours is some slutty looking skank in a pink fluffy thing |
| 02:28.58 | [TK]D-Fender | TripleFFFF : Where are you located? |
| 02:29.10 | dlynes_home | mitcheloc: and she's 27 |
| 02:29.11 | mitcheloc | dlynes_office: it's different results then yours |
| 02:29.22 | dlynes_home | mitcheloc: and she's a swinger |
| 02:29.29 | TripleFFFF | canada |
| 02:29.33 | TripleFFFF | qc,near mtl |
| 02:29.50 | TripleFFFF | http://photos.pop6.com/photo-ffadult-r20-s2-75698752_84100.22028479.gallery.gif |
| 02:29.51 | TripleFFFF | here |
| 02:29.53 | TripleFFFF | thats her |
| 02:29.56 | dlynes_home | and TripleFFFF's is an 18 yr old kid, that's a swinger |
| 02:29.56 | [TK]D-Fender | TripleFFFF : Cool, been to an AMUG meetup? |
| 02:30.15 | mitcheloc | *sigh* |
| 02:30.20 | TripleFFFF | amug ? |
| 02:30.20 | dlynes_home | TripleFFFF: still slutty looking |
| 02:30.27 | TripleFFFF | ;) i like slyutty loking |
| 02:30.35 | [TK]D-Fender | TripleFFFF : Asterisk Montreal Users Group |
| 02:31.08 | TripleFFFF | oh |
| 02:31.10 | TripleFFFF | nop |
| 02:31.48 | dlynes_home | hahahahhaa |
| 02:31.52 | dlynes_home | moncton got their ass kicked |
| 02:31.53 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 02:32.05 | file | oh, the Wildcats? |
| 02:32.08 | dlynes_home | yeah |
| 02:32.15 | file | I'm not surprised |
| 02:32.35 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:32.45 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:33.39 | dlynes_home | file: dood |
| 02:33.47 | dlynes_home | file: you guys major iv users there, too? |
| 02:34.00 | dlynes_home | file: in saint john? |
| 02:34.10 | file | I'm not in Saint John |
| 02:34.19 | dlynes_home | same province though |
| 02:34.19 | TripleFFFF | A TROPICAL STORM WARNING IS IN EFFECT FOR THE PACIFIC COAST OF MEXICO |
| 02:34.19 | TripleFFFF | FROM PUNTA MALDONADO WESTWARD TO ZIHUATANEJO. |
| 02:34.21 | TripleFFFF | alleta |
| 02:34.23 | TripleFFFF | hmm ok |
| 02:34.59 | dlynes_home | file: you can drive across the whole province in less than four hours, right? |
| 02:35.08 | file | well, yes |
| 02:35.20 | [TK]D-Fender | TripleFFFF : Reminds me of a nifty quote "Georgia : If you don't like the weather, wait 10 minutes" |
| 02:35.32 | file | [TK]D-Fender: dead like me? |
| 02:35.40 | [TK]D-Fender | 4 hours for all of NB? Don't think so..... |
| 02:35.45 | dlynes_home | so, saint john, fredericton, moncton, oromocto, ..., .. what's the difference? |
| 02:35.56 | file | well, it depends which part you're going to from where... |
| 02:36.03 | file | which corner to corner... that sort of stuff |
| 02:36.28 | [TK]D-Fender | Moncton is closer to Shediac, lobster capital of the UNIVERSE (fuck Maine) |
| 02:36.41 | file | indeed |
| 02:36.57 | dlynes_home | dood...i thought moncton was the pot smuggling capital of canada? |
| 02:36.57 | file | espically when it's lobster festival time |
| 02:37.04 | [TK]D-Fender | Thats something I definately miss... fresh lobster |
| 02:37.06 | file | Moncton is a lot of things |
| 02:37.18 | [TK]D-Fender | file : Festivals : the leading cause of death in lobsters :D |
| 02:37.29 | dlynes_home | the americans get all pissed about all the pot being smuggled through new brunswick |
| 02:37.30 | dlynes_home | heh |
| 02:37.45 | dlynes_home | i don't know why they get so pissed |
| 02:37.48 | dlynes_home | it's not even good pot |
| 02:38.16 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:38.55 | file | [TK]D-Fender: you should come to Moncton! |
| 02:39.10 | [TK]D-Fender | file : An idea. |
| 02:39.14 | TripleFFFF | Sun's Dual Core x64 Server |
| 02:39.18 | TripleFFFF | any one try that ? |
| 02:39.24 | TripleFFFF | can we compile asterisk on that ? |
| 02:39.25 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:39.31 | dlynes_home | TripleFFFF: i think qwell's running some |
| 02:39.37 | TripleFFFF | lol |
| 02:39.37 | dlynes_home | TripleFFFF: They're called Sunfires |
| 02:39.45 | file | [TK]D-Fender: I don't know what exactly you would do here... but whatever |
| 02:39.50 | TripleFFFF | We bring americans the pot tey bring the coca from the south |
| 02:39.55 | dlynes_home | TripleFFFF: but i think he's running solaris on them |
| 02:40.26 | TripleFFFF | ps im my opinion todays pot is alot STROINGER then the coke |
| 02:40.35 | TripleFFFF | sunfire ok |
| 02:40.35 | hansin321 | question?: Does anyone have any info/opinions on add-on cards that supply digital signal processing for Asterisk so as to off-load some of this from the main CPU and to do so more efficiently through the use of specialized DSP chips or the like? I am curious about this in a pure ethernet/IP setup, so not any of this included on any traditional telephony cards. Is this done? |
| 02:40.44 | TripleFFFF | hansin |
| 02:40.49 | TripleFFFF | check lyrtech.com |
| 02:40.58 | hansin321 | thanks. |
| 02:41.04 | TripleFFFF | they do DSP's my bro is engeneer there. ;) lots of audio stuff they do |
| 02:41.13 | file | Digium is making a transcoding board as well for Asterisk... |
| 02:41.15 | TripleFFFF | heehe wait |
| 02:41.19 | TripleFFFF | its not really for retail |
| 02:41.22 | file | to offload G729 transcoding to it, and add G723.1 transcoding |
| 02:41.29 | dlynes_home | hansin321: sangoma is working on some cards that will have a g729 chip on them |
| 02:41.38 | TripleFFFF | well |
| 02:41.38 | file | plus whatever else we can get |
| 02:41.49 | TripleFFFF | illa sk my bro in morning.. |
| 02:42.34 | hansin321 | ok. thanks all. I am no expert on thi stuff, but I suppose you would have kernel driver that would allow Asterisk to hokk into these cards? |
| 02:42.48 | TripleFFFF | weird |
| 02:42.53 | *** join/#asterisk _daver_ (n=daver@ns1.tmok.com) |
| 02:43.29 | *** join/#asterisk oej (n=oej@65.246.174.67) |
| 02:44.39 | file | hansin321: aye |
| 02:45.40 | dlynes_home | file: any expected release date on those cards? |
| 02:46.14 | file | info have I not! |
| 02:46.35 | dlynes_home | file: do the tdm400p's or the tdm2400p's have any hardware echo cancellers? |
| 02:46.52 | file | the TDM2400P has the capability to have an echo canceller card installed |
| 02:47.17 | dlynes_home | file: ah...so it's nothing that attaches to the card, eh? it would actually take up an additional pci slot? |
| 02:47.34 | file | no, it's a daughter board that attaches to the TDM2400P board |
| 02:47.35 | nextime | and the echo canceller on the 2400p is working good ( almost for me ) |
| 02:47.47 | dlynes_home | ah |
| 02:47.48 | file | http://www.digium.com/en/products/hardware/tdm2400p.php |
| 02:47.49 | dlynes_home | thx |
| 02:47.54 | file | if you look at the middle, on the bottom... |
| 02:48.00 | file | there's a board with two chips and an Asterisk |
| 02:48.03 | file | that's the echo canceller |
| 02:48.38 | dlynes_home | aha |
| 02:49.48 | dlynes_home | you'd think sangoma would design their cards similarly |
| 02:49.56 | file | as for the transcoder board... I just know how it works, not release dates or pricing |
| 02:49.59 | dlynes_home | instead of having to wire up rj11 plugs for 24 jacks |
| 02:50.10 | dlynes_home | you can use an amphenol tail instead |
| 02:50.44 | dlynes_home | 24 rj11 connectors is a bit retarded |
| 02:55.43 | *** join/#asterisk voipaster (i=25x8supp@203.192.191.36) |
| 03:01.52 | *** part/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 03:20.15 | *** join/#asterisk L|NUX (n=linux@202.5.145.57) |
| 03:25.51 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 03:26.23 | mpruett | . |
| 03:27.02 | orlock | ... . . . ... . . . |
| 03:28.23 | mpruett | anyone here? |
| 03:28.47 | mpruett | awful quiet? |
| 03:31.25 | *** part/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 03:32.26 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 03:33.25 | mpruett | Hello? |
| 03:35.11 | bigmac4444 | hi |
| 03:35.18 | file | it's a Sunday night, what'cha want? :P |
| 03:35.24 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 03:35.26 | mpruett | Hey I was wondering if this was working |
| 03:35.26 | bigmac4444 | monday here |
| 03:35.33 | bigmac4444 | seems like it is |
| 03:35.34 | file | well, it's Monday here too... |
| 03:35.36 | file | minor point |
| 03:36.14 | mpruett | Are any of you guys familiar with the finer points of meetme? |
| 03:36.30 | file | define finer points |
| 03:37.35 | mpruett | I need to know how to pass which conference room was used on the cdr |
| 03:37.53 | mpruett | I have a common number that sends users to meetme |
| 03:38.03 | file | well, you've got CDR variables... |
| 03:38.26 | file | plus the application info is available in the CDR too... so you could parse that if you really wanted... |
| 03:38.44 | mpruett | Yeah but I can't figure out how to pass the value the entered in Meetme - I let them chose that when they dial in |
| 03:39.34 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 03:39.36 | tengulre | hi,all |
| 03:40.26 | mpruett | I see in lastapp "meetme" - but nothing is in lastdata |
| 03:41.00 | mpruett | Maybe I should start by telling you what I am trying to do - and you guys may have a better way |
| 03:41.08 | file | why don't you do it outside of meetme? because once it is in there... it's out of your hands |
| 03:41.41 | mpruett | THat is what I am doing now - thought I would try here to see if there is a better way |
| 03:42.14 | mpruett | What I am trying now is to use MySql and handle authentication upfront then past the con room to the cdr |
| 03:42.41 | file | it's too late to think about this, but I've given you some info to think about and incorporate perhaps... |
| 03:44.08 | mpruett | Can I tell what I am trying to do? Maybe there is a better way? |
| 03:44.15 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 03:44.30 | file | I can't stop you... |
| 03:44.36 | mpruett | lol |
| 03:44.42 | file | well, I could - but I'm in bed and it would require a phone call... and my cell is at my desk |
| 03:45.19 | file | I think I have it set on vibrate |
| 03:45.27 | file | so it would probably just fall off my desk |
| 03:45.42 | Corydon76-home | You should wear that on your waistband... front... inside... |
| 03:45.50 | file | or not |
| 03:46.09 | file | remind me not to tell you when I'm in BNA |
| 03:46.19 | Corydon76-home | lol |
| 03:46.37 | mpruett | I have one number I want users to dial in on and chose thier conf room and authenticat with thier pin |
| 03:46.49 | mpruett | s/chose/choose |
| 03:47.02 | Corydon76-home | Actually, you'd rather I call that number while you're in the US, not your other one |
| 03:47.40 | Corydon76-home | and one day, we'll get you a 256 number |
| 03:47.47 | file | I already have two |
| 03:47.58 | mpruett | I use MeetMe without any Parameters "MeetMe()" |
| 03:48.12 | Corydon76-home | Your Digium number and what else? |
| 03:48.16 | file | cell |
| 03:48.28 | Corydon76-home | Ah |
| 03:48.32 | mpruett | This prompts the user for room and pin and they do enter room |
| 03:48.36 | Corydon76-home | 3 cell phones now? |
| 03:48.40 | file | SIM card |
| 03:48.44 | Corydon76-home | Ah |
| 03:48.58 | mpruett | but I have no idea from the CDR which room they entered |
| 03:49.06 | Corydon76-home | You're starting to have more numbers than a CIA agent |
| 03:49.12 | file | mpruett: so do it outside of meetme |
| 03:49.21 | file | (the room part) |
| 03:49.51 | file | Corydon76-home: yeah, I'm reluctant to let go of my Canadian cell number though... and my plan... |
| 03:50.01 | file | I was going to switch providers but I can't bring myself to do it |
| 03:50.06 | Corydon76-home | Read(conf,conf-getpin) |
| 03:50.21 | Strom_C | also, dongs |
| 03:50.35 | file | ding dong Strom's analog line is dead |
| 03:50.36 | Corydon76-home | or Read(conf,conf-getconfno) |
| 03:51.15 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:51.16 | Strom_C | file: dont scare me like that. you made me walk over to the payphone to verify that i can still get a dialtone |
| 03:51.22 | mpruett | OK - Thanks guys this was the route I was heading down - just didn't know if there was an better/standard way |
| 03:51.46 | *** join/#asterisk bmg505 (n=leon@c1-13-15.rndf.isadsl.co.za) |
| 03:51.46 | file | Strom_C: the dialtone is imaginary, you've just heard it so long |
| 03:52.13 | Strom_C | :( |
| 03:52.22 | litage | my users register with ser, and ser forwards calls to asterisk. how can i prevent my users from registering directly with my asterisk server? |
| 03:52.40 | file | tell them not to register with it? |
| 03:53.10 | litage | file: that doesn't prevent them from doing so |
| 03:53.23 | file | welp, you can just use user entries... |
| 03:53.49 | file | so Corydon, going to have an eventful day tomorrow? |
| 03:54.01 | litage | file: not sure what you mean |
| 03:54.26 | Corydon76-home | file: yeah, big orgy planned |
| 03:54.31 | file | litage: chan_sip has a concept of users and peers... a peer is who you send calls to and what devices register to... users are only used for incoming calls... |
| 03:54.33 | Corydon76-home | file: NOT |
| 03:54.38 | file | if there's only a user entry, they can't register |
| 03:55.43 | file | Corydon76-home: that's disappointing! :P |
| 03:56.05 | Corydon76-home | Isn't it? If you were here, you could change that. |
| 03:56.11 | file | but I'm not! |
| 03:56.23 | Corydon76-home | Nor is Qwell |
| 03:56.41 | Qwell | ? |
| 03:56.56 | Corydon76-home | Spoonage |
| 03:57.01 | Qwell | ahh |
| 03:57.26 | Corydon76-home | Your wife must have seen the spoon reference by now |
| 03:57.39 | Qwell | not so much, no |
| 03:57.40 | Qwell | :p |
| 03:57.48 | file | he's put up a firewall! |
| 03:57.54 | Corydon76-home | I'm going to have to call her, then... :-P |
| 03:59.16 | file | Qwellllllllllllll |
| 03:59.37 | *** join/#asterisk annonimous (n=annonimo@dsl-201-129-251-45.prod-infinitum.com.mx) |
| 03:59.45 | annonimous | hiya! |
| 03:59.50 | file | ...hello |
| 04:00.03 | annonimous | how are you? |
| 04:00.12 | file | sleepy, u? |
| 04:00.22 | Corydon76-home | and fabulous |
| 04:00.28 | Corydon76-home | file is always fabulous |
| 04:00.34 | file | indeed |
| 04:00.39 | annonimous | im so so, working =/ |
| 04:00.47 | annonimous | jeje |
| 04:02.12 | *** join/#asterisk nvrs (i=RUR@Quebec-HSE-ppp3613721.sympatico.ca) |
| 04:02.32 | annonimous | by the way, where can i found some dialplans examples? (not voipinfo-org) cause i need to input to my boss to dialplans one for transfer extensions to pstn lines an one for the user of the extension who is lefting to dial his cellphone? |
| 04:02.49 | Qwell | why not voipinfo? |
| 04:02.55 | file | so Corydon, how's life? |
| 04:03.18 | *** join/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 04:03.25 | annonimous | Qwell, cause i saw it and i need to understand a little more about dialplans |
| 04:03.36 | Qwell | ~book |
| 04:03.37 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:04.03 | annonimous | oh ok thanks! =) |
| 04:04.07 | Corydon76-home | file: it's busy |
| 04:04.11 | annonimous | let me see it =D! |
| 04:04.21 | file | busy can be good |
| 04:04.43 | file | it's been such a nice stress free weekend |
| 04:05.06 | Corydon76-home | I have an event for next weekend that I need to postpone, I need some 501(c)3 tax advice before the end of June, and I have a project due on the 1st |
| 04:05.13 | file | pfft you can't do that |
| 04:05.20 | Qwell | file: can too! |
| 04:05.30 | file | nope! |
| 04:05.32 | Qwell | Digium sold me the rights to your weekends. ha! |
| 04:05.40 | file | the only person who can do that is gone till Tuesday |
| 04:05.41 | file | so HA |
| 04:05.48 | Qwell | He sold it to me! |
| 04:05.52 | Qwell | for cheap, too... |
| 04:05.58 | file | oh noes! |
| 04:06.01 | Qwell | yep |
| 04:06.07 | file | I bet you're going to make me... do naughty things... |
| 04:06.19 | Qwell | file: only if Corydon76-home is willing to pay... |
| 04:06.23 | Corydon76-home | ...on the webcam... |
| 04:06.33 | Qwell | ...and we both know he is |
| 04:06.45 | file | for some geek on geek action? |
| 04:07.14 | file | this is #asterisk after hours all btw :P |
| 04:09.27 | file | Qwell: so can I... buy it back? |
| 04:10.05 | Qwell | file: not yet |
| 04:10.08 | file | :( |
| 04:10.15 | file | I've beaten you though :) |
| 04:10.17 | file | it's Monday here! |
| 04:10.29 | Qwell | but it's a US holiday! |
| 04:10.31 | Qwell | ha! |
| 04:10.41 | file | still not a weekend! |
| 04:10.46 | Qwell | yep! |
| 04:10.57 | file | I r teh winna |
| 04:10.58 | Qwell | three day weekends count. It's in the contract |
| 04:11.17 | file | I'd like to see this contract |
| 04:11.29 | Qwell | can't, it's under NDA |
| 04:11.35 | file | but but but |
| 04:11.50 | Qwell | I saw this tv show once... |
| 04:11.58 | file | WOW! THAT'S AMAZING |
| 04:11.59 | Qwell | where they had an nda... |
| 04:12.08 | file | equally as amazing |
| 04:12.12 | Qwell | but, the nda was under another nda, so it couldn't be discussed with law enforcement |
| 04:12.35 | file | this IRC conversation is under NDA btw |
| 04:12.38 | Qwell | good |
| 04:13.40 | file | so Qwell, I heard you're opening up a new telephony pr0n website... any truth to that? |
| 04:13.43 | Corydon76-home | But the spoonage is not under NDA |
| 04:13.46 | *** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 04:14.02 | Qwell | Corydon76-home: No, it's under the "what happens in san jose, stays in san jose" rule |
| 04:14.07 | file | pfft |
| 04:14.10 | file | it's under an MDA |
| 04:14.13 | file | Must Disclose Agreement |
| 04:14.52 | file | Qwell: so if we're ever back in SJC... |
| 04:14.59 | chino | whats up ? |
| 04:15.15 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 04:15.16 | Qwell | file: nope, once any party leaves sjc, it is null and void |
| 04:15.56 | file | well ic |
| 04:16.07 | Corydon76-home | We'll get you drunker next time |
| 04:16.32 | Qwell | not gonna happen :p |
| 04:16.41 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 04:17.19 | file | I'll join in... see how drunk we can get you |
| 04:17.24 | Qwell | ... |
| 04:17.32 | Qwell | join in...what...exactly? |
| 04:17.41 | file | the task of seeing how drunk we can get you |
| 04:17.43 | file | :P |
| 04:17.49 | file | although I warned you, I warned you... |
| 04:17.54 | Qwell | umm |
| 04:17.55 | file | (@ the party) |
| 04:18.28 | Qwell | So, I need to get my own room next time, obviously. :P |
| 04:19.12 | file | haha |
| 04:19.41 | Corydon76-home | No, of course not... |
| 04:19.49 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 04:19.59 | Corydon76-home | Remember, you touched me... I didn't touch you... |
| 04:20.18 | Corydon76-home | other than to shake you when you were SNORING |
| 04:20.21 | Qwell | :p |
| 04:20.36 | voipaster | hello |
| 04:20.56 | Qwell | voipaster: Hi! (nice timing) |
| 04:21.08 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 04:21.18 | *** join/#asterisk operat0r (i=operator@adsl-152-132-93.asm.bellsouth.net) |
| 04:21.19 | voipaster | anyone here has a working system for a 10 seats callcenter? |
| 04:21.19 | znoG | are we still talking about singapore chicks? |
| 04:21.28 | voipaster | hello Qwell |
| 04:21.40 | voipaster | outbound |
| 04:21.50 | Corydon76-home | znoG: no, we're talking about Qwell predispositions... |
| 04:22.29 | operat0r | HEY,Ops just as a notice FWD AIX they are working on it now I spent the past few days and on there .com they say they are working on it |
| 04:22.52 | Qwell | umm, okay |
| 04:23.03 | Qwell | NEWSFLASH: FWD doesn't work! |
| 04:23.32 | Corydon76-home | znoG: especially all the interesting things he does when he's drunk |
| 04:23.41 | bkw_ | Qwell works fine on SIP |
| 04:24.26 | Qwell | bkw_: was being humorous...trying to avoid the topic at hand :p |
| 04:24.31 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:24.32 | voipaster | pls send me proposal astvoip@gmail.com |
| 04:25.35 | voipaster | on a 10seats callcenter system outbound |
| 04:26.01 | Corydon76-home | Qwell: I can't imagine why. ;-) |
| 04:28.04 | znoG | voipaster: would the proposal be to get a good idea on how to do it and then screw the person that sent it to you and do it yourself to save you from all the hassle of having to investigate???? |
| 04:28.30 | voipaster | hmmm |
| 04:28.35 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 04:28.57 | voipaster | nope |
| 04:29.01 | x86 | where can i get Asterisk::Manager? |
| 04:29.11 | Qwell | x86: google |
| 04:29.14 | voipaster | im intersted on buying the system that he will make |
| 04:29.34 | voipaster | not scew the person after the effort he've done.. |
| 04:29.38 | *** join/#asterisk hacked`` (n=lol@modemcable226.130-37-24.mc.videotron.ca) |
| 04:29.39 | hacked`` | guys |
| 04:29.44 | hacked`` | can anyone give me some advice |
| 04:29.48 | x86 | Qwell: yeah I did |
| 04:29.52 | hacked`` | what i want is 4 voip "lines" via 1 phone number, when customer calls in, gives him automated voice saying "for customer service press #1" and it would transfer the call to a specific ip phone, and also an option for the customer to enter their account # and it will read out to them when their payment is due which is taken from a mysql table |
| 04:29.53 | Qwell | hacked``: Don't get drunk at VON. |
| 04:30.00 | hacked`` | qwell, k |
| 04:30.04 | x86 | Qwell: do you know or are you just being an ass (again) ? |
| 04:30.48 | Qwell | x86: No, I'm saying search google |
| 04:30.48 | Sedorox | ahah |
| 04:30.49 | x86 | Qwell: right, i did |
| 04:30.52 | file | hacked``: so do it :) |
| 04:31.05 | Corydon76-home | x86: the package you want is called asterisk-perl and is located at http://gnuinter.net |
| 04:31.39 | hacked`` | file, ya but i dont know which provider will give me 4 lines on 1 phone #, know what i mean? |
| 04:31.49 | Qwell | hacked``: any per-minute provider |
| 04:31.53 | file | hacked``: so research... usually per minute ones do |
| 04:32.03 | Qwell | almost any, anyhow |
| 04:32.22 | file | you have to research based on where you want the number and what you want exactly... we're not going to do that for you :) need to read comments/reviews/poke around... |
| 04:32.29 | hacked`` | what do you mean per minute, i want unlimited US |
| 04:32.31 | voipaster | znoG:im intersted on buying the system that he will make |
| 04:32.54 | chino | get a new name |
| 04:33.41 | file | hacked``: on incoming? |
| 04:35.27 | voipaster | znoG: not scew the person after the effort he've done.. |
| 04:35.31 | *** part/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 04:36.25 | voipaster | znoG:that is why i tried to join this forum, u guys are the expert, im not |
| 04:37.06 | voipaster | znoG:and im trying also to workout my on asterisk setup |
| 04:37.26 | CunningPike | Bit testy tonight, aren't we? :) |
| 04:40.49 | Corydon76-home | voipaster: try the -biz list |
| 04:41.20 | Corydon76-home | This isn't exactly the best forum for evaluating business proposals |
| 04:42.08 | voipaster | Corydon76-home:im so sorry for posting that kind of proposal |
| 04:42.32 | Corydon76-home | Don't be sorry. Just go post on the -biz list |
| 04:43.09 | voipaster | -biz how? |
| 04:43.35 | voipaster | ok thanks got it |
| 04:45.47 | annonimous | well i have to go to sleep thanks for the help and the link of the ook =) |
| 04:45.49 | annonimous | *book |
| 04:47.48 | x86 | Corydon76-home: thanks :) |
| 04:48.11 | file | Strom_C: don't you have stuff to do? :P |
| 04:48.25 | hacked`` | guys, how does broadvoice compare, i dont know how many virtual lines they give me though |
| 04:48.50 | Strom_C | file: yes, I do |
| 04:51.37 | file | Strom_C: well - hop to it |
| 04:52.00 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 04:52.10 | kernel20 | hi there party peeps |
| 04:52.24 | Strom_C | i started hopping, but my downstairs neighbors complained |
| 04:52.37 | file | darn |
| 04:52.38 | kernel20 | would it be possible to change the voicemail attendant?, is so how? |
| 04:52.50 | kernel20 | would it be possible to change the voicemail attendant?, if so how? |
| 04:52.53 | kernel20 | i mean |
| 04:53.13 | CunningPike | kernel20: You mean the actual voice? |
| 04:53.34 | CunningPike | kernel20: There are alternatives out there - or you could record your own |
| 04:53.51 | kernel20 | pre-recorded voicew |
| 04:54.10 | CunningPike | kernel20: Complete sentences, please |
| 04:54.26 | file | those cost extra |
| 04:54.32 | CunningPike | lol |
| 04:54.43 | kernel20 | exten => 1000,1,VoiceMailMain(202@barnvoicemail) |
| 04:54.52 | kernel20 | would play the attendant |
| 04:55.17 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:55.37 | kernel20 | i want to replace the attendant to my customized voice, but that pre-recorded voice is only true to 202@barnvoicemail |
| 04:55.40 | kernel20 | any ideas? |
| 04:59.25 | CunningPike | kernel20: I'm still not sure what you are asking....... if you are talking about "the person at extension 202 is not available", a personal greeting needs to be recorded for each mailbox |
| 05:05.02 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-140-26.dsl.irvnca.pacbell.net) |
| 05:08.57 | *** part/#asterisk [LiFE] (n=LiFE@toronto-HSE-ppp4020917.sympatico.ca) |
| 05:09.16 | acehunky | any one over here who can help me with ZT_CHANCONFIG failed on channel 1: No such device or address (6) ... on a X100P card .. |
| 05:09.39 | acehunky | i tried changing to different PCI slots (my mobo just have 2 pci slots) |
| 05:10.04 | acehunky | ACPI: PCI Interrupt 0000:02:14.0[A] -> GSI 17 (level, low) -> IRQ 17 |
| 05:10.04 | acehunky | Failed to initailize DAA, giving up... |
| 05:10.04 | acehunky | wcfxo: probe of 0000:02:14.0 failed with error -5 |
| 05:10.10 | acehunky | this is what dmesg says |
| 05:13.35 | kernel20 | i want to replace the attendant to my customized voice, but that pre-recorded voice is only true to 202@barnvoicemail |
| 05:13.51 | kernel20 | the auto attendant i want to delete it |
| 05:13.58 | kernel20 | the auto attendant i want to replace it |
| 05:16.18 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 05:18.15 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 05:19.51 | *** part/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 05:29.56 | kernel20 | why is it at xlite if my sip client puts me on hold i cant hear any sound files which is have set in musiconhold.conf |
| 05:29.58 | kernel20 | any ideas? |
| 05:30.16 | bigmac4444 | ver 1.2 ? |
| 05:30.44 | bigmac4444 | mode=files ? |
| 05:33.13 | kernel20 | 1.2.7 |
| 05:33.34 | bigmac4444 | and you will just be playing mp3 files? |
| 05:34.16 | kernel20 | yeap |
| 05:34.45 | kernel20 | where should i place this one default => custom:/var/lib/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000 --output=raw:- |
| 05:35.04 | bigmac4444 | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 05:35.27 | bigmac4444 | if thats where you have them |
| 05:35.39 | bigmac4444 | then configure extensions to suit |
| 05:36.30 | kernel20 | [default] |
| 05:36.30 | kernel20 | mode=quietmp3 |
| 05:36.30 | kernel20 | directory=/var/lib/asterisk/mohmp3 |
| 05:36.52 | kernel20 | at musiconhold.conf |
| 05:37.17 | *** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 05:37.31 | bigmac4444 | i find it MUCH easier to use the old way |
| 05:37.52 | kernel20 | what old way? |
| 05:37.58 | bigmac4444 | via mpg123 |
| 05:38.06 | kernel20 | ahh i need to install it? |
| 05:38.14 | bigmac4444 | best to, yes |
| 05:38.19 | kernel20 | k wait |
| 05:42.54 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 05:43.25 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 05:47.49 | stephane_ | jour |
| 05:48.10 | bigmac4444 | hi |
| 05:50.33 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 05:53.28 | satlan32 | does anyone know the xorcom ts-1 system? |
| 05:53.33 | *** join/#asterisk clive- (n=pirch@dsl-165-172-117.telkomadsl.co.za) |
| 05:53.45 | satlan32 | how do i connect to the mtsql? |
| 05:53.49 | satlan32 | mysql? |
| 05:54.17 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 05:56.43 | *** join/#asterisk BugKham (i=BugKham@202.8.86.168) |
| 05:59.26 | *** join/#asterisk Kis (i=vlad@p5080FDF2.dip.t-dialin.net) |
| 05:59.57 | gnosys_ | General question for the room: what IAX2 gateways to PSTN is everyone using? I've been using VoicePulse, but they've recently changed their terms of service and I'm really unhappy with those so I'm considering dropping them in favor of another gateway. Recommendations? (for USA) |
| 06:01.29 | Strom_C | which part of the new terms of service are you unhappy with? |
| 06:03.57 | gnosys_ | I guess I could answer that most accurately by saying: (1) all 14 pages of it, (2) the fact that they are responsible for nothing and I am responsible for everything, (3) the fact that they disabled autopay for me and now want me to agree to these TOS in order for me to use it again, and (4) the fact that they are pressuring me into accepting it with prices, and (5) the fact that if I want to pay them even once, then I must agree to the 14 |
| 06:05.22 | clive- | nufone is prety good |
| 06:08.16 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 06:08.28 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 06:08.35 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 06:08.46 | gnosys_ | ok. thanks. any other suggestions? I'd like to collect a few and try them before I make the move. I have an account with nufone, teliax, voipjet, exgn. someone here mentioned plainvoip but i worry about that place because it uses a self-signed certificate for host: localhost.localdomain. Not sure if I would be safe entrusting my credit card to this guy docelmo if he's invested so little in infrastructure that his self-signed cert mak |
| 06:10.30 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:22.08 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 06:30.18 | orlock | Hmm, i am having issues getting asterisk to load either chan_sccp or chan_skinny |
| 06:30.39 | orlock | Dos anybody have it working with svn source |
| 06:30.40 | orlock | ? |
| 06:35.06 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 06:37.32 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 06:43.54 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 06:44.29 | bigmac4444 | lol |
| 06:45.33 | dlynes_home | orlock: no point trying to use chan_sccp |
| 06:45.39 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 06:45.52 | dlynes_home | orlock: chan_skinny has now surpassed chan_sccp |
| 06:46.19 | bigmac4444 | went on a seefood diet |
| 06:46.21 | dlynes_home | orlock: there's a few people that have it working, but it's about as stable as the h323 module |
| 06:49.22 | *** join/#asterisk lorinc (n=ang@caracas-1824.adsl.interware.hu) |
| 06:49.48 | Qwell | I don't know about "surpassed"... |
| 06:49.54 | Qwell | but, the maintainer sure is a cool guy |
| 06:49.56 | Qwell | :D |
| 06:50.36 | Qwell | orlock: http://svn.digium.com/svn/asterisk/team/north/chan_skinny-fixup/ |
| 06:50.44 | Strom_C | yeah, I'd buy him beers any day |
| 06:50.54 | kernel20 | bigmac4444: |
| 06:51.01 | Qwell | Strom_C: you do realize I'm just up the street, right? :P |
| 06:51.03 | bigmac4444 | yes mate? |
| 06:51.04 | Qwell | I mean..he |
| 06:51.21 | Strom_C | seriously Qwell, I can't wait till chan_skinny is stable enough for me to implement in production environments |
| 06:51.27 | kernel20 | can i distub for a while again? |
| 06:51.28 | Qwell | Strom_C: yeah...me too |
| 06:51.35 | Strom_C | how up the street are you? I forget. SFV, right? |
| 06:51.39 | Qwell | Strom_C: but until people give me their test results... |
| 06:51.42 | Qwell | Strom_C: wsco |
| 06:51.51 | Strom_C | wsco? |
| 06:51.54 | Qwell | west covina |
| 06:51.59 | Strom_C | ah ok |
| 06:52.06 | Qwell | probably a charter abbrev |
| 06:52.17 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 06:52.26 | Strom_C | yeah, i saw wsco and I thought "Wasco?! That's not up the street!!" |
| 06:52.28 | Qwell | heh |
| 06:52.56 | Qwell | on a good note, I think I've gotten the last few reset bugs fixed |
| 06:53.07 | Strom_C | Qwell: once I get another cisco phone, I'll run it fulltime at home and help you rest it |
| 06:53.11 | Strom_C | er, test it |
| 06:53.32 | orlock | Qwell: svn that? |
| 06:53.45 | Qwell | orlock: yep, that'll get you a "better" chan_skinny |
| 06:54.01 | Qwell | slightly, anyhow.. |
| 06:54.09 | Qwell | what device is this? |
| 06:54.44 | Qwell | 7910 you said, right? |
| 06:54.51 | orlock | here, yeah |
| 06:55.03 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:55.06 | Qwell | nobody has tested those, but it's a basic phone, and should work great |
| 06:55.25 | dlynes_home | How is West Covina up the street from Regina? |
| 06:55.27 | orlock | out current voip provider is 100% cisco, i'd like to try these phones with asterisk without migrating to sip (we have a pile of 7940's and 7960's too) |
| 06:55.44 | Qwell | orlock: If you wouldn't mind, I'd really appreciate it if you could post your findings at http://bugs.digium.com/view.php?id=6859 |
| 06:55.44 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a222.nwlnnh.tds.net) |
| 06:56.01 | Qwell | dlynes_home: *shrug*, it isn't that far |
| 06:56.08 | orlock | yeah, i'm sure i need to fgure out the cisco config files and the like as well though :-( |
| 06:56.20 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 06:56.26 | Qwell | orlock: They're fairly easy...once you've got one, the rest kinda "Just Work" |
| 06:56.34 | dlynes_home | West Covina's california or something? |
| 06:56.42 | Strom_C | yeah, los angeles metro area |
| 06:56.54 | Qwell | I'm only about 15-20 minutes from downtown |
| 06:57.01 | Strom_C | assuming no traffic on I-10 |
| 06:57.03 | dlynes_home | yeah...it's only maybe 1-1/2 to 2 days drive |
| 06:57.15 | orlock | Qwell: yeah, its purely how the asterisk config/uersname/password lines up with the cisco one.. i cant see where you specify the name/password in the cisco's sepmac.conf |
| 06:57.20 | Qwell | dlynes_home: from Strom_C to me? |
| 06:57.33 | dlynes_home | Strom_C: didn't you say you lived in Regina? |
| 06:57.34 | Strom_C | Qwell: with enough sigalerts, sure :) |
| 06:57.37 | Qwell | maybe half an hour, heh |
| 06:57.44 | Strom_C | dlynes_home: no, I said I lived in Los Angeles |
| 06:57.48 | dlynes_home | ah |
| 06:57.49 | Qwell | Strom_C: You're just ~in holywood, right? |
| 06:57.54 | Strom_C | I made a joke about scenic downtown Regina |
| 06:57.54 | Qwell | hollywood even |
| 06:57.59 | Strom_C | Qwell: Los Feliz |
| 06:58.05 | dlynes_home | Strom_C: yeah...that's why i thought you lived there :0 |
| 06:58.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 06:58.14 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:58.14 | Strom_C | dlynes_home: I've never been to saskatchewan |
| 06:58.18 | Qwell | bah! |
| 06:58.26 | Qwell | yeah, that's right around the corner, heh |
| 06:58.33 | Strom_C | Qwell: it's close enough to Hollywood anyway |
| 06:58.47 | Strom_C | I can just barely see hollywood blvd from where I'm sitting |
| 06:59.09 | Qwell | If I go about a mile east (up the hill), I can see downtown |
| 07:00.03 | Qwell | wrong hill, methinks |
| 07:01.36 | Strom_C | I'm sitting at a coffeshop |
| 07:01.39 | Strom_C | beautiful night out |
| 07:01.51 | Strom_C | I'm on the sidewalk |
| 07:01.57 | Strom_C | (well, in a chair) |
| 07:03.31 | orlock | cpp? wtf |
| 07:04.07 | Strom_C | orlock: ??? |
| 07:04.42 | dlynes_home | ftw? |
| 07:05.06 | Strom_C | dogballs? |
| 07:05.36 | KaBewM | Dog Bollocks I believe |
| 07:06.19 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:06.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:08.13 | Strom_C | Feline copulation, Canine genitals, and deceased prostitutes |
| 07:09.37 | dlynes_home | Strom_C: is that what you're eating lately? |
| 07:09.56 | orlock | Qwell: should that chan_skinny need anything asterisk doesnt to compile? |
| 07:10.04 | orlock | its complaning that cpp is faling a sanity check |
| 07:10.06 | Qwell | orlock: no.. |
| 07:11.41 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 07:12.32 | Strom_C | dlynes_home: yes, it's the new fad diet |
| 07:17.47 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 07:29.13 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:30.32 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 07:30.40 | tparcina | good morning group! |
| 07:30.45 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:30.54 | tparcina | at saturday i head weary successful day :)) |
| 07:31.21 | tparcina | i have menaged to connect * with 2E1 ports - one to provider and another to Ericsson BP250 |
| 07:31.55 | tparcina | and afther that, i have connect this asterisk to another asterisk over 2 Mb frame relay link |
| 07:32.04 | tparcina | I'm proud now :)) |
| 07:32.34 | Strom_C | congrats, tparcina |
| 07:32.51 | tparcina | thank you strom :) |
| 07:33.43 | tparcina | but i need to thank to people from the group. in few situations they helped me |
| 07:33.48 | Strom_C | :) |
| 07:34.56 | Strom_C | I should have brought my headphones with to the coffeeshop |
| 07:35.37 | tparcina | on www.asterisk.org was a page with list of packages that * needs. i can't find that page now. has anybody have a link? |
| 07:36.07 | Supaplex | tparcina: what platform/os/distro etc are you on? |
| 07:36.17 | tparcina | Strom, why? newer bring soch things with you. in coffeeshop you should relay yourself and not think on job |
| 07:36.54 | tparcina | supaplex, i'm on fedora core 4, but i'm trying to install it on SME setver 7.0.rc1 |
| 07:37.07 | Strom_C | tparcina: I came to the coffeeshop to work actually |
| 07:37.33 | Strom_C | I spend so much time relaxing in front of the PC at home that it's difficult for me to concentrate on work if I'm sitting at my desk |
| 07:37.34 | tparcina | strom, to work on asterisk or to work something else |
| 07:37.53 | Strom_C | tparcina: work on other things |
| 07:38.28 | Strom_C | tparcina: the list of packages needed is at http://www.asterisk.org/download |
| 07:38.32 | Strom_C | bottom of the page |
| 07:39.47 | tparcina | strom, ncurses, openssl, zlib and bison - is this all? what about gss and other stuf? |
| 07:40.08 | tparcina | strom, i have seen this before, but i didn't think this is complete list... |
| 07:40.27 | tparcina | strom, ncurses, openssl, zlib and bison - is this all? what about GCC and other stuf? |
| 07:41.12 | Strom_C | well I think gcc is kind of a given |
| 07:41.45 | Strom_C | what, you want hte page to also specify that you need a computer? |
| 07:41.53 | *** join/#asterisk holaaa (n=a@217.11.120.84) |
| 07:42.35 | Strom_C | or maybe electricity as well? :) |
| 07:42.35 | tparcina | strom, :)) ok, i get it |
| 07:44.43 | holaaa | I hear some "noises" in my phone. Monitoring traffic, I discover it is due to a traffic "rush". Analyzing traffic with ethereal y discover the noises occurs when there is a hing "Rvr Jitter" value from time to time, from the server to the phone. No codec problem, just high Rvr Jitter value (named by ethereal) in the traffic... any ideas? |
| 07:45.32 | Strom_C | holaaa: describe your network setup from end-to-end please |
| 07:45.43 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:47.06 | holaaa | network setup?... just a voip phone connected to a switch... the suitch to the server... and a TDM400 2FXO card... sorry don't know exactly the data you need... |
| 07:47.45 | Strom_C | so ok, you've got an analog telephone line, an asterisk box, and the asterisk box connects to the voip phone over a LAN? |
| 07:47.59 | Strom_C | there's no internet transport in this situation, right? |
| 07:48.19 | holaaa | that is |
| 07:48.41 | holaaa | no voip providers or anything |
| 07:48.45 | Strom_C | ok |
| 07:48.59 | Strom_C | are you sharing your LAN with any data-intensive applications? |
| 07:49.16 | Strom_C | and is the asterisk box only functioning as a PBX, or are you doing other things with it as well? |
| 07:50.13 | holaaa | Im sharing the lan, but with no with intensive things... asterisk machine only works as a pbx... in fact, it is 98% idle cpu all the time.. and network traffic is low, even calling or beeing called |
| 07:50.47 | holaaa | The "traffic rushes" comes from the pbx to the phone not the other way |
| 07:51.09 | Strom_C | run a ping test to the phone for a few minutes and pastebin the result |
| 07:51.13 | Strom_C | ~pb |
| 07:51.17 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 07:52.25 | holaaa | IStrom_C, I will do it... wait a second :-) |
| 07:54.31 | holaaa | But, I *think* the problem is the asterisk machine sending some kind of wierd traffic (named as "Rvr Jitter" by ethereal). Usually this value is 5ms all the time but there are peaks of 20ms.. it is then when i hear the noises. |
| 07:55.06 | holaaa | In fact, the other side of the phone (not my ip phone) can NOT hear the "noises"... just me. |
| 07:56.05 | holaaa | pint test is just less than 1ms all the time. |
| 07:56.16 | Strom_C | odd, because 15ms jitter every now and again shouldnt be a problem |
| 07:56.50 | Strom_C | what happens if you take everything off your network except the asterisk box and the IP phone? |
| 07:57.33 | holaaa | Strom_C... jitter value from phone to pbx is 5-7ms all the time... from pbx to phone is 20-25ms from time to time... that are the noises" (like a" beep") |
| 07:58.11 | Strom_C | you hear beeps? |
| 07:58.15 | Strom_C | that's weird |
| 07:58.22 | Strom_C | what protocol are you using? what phones are you using? |
| 07:58.41 | *** join/#asterisk fuzza (n=andrew@www.terminus.net.au) |
| 07:58.46 | fuzza | hi all |
| 07:58.51 | Strom_C | good afternoon |
| 07:59.08 | holaaa | Strom_C... not triyed to disconect it all... just don't thing it is the problem... no heavy traffic.. i am monitoring it all the time... in a conversation it never goes more than 15-20k |
| 07:59.31 | holaaa | protocal... ulaw |
| 07:59.55 | holaaa | thomsom speedtouch 2030 |
| 08:00.39 | fuzza | setting up a box on sarge with voip gatewaying (among other things), trying to connect to engin. the outgoing calls work fine (the [engin] snippet in sip.conf) but no matter what I put in the register=> line, it doesn't even seem to be trying to register (sip show registry is always empty)... am I missing something? |
| 08:00.55 | fuzza | oh, on 1.2.1 |
| 08:01.05 | holaaa | My guess is that pbx does have some kind o wrong jitter value, and every x seconds, it sends a rush of traffic to the voip phone... but jitter values i have seen for asterisk are for iax protocol. |
| 08:01.39 | Strom_C | holaaa: but what is the phone speaking? ulaw is the codec. I need to know the protocol. |
| 08:01.59 | Strom_C | fuzza: why 1.2.1? we're at 1.2.7.1 now |
| 08:02.36 | holaaa | Strom_C RTP... thats the way I can analyze it via Ethereal (it makes graphs and lot of cool things) |
| 08:02.42 | fuzza | ISTR that's what was in sarge (actually testing, sarge is still on 1.0.7 or somesuch) |
| 08:02.51 | Strom_C | holaaa: no no. SIP? IAX? H323? |
| 08:02.53 | fuzza | compiled manually from the src deb |
| 08:02.56 | holaaa | SIP |
| 08:03.01 | holaaa | sorry |
| 08:03.25 | Strom_C | fuzza: I love debian, but don't use the debian packages. Download asterisk stable using svn directly from digium and compile that from source |
| 08:03.50 | Strom_C | holaaa: what OS are you running asterisk on? |
| 08:03.52 | holaaa | Strom_C i triyed with a softphone as well... and it happens the same... the problem is not with the voip phones. |
| 08:04.06 | holaaa | CentOS |
| 08:04.33 | Strom_C | holaaa: the problem is either with your box or with your network. Disconnect everything but the asterisk box and the IP phone from your network and see if you still have the same problems. |
| 08:04.36 | fuzza | Strom_C: oh? are there known problems? or is it just a "keep up with the latest" issue? |
| 08:04.52 | Strom_C | fuzza: there are LOTS of bugfixes from 1.2.1 to 1.2.7.1 |
| 08:05.11 | Strom_C | debian packages always lag way behind source releases |
| 08:05.24 | Strom_C | asterisk is evolving very rapidly :) |
| 08:05.41 | fuzza | Strom_C: fair enough |
| 08:05.51 | holaaa | Strom_C, but traffic rushes come extricly from the pbx to the voip phone (soft or hard phone), i already discovered that analyzing the traffic. |
| 08:06.21 | holaaa | and that "rushes" are interpreted by an analyzer as "rvr jitter" value rising up to 20ms. |
| 08:06.24 | Strom_C | holaaa: look, forget what ethereal is telling you. Just try it please. |
| 08:06.28 | fuzza | Strom_C: I remember now why I was on that, cause that was all I could find compilable source for, since at the time I thought I needed a patch for part of it. I turned out not to need it, so I'll probably either get official or grab from peen.net (which I've done elsewhere) |
| 08:06.40 | Strom_C | um |
| 08:06.41 | Strom_C | use digium |
| 08:06.43 | Strom_C | nothing else |
| 08:06.50 | fuzza | heh |
| 08:06.55 | Strom_C | and dont use the tarballs; use svn |
| 08:07.14 | Strom_C | the stable svn downloads tend to have minor fixes |
| 08:08.11 | holaaa | Ok Strom_C, I will try, but can not dow it right now... In the case traffic is not the problem... what else should I try...any idea? (just to save time) |
| 08:08.39 | Strom_C | holaaa: is the asterisk box a fresh build from scratch? is anything but asterisk running on the box? |
| 08:08.46 | holaaa | fresh |
| 08:09.47 | Strom_C | out of curiosity, have you tried using an iax softphone? |
| 08:10.24 | Strom_C | oh, and what version of asterisk are you running? |
| 08:11.04 | holaaa | no... can you recommend any for windows? version 1.2.5 |
| 08:11.18 | Strom_C | holaaa: upgrade to 1.2.7.1 |
| 08:11.37 | holaaa | Is is stable enough? |
| 08:11.45 | holaaa | I dont like upgrades :D |
| 08:11.45 | Strom_C | holaaa: it's the stable release |
| 08:12.03 | holaaa | ok |
| 08:12.48 | holaaa | Any other advice you think can be usefull? |
| 08:16.09 | Strom_C | sorry, im back |
| 08:16.17 | Strom_C | the network connection here took a dump |
| 08:16.26 | holaaa | no problem |
| 08:17.10 | Strom_C | I'd upgrade the asterisk box...I do remember there were issues with the SIP stack at some point not terribly long ago, though I dont remember if it was a problem in 1.2.5 |
| 08:17.26 | Strom_C | but I've had no SIP issues in 1.2.7.1 |
| 08:17.43 | Strom_C | this is straight asterisk, right? it's not asterisk@home or anything? |
| 08:17.58 | holaaa | Sorry.. it is A@H |
| 08:18.01 | Strom_C | ugh |
| 08:18.07 | holaaa | forgot to mention |
| 08:18.17 | holaaa | Im used to work with it.. |
| 08:18.21 | Strom_C | that's probably 75% of the problem right there |
| 08:18.27 | Strom_C | :_ |
| 08:18.29 | Strom_C | er :) |
| 08:18.42 | holaaa | I see :D |
| 08:18.57 | holaaa | With A@H "you can never tell" |
| 08:19.21 | Strom_C | it's not a bad tool for getting your feet wet with ip telephony, but for any serious production system I would personally not use it. |
| 08:19.58 | holaaa | Now I have some more exprecience I will change, but it was my first contact so... |
| 08:20.04 | holaaa | i installed it. |
| 08:20.27 | Strom_C | what are the specs of the machine it's running on? |
| 08:20.59 | holaaa | usual clonic PC 512 ram... 3GHrz |
| 08:21.19 | Strom_C | hmm, ok...not an anemic machine |
| 08:21.23 | holaaa | quite enough power.. most of the time it is idle |
| 08:21.45 | Strom_C | but yeah, I don't know. a@h is not exactly what I'd call an efficient system |
| 08:22.07 | holaaa | I know, but for some reason in this case i think it is not the cause of the problem. |
| 08:24.53 | *** join/#asterisk tuorpeZ (n=asrm1-09@ns.info.univ-evry.fr) |
| 08:26.15 | *** part/#asterisk fuzza (n=andrew@www.terminus.net.au) |
| 08:30.18 | holaaa | Just a last question Strom_C... why do you think I should "forget" about ethereal analysis? Can it be inaccurate or wrong for some reason? |
| 08:30.44 | kay2 | someone has ever experienced RealTime ? |
| 08:31.39 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 08:32.24 | bdunn | HELP... I am now having this problem after restart my * box. When I run asterisk -vvvvvvvvvvvd, I end up with this: Ouch ... error while writing audio data: : Broken pipe |
| 08:32.37 | *** join/#asterisk kristalino (n=kristali@230.Red-83-32-123.dynamicIP.rima-tde.net) |
| 08:32.57 | bdunn | asterisk doesn't load - but mpg123 does load. I think this has something to do with a problem with mpg123, but I can't pinpoint it. |
| 08:37.37 | kay2 | bdunn: make mpg123 |
| 08:38.31 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 08:40.30 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 08:40.54 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 08:41.15 | kernel20 | bigmac4444: hi |
| 08:41.33 | kernel20 | Œê‚¨‚„‚¾‚™ |
| 08:41.42 | kernel20 | hello |
| 08:42.22 | kernel20 | bigmac4444: i git dc earlier |
| 08:42.29 | kernel20 | bigmac4444: i got dc earlier |
| 08:42.43 | kernel20 | would it be fine if am going to ask for the link now? |
| 08:43.42 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-245-108.telkomadsl.co.za) |
| 08:43.44 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 08:43.55 | Strom_C | well that was fun |
| 08:44.01 | X-Gen | hey freaks |
| 08:45.25 | Strom_C | yo |
| 08:47.38 | kernel20 | ? |
| 08:48.52 | acehunky | <PROTECTED> |
| 08:49.02 | acehunky | any one faced strange issues with X100P card ? |
| 08:49.16 | acehunky | like No such device (6) |
| 08:53.22 | kernel20 | hi |
| 08:53.30 | kernel20 | any ideas where can i download eyebeam? |
| 08:53.41 | kernel20 | i wanna test it first before i will buy |
| 08:55.09 | holaaa | Just a last question Strom_C... why do you think I should "forget" about ethereal analysis? Can it be inaccurate or wrong for some reason? |
| 09:01.08 | *** join/#asterisk muppetmaster (n=jasongoe@27.Red-213-97-53.staticIP.rima-tde.net) |
| 09:02.14 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.10 | Strom_C | holaaa: because you become too sensitive to the data and not sensitive enough to real-world problem solving |
| 09:04.12 | holaaa | Ok. I see. Thaks a lot. |
| 09:06.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 09:07.23 | tparcina | I have stranke problem, when someone tries to call out i get this message -- OH323/484732@85.114.35.42-57383918 is making progress passing it to SIP/301-449f and user waits for 20 sec but call doesn't establish. where should i look for problem? |
| 09:11.01 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.13 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 09:12.21 | mr_horsepower | hi, morning all |
| 09:17.38 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:18.34 | *** join/#asterisk Teeli (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 09:19.33 | mr_horsepower | anyone here, working with disa and have some problems, when dialling faster? |
| 09:31.10 | kay2 | In realtime, where do I have to put the login/pwd for the sql access ? |
| 09:34.07 | *** join/#asterisk InfraRed (n=subhi@arpa-addr.in) |
| 09:34.12 | InfraRed | hi all |
| 09:34.28 | Strom_C | good afternoon |
| 09:37.51 | RoyK | Strom_C: good morning |
| 09:39.06 | Strom_C | good afternoon, RoyK |
| 09:57.30 | *** join/#asterisk jahani (n=k@adsl196-76-239-217-196.adsl196-16.iam.net.ma) |
| 10:01.38 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:08.08 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-112-179.telkomadsl.co.za) |
| 10:15.35 | *** join/#asterisk Creperum (n=Ilya@tex.tsua.net) |
| 10:17.00 | Creperum | hi, i have 2 trunks ZAP/g0 and ZAP/g1... how can i tell asterisk to route all calls from ZAP/g1 to ZAP/g0??? |
| 10:17.06 | nettie | hey guys, anyone now if could be possible increase the jitter for sip clients please? |
| 10:17.18 | *** join/#asterisk azeteg (n=azeteg@c115.brewhouse.se) |
| 10:17.41 | RoyK | nettie: why the fsck would you want to increasse jitter? |
| 10:17.44 | azeteg | I have a problem with an ATA-186 using SIP, when I dial its extension - I get this: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' |
| 10:17.57 | azeteg | anyone know what kind of problem it might be? |
| 10:18.00 | Creperum | nettie, hey, you can enlarge your jitter just for $10! %) |
| 10:18.24 | azeteg | or this in fact: May 29 10:12:14 NOTICE[24607]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 10:18.27 | RoyK | azeteg: pastebin a full sip debug and ask again |
| 10:18.27 | azeteg | <PROTECTED> |
| 10:18.30 | azeteg | oops - sorry for the paste |
| 10:18.32 | azeteg | <PROTECTED> |
| 10:18.33 | RoyK | ~pb |
| 10:18.36 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:18.54 | nettie | guys sorry I badly explained it |
| 10:19.10 | nettie | I'll retry |
| 10:19.20 | azeteg | http://pastebin.com/744474 |
| 10:19.22 | azeteg | there it is |
| 10:20.01 | nettie | under heavy load my internet link starts to lag |
| 10:20.15 | azeteg | nettie: better do some shaping then |
| 10:21.20 | nettie | and the jitter increase badly |
| 10:21.29 | nettie | I'm actually prioritizing traffic |
| 10:21.35 | nettie | and this definitely help |
| 10:21.48 | *** join/#asterisk Ecio (n=eciostar@194.105.59.42) |
| 10:21.48 | RoyK | nettie: you need an RTP jitterbuffer |
| 10:22.09 | nettie | but I would like to know if there's a setting in asterisk that will help in case of RTP dispersion |
| 10:22.09 | RoyK | nettie: http://bugs.digium.com/view.php?id=3854 |
| 10:22.16 | Ecio | hi all, i have a problem with a SIP trunk between cisco CM 4 and asterisk, can anybody help me? |
| 10:22.23 | RoyK | nettie: see that bug. there's a patch available |
| 10:22.36 | nettie | or a suggestion on how to improve the fragmnentation delay on my cisco router :) |
| 10:22.40 | nettie | ahh |
| 10:22.44 | nettie | RoyK: looking |
| 10:23.12 | Ecio | i can call from CM to asterisk but not viceversa, when the CM receive the sip call it replies with 404 not found |
| 10:23.35 | azeteg | RoyK: do you have any idea why I have that error? |
| 10:24.56 | nettie | RoyK I'm running 1.2.6 sure it's not already there? |
| 10:25.05 | nettie | (just courious) |
| 10:25.13 | Ecio | or, takin it from another point of view, has anybody a working trunk between CM and * where i could inspect the debug/packet dump? |
| 10:25.48 | nettie | RoyK: are you actually using it? |
| 10:26.06 | RoyK | nettie: there'll never be an rtp jitterbuffer in 1.2.x |
| 10:26.16 | RoyK | nettie: i am. i paid for it to be written |
| 10:27.09 | nettie | RoyK oh really? |
| 10:27.21 | RoyK | yes, really :) |
| 10:27.30 | nettie | RoyK does it makes huge difference :) |
| 10:28.38 | RoyK | we took a 704/128 ADSL link, started ~10 downloads and a few uploads, started a voip call and the audio was crystal, with g.711a |
| 10:28.55 | nettie | GOD |
| 10:28.59 | RoyK | without the jb audio was beyond crap |
| 10:29.15 | RoyK | by gods, yes, it works |
| 10:29.25 | RoyK | a little memleak in there somewhere, though |
| 10:30.03 | nettie | RoyK why you had someone write it? u needed it for some special project? |
| 10:30.13 | nettie | RoyK what you say is just great anyway |
| 10:30.19 | RoyK | yes, for using asterisk large-scale |
| 10:31.02 | nettie | with it the customer wont need QoS anymore on their link I suppose |
| 10:31.06 | RoyK | nettie: also, uncomment abstract_jb.c line 347. something strange happens every now and then and that is called 50 times a second, which fscks up things. just remove the warning log |
| 10:31.21 | RoyK | nettie: they won't, if they have a good jitterbuffer on their side |
| 10:31.23 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
| 10:31.48 | nettie | RoyK I use polycoms phones jitter buffer goes up to 160ms |
| 10:32.07 | RoyK | nettie: it's jitterbuffer and PLC, packet-loss concealment. if packet 33 is lost, the jb code interpolates packet 32 and 34 to generate the lost packets. |
| 10:32.23 | RoyK | nettie: all major voip equipment have good jb's |
| 10:32.25 | nettie | RoyK I'm farly happy with Cisco LLQ QoS but I understantd the rtp buffer inherits.. looks great |
| 10:32.54 | nettie | RoyK works like an upscaler |
| 10:33.14 | nettie | of course it has limits |
| 10:33.35 | nettie | but then I think it's up to the user be smart enought to use a better optimized codec |
| 10:34.01 | nettie | enought == enough |
| 10:34.01 | mr_horsepower | why my matra pbx, when making a call, it sends the first digit, and all the others are dtmfs? this is a normal behaviour? |
| 10:34.19 | nettie | RoyK so you actually use the SVN version of * ? |
| 10:34.28 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 10:35.10 | RoyK | nettie: not at all - i'm not that sick :) |
| 10:35.22 | RoyK | nettie: I can send you a jb patch for 1.2.6 |
| 10:36.06 | nettie | RoyK :) |
| 10:36.11 | nettie | than woul dbe great thanx a lot |
| 10:36.26 | nettie | please sent to nettie@apple.2.com |
| 10:36.29 | nettie | please sent to nettie@apple2.com |
| 10:38.43 | RoyK | sent |
| 10:42.37 | nettie | RoyK thank you very much.. I'll definitely give it a try :) |
| 10:42.51 | nettie | RoyK how many users you have |
| 10:42.51 | nettie | ? |
| 10:43.08 | azeteg | himself and his mom |
| 10:43.14 | nettie | ehehe |
| 10:43.18 | azeteg | ;) |
| 10:43.24 | nettie | I have more then :) |
| 10:43.32 | azeteg | your auntie as well? |
| 10:43.34 | nettie | at least I have my office collegues |
| 10:43.34 | nettie | ehehe |
| 10:43.38 | nettie | me? |
| 10:43.40 | nettie | nah not me |
| 10:45.11 | azeteg | nettie: do you know what my problem could be? |
| 10:45.21 | nettie | no idea |
| 10:45.24 | azeteg | http://pastebin.com/744474 |
| 10:45.36 | nettie | let's see |
| 10:45.37 | azeteg | no route to destination - is weird |
| 10:46.12 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 10:46.19 | azeteg | calling other SIP phones work fine |
| 10:46.30 | azeteg | it just doesn't work calling the ATA-186 |
| 10:46.38 | azeteg | calling FROM the ATA-186 works fine |
| 10:47.53 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 10:50.35 | nettie | uhmm |
| 10:50.39 | nettie | from what I Can see |
| 10:50.49 | nettie | it doesnt match an extension |
| 10:50.57 | azeteg | but it does |
| 10:51.29 | nettie | sure it's autheticated? |
| 10:51.37 | nettie | try |
| 10:51.40 | nettie | sip show registry |
| 10:52.00 | nettie | and see if that extension is registered |
| 10:52.03 | *** join/#asterisk wilane_ (n=user@196.207.218.107) |
| 10:52.15 | azeteg | sip show registry -> shows nothing |
| 10:54.01 | nettie | are you trying to call a sip extensions? |
| 10:54.20 | nettie | sorry |
| 10:54.22 | nettie | sip show peers |
| 10:54.23 | nettie | my bad |
| 10:55.00 | azeteg | I think I might have forgot an authentication name in the ATA conf |
| 10:55.21 | mr_horsepower | azeteg: nat. |
| 10:55.31 | azeteg | nonat |
| 10:55.40 | mr_horsepower | i have the same problem with zyxel |
| 10:55.42 | mr_horsepower | nat issues |
| 10:55.47 | azeteg | I have no NAT |
| 10:56.18 | mr_horsepower | i dont have too |
| 10:56.20 | *** join/#asterisk syneus (n=syneus@81.88.224.6) |
| 10:56.38 | syneus | hi * |
| 10:56.40 | azeteg | I have all sip peers set to no nat |
| 10:57.25 | syneus | I would like to know if is it possible that asterisk doesn't control the RTP traffic. |
| 10:57.29 | syneus | any suggestions? |
| 10:58.00 | syneus | I would that the end point control the RTP flow |
| 10:58.08 | mr_horsepower | azeteg: thats ata configurations, no asterisk configurations. |
| 10:58.18 | azeteg | what do you mean then? |
| 10:58.26 | mr_horsepower | search for nat related configurations in ata. |
| 10:58.43 | azeteg | NATIP in ATA config? |
| 10:58.51 | mr_horsepower | no ip |
| 10:59.22 | mr_horsepower | w8 a min |
| 11:00.24 | mr_horsepower | azeteg: something like "NAT Keep Alive |
| 11:00.26 | mr_horsepower | ? |
| 11:02.56 | azeteg | hmmm ok |
| 11:06.06 | SheriF_WorK | how to turn cdr debug ? |
| 11:07.23 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 11:08.52 | MrChimpy | "one hundred one" indeed. phah! |
| 11:12.51 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 11:14.23 | bmg505 | hi all |
| 11:14.48 | bmg505 | I jsut blew my whole install by installing amportal on top of my current config :) not very clever |
| 11:15.19 | bmg505 | but now that I've figured out the gui setup, I find it kinda easy to setup * |
| 11:15.37 | X-Rob_ | bmg505, it hasn't been called amportal for about 6 months now |
| 11:15.40 | *** join/#asterisk jahani (n=k@41.250.32.254) |
| 11:15.41 | X-Rob_ | you want freepbx |
| 11:15.48 | bmg505 | soz man |
| 11:15.49 | X-Rob_ | (also, see #freepbx 8) |
| 11:15.59 | bmg505 | but the daemon start is still called amportal |
| 11:16.03 | X-Rob_ | yeah |
| 11:16.12 | X-Rob_ | I'll fix that in 2.2, honest 8) |
| 11:16.30 | bmg505 | yea and warn the users that u going to destroy his current config |
| 11:16.31 | SheriF_WorK | how to turn cdr debug ? |
| 11:16.41 | X-Rob_ | bmg505, well, we do. |
| 11:16.51 | SheriF_WorK | how to turn cdr debug on ? |
| 11:16.58 | bmg505 | well I followed the instructions, and no where did it warn me |
| 11:17.08 | X-Rob_ | which instructions? |
| 11:17.10 | bmg505 | but its a test system so no harm done |
| 11:17.15 | bmg505 | INSTALL |
| 11:17.29 | X-Rob_ | Mmmm. |
| 11:17.43 | bmg505 | Yea i'm one of those idiots that actually foloow the stuff in the INSTALL file |
| 11:18.23 | X-Rob_ | most people follow the wiki |
| 11:18.24 | X-Rob_ | it's got warnings all over the place there |
| 11:18.29 | X-Rob_ | I'm just updating INSTALL now. |
| 11:19.48 | bmg505 | I should have taken a picture of myself when kpsi came up and said it cannot register |
| 11:20.04 | X-Rob_ | ### |
| 11:20.04 | X-Rob_ | Important Warning! |
| 11:20.04 | X-Rob_ | freePBX _will_ overwrite any exisiting asterisk configurations you may have. This project attempts |
| 11:20.04 | X-Rob_ | to manage as much of asterisk as it can, and this means lots of automatically generated dialplans. |
| 11:20.04 | X-Rob_ | Please visit both the Documentation wiki (http://www.aussievoip.com.au/wiki/freePBX) and the Dev |
| 11:20.05 | X-Rob_ | wiki (http://www.freepbx.org/wiki) for instructions, hints and tips. |
| 11:20.32 | bmg505 | where did I miss that? |
| 11:20.39 | X-Rob_ | I just put it in |
| 11:20.49 | bmg505 | lol o ok |
| 11:21.38 | bmg505 | at least I had a backup of the 30+ line menu I made |
| 11:22.21 | marl | hi, can someone tell me if i have the following wrong? .call files can be setup to call an internal extesnsion and when its answered then transfer the call to an outgoing line (eg. only make the external call if the internal extesnion has been asnwered)? |
| 11:26.47 | marl | as all the examples i have seen so far (and the documentation ive read) implies that .call files are normally setup to dial the external number first and THEN the internal extenion |
| 11:27.20 | hwt | how do i uninstall asterisk? |
| 11:27.23 | hwt | to start all fresh. |
| 11:28.14 | hwt | will a: |
| 11:28.15 | hwt | for i in `find / -name asterisk -type d`; do rm -rf $i; done |
| 11:28.17 | hwt | do? |
| 11:28.43 | marl | hwt, one tip, if there may be anything in your current install you might want to referance back to, backup all your data ( i forgot one time and had to repate a ton of work) thinngs like sound files and zap conf files |
| 11:29.00 | marl | did u compile from source or use a package? |
| 11:29.44 | hwt | marl: i have everything documented. |
| 11:29.56 | marl | and a backup of any custom sound files? |
| 11:30.21 | marl | that and my old call logs/recorded calls, was what i forgot :( |
| 11:30.49 | hwt | marl: nah, i have those somewhere else. |
| 11:31.07 | marl | so, source or package install? |
| 11:39.30 | hwt | source. |
| 11:40.05 | marl | does the make file not have an uninstall target? i thought it did |
| 11:42.53 | hwt | don't think so. |
| 11:47.18 | mr_horsepower | why my matra pbx, when making a call, it sends the first digit, and all the others are dtmfs? this is a normal behaviour? |
| 11:47.31 | MrChimpy | excellent. asterisk says numbers properly now. |
| 11:49.12 | clive- | mrchimpy wht does yoru patch do ? |
| 11:50.05 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 11:52.30 | *** join/#asterisk sturmflut (n=sraffein@mail.app.leitwerk.net) |
| 11:54.06 | MrChimpy | makes "say number" read numbers out with the correct english grammar |
| 11:54.32 | MrChimpy | it's nothing, but we couldn't have our system go live with balances being read as "one hundred twelve" |
| 11:55.24 | Strom_C | that's how you're supposed to read "112" |
| 11:55.29 | Strom_C | one hundred twelve |
| 11:55.37 | Strom_C | "one hundred and twelve" is technically incorrect |
| 11:55.56 | MrChimpy | no it isn't, if you actually speak english. |
| 11:56.11 | Strom_C | I speak English quite fluently, thank you very much |
| 11:56.32 | MrChimpy | presumably not the original version then |
| 11:57.07 | MrChimpy | one hundred twelve is as wrong as "color" :) |
| 11:57.25 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 11:57.45 | MrChimpy | if you speak US english, of course, that's fine. just don't call it english. |
| 11:58.29 | azeteg | mouhahah |
| 11:58.31 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:59.52 | Strom_C | balls. "six hundred seventy-three dollars and forty-nine cents" is more efficient and less ambiguous sounding than "six hundred and seventy three dollars and forty-nine cents." Granted, I despise a lot of the conventions of north american English, but this is one I happen to think is much cleaner in design. |
| 11:59.54 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-140-26.dsl.irvnca.pacbell.net) |
| 12:01.35 | MrChimpy | may have escaped your notice but language isn't "designed". it evolves. and at some point grammar is defined and that's what is correct for that language. not what saves time or is convenient - that's dialect or slang. |
| 12:04.02 | syneus | is possible to configure asterisk so that it doesn't control the RTP traffic? |
| 12:08.15 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:10.25 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 12:10.47 | [TK]D-Fender | syneus :Yes and no. Asterisk controls it by determining if the clients are ALLOWED to re-invite and pass RTP direct or not. |
| 12:12.44 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:14.09 | *** join/#asterisk holaaa (n=a@217.11.120.84) |
| 12:15.02 | syneus | [TK]D-Fender: sorry but i'm not very skilled; my collegues tell me that during some tests they noticed that RTP traffic pass through the asterisk-sip-server but they'd like RTP traffic pass only between the sip clients: is it possible? |
| 12:15.21 | tzanger | morning |
| 12:15.30 | holaaa | I hear some "noises" in my phone. Monitoring traffic, I discover it is due to a traffic "rush". Analyzing traffic with ethereal y discover the noises occurs when there is a hing "Rvr Jitter" value from time to time, from the server to the phone. No codec problem, just high Rvr Jitter value (named by ethereal) in the traffic... any ideas? The problem is not heavy network traffic itself, becaus I solated the system (phone, asterisk, pots) and still |
| 12:15.31 | holaaa | happens... any "jitter" configuration for RTP protocol? something? |
| 12:15.54 | tzanger | holaaa: your network has jitter |
| 12:16.01 | tzanger | it's a fact of life with VOIP |
| 12:16.03 | hwt | syneus: yes. that's re-inviting. |
| 12:16.10 | tzanger | your endpoints need to implement jitter buffers |
| 12:16.18 | tzanger | Asterisk has one for IAX2 and the one for SIP is in development |
| 12:16.30 | tzanger | you can help test it, search for Olle's posts about testing his SIP branch |
| 12:16.42 | puzzled | hi |
| 12:16.45 | [TK]D-Fender | syneus : Yes. Just set allt he related phones to "canreinvite=yes" and they will renegotiate RTP between them. |
| 12:16.56 | holaaa | tzanger: The voip phone has jitter configuration.. but I don't know how to do it with Asterisk it self |
| 12:17.10 | holaaa | Because I only see in google it works for IAX |
| 12:17.21 | tzanger | holaaa: I *just* said to search for Olle's posts about testing his SIP branch |
| 12:17.22 | hwt | syneus: watch out for NAT-problems, though. |
| 12:17.53 | SheriF_WorK | hum at last my CDR is working ... but there is something if i want to add start: Start of call (date/time) any idea? |
| 12:18.18 | puzzled | afiak that's included in the cdr |
| 12:18.37 | holaaa | tzanger you mean in google or any specific site... |
| 12:19.27 | syneus | [TK]D-Fender: thx a lot |
| 12:19.31 | syneus | hwt: thank U |
| 12:22.13 | *** join/#asterisk ness (n=Tom@pppin-10-b6.pop-kaltenengers.rz-online.NET) |
| 12:24.49 | ness | hi, I have a strange problem: http://forums.digium.com/viewtopic.php?t=6937. ideas? |
| 12:28.15 | hwt | ness: probably some <cr>-issues. |
| 12:28.33 | *** join/#asterisk coppice (n=chatzill@66.155.17.210.dyn.pacific.net.hk) |
| 12:30.48 | ness | a) I looked in the debugger and I'm pretty sure the string is correct, b) if I issue the same call twice, the same symptoms occur |
| 12:31.00 | SheriF_WorK | puzzled: yes but should i do something ? or just add the colum start ? |
| 12:31.43 | znoG | hey does anyone know if I can tweak something in Asterisk for it to look for Caller ID info when a Lucent Definity PBX connected via FXO/FXS ports dials out via *? |
| 12:32.23 | znoG | like this: <lucent definity> --> <TDM2400 card with FXS ports><Asterisk> --> <remote asterisk box> |
| 12:32.48 | znoG | when I dial from a lucent definity extension through the FXS port on the Asterisk box (i have usecallerid=yes in zapata.conf), no caller ID info is received by Asterisk |
| 12:33.44 | coppice | znoG: most PBXes don't generate caller ID. are you sure your Lucent does? |
| 12:34.29 | znoG | coppice: from what I'm told by a Lucent tech, it does. |
| 12:34.43 | znoG | coppice: what I mean by Caller ID is simply the extension that made the call |
| 12:35.08 | coppice | znoG: if you plug a phone into that Lucent port, does it decode caller ID properly? |
| 12:35.36 | znoG | hrm, nope, good point. I haven't tried but I'm pretty sure it doesn't. |
| 12:36.29 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 12:36.43 | Dovid | Morning all |
| 12:36.49 | ness | are the requests coming in via the manager api stored in a log file? |
| 12:38.10 | Dovid | . |
| 12:38.19 | SheriF_WorK | how to activate start in CDR ?? |
| 12:38.51 | marl | hi, can someone tell me, if i have iax enabled properly on my * box, should nmap -sS ip-of-*-box show that port as being open?, im trying to find out if my iax is working, as i cant connect to it via iax |
| 12:39.31 | sturmflut | You have to use -sU instead of -sS |
| 12:39.36 | sturmflut | IAX uses UDP |
| 12:39.36 | Strom_C | marl: what does the console say when you try to connect? |
| 12:40.14 | marl | its saying nothing, but im trying to work out if its a firewall problem or not |
| 12:40.24 | Strom_C | what's your verbosity level set at |
| 12:40.30 | marl | and the -sU shows nothing on the port :( |
| 12:40.34 | Dovid | marl: are you forwarding the ports ? |
| 12:40.43 | *** part/#asterisk ness (n=Tom@pppin-10-b6.pop-kaltenengers.rz-online.NET) |
| 12:40.52 | marl | very hi verb, and yup im forwarding |
| 12:40.58 | sturmflut | marl: "iax2 show peers" on your asterisk console should say something like Status OK |
| 12:41.03 | marl | but was running the nmap on the asterisk box itsself |
| 12:41.08 | *** join/#asterisk ness (n=Tom@pppin-10-b6.pop-kaltenengers.rz-online.NET) |
| 12:41.50 | coppice | is it a holiday in .us today? |
| 12:42.03 | Dovid | coppice: yes it is |
| 12:42.21 | coppice | i guess that explains the huge drop in spam |
| 12:42.26 | Dovid | coppice: I believe it is memorial day |
| 12:42.26 | RoyK | rotfl |
| 12:42.32 | Dovid | hehe |
| 12:42.39 | Dovid | Nah. They work 24/7 |
| 12:42.51 | Dovid | I know that I the us they are trying to make it a felony |
| 12:42.59 | sturmflut | marl: "nmap -sU -p 4569 localhost" should report the port as being closed because you usually run asterisk on some interface different than lo |
| 12:43.07 | coppice | actually they don't. I receive far less spam at weekends |
| 12:43.23 | Dovid | As long as there are fines it is worth it for them they still make money after paying the fines |
| 12:43.30 | marl | running nmap with the interface ip addy that is being used for normal network access |
| 12:43.35 | marl | not the lo interface |
| 12:43.35 | Dovid | If there is jail time some of them will think twoce |
| 12:43.41 | Ecio | coppice: maybe zombie pcs are off during the weekend :) |
| 12:43.50 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:44.02 | sturmflut | marl: The TCP/IP stack will recognize that this is your own IP adress and redirect the traffic to the lo interface |
| 12:44.14 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 12:44.21 | coppice | Ecio: yeah. all those fortune 500 corporate desktops get switched off :-) |
| 12:44.33 | Ecio | yep :) |
| 12:45.00 | AsteriskAlbania | <PROTECTED> |
| 12:45.36 | AsteriskAlbania | I need to use G729 codec for low bandwidth consumption |
| 12:46.03 | Dovid | Stupid question to ask but do u have the licence for it ? |
| 12:46.40 | RoyK | AsteriskAlbania: zap is not a codec |
| 12:46.45 | RoyK | eh |
| 12:46.55 | RoyK | AsteriskAlbania: have you bought g.729a? |
| 12:47.10 | AsteriskAlbania | no |
| 12:47.20 | AsteriskAlbania | should I buy it :) |
| 12:47.25 | *** join/#asterisk aze (n=aze@ACayenne-101-1-10-171.w81-248.abo.wanadoo.fr) |
| 12:47.25 | Dovid | yes |
| 12:47.25 | marl | scanning the *'s ip addy from another machine on the local network, shows the port as closed |
| 12:47.33 | Dovid | G729 will not work without it |
| 12:47.56 | AsteriskAlbania | is G729 the best codec , and how much does it costs |
| 12:47.57 | RoyK | AsteriskAlbania: it doesn't come with asterisk. there is a 'free' g.729a codec out there, but not legal in some countries (US, UK + +) |
| 12:47.58 | marl | im sure ive missed something stupid here, but cant work out what it is :( could it be anything to do with iax2? |
| 12:48.04 | Dovid | It depends for what |
| 12:48.18 | Dovid | It does more transcoding so it is more cpu intensive |
| 12:48.22 | RoyK | AsteriskAlbania: it's the best low-bandwidth codec, and costs, from digium, $10 per channel |
| 12:48.23 | Dovid | But it saves on bandwith |
| 12:48.36 | coppice | AsteriskAlbania: how do you define best? |
| 12:48.43 | RoyK | coppice: coolest |
| 12:48.49 | AsteriskAlbania | comparing to free ones |
| 12:48.53 | coppice | its not the coolest |
| 12:48.54 | RoyK | AsteriskAlbania: use gsm |
| 12:49.02 | AsteriskAlbania | let me try it |
| 12:49.10 | RoyK | AsteriskAlbania: gsm is free and low bandwidth and sounds like shit but it works |
| 12:49.11 | Dovid | If bandwith isnt an issue then u dont need it |
| 12:49.24 | Dovid | Not allways shit |
| 12:49.30 | coppice | that doesn't define best. best quality? lowest bit rate? widest compatibility? |
| 12:49.32 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a222.nwlnnh.tds.net) |
| 12:49.58 | RoyK | coppice: worst audio, most bandwidth? instant tom waits voice?? |
| 12:50.15 | Dovid | Not the worst audio |
| 12:50.19 | Dovid | It works |
| 12:50.27 | RoyK | Dovid: gsm works, yes |
| 12:50.30 | Ecio | guys i have problems with SIP trunk between Cisco Call Manager 4 and Asterisk, can someone give me some hint? |
| 12:50.49 | AsteriskAlbania | I am testing with GSM and it seems good |
| 12:51.09 | Dovid | Over the internet or localy ? |
| 12:51.17 | coppice | the main reason to use G.729 is because the other end does |
| 12:51.50 | AsteriskAlbania | PC -> Asterisk -> Phone |
| 12:51.52 | Dovid | And low bandwith |
| 12:52.00 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a222.nwlnnh.tds.net) |
| 12:52.01 | Dovid | I use it when I connect via grps |
| 12:52.19 | coppice | G.729 isn't much lower than GSM, when you add in all the overheads |
| 12:52.36 | Dovid | But then gsm has overheads too |
| 12:53.00 | AsteriskAlbania | I am interesting more regardin the quality that can work on 64 kbps |
| 12:53.10 | *** join/#asterisk nags (n=nags@125.16.129.16) |
| 12:53.31 | *** part/#asterisk ness (n=Tom@pppin-10-b6.pop-kaltenengers.rz-online.NET) |
| 12:53.35 | coppice | if you compare 13.2K with 8K the difference looks big. comparing 29K wiith 24K doesn't look so different |
| 12:54.38 | AsteriskAlbania | :) |
| 12:56.20 | *** part/#asterisk holaaa (n=a@217.11.120.84) |
| 12:56.37 | AsteriskAlbania | what is the difference in quality between GSM and G729 |
| 12:56.53 | Dovid | I have not seen a diffrence |
| 12:56.57 | Dovid | But that was on a LAN |
| 12:56.58 | AsteriskAlbania | is there any loss on G729 since it is 8 kbps |
| 12:57.06 | Dovid | G729 is more cpu intensive |
| 12:57.20 | AsteriskAlbania | got the point thankyou |
| 12:57.33 | Strom_C | what do you mean "loss"? |
| 12:57.40 | Dovid | U got play and see what u get. |
| 12:57.40 | AsteriskAlbania | on voice quality |
| 12:57.54 | Strom_C | gsm and g729 sound equally abysmal to me |
| 12:58.38 | Strom_C | different flavors of abysmal, yes...but abysmal. :) |
| 13:04.00 | *** join/#asterisk coppice (n=chatzill@187.197.17.210.dyn.pacific.net.hk) |
| 13:04.02 | Strom_C | :43 < Dovid> As long as there are fines it is worth it for them they still make money after paying the fines |
| 13:04.07 | Strom_C | 05:er |
| 13:04.08 | Strom_C | er |
| 13:04.09 | Strom_C | what the hell |
| 13:04.23 | Dovid | ? |
| 13:04.38 | Strom_C | copy/paste weirdness |
| 13:04.44 | Dovid | We were talkin b4 about spammers |
| 13:04.59 | Strom_C | it was a mistake. ignore the bell on your terminal. |
| 13:05.05 | Dovid | Lol. ok |
| 13:06.05 | *** join/#asterisk fnordian (i=fnord@spaceboyz.net) |
| 13:06.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 13:07.52 | fnordian | hi |
| 13:08.05 | Dovid | hello |
| 13:08.39 | fnordian | i've got a problem with my generator and a client using silence supression |
| 13:08.41 | Ecio | is there some SIP tool that can be used to determine why i got a 404 not found ? |
| 13:09.07 | Dovid | Ecio: when you try to make a call u get the error ? |
| 13:09.20 | Dovid | fnordian: generator ? |
| 13:10.07 | Ecio | dovid: i have this SIP trunk between cisco call manager and asterisk, if i call from a phone behind CCM i can reach asterisk (both conference number and xlite softphone number too) |
| 13:10.18 | Ecio | if i call back from xlite to my number on the CM |
| 13:10.21 | Ecio | i got a 404 |
| 13:10.25 | fnordian | Dovid: ast_generator |
| 13:10.40 | Dovid | Ecio: I dont know cisco sorry :( |
| 13:10.52 | Ecio | i can see from ethereal that * is calling CM using my_cisco_phone_number@callmanager_IP |
| 13:10.53 | Dovid | Ah ok |
| 13:10.59 | Ecio | and CM says "not found" :/ |
| 13:11.20 | fnordian | Dovid: as far as i understood it, channel.c polls my generator regularly |
| 13:11.22 | Dovid | This is when u are connection to ur ast. Box and u want it to call the CM ? |
| 13:11.35 | [TK]D-Fender | Ecio : 404 means whatever # you are dialing is simply not valid. PERIOD |
| 13:11.38 | Dovid | fnordian: dont know that function well |
| 13:11.43 | Dovid | Try voip-info.org |
| 13:12.02 | fnordian | Dovid: either triggered by a scheduler or by i thing called "phase locked mode" |
| 13:12.05 | Ecio | d-fender: i see.. but it's strange |
| 13:12.13 | Dovid | Ecio: paste your dial plan to call the CM in pastebin.com and put the link here |
| 13:12.20 | Ecio | k |
| 13:12.40 | Dovid | Either you didnt code it right in asterisk or the CM Is rejecting it for some reason |
| 13:13.27 | fnordian | huh |
| 13:13.30 | Dovid | You are connecting to it via SIP ? |
| 13:13.41 | fnordian | Dovid: talking to me? |
| 13:13.41 | Ecio | http://pastebin.ca/59564 |
| 13:13.42 | *** join/#asterisk Splat (n=Splat@220-253-102-19.TAS.netspace.net.au) |
| 13:13.50 | Dovid | Was tallin to ecio |
| 13:13.54 | fnordian | a |
| 13:13.57 | fnordian | ok, sorry |
| 13:14.04 | Ecio | dovid: we have 4 digit numbers on the CM |
| 13:14.09 | Ecio | my society is 4xxx |
| 13:14.35 | Ecio | in the debug i see asterisk sayin "SIP/callman02-3c4f is circuit-busy" |
| 13:14.38 | Dovid | Ecio: how are you connecting asterisk to it ? Via sip ? |
| 13:14.43 | Ecio | yes SIP trunk |
| 13:14.48 | kay2 | Do I have to install asterisk-addon if I want to access to the RealTime using mysql ? |
| 13:14.52 | Dovid | And u have it set to peer ? |
| 13:14.59 | Ecio | i've tried some configurations... i've copied the one found on voip-user.org |
| 13:15.00 | Dovid | kay2: yes |
| 13:15.14 | Ecio | callman01 and callman02 from http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration |
| 13:15.18 | Ecio | obviously changing the ips |
| 13:15.30 | Ecio | and i've tried also with CCM1 that is a peer |
| 13:15.54 | coppice | 2 days to the feature freeze for 1.4, and most interesting things don't seem to be in the SVN trunk right now. |
| 13:16.02 | Ecio | as u can see i've created a dialplan with 4XXX but i've tried also hard-coding the SIP call on some "fast access" Sip numbers |
| 13:16.09 | Dovid | Based on circuit-busy it seems the CM is rejecting the call |
| 13:16.16 | tzafrir | coppice, what do you mean? |
| 13:16.23 | Dovid | Can u try to have xlite connect to it directly and see if it works ? |
| 13:17.03 | Dovid | It seems CM is not letting the call in |
| 13:17.05 | coppice | tzafrir: what I said. jitterbuffer, t.38, nothing interesting has gone into the trunk yet |
| 13:17.26 | Ecio | dovid: that's what i think too... but once it worked (some days ago, doing a quick test with a@h) |
| 13:17.38 | Dovid | Ecio: when you call exten 5,6,7 do u get a busy too ? |
| 13:18.01 | Dovid | Ecio: try connecting direct via ur sip phone and see what happens |
| 13:18.03 | Ecio | on xlite i got a 503 service not available |
| 13:18.09 | coppice | RoyK: I see you are thinking the same as me :-) |
| 13:18.14 | Dovid | Yes. Then it is a CM issut |
| 13:18.18 | Ecio | on debug i see circuit busy |
| 13:18.38 | Dovid | yup |
| 13:18.39 | Ecio | dovid: that's why i was wondering if there is some tool do some test on sip port on the cisco |
| 13:18.47 | Ecio | do = to do |
| 13:18.51 | Dovid | I dont know cisco :( |
| 13:18.59 | Ecio | and some general SIP tool? |
| 13:19.22 | Dovid | Do a google search or voip-info.org |
| 13:19.26 | Ecio | something like /trytoconnecttothatbitch.sh CCM_ip |
| 13:19.26 | Ecio | :D |
| 13:20.41 | *** join/#asterisk _omer (i=_omer@203.215.180.247) |
| 13:23.58 | Ecio | uhm.. |
| 13:26.42 | Ecio | crappy cisco documentation... i cant find anything useful... |
| 13:26.57 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 13:28.37 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 13:28.58 | Dovid | Thats why we go only with asterisk |
| 13:33.02 | Dovid | <PROTECTED> |
| 13:34.51 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:35.16 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 13:38.46 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:38.54 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 13:42.36 | fnordian | Ecio: sipsak might be your friend |
| 13:47.09 | kay2 | <PROTECTED> |
| 13:47.15 | kay2 | Someone has an idea ?/ |
| 13:48.54 | fnordian | echo 'noload => format_mp3' >> /etc/asterisk/modules.conf |
| 13:49.12 | fnordian | .so |
| 13:50.17 | Ecio | thx fnordian |
| 13:50.57 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.224.Dial1.SanJose1.Level3.net) |
| 13:51.11 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.224.Dial1.SanJose1.Level3.net) |
| 13:53.05 | azeteg | I have a little problem with music on hold that doesn't want to work. Here is my log: http://pastebin.com/744780 |
| 13:53.13 | azeteg | anyone knows what might cause this? |
| 13:53.48 | azeteg | I never hear any music |
| 13:56.31 | *** join/#asterisk assert_true (n=Sunil@59.176.43.38) |
| 13:56.36 | [TK]D-Fender | azeteg : a few things, either you have no files to in the proper folder, you are using Native MoH and have no files of a compatible format (MP3's require you to have compiled format_mp3.so from the Asterisk Add-ons pack), or you are NOT using native and your mpg123 is no good. |
| 13:56.58 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 13:57.00 | azeteg | I'm using native |
| 13:57.10 | azeteg | it doesn't compile mp3 module as default? |
| 13:57.16 | mosty | why is it better to use _X. than _. ? |
| 13:57.34 | [TK]D-Fender | azeteg : Its not included with the base * tarball for legal reasons |
| 13:57.35 | Nugget | _. will match all the special meta-extensions like s, i, h, and t. |
| 13:57.56 | azeteg | ah ok |
| 13:57.57 | Nugget | you can use it, but be very careful otherwise your asterisk will do wonky things |
| 13:58.00 | azeteg | where do I get it? |
| 13:58.04 | azeteg | (google) |
| 13:59.05 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:59.58 | azeteg | *compiling* |
| 14:00.20 | [TK]D-Fender | Nugget : We need an alpha or "complete" wildcard indicator for dial-plan matching like _X!?X or so where ! might represent ANY character, and ? and CHARACTER. that would also allow matches like [A-F,R] as well |
| 14:00.39 | [TK]D-Fender | azeteg : on asterisk.org there is a link to Digium's ftp for it. |
| 14:01.00 | azeteg | compiled and installed already thankx |
| 14:01.20 | [TK]D-Fender | azeteg : You're welcome |
| 14:02.25 | *** join/#asterisk awad (n=naoshige@avtomat.probsd.net) |
| 14:06.49 | azeteg | [TK]D-Fender: now I have the format_mp3.so module installed, but I have the same behavior. What could it be? |
| 14:07.26 | [TK]D-Fender | azeteg : Have you verified the presence of appropriate files in the MoH folder you specified? Pastebin your musiconhold.conf files jsut to be sure as well. |
| 14:07.44 | azeteg | http://pastebin.com/744798 |
| 14:07.46 | [TK]D-Fender | And have you manually loaded the module or completely restarted * to put it in effect? |
| 14:08.01 | azeteg | completely restarted |
| 14:08.04 | [TK]D-Fender | azeteg : I didn't ask for CLI out, I asked for the config file... |
| 14:09.27 | azeteg | is not much in it |
| 14:09.47 | azeteg | http://pastebin.com/744801 |
| 14:12.04 | [TK]D-Fender | azeteg : that is NOT set up for Native Moh. To do that you need to change your mode to "files" |
| 14:12.11 | azeteg | ah |
| 14:12.13 | azeteg | thanks |
| 14:12.17 | *** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 14:14.20 | azeteg | I put mode as file |
| 14:14.21 | azeteg | files |
| 14:14.24 | azeteg | no difference |
| 14:14.58 | fnordian | does anyone know what has happened to ast_silence_suppression_enabled? |
| 14:16.06 | azeteg | ok, I had some weirdness in that dir |
| 14:16.08 | azeteg | now works |
| 14:16.12 | azeteg | thank you |
| 14:18.18 | [TK]D-Fender | np... make sure your MP3's don't have ID3 tags either... |
| 14:23.01 | *** join/#asterisk jpbotelho (n=jpbotelh@201.7.108.130) |
| 14:23.21 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 14:23.39 | ceeto | Hi all. Are you guys compiling spandsp from source or are you using precompiled packages as that comes with debian, redhat, etc.? |
| 14:23.54 | ceeto | I'm trying to "improve" my faxing capabilities. |
| 14:24.18 | [TK]D-Fender | ceeto : Compile all the way... |
| 14:24.30 | [TK]D-Fender | ceeto : The only way to fly for all things * |
| 14:24.58 | ceeto | ;) yea, figured as much... |
| 14:25.05 | ceeto | Do you use inbound faxing? |
| 14:25.37 | *** join/#asterisk stevej (n=stevej@mail.joneslinux.com) |
| 14:26.03 | [TK]D-Fender | ceeto : Yup, works pretty decent. |
| 14:26.11 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 14:26.43 | [TK]D-Fender | ceeto : I had mine at work set up by my solution provider, but I've done it for a client of mine with X100P's and seems ok there too. |
| 14:27.15 | ceeto | It's working "ok" with TDM400P's.. sometimes the faxes get corrupted... |
| 14:27.38 | ceeto | I'm using spandsp and rxfax from debian packages, but asterisk 1.2 compiled from source.. |
| 14:28.03 | ceeto | I get stuff like: channel.c:2326 set_format: Unable to find a codec translation path from unknown to unknown |
| 14:28.09 | ceeto | And: app_rxfax.c:305 rxfax_exec: Unable to restore read format on 'Zap/2-1' |
| 14:28.28 | [TK]D-Fender | :/ |
| 14:28.48 | [TK]D-Fender | I wouldn't try to mix & match if I were you... bad things happen |
| 14:29.18 | ceeto | Is spandsp and the rxfax sources available through digium or third party? |
| 14:29.28 | ceeto | i.e., is there 'official' versions for the 1.2.x trees? |
| 14:29.34 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 14:30.12 | [TK]D-Fender | ceeto : No, SpanDSP is completely 3rd party |
| 14:30.37 | [TK]D-Fender | ceeto : Its all on http://www.soft-switch.org |
| 14:30.39 | ceeto | What about rxfax? |
| 14:30.42 | ceeto | (thanks, btw) |
| 14:30.45 | [TK]D-Fender | ceeto : All of it. |
| 14:30.57 | RoyK | ceeto: rxfax just uses spandsp |
| 14:31.28 | [TK]D-Fender | ceeto : I found the instruction a little lacking but once you browse through their FTP you get to realize there are a NUMBER of steps involved and likely some manual patching of the Asterisk makefile. |
| 14:31.57 | ceeto | k, thanks. I'll go check it out. |
| 14:32.00 | [TK]D-Fender | ceeto : I am a non-programmer as far as most things Linux related is concerned but figured it out at a decent rate. |
| 14:32.03 | RoyK | [TK]D-Fender: you mean for app_[tr]xfax to work? |
| 14:32.34 | [TK]D-Fender | RoyK : Those 2 apps have to be downloaded from their site and manually placed into the * apps folder and the makefile patched to compile. |
| 14:32.47 | [TK]D-Fender | RoyK : A bit of work for sure, but not too serious. |
| 14:33.01 | ceeto | What about the source for app_rxfax.so? |
| 14:33.03 | RoyK | you only copy app_[tr]xfax.c into apps/ and apply the apps/Makefile patch from the download area |
| 14:33.12 | RoyK | ceeto: just a sec |
| 14:33.16 | [TK]D-Fender | ceeto : Available on soft-switch.org |
| 14:34.06 | RoyK | ceeto: grab spandsp from here: http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/ and the apps from http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/ and apply http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/apps_Makefile.patch to apps/Makefile, make install, done |
| 14:34.15 | RoyK | ~spandsp? |
| 14:34.18 | jbot | it has been said that spandsp is cool : http://www.soft-switch.org/installing-spandsp.html |
| 14:34.39 | [TK]D-Fender | RoyK : Applying the patch doesn't always work so well as I've discovered, but the effect its supposed to do seemed pretty evident to me (all on instinct) do I just started cut&pasting my way through it and everything worked like a charm. |
| 14:36.19 | ceeto | Thanks, all. |
| 14:36.41 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 14:37.24 | RoyK | [TK]D-Fender: anyway, it consists of a 10 lines addition, so hand-patching isn't too hard :) |
| 14:38.17 | Dovid | Anyone know the max channels voipjet allows ? |
| 14:38.26 | Dovid | I need a provider that will allow 50 channels per minute |
| 14:38.41 | Dovid | And we burst 50 - 100 calls at a time and random times of the day |
| 14:40.01 | kay2 | someone could tell me what's wrong with that : |
| 14:40.01 | kay2 | res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk_w2 on localhost. Check debug for more info. |
| 14:40.02 | kay2 | May 29 16:29:22 WARNING[791]: res_config_mysql.c:450 load_module: MySQL RealTime: Couldn't establish connection. Check debug. |
| 14:41.03 | [TK]D-Fender | RoyK : I didn't think so even with my lack of any experience in doing so. |
| 14:42.18 | zoa | did somebody ever hear about caller id time ? |
| 14:42.43 | coppice | what about caller ID time? |
| 14:45.21 | zoa | im looking for the specs for that |
| 14:45.25 | zoa | any idea where to look for it ? |
| 14:45.31 | zoa | or how its called officially ? |
| 14:45.41 | zoa | cant really find anything on google |
| 14:45.45 | coppice | the caller ID message in many places contains the date and time |
| 14:45.50 | zoa | aha |
| 14:45.55 | zoa | so that is the same field as the name |
| 14:45.57 | *** join/#asterisk edguy3 (n=edguy@host-24-149-134-164.patmedia.net) |
| 14:46.11 | coppice | no. its the date and time field :-) |
| 14:46.35 | zoa | k thanks :) |
| 14:47.19 | coppice | look at the code in spandsp for ADSI processing. All the fields I know about are handled in that |
| 14:48.03 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 14:49.16 | zoa | oki thanks |
| 14:49.36 | mitcheloc | oi zoa ;) |
| 14:51.03 | zoa | you saved me a day of googling again |
| 14:51.04 | zoa | hey ho |
| 15:01.36 | Ecio | bye all |
| 15:03.39 | [TK]D-Fender | zoa : I used to have docs on the full spec... |
| 15:04.15 | [TK]D-Fender | zoa : and on a manual CID module I had I needed to aprse it out of the string which would be dumped back as raw datastream from FSK. |
| 15:06.11 | coppice | i love the pricing for those specs. something like $100 for the MWI spec, if I recall, and its 3 pages :-) |
| 15:06.47 | RoyK | coppice: ? |
| 15:06.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 15:07.10 | mitcheloc | which specs are you guys referring to? |
| 15:07.14 | coppice | the US analogue caller ID spec is pretty simple. some countries have a lot more possible fields |
| 15:07.20 | coppice | caller ID specs |
| 15:07.37 | *** join/#asterisk PoWeRKiLL (i=PoWeRKiL@193.189.125.8) |
| 15:08.06 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-124.prod-infinitum.com.mx) |
| 15:08.49 | RoyK | I'd writing three pages and then charge $100 for them..... |
| 15:09.03 | *** join/#asterisk sturmflut (n=sraffein@mail.app.leitwerk.net) |
| 15:09.09 | sturmflut | Hi |
| 15:09.23 | *** join/#asterisk normast (n=Norm@CPE0014bf80aeff-CM0012c90d3496.cpe.net.cable.rogers.com) |
| 15:09.23 | coppice | i think the 3 pages included a title page and a revision history page too. |
| 15:10.03 | RoyK | quite dilbertish |
| 15:10.30 | sturmflut | Anybody ever successfully connected a SwyxPhone (VoIP Phone from Siemens) to Asterisk? People keep telling me that Swyx talks SIP but the packets coming from this phone here do not look like SIP |
| 15:10.44 | coppice | well, the spec only has to document the contents of one tiny message. everything else comes from the other specs |
| 15:11.47 | coppice | sturmflut: most of these phones can be reflashed with different software for different protocols. you might not have a SIP one |
| 15:12.02 | coppice | siemens love MGCP |
| 15:12.25 | sturmflut | coppice: Oh, is there a place where I can get new flash images? Or do I have to request them from Siemens? |
| 15:12.39 | blitzrage | [TK]D-Fender: file is an anagram for Leif eh? :) |
| 15:12.47 | coppice | i don't know the answer to that |
| 15:12.48 | blitzrage | you trhink you're soooooo clever that you figured it out :) |
| 15:13.06 | sturmflut | coppice: Okay, thanks for the hint |
| 15:16.07 | *** join/#asterisk momelod (n=momelod@HSE-London-ppp290865.sympatico.ca) |
| 15:16.12 | momelod | hello people |
| 15:16.42 | momelod | i have a question about echo cancelation, how do i enable this feature on my digium card? |
| 15:17.31 | salviadud | i think its in a config file, zaptel.conf |
| 15:17.44 | salviadud | or zapata.conf |
| 15:17.51 | salviadud | i can't remember, i don't use those cards |
| 15:17.54 | kay2 | someone could tell me why I get that error with asterisk realtime: WARNING[995]: res_config_mysql.c:551 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. |
| 15:18.12 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:21.10 | [TK]D-Fender | blitzrage : ! ! ! |
| 15:21.25 | [TK]D-Fender | blitzrage : I am SMRT |
| 15:21.29 | blitzrage | w00t |
| 15:21.46 | InfraRed | SMRT |
| 15:21.58 | InfraRed | UR DUMB |
| 15:22.09 | gaupe | kay2: it's not a ERROR it's a WARNING |
| 15:23.38 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-112-179.telkomadsl.co.za) |
| 15:23.48 | *** part/#asterisk assert_true (n=Sunil@59.176.43.38) |
| 15:24.39 | [TK]D-Fender | InfraRed : Sorry... I haven't validated your qualification to trash talk with me :) |
| 15:24.53 | *** join/#asterisk coppice (n=chatzill@187.197.17.210.dyn.pacific.net.hk) |
| 15:25.35 | flujan | I set up asterisk to work with a E1. The incomming calls aren't entering the context nor executing the dialplan. I'm having just this in the console: http://pastebin.com/744931 |
| 15:25.41 | flujan | How can I diagnose the problem? |
| 15:25.44 | InfraRed | this is my qualification |
| 15:25.56 | coppice | RoyK: did you see the reply to your question on the mailing list? |
| 15:26.00 | flujan | The dialplan, is simple... It just playback the hello-world sound. |
| 15:26.03 | [TK]D-Fender | InfraRed : Yup.... thats trash.. you should throw it out :D |
| 15:26.50 | salviadud | pastebin the dialplan flujan |
| 15:30.09 | flujan | salviadud, http://pastebin.com/744943 |
| 15:30.16 | flujan | salviadud, it is above the errro message |
| 15:30.37 | *** join/#asterisk assert_true (n=anil@59.176.43.38) |
| 15:32.46 | salviadud | flujan, what channel are you using? |
| 15:33.14 | flujan | salviadud, I put the context to the entire channel. |
| 15:33.41 | ceeto | Man, compiling software is easy... I used to be so scared of it... |
| 15:33.41 | salviadud | what channel? |
| 15:33.45 | salviadud | zap? |
| 15:33.46 | salviadud | sip? |
| 15:33.49 | flujan | salviadud, or I should specify a dialplan for each channel? |
| 15:33.57 | salviadud | just the context |
| 15:33.58 | flujan | salviadud, I'm using Unicall |
| 15:34.08 | salviadud | unicall, is that even a channel? |
| 15:34.23 | Juggie | flujan, make sure you have a context assigned to the unicall driver |
| 15:34.27 | flujan | salviadud, with a E1 link. It works like Zap channels... |
| 15:34.30 | flujan | Juggie, I have |
| 15:34.35 | flujan | I will paste bin the unicall.conf |
| 15:34.45 | Juggie | flujan, none of those errors tell me asterisk is receiving a call |
| 15:34.51 | Juggie | er, warnigs. |
| 15:34.56 | Juggie | theyt just show me unicall events |
| 15:35.06 | Juggie | you shuold see something with asterisk being unable to find a context, etc. |
| 15:35.18 | flujan | Juggie, This events appears when I make a call. |
| 15:35.35 | flujan | Juggie, I didn't receive this erros messages. |
| 15:35.45 | flujan | Juggie, saying about a missing context and stufff |
| 15:36.08 | Juggie | flujan, why are you showing me messages from maknig a call, with your incomming context |
| 15:36.12 | Juggie | your looknig at two different thnigs |
| 15:36.14 | Juggie | pick one. |
| 15:36.24 | Juggie | either incomming calls or outgoing calls |
| 15:36.47 | flujan | Juggie, salviadud http://pastebin.com/744960 |
| 15:37.11 | flujan | Juggie, I'm trying to put asterisk working with a legacy pbx |
| 15:37.19 | Juggie | what happens when you dial into asterisk |
| 15:37.39 | flujan | Juggie, I make a call. the legacy pbx receive it and then place the call to asterisk. |
| 15:37.45 | Juggie | right |
| 15:37.49 | Juggie | what messages does asterisk say |
| 15:39.38 | flujan | Juggie, when I place a call direct from asterisk I recieve a CHANUNAVAIL message |
| 15:39.44 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 15:39.55 | Juggie | thats not what i asked |
| 15:39.55 | flujan | I will pastebin it... |
| 15:39.55 | Juggie | what happens when you dial into asterisk |
| 15:40.12 | Juggie | if did 4000 is assigned to that E1 and you dial it |
| 15:40.13 | Juggie | what happens |
| 15:41.34 | flujan | Juggie, http://pastebin.com/744969 |
| 15:41.53 | Juggie | sooo |
| 15:41.56 | Juggie | what do you see here. |
| 15:42.18 | Juggie | look closely @ line 10 and 13-14 |
| 15:42.22 | Juggie | theres a link level problem |
| 15:42.30 | Juggie | well, i shoudnt say link level |
| 15:42.33 | Juggie | theres a E1 problem |
| 15:42.55 | Juggie | i dont know anything about unicall but it seems something is misconfigured |
| 15:43.11 | Juggie | and the two pbx's (asterisk & legacy) arnt talknig properly |
| 15:44.10 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 15:44.26 | dlynes_office | jbot: wake CunningPike |
| 15:44.27 | jbot | CunningPike: GOOD MORNING!!! |
| 15:44.40 | CunningPike | Good morning, dlynes_office |
| 15:44.44 | dlynes_office | heh |
| 15:45.04 | CunningPike | Wassup? |
| 15:45.08 | CunningPike | Still quiet in here? |
| 15:45.13 | dlynes_office | I found out what the problem with my pri was :0 |
| 15:45.23 | CunningPike | dlynes_office: Oh? |
| 15:45.31 | dlynes_office | The jack for span1 was faulty |
| 15:45.42 | Juggie | my problem was telus :) |
| 15:45.47 | Juggie | they fixed it friday |
| 15:45.48 | CunningPike | dlynes_office: Wow - well, at least you found it |
| 15:45.50 | dlynes_office | Juggie: ah...you're in BC, too? |
| 15:45.54 | Juggie | no |
| 15:45.56 | Juggie | ontario |
| 15:45.59 | dlynes_office | ah |
| 15:46.02 | CunningPike | Juggie: All our problems are Telus |
| 15:46.03 | flujan | Juggie, I will try to connect the E1 direct in the asterisk port an see what happen. |
| 15:46.13 | dlynes_office | Freaking Telus infests everywhere they go to |
| 15:46.36 | Juggie | flujan, sorry i dont know anything about unicall but those errors are by the unicall driver itself meaning its not able to talk to the pbx. |
| 15:46.37 | dlynes_office | Juggie: so have you had the displeasure of dealing Telus' ivr hell? |
| 15:46.53 | Juggie | dlynes_home, what do you mean calling their support line? |
| 15:46.56 | salviadud | i called telus, and they didn't even answer |
| 15:47.02 | dlynes_office | Juggie: yeah :0 |
| 15:47.05 | flujan | flujan, No problem... I will try to find coppice here latter. He wrote the unicall driver. :) |
| 15:47.08 | Juggie | we had a direct number for the tech working on oru ticket. |
| 15:47.22 | Juggie | flujan, coppice was active abotu 1hour ago |
| 15:47.24 | flujan | Juggie, thank you for your helpp |
| 15:47.25 | Juggie | he might still be around. |
| 15:47.31 | dlynes_office | Juggie: but you didn't do an initial callin to get that number? |
| 15:47.43 | dlynes_office | salviadud: what number did you call? |
| 15:47.49 | salviadud | their toll free number |
| 15:47.50 | Juggie | no problem, sorry i coudnt be of more assistance, seems to be a e1 issue thats all i can say for sure. |
| 15:47.54 | flujan | Juggie, so sad... I alredy ping he... No answer... I will try later. :) |
| 15:48.04 | dlynes_office | salviadud: i can't see that working for you |
| 15:48.11 | Juggie | dlynes_home, not that i know of, my boss delt with it |
| 15:48.12 | dlynes_office | salviadud: that number's probably only valid from Canada |
| 15:48.16 | De_Mon | any FOP users know why it's unable to listen on port 4445? |
| 15:48.18 | Juggie | he just called and asked for him |
| 15:48.25 | salviadud | FWD allows me to dial toll free numbers |
| 15:48.39 | dlynes_office | salviadud: yeah, but fwd is an american provider |
| 15:48.40 | salviadud | i don't see why i shouldn't be able to call there |
| 15:48.43 | Juggie | so does skype |
| 15:48.53 | dlynes_office | salviadud: so unless it's an american 1-800 number, it won't be able to do anything |
| 15:49.10 | salviadud | fwd does toll free from japan too |
| 15:49.24 | dlynes_office | salviadud: really? |
| 15:49.35 | salviadud | yeah, it's crazy |
| 15:49.57 | CunningPike | De_Mon: Is there another service on that port, maybe? |
| 15:49.59 | dlynes_office | salviadud: does it have a setting that allows you to force a 1-800 call to be through a Canadian local? |
| 15:50.30 | salviadud | dlynes_home, I don't know how it works, I just know it does |
| 15:50.35 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 15:50.43 | dlynes_office | salviadud: some 1-800 numbers are only available in Canada, some are only available in the US, some are available in both, and some are only available in certain provinces and/or states |
| 15:51.06 | Juggie | dlynes, the 1-800 network does alot more then that. |
| 15:51.10 | salviadud | i wanted to have a laugh at the ivr :( |
| 15:51.20 | *** join/#asterisk oceanlan|dustin (i=Iam8up@rrcs-24-172-153-135.central.biz.rr.com) |
| 15:51.28 | dlynes_office | Juggie: Well, it hasn't started making breakfast for me yet...what did I miss? |
| 15:51.30 | salviadud | maybe because i called them at 2 am... |
| 15:51.33 | Juggie | salviadud, its some stupid fustrating voice rec ivr. |
| 15:51.47 | dlynes_office | salviadud: no, it's manned 24 hours |
| 15:52.01 | Juggie | dlynes, for example, i (we) have a direct feed from bell |
| 15:52.07 | salviadud | i got 2 toll free number services |
| 15:52.09 | Juggie | which provides instant events for all our 1-800 numbers |
| 15:52.18 | salviadud | fwd and trxtel |
| 15:52.23 | Juggie | eg, someone dials a 1-800 number, we get an event that instant |
| 15:52.34 | salviadud | but, trxtel isn't working, i get some g729 error |
| 15:52.49 | dlynes_office | Juggie: what's an instant event? |
| 15:53.14 | Juggie | i mean that the stats comes in instantaniously |
| 15:53.21 | Juggie | eg, we get a record of the call the instant its placed |
| 15:53.35 | Juggie | we get a record of where it goes, if its transfered, etc. |
| 15:53.48 | dlynes_office | Juggie: ah |
| 15:53.53 | sturmflut | Wow, when a Cisco IP Phone 7914G connect to my Asterisk 1.2.1 via Skinny Asterisk dies |
| 15:53.55 | Juggie | how long the 1-800 call was |
| 15:54.02 | Juggie | we know when the 1-800 network returns busy |
| 15:54.04 | Juggie | even |
| 15:54.17 | Juggie | anything you could imagine :) |
| 15:54.28 | dlynes_office | sturmflut: why don't you upgrade to a version of asterisk sometime this decade? |
| 15:54.43 | dlynes_office | sturmflut: 1.2.1 is pretty damned old |
| 15:55.00 | De_Mon | CunningPike er.. ya actually another op-panel process that didn't die like it was supposed to //me turns in his guru badge and sits in the corner |
| 15:55.04 | Juggie | dlynes, we get about i dunno 75-100k call records a day. |
| 15:55.09 | dlynes_office | sturmflut: it's so old 1.4 is already in beta 1 |
| 15:55.14 | dlynes_office | Juggie: holy crap |
| 15:55.21 | CunningPike | De_Mon: We've all done it ;) |
| 15:55.24 | file | what, we aren't in beta 1 yet |
| 15:55.24 | dlynes_office | Juggie: where the heck do you work? |
| 15:55.40 | Juggie | a departement of the canadian goverment. |
| 15:55.47 | De_Mon | how many betas does it usualy take before a release is tags stable |
| 15:55.54 | dlynes_office | Juggie: ah |
| 15:55.54 | salviadud | when is 1.4 coming out? |
| 15:56.03 | file | end of June start of July for 1.4 |
| 15:56.06 | Juggie | dlynes, the stats is mostly used for agent forcasting. |
| 15:56.09 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:56.11 | dlynes_office | file: it said 1.4 beta 1 was out now in the 1.4 topic last night |
| 15:56.23 | sturmflut | dlynes_office: It comes with Debian Backports |
| 15:56.24 | Juggie | did we have enough agents, how many people got busy signals, did we haev agents idle, etc. |
| 15:56.31 | dlynes_office | Juggie: ah...for determining how many people to schedule on each shift? |
| 15:56.31 | salviadud | any major changes in 1.4? |
| 15:56.41 | file | I find that hard to believe :) |
| 15:56.42 | De_Mon | salviadud always! |
| 15:56.44 | Juggie | dlynes, yep. thats one part of it. |
| 15:56.46 | file | but whatever |
| 15:56.53 | dlynes_office | file: Ask JerJer...he changed it |
| 15:57.01 | file | I have a log |
| 15:57.08 | dlynes_office | so do i :) |
| 15:57.08 | file | X days until Asterisk 1.4 beta! |
| 15:57.13 | file | 3.14159265 days until Asterisk 1.4 beta! |
| 15:57.13 | dlynes_office | ah.. |
| 15:57.17 | dlynes_office | maybe that's what it was :) |
| 15:57.24 | file | almost gave me a heart attack |
| 15:57.25 | [TK]D-Fender | file : Mmmmm Pie... |
| 15:57.26 | salviadud | i can imagine the extension language is going to stay the same, and something about sip b |
| 15:57.41 | dlynes_office | salviadud: AEL has been replaced with AEL2 |
| 15:58.03 | dlynes_office | salviadud: but nice try :) |
| 15:58.16 | salviadud | o_O |
| 15:58.33 | salviadud | what will AEL2 do now? |
| 15:58.37 | dlynes_office | salviadud: it's impossible to make predictions when you're talking about Asterisk :0 |
| 15:58.55 | file | we'll have a full feature list/change thing... |
| 15:58.58 | dlynes_office | salviadud: beats me...I don't use AEL |
| 15:59.01 | file | it's just not released yet :D |
| 15:59.07 | dlynes_office | salviadud: i just know it's a complete rewrite of AEL2 |
| 15:59.07 | file | so the document isn't written yet |
| 15:59.16 | dlynes_office | erm complete rewrite of AEL I mean |
| 16:00.38 | salviadud | i wish 1.4 could mixmonitor into mp3 |
| 16:01.22 | *** join/#asterisk theorem_ (n=theorem@pool-71-251-196-97.nwrknj.fios.verizon.net) |
| 16:01.26 | theorem_ | fun fun fun |
| 16:01.38 | file | time to... |
| 16:01.38 | file | CLEAN! |
| 16:01.45 | dlynes_office | Until theorem_ showed up to spoil the fun :((( |
| 16:01.53 | theorem_ | hoo ha ? |
| 16:02.38 | file | yup, all that fun we're having |
| 16:03.53 | file | who wants to go run my errands for me? |
| 16:03.58 | file | I know you all do! |
| 16:04.02 | theorem_ | got $ ? |
| 16:04.14 | file | yes, $5 |
| 16:04.14 | theorem_ | or does any one stop involve an ATM ? :) |
| 16:04.56 | theorem_ | I have a feeling that $5 is unlikely to even cover gas .. |
| 16:05.08 | file | pfft I'm not forcing you |
| 16:05.12 | theorem_ | hehe |
| 16:05.35 | file | I have these big file folders for business documents/bills/etc... and no place to put them |
| 16:05.38 | CunningPike | file: Just pm me your bank account details and I'll get right on it |
| 16:05.42 | mr_horsepower | dial() should be rewritten |
| 16:05.47 | dlynes_office | theorem_: $5USD, or $5Cdn? |
| 16:05.48 | file | mr_horsepower: yes we know, moving on |
| 16:06.01 | dlynes_office | theorem_: yeah...$5 USD won't get you much of anywhere, anymore :) |
| 16:06.07 | theorem_ | USD > ca |
| 16:06.08 | CunningPike | dlynes_office: Much the same thing, these days |
| 16:06.11 | *** join/#asterisk RoyK (n=roy@ti211310a080-3110.bb.online.no) |
| 16:06.34 | dlynes_office | mr_horsepower: you're volunteering? |
| 16:06.38 | mr_horsepower | file: it should be better to suport billing systems. calls should run on a diferent thread or something, so we can bill correct the call. |
| 16:06.39 | theorem_ | yeah I $5 regular in the tank the other day .. |
| 16:06.41 | theorem_ | 1.7 gallons |
| 16:06.43 | theorem_ | :( |
| 16:06.48 | mr_horsepower | dlynes_office: maybe i have to! :D |
| 16:06.57 | dlynes_office | mr_horsepower: nobody's stopping you |
| 16:06.58 | theorem_ | *I put (rther) |
| 16:07.10 | mr_horsepower | we need it over here, but, you need another things before. |
| 16:07.37 | mr_horsepower | s/you/we |
| 16:08.22 | mr_horsepower | re-write dial() will change everything, and break almost everything, not a easy task i think. |
| 16:08.23 | file | right now the core group is in the middle of working on getting a new 1.2 release and the 1.4 beta out |
| 16:08.40 | file | so new things are not a huge priority |
| 16:08.46 | dlynes_office | file: ever notice how people bitch about deficiencies in open source, but they don't want to help fix/improve what they complain about? |
| 16:09.14 | file | don't remind me |
| 16:09.20 | mr_horsepower | sip implementation in asterisk its very bad, i dont know about the new sip implementation that's on oej branch |
| 16:09.31 | file | why is it very bad? |
| 16:09.48 | file | do you have ANY idea how complicated SIP is to implement with maximum interoperability and compatibility? |
| 16:09.49 | theorem_ | I've only run into problems with AIX |
| 16:09.50 | mr_horsepower | let me correct it, not very bad, its not very god. |
| 16:10.07 | file | what's not good about it? |
| 16:10.09 | theorem_ | mr_horsepower - I suggest you work on it then. |
| 16:10.10 | file | give me solid reasons. |
| 16:10.12 | dlynes_office | theorem_: yeah...AIX is a pretty crappy UNIX |
| 16:10.29 | salviadud | what was ibm thinking right? |
| 16:10.41 | mr_horsepower | theorem_: we have, i have here the patch, but never sended it. |
| 16:10.50 | theorem_ | dlynes_office - oops, I mistyped -- I meant IAX. |
| 16:10.51 | file | patch to do what? |
| 16:11.14 | file | (as you can tell I'm rather... bitter at the moment due to all of the complaining people have been doing the past 2 days) |
| 16:11.32 | theorem_ | file - take everything with a grain of salt. |
| 16:12.01 | dlynes_office | theorem_: more like a couple mickeys of vodka :) |
| 16:12.07 | file | theorem_: I usually do but when people complain and complain about something you work on every day, and don't give valid reasons... it's hard :) |
| 16:12.08 | mr_horsepower | file: send the from domain in a call, and use it. |
| 16:12.12 | theorem_ | it's common for people to see deficiencies, they're not always qualified to fix or even know wtf they are talking about ;-) |
| 16:12.30 | dlynes_office | theorem_: so then they can pay someone that can |
| 16:12.43 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-4-122.w81-248.abo.wanadoo.fr) |
| 16:13.11 | *** join/#asterisk ToTo (n=ToTo@host107-158.pool874.interbusiness.it) |
| 16:13.13 | salviadud | i was thinking about the narco market here in mexico |
| 16:13.18 | theorem_ | mr_horsepower - san you rrphrase ? you seem to be missing ome words in that sentence. |
| 16:13.24 | theorem_ | *can |
| 16:13.26 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 16:13.29 | theorem_ | *rephrase |
| 16:13.37 | theorem_ | jeeze, typing skills are horrendous. |
| 16:13.38 | dlynes_office | theorem_: no...he's not missing anything |
| 16:13.38 | salviadud | if i were to do a couple of asterisk installs on some drug dealers, i could get lots of dough... |
| 16:13.43 | dlynes_office | theorem_: i understood it just fine |
| 16:13.55 | salviadud | mexico is full of 'em criminals |
| 16:13.58 | file | this is not helping me clean my desk and stuff, dang nabbit |
| 16:14.02 | theorem_ | oh.. ... |
| 16:14.38 | mr_horsepower | asteriskA, user1 calls user2@asteriskB, the from call, will be seen as user1@asteriskB. |
| 16:14.40 | dlynes_office | theorem_: however, I'm not so sure that's an asterisk deficiency as it is a hardphone deficiency |
| 16:14.50 | dlynes_office | mr_horsepower: ah...that one |
| 16:15.09 | mr_horsepower | y |
| 16:15.16 | dlynes_office | mr_horsepower: where someone logs into sip with an extension name, but you don't know if it's local to domain a or domain b |
| 16:15.35 | file | and just fyi, chan_sip is going to be rewritten... |
| 16:15.48 | dlynes_office | mr_horsepower: and so 221@domainA logs in, and then 221@domainb logs in and kicks out 221@domaina |
| 16:15.58 | theorem_ | hmm |
| 16:16.01 | theorem_ | that's not so good ! |
| 16:16.22 | dlynes_office | but, sip is damned complicated |
| 16:16.32 | dlynes_office | I have no desire to look at that code, myself |
| 16:16.32 | mr_horsepower | dlynes_office: yes, because you dont have full domain suport in asterisk. |
| 16:16.51 | file | we don't claim to have full domain support :) |
| 16:16.54 | theorem_ | mr_horsepower - you mentioned you had a patch for that support ? |
| 16:17.15 | dlynes_office | mr_horsepower: i don't know if you noticed or not, but asterisk has never claimed to be 100% SIP compliant, either |
| 16:17.21 | mr_horsepower | theorem_: yes i have, but we dont implement multiple domains. |
| 16:17.33 | file | the motto for chan_sip is be lenient in what we accept, and strict in what we send |
| 16:17.44 | file | so in order to achieve maximum compatibility and interoperability - we break some rules |
| 16:17.47 | mr_horsepower | dlynes_office: yes i know, but to be usefull for us, it has to do some things. |
| 16:17.47 | dlynes_office | mr_horsepower: then wherein lies the problem that asterisk can't support multiple domains? |
| 16:17.52 | mr_horsepower | that dont, yet. |
| 16:18.15 | theorem_ | file - is it worthwhile for mr_horsepower to submit his patch to support multiple domains in SIP to you guys ? |
| 16:18.22 | dlynes_office | seems kinda silly to point out deficiencies that don't even affect you |
| 16:18.37 | file | theorem_: yes, but it won't get looked at immediately... |
| 16:18.47 | file | we're approaching the time of total freeze and only bug fixes |
| 16:18.56 | theorem_ | ok, makes sense. |
| 16:19.09 | mr_horsepower | multiple domains |
| 16:19.16 | mr_horsepower | should be the great in asterisk |
| 16:19.23 | dlynes_office | file: so digium is going to do the "right" thing on 1.4, then? |
| 16:19.36 | theorem_ | mr_horsepower - if you'd like to contribute I am sure everyone will welcome your addition. file - how is best for him to submit his patch ? |
| 16:19.37 | *** join/#asterisk killfill (n=killfill@pc-200-74-99-214.asturias2.pc.metropolis-inter.com) |
| 16:19.42 | killfill | hey what does this mean? |
| 16:19.45 | file | http://bugs.digium.com/ |
| 16:19.45 | killfill | 9 12:19:33 WARNING[626] chan_zap.c: Call specified, but not found? |
| 16:19.47 | killfill | May 29 12:19:33 WARNING[626] chan_zap.c: Unable to move channel 3! |
| 16:19.49 | file | needs to be disclaimed |
| 16:19.55 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 16:19.57 | dlynes_office | file: I just remember a while back, it seems new features were being added up until 1 or 2 days before a release or something like that |
| 16:20.01 | *** join/#asterisk cypromis (n=michal@voiceworks.pl) |
| 16:20.07 | file | dlynes_office: ah... we have a schedule now |
| 16:20.33 | ManxPower | Some days my job really sucks |
| 16:20.34 | mr_horsepower | file: are you thinking in adding multiple domains suport in chan_sip in 1.4? |
| 16:20.40 | dlynes_office | file: so iow, new releases (1.4 and higher) should be much more stable, then |
| 16:20.44 | mr_horsepower | you really should do it :D |
| 16:20.49 | file | mr_horsepower: no it's too late |
| 16:21.02 | file | well... |
| 16:21.02 | mr_horsepower | file: what's new in sip? |
| 16:21.13 | file | it would be very difficult, and I don't have the exact schedule in front of me |
| 16:21.19 | dlynes_office | file: btw, are you sick lately? |
| 16:21.20 | file | look at the commit list... I'm not a walking feature list :) |
| 16:21.25 | *** join/#asterisk javaTard (n=javaTard@cpe-69-207-34-244.twcny.res.rr.com) |
| 16:21.30 | file | dlynes_office: no |
| 16:21.42 | killfill | anyone happend to run asterisk on freebsd?.. its terrible unstable in here.. :-S |
| 16:21.46 | theorem_ | file - you're never going to get your errands done :) |
| 16:21.47 | dlynes_office | file: I've actually seen you say more than two things a day lately |
| 16:22.00 | file | oh, in here? |
| 16:22.01 | salviadud | i think mr_horspower should be heard |
| 16:22.10 | mr_horsepower | file: where is the commit list for chan_sip? |
| 16:22.18 | theorem_ | check the wiki mr_horsepower |
| 16:22.21 | dlynes_office | file: in any of the asterisk chat channels :) |
| 16:22.36 | file | mr_horsepower: http://lists.digium.com/pipermail/asterisk-commits/ |
| 16:22.41 | mr_horsepower | file: tks. |
| 16:22.44 | file | dlynes_home: ah I'm usually active in dev |
| 16:23.11 | file | it takes more thought to contribute and talk in here, because this channel is for people needing help |
| 16:23.31 | theorem_ | and hte bleeding newbies ;-) |
| 16:23.37 | mr_horsepower | we have sent a patch for rawplayer too. |
| 16:23.52 | mr_horsepower | i hope 1.4 have it, i'm tyred to patch! :P |
| 16:23.55 | theorem_ | mr_horsepower - thanks for the contribution ! |
| 16:24.15 | file | I've lost track of where I was cleaning... |
| 16:24.31 | RoyK | zoa: ping |
| 16:25.03 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 16:25.04 | killfill | file: you ever seen this? chan_zap.c: Call specified, but not found? chan_zap.c: Unable to move channel 1! |
| 16:25.04 | mr_horsepower | i send another week, one email, because i dont understand why "." dots are taked of the sip url, have anyone seen the email? no one awnsered me. |
| 16:25.09 | mr_horsepower | theorem_: np |
| 16:25.19 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 16:25.19 | file | killfill: I don't do zaptel... sorry |
| 16:25.35 | killfill | ok.. |
| 16:25.47 | mr_horsepower | cristian from beronet have some god code that should be included in 1.4 too. |
| 16:25.58 | mr_horsepower | he has no time to propose it. |
| 16:26.17 | theorem_ | well, again as file said, deadlines are fast approaching |
| 16:26.24 | mr_horsepower | yes |
| 16:26.37 | file | see this is one of the difficult things - people are happy when we have this deadline and schedule thing, and people are unhappy |
| 16:26.39 | JASON99 | Hello, if two sip phones are calling each other, how would I make the RTP packets go direct instead of going through the asterisk server? Is this possible? |
| 16:26.53 | mr_horsepower | i think most of the ppl that DO some stuff, dont have time to send it |
| 16:27.05 | theorem_ | file - deadlines are good, they keep you on track and focused |
| 16:27.23 | mr_horsepower | yes, deadlines are god |
| 16:27.24 | theorem_ | without it , you'd be making some messy code that would quickly beomce unmaintainable (imho ) |
| 16:27.43 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 16:28.19 | file | http://www.asterisk.org/developers/releasecycle |
| 16:28.34 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 16:28.40 | theorem_ | JASON99 - you would lose the benefit of having asterisk as a middleman --- what are you trying to acheive ? |
| 16:29.13 | file | 2 more days and we hit month 6 |
| 16:29.32 | InfraRed | you're pregnant? |
| 16:29.38 | theorem_ | lol |
| 16:29.42 | InfraRed | :) |
| 16:29.43 | file | last I checked... no |
| 16:29.49 | theorem_ | yes, with an asterisk baby :) |
| 16:30.13 | InfraRed | what about now? |
| 16:30.14 | JASON99 | theorem_: I'm trying to save resources on the system and bandwidth for local calls. If I have 100 calls coming to my server when they could stay local, thats a waste of server resources and bandwidth.. |
| 16:30.16 | theorem_ | it called him w/ SIP over WiFI last night. |
| 16:30.23 | file | eep |
| 16:30.25 | theorem_ | JASON99 - true ... |
| 16:30.46 | theorem_ | JASON99 - you'd need to open up a fresh SIP connection between the phones .. |
| 16:30.55 | theorem_ | what you're discussing is a true P2P approach. |
| 16:31.18 | InfraRed | JASON99: look at sip reinvite |
| 16:31.19 | Ahrimanes | InfraRed: 2 words.. hamster and ducttape |
| 16:31.22 | theorem_ | asterisk then would in fact act like a bittorrent tracker for hte calls. |
| 16:31.33 | JASON99 | theorem_: but asterisk should be able to make the connection point to point instead of taking on all the rtp sessions |
| 16:31.51 | *** join/#asterisk ToyMan (n=stuq@adsl-71-158-156-177.dsl.applwi.sbcglobal.net) |
| 16:31.55 | JASON99 | InfraRed: ok, I will look at that |
| 16:31.56 | mr_horsepower | file: chan_sip have been re-written? |
| 16:32.04 | file | mr_horsepower: not yet, I said for the next version |
| 16:32.05 | theorem_ | It's an intriguing idea .. follow InfraRed's idea ... I have never tried before. |
| 16:32.12 | mr_horsepower | hooo next version, ok. |
| 16:32.20 | JASON99 | theorem_: Thanks |
| 16:32.26 | mr_horsepower | sorry i'm a litle busy dont follow... |
| 16:33.10 | file | okay I'm cleaning, yup... cleaning |
| 16:33.31 | InfraRed | JASON99: it's seperating the media from the control on sip |
| 16:33.51 | InfraRed | wont work with nat |
| 16:34.51 | mr_horsepower | file: if i want to submit the patch, i have to patch for the svn version or to the stable version? |
| 16:35.09 | file | trunk |
| 16:35.34 | file | new features = trunk, bug fixes = trunk or 1.2... depends sorta thing |
| 16:36.11 | JASON99 | InfraRed: I plan on leaving all nat go through the server but anything public wont. We do lots of transfers to public IPs and if we dont have to go through the server we would rather not. I think you are strearing me in the right direction.. thanks for the help.. |
| 16:36.19 | [TK]D-Fender | file : What the rought target month/year for 1.6? |
| 16:36.42 | file | start of next year I _think_ |
| 16:37.47 | *** join/#asterisk suma (n=suma@222.165.116.228) |
| 16:37.49 | suma | hi |
| 16:38.05 | suma | My echo test is not working |
| 16:38.15 | suma | can anyone please help me how to solve it ? |
| 16:38.18 | file | I found a collection of floppy disks... |
| 16:38.20 | file | do I dare throw them out |
| 16:39.48 | [TK]D-Fender | file : Thats the kind of answer I was looking for. |
| 16:40.23 | *** part/#asterisk assert_true (n=anil@59.176.43.38) |
| 16:40.44 | theorem_ | file - they're antiquated now |
| 16:40.58 | theorem_ | I noticed that qualiy of floppy disks have seriously degraded over the years |
| 16:41.25 | theorem_ | my floppy from 12 years ago is still good and has gotten a lot of use, but ones that are < 2 years old are f* beyond repair. |
| 16:41.35 | file | yuck, a DVD for Fedora Core 4 |
| 16:41.47 | theorem_ | the microwave is prettier |
| 16:41.50 | distortion | keep your floppies to yourself |
| 16:41.53 | theorem_ | *microwave trick |
| 16:42.53 | distortion | suma: are you first using "Answer()" then "Echo()"? |
| 16:44.10 | suma | distortion: It works fine with iax |
| 16:44.14 | suma | or soft channels |
| 16:44.21 | suma | It is not working with zaptel |
| 16:44.25 | *** join/#asterisk jpbotelho (n=jpbotelh@201.7.108.130) |
| 16:44.28 | suma | i have x100p installed |
| 16:45.40 | InfraRed | wildcard |
| 16:45.41 | distortion | suma: sorry, i am not familiar with zaptel. The concept should be the same if you have it working on iax. |
| 16:45.45 | InfraRed | you make my heart sing |
| 16:45.49 | InfraRed | you make everything |
| 16:45.53 | InfraRed | except phonecalls |
| 16:45.54 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 16:45.56 | InfraRed | wildcard |
| 16:46.24 | boch | is g723 codec supported by asterisk ? |
| 16:46.25 | suma | yes, distortion, not sure why that is not working |
| 16:46.40 | suma | boch: you need get license for using g723.1 |
| 16:46.54 | distortion | boch: yes, but you need a decoder if you wish to use anything other than passthrough |
| 16:47.27 | boch | suma: even if g723.1 passthrough ? |
| 16:47.40 | suma | passthrough you can use any codec |
| 16:47.44 | distortion | boch: so, ua g723 -> * -> g723 to ? will work. But if you use tdm hardware or need to transcode you will need a license |
| 16:47.49 | suma | not restricted to g723 |
| 16:48.03 | suma | asterisk has nothing to do with passthrough |
| 16:48.43 | distortion | well, it still proxies the rtp packets in passthrough, it just doesnt decode/encode them |
| 16:49.01 | suma | yep |
| 16:50.05 | JASON99 | That's good to know.. :P |
| 16:50.20 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 16:51.48 | [TK]D-Fender | distortion : THAT would be nice... leaving * the final choice to negotiate again for transcoding if no compatible match found.... |
| 16:52.03 | [TK]D-Fender | distortion : So "bridge if necessary only" |
| 16:53.26 | distortion | exactly! it looks like its getting close, the patch on bug id: 4825 seems like people have gotten it to kinda work. |
| 16:55.17 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:55.37 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:56.10 | *** join/#asterisk saftsack (n=saftsack@p54A7E16C.dip.t-dialin.net) |
| 16:56.27 | distortion | hopefully it wont be long to have that added to the main trunk. |
| 16:57.06 | distortion | cause it would be nice to have passthrough codec negotiation and then be able to use other patches ie: t38 |
| 16:57.28 | *** join/#asterisk _omer (i=_omer@203.215.180.247) |
| 16:57.34 | _omer | Hi, |
| 16:57.39 | boch | distortion: very clear, thanks |
| 16:57.41 | distortion | one can only dream i guess |
| 16:57.55 | _omer | what should I type to install "asterisk-perl-0.08" |
| 16:57.58 | _omer | ?? |
| 16:58.12 | _omer | make, make install ..dont work |
| 16:58.34 | distortion | look at the install readme, make sure you have the dependencies installed |
| 16:58.39 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 16:58.57 | *** join/#asterisk eluizbr (n=eluizbr@200.251.32.8) |
| 16:59.03 | _omer | okey.. |
| 16:59.16 | distortion | if you use a redhat style os, and have yum installed, you will need to type "yum install XXX" where xxx is the missing package |
| 16:59.52 | distortion | you can list the installed packages with "rpm -qa |grep XXX" to check against the packages listed in the install readme |
| 17:00.45 | eluizbr | hi, |
| 17:00.46 | eluizbr | how I make to improve the quality of voice in operators voip? |
| 17:01.25 | eluizbr | my linkings are generating many noises |
| 17:01.37 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
| 17:04.02 | *** join/#asterisk justnulling2 (i=justnull@ool-182e45b6.dyn.optonline.net) |
| 17:04.13 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 17:08.22 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 17:08.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 17:09.00 | Dr-Linux | how can i add a member to a queue from CLI? i saw example for didn't understand. any help |
| 17:09.18 | Dr-Linux | LHR-PBX*CLI> add queue member sip support 4092 |
| 17:09.19 | Dr-Linux | Usage: add queue member <channel> to <queue> [penalty <penalty>] |
| 17:09.21 | *** join/#asterisk fugitivo (n=ajf@190.48.167.142) |
| 17:09.28 | fugitivo | hi |
| 17:10.34 | eluizbr | o fugitivo |
| 17:10.36 | [TK]D-Fender | Dr-Linux : SIP is not a channel, its a technology. SIP/100 is a channel. |
| 17:11.04 | [TK]D-Fender | Dr-Linux : And you missed the "to" in your line. |
| 17:11.36 | eluizbr | how I make to improve the quality of voice in operators voip? my linkings are generating many noises |
| 17:12.23 | *** join/#asterisk redondos (n=redondos@190.48.58.11) |
| 17:12.31 | [TK]D-Fender | eluizbr : Please don't keep spamming the same question. I fsomeone knows they will answer you, but I will say that your question is very hard to understand. |
| 17:12.44 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 17:12.48 | fnordian | are their different trunk-repositories for asterisk? |
| 17:13.17 | Dr-Linux | [TK]D-Fender: thanks, its done |
| 17:13.18 | Dr-Linux | LHR-PBX*CLI> add queue member sip/4092 to support |
| 17:13.18 | Dr-Linux | Added interface 'sip/4092' to queue 'support' |
| 17:13.29 | [TK]D-Fender | Dr-Linux : Much better.... |
| 17:13.36 | Dr-Linux | [TK]D-Fender: what's [penalty <penalty>] ? |
| 17:13.43 | fugitivo | skills |
| 17:14.07 | fugitivo | with an inverse meaning :) |
| 17:14.15 | Dr-Linux | [TK]D-Fender: also i wanna know, if it's permanent added or i need to save/reload queue module? |
| 17:14.16 | fnordian | a thread in the bugtracker references rev 16473, but the trunk i checked out is at r30744 |
| 17:14.22 | [TK]D-Fender | Dr-Linux : If 2 agents are available and could be chosen equally this lets you say that one person is better than another for that call. |
| 17:14.25 | Dr-Linux | fugitivo ? |
| 17:14.32 | [TK]D-Fender | Dr-Linux : Temporary |
| 17:14.43 | *** join/#asterisk pollo (n=a@87.219.128.65) |
| 17:14.48 | fugitivo | Dr-Linux: it's called "skills" on a tradicional pbx |
| 17:14.57 | eluizbr | as I make to improve the quality of the sound in linkings SIP |
| 17:15.08 | pollo | hi |
| 17:15.20 | [TK]D-Fender | eluizbr : Sorry, the language barrier is pretty big here.... |
| 17:15.40 | Dr-Linux | [TK]D-Fender: is there any way that i can save it permanently? |
| 17:16.20 | Dr-Linux | [TK]D-Fender: as you said "Temporary" , it's mean it will be remove after next reload? |
| 17:16.28 | eluizbr | as I can improve the quality of voice in a canal SIP |
| 17:16.35 | fugitivo | Dr-Linux: it's like dfender said, for example you have two queues, one for sales, and another one for support, you have agent A in support and you'd like agent A to answer calls for queue sales if all agents are busy, you add agent A in queue sale with a penalty of 1 for example |
| 17:17.25 | Dr-Linux | fugitivo: cool, i understand |
| 17:19.11 | *** part/#asterisk eluizbr (n=eluizbr@200.251.32.8) |
| 17:19.34 | [TK]D-Fender | What a poor poor sap.... |
| 17:20.38 | *** join/#asterisk coolhp (n=crap@modemcable240.139-203-24.mc.videotron.ca) |
| 17:20.46 | coolhp | Good day everyone. |
| 17:20.54 | pollo | hi , have de these error on asterisk y can call to my internal sip extensions it return me chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call when i want to do a call , my error and config files are there http://pastebin.com/745115 |
| 17:20.56 | Dr-Linux | fugitivo: have you any clue about my last question |
| 17:21.18 | coolhp | Would any of you happen to have ever used an AdTran TA750 channel bank with asterisk ? |
| 17:21.19 | Dr-Linux | how long it will work if i add member frrom the CLI, till next reload? |
| 17:21.32 | Dr-Linux | if yes then how can i save it permanently? |
| 17:22.12 | CunningPike | Dr-Linux: You can specify queue agents in agents.conf |
| 17:22.37 | [TK]D-Fender | Dr-Linux : run time changes are temporary.. you want it permanent, modify the config file. |
| 17:22.43 | CunningPike | Dr-Linux: I don't think a reload resets the status of dynamic agents - at least I sure hope not :) |
| 17:23.06 | Dr-Linux | hhm.. |
| 17:23.41 | Dr-Linux | well, i can do easily from configs, but we are creating an user base application that that we need this. |
| 17:24.22 | Dr-Linux | hhm. |
| 17:24.37 | Dr-Linux | CunningPike: then how long it will work? |
| 17:24.52 | Dr-Linux | CunningPike: untill restat the asterisk? |
| 17:24.55 | CunningPike | Dr-Linux: Until the next restart, I should imagine |
| 17:25.22 | Dr-Linux | hhmm... |
| 17:25.48 | CunningPike | Dr-Linux: But you seem to be wanting a permanent agent, without using agents.conf....... not sure why? |
| 17:25.57 | fnordian | can anybody tell me, where the changes from http://bugs.digium.com/view.php?id=5374 got merged to? |
| 17:26.10 | CunningPike | fnordian: Maybe ask on -dev |
| 17:26.21 | Dr-Linux | CunningPike: forget about agents.conf, it's queues.conf game though |
| 17:26.32 | fnordian | CunningPike: thx, i will try that |
| 17:26.48 | CunningPike | fnordian: That's where all the propellor-heads hang out ;) |
| 17:26.53 | Dr-Linux | CunningPike: we are developing an user web based applications. we need for that |
| 17:27.25 | fnordian | CunningPike: here i come ;-) |
| 17:27.53 | kay2 | when My asterisk is connected to an other one and I place a call, I can only hear one word out of two |
| 17:28.07 | CunningPike | Dr-Linux: Not sure what you need then..... |
| 17:28.23 | kay2 | could someone tell me what could be the reason ? |
| 17:29.08 | CunningPike | kay2: Lots of reasons - need more information. How are the two servers connected> |
| 17:29.11 | CunningPike | ? |
| 17:29.19 | Dr-Linux | CunningPike: i'm not sure how can you add an agents from CLI , i can't see any command with add an agent |
| 17:29.42 | kay2 | CunningPike: using dsl |
| 17:29.55 | CunningPike | Dr-Linux: Oh, I see - I understand now - sorry |
| 17:29.58 | kay2 | none of them is behind a nat |
| 17:30.04 | CunningPike | kay2: IAX? |
| 17:30.09 | Ahrimanes | Dr-Linux: add queue member SIP/123456 to <queue> |
| 17:30.26 | kay2 | CunningPike: yeah |
| 17:30.28 | CunningPike | Dr-Linux: What he said |
| 17:30.36 | CunningPike | kay2: Are you trunking? |
| 17:30.48 | kay2 | CunningPike: no |
| 17:30.56 | Dr-Linux | Ahrimanes: yes, i got that already, but i wanna save it permanetly. |
| 17:30.57 | dlynes_office | fnordian: if anything, it's getting posted to 1.4 |
| 17:31.09 | Dr-Linux | CunningPike: he is adding memeber, but not agent |
| 17:31.21 | Ahrimanes | Dr-Linux: set persistentmembers=yes in queues.conf ? |
| 17:31.27 | CunningPike | kay2: You should try trunking and play around with your jitterbuffer |
| 17:31.35 | kay2 | CunningPike: how do I do that ? |
| 17:31.44 | Dr-Linux | Ahrimanes: hhm.. what that will do? |
| 17:31.47 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 17:31.53 | kay2 | CunningPike: trunk=yes ? |
| 17:31.58 | CunningPike | kay2: Yes |
| 17:32.05 | dlynes_office | fnordian: it's in trunk |
| 17:32.06 | kay2 | what would that change ? |
| 17:32.20 | kay2 | CunningPike: it's only on one single call |
| 17:32.23 | Ahrimanes | Dr-Linux: it saves the members add with add queue member in astdb.. thus those added will stay in the queue across crashes/restarts |
| 17:32.25 | dlynes_office | fnordian: Check out rev 16473 from trunk and you'll have it, or just grab the latest trunk |
| 17:32.54 | CunningPike | kay2: Then it won't help :) Experiment with different jitterbuffer settings, but also measure the latency between the two servers |
| 17:33.04 | fugitivo | Dr-Linux: what question? |
| 17:33.12 | CunningPike | kay2: What codec are you using |
| 17:33.27 | kay2 | CunningPike: alaw |
| 17:33.36 | kay2 | CunningPike: what should I put for the jitterbuffer |
| 17:33.47 | kay2 | CunningPike: to start |
| 17:33.54 | CunningPike | kay2: Ah - try a smaller codec - try gsm and see if it helps first off |
| 17:34.01 | *** join/#asterisk ramo (n=ramo@59.92.167.158) |
| 17:34.04 | kay2 | CunningPike: already tried |
| 17:34.05 | kay2 | same |
| 17:34.06 | Dr-Linux | Ahrimanes: after restarting asterisk, that memeber will be still in the queue? |
| 17:34.10 | Ahrimanes | Dr-Linux: yes |
| 17:34.13 | kay2 | CunningPike: it's not a bw issue |
| 17:34.20 | kay2 | it's really like a buffer issue |
| 17:34.27 | CunningPike | kay2: Sounds like a latency issue then - bandwidth and latency aren't the same....... |
| 17:34.33 | Ahrimanes | Dr-Linux: i use it to add/remove queue members via agi |
| 17:34.45 | kay2 | CunningPike: and what should I put for jitterbuffer |
| 17:34.46 | kay2 | to try |
| 17:34.53 | Dr-Linux | persistentmembers - if this option is set to yes, it will cause the system to store each dynamically logged in agent, from each separate queue, in the Asterisk`s database. In this way, in case of restarting the Asterisk PBX, the agents will be automatically readded into their recorded queues. By default the option is set to yes. |
| 17:35.04 | CunningPike | kay2: Not sure.......... experiment :) |
| 17:35.22 | kay2 | CunningPike: I start with 20 or 200 or 20000 ? |
| 17:35.30 | Ahrimanes | Dr-Linux: there's a slight bug, it seems to be set default to no.. but setting it in queues.conf works |
| 17:35.31 | Dr-Linux | Ahrimanes: what agi script you use |
| 17:35.36 | Dr-Linux | Ahrimanes: can i see it? |
| 17:35.57 | Ahrimanes | Dr-Linux: i made one myself, it just toggles an extension in and out of the queue |
| 17:36.39 | Dr-Linux | i see |
| 17:38.40 | CunningPike | kay2: Try =yes to start |
| 17:40.08 | De_Mon | what in the hell> |
| 17:40.16 | De_Mon | where did the agent function go? |
| 17:41.25 | dlynes_office | it got busted by the CIA |
| 17:41.35 | *** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 17:42.04 | De_Mon | show function agent use to exist, now it doesnt.. chan_agent is the only thingI see that would include it |
| 17:44.12 | Dr-Linux | Ahrimanes: May 29 22:56:09 WARNING[17206]: app_queue.c:714 queue_set_param: Unknown keyword in queue 'support': persistentmembers at line 20 of queues.conf |
| 17:45.15 | Dr-Linux | sorry, it's already there |
| 17:45.19 | Ahrimanes | Dr-Linux: hehe ok |
| 17:45.33 | Ahrimanes | Dr-Linux: it should be in [general] not in a specific queue i believe |
| 17:47.42 | Dr-Linux | Ahrimanes: yeah,i just saw, it's already there |
| 17:48.28 | Dr-Linux | Ahrimanes: but now sure, if restart the asterisk, this extension will be still in the queue or not :S |
| 17:49.53 | justnulling2 | any cisco7960 gurus here? is there a way to use alternative tftp server to load the data files instead of a locale one without touch the phone (only through config files)? |
| 17:52.28 | ManxPower | justinu, you can specify that info in the DHCP options |
| 17:55.45 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 17:56.08 | justnulling2 | manxpower: the idea is that the phone will be located off side and there for i will not have access to local dhcp config options |
| 17:56.09 | hacked`` | guys, you know voip providers, i just emailed one to ask if they support asterisk, and they said they do but they have no documentation on their site, what info do i need from them to set up asterisk? |
| 17:56.33 | ManxPower | justinu, configure it before you send the phone to the end site. |
| 17:56.39 | ManxPower | that is your only other option |
| 17:57.27 | justnulling2 | manxpower: that i will do but in case there are changes i wanted it to auto update from my tftp server, oh well |
| 17:57.59 | ManxPower | justnulling2, I'm sure the cisco docs talk about that sort of stuff |
| 18:01.50 | justnulling2 | manxpower: it is not listed as config files param in here http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a00801d1977.html so wanted to know if there is some hidden feature or something |
| 18:03.41 | robl^ | justnulling2: sorry, I missed the start of the conversation. what are you trying to do? I used to have a bunch of Cisco 7960s here |
| 18:05.14 | justnulling2 | bobl^: the idea is to use alternative tftp server automagicly without going into the phone settings options but from a config file |
| 18:05.37 | justnulling2 | rolb^ see up |
| 18:06.25 | justnulling2 | rolb^ the point is so that the phone will be off site and still be able to be auto update |
| 18:09.17 | robl^ | ohh... you have to set the tftp server manually OR via dhcpd. if I remember. I had a remote extension.. I had to set the tftp before I shipped the phone offsite. then it would read the config file |
| 18:09.53 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 18:10.09 | mpruett | Hello Everyone!!! |
| 18:10.59 | mpruett | I believe I have an easy one for you guru's |
| 18:11.16 | robl^ | justnulling2: it only has to be set one time.. them it will remember the setting and re-read the configuration and ringers off the remote server when the phone reboots |
| 18:12.33 | CunningPike | mpruett: Well, don't keep us in suspense....... |
| 18:12.42 | justnulling2 | robl^ what is this remote extension and how do i set it to re-read config files from remote server? |
| 18:13.21 | robl^ | remote extension == phone placed offsite |
| 18:13.47 | mpruett | I am using MYSQL() to get some info out of a DB. I set a variable with my Fetch statement. I do get a value for the variable upon fetching, but the variable is empty if I do the Clear & Disconnect. |
| 18:14.46 | mpruett | I need to use that value to use in MeetMe. If I do the MeetMe before the CLear & DIsconnect I get the hung processes you don't want. |
| 18:14.58 | robl^ | justnulling2: you have to set the tftp server's IP before you take the phone off site. how else would the phone know how to find the tftp server? |
| 18:17.53 | justnulling2 | robl^ i was using dyn_tftp_addr let me try it with primary_tftp_addr |
| 18:18.53 | mpruett | If I do the MeetMe after the CLear & Disconnect the variable is empty and I get a fast busy when I call MeetMe |
| 18:19.52 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:20.34 | flujan | hi all, Do someone have experience using the Unicall driver? |
| 18:21.18 | robl^ | justnulling2: its AlternateTFTP, and I don't think it can be set from a config files.. because it would need to know about the Alt TFTP before it can load the file. |
| 18:21.25 | *** join/#asterisk madd (n=madd@p15169043.pureserver.info) |
| 18:21.30 | madd | moin |
| 18:23.56 | Ahrimanes | Dr-Linux: after restart the extension will still be in the queue |
| 18:23.57 | [TK]D-Fender | plus |
| 18:24.30 | justnulling2 | robl^: well it saves the previous config file so it can use tftp seting from there? |
| 18:26.17 | Dr-Linux | Ahrimanes: okey, thanks |
| 18:26.35 | Ahrimanes | Dr-Linux: np |
| 18:28.40 | kay2 | someone has ever experianced Asterisk Realtime with queue ? |
| 18:29.52 | fugitivo | anyone knows if the motherboard Intel SE7320SP2 is compatible with digium cards_ |
| 18:30.38 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:34.24 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 18:36.23 | mpruett | Anybody know why my Varible in my fetch statement using MYSQL() is empty AFTER my CLear & Disconnect Statement? |
| 18:37.18 | mpruett | Anybody know how to use the value stored in the Variable AFTER my Clear & Disconnect Statements |
| 18:37.30 | *** join/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 18:37.52 | AJmn | Anyone using a PAP2 connected to there * box? |
| 18:38.08 | Strom_C | AJmn: i have one |
| 18:38.22 | *** join/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 18:38.38 | *** part/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 18:39.17 | *** join/#asterisk _Paulo_ (n=Paulo@c9064c64.virtua.com.br) |
| 18:42.32 | *** join/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 18:42.41 | AJmn | Storm_C Im having an issue with one that is at another office. It registers with * but if you try to call out you get nothing. and if u dial its # it rings but you dont the phone never rings. |
| 18:43.26 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:43.33 | AJmn | Storm_C oh i just noticed I have a status of : UNREACHABLE on it... Any ideas? its behind a linksys router, but i had the ports forwarded to it |
| 18:46.29 | InfraRed | AJmn: run sip debug on * |
| 18:46.39 | InfraRed | then log for later consumption |
| 18:48.01 | fugitivo | i hate hardware problems |
| 18:48.10 | Strom_C | AJmn: is the asterisk box also behind NAT>? |
| 18:48.18 | AJmn | what am i looking for?! im so lost to trouble shooting these issues. |
| 18:48.22 | InfraRed | i hate people with hardware problems |
| 18:48.28 | stephane_ | re |
| 18:48.31 | AJmn | NO |
| 18:48.40 | AJmn | InfraRed HEy now :P |
| 18:48.44 | InfraRed | AJmn: thats the whole point of debugging. good luck :) |
| 18:48.53 | InfraRed | log as you make a call |
| 18:49.01 | InfraRed | then read the log later with a nice cup of tea |
| 18:49.04 | blitzrage | AJmn: what does your topology look like? |
| 18:49.13 | blitzrage | AJmn: what are the symtoms? |
| 18:49.27 | Strom_C | AJmn: if it's listed as "UNREACHABLE" then perhaps you have your qualify setting too low. What does the qualify= line say in the PAP2's sip.conf entry? |
| 18:49.27 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 18:49.46 | blitzrage | just turn off qualify |
| 18:49.51 | Strom_C | or that too |
| 18:50.07 | blitzrage | holy crap its hot out there |
| 18:50.16 | blitzrage | the bike ride to volleyball tonight is going to be .... fun |
| 18:50.20 | *** join/#asterisk ToTo (n=ToTo@host105-142.pool878.interbusiness.it) |
| 18:50.22 | *** join/#asterisk kristalino (n=kristali@230.Red-83-32-123.dynamicIP.rima-tde.net) |
| 18:50.26 | blitzrage | only 20km one way :) |
| 18:50.46 | *** join/#asterisk assert_true (n=Sunil@59.176.43.38) |
| 18:51.34 | AJmn | sip.conf tried Qualify = YES and tried NO |
| 18:51.49 | [TK]D-Fender | blitzrage : Uphill, in snow 10' high.... BOTH WAYS. |
| 18:52.07 | *** part/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 18:52.47 | AJmn | im thinking it has to be something with the firewall/router on there end cause i have 2 other PAP2's running and they connect fine. also others using X-Lite and all connect |
| 18:53.44 | file | blitzrage: you will bike... AND YOU WILL LIKE IT |
| 18:54.29 | Ahrimanes | haha |
| 18:54.36 | fugitivo | great, i have this exact problem http://forums.digium.com/viewtopic.php?p=21722&sid=28a7a0baadf3a44e494d74c65902d602 |
| 18:55.12 | Qwell | fugitivo: call digium |
| 18:55.53 | fugitivo | Qwell: i did last week, but that guy says it's the board |
| 18:56.23 | fugitivo | well, all my tests points to that direction |
| 19:01.26 | *** join/#asterisk saftsack (n=saftsack@p54A7E16C.dip.t-dialin.net) |
| 19:06.57 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 19:07.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:08.05 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:08.59 | coolhp | Would anyone have any experience with Adtran Channel banks with FXS cards and Asterisk ? |
| 19:09.17 | *** join/#asterisk darby_t (i=darby_t@aaoz49.neoplus.adsl.tpnet.pl) |
| 19:09.24 | Qwell | coolhp: analog is analog is analog.. |
| 19:09.38 | Qwell | Just make sure that one side is fxo, and the other is fxs |
| 19:09.41 | Strom_C | coolhp: I've done an install with 48 FXS ports on two adtran channel banks |
| 19:09.51 | coolhp | I've got a TA750 connected to an asterisk server through a crossover cable... I've set the signaling to E&M.... is that wrong ? |
| 19:10.02 | Qwell | oh, pfft |
| 19:10.05 | coolhp | LOL |
| 19:10.06 | coolhp | Sorry. |
| 19:10.10 | coolhp | I'm really new at this. |
| 19:10.18 | coolhp | FXO on one side, FXS on the other. |
| 19:10.27 | Qwell | So it isn't t1 to asterisk? |
| 19:10.28 | InfraRed | ~fxofxs |
| 19:10.32 | jbot | well, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 19:10.43 | coolhp | It is a T1 to asterisk. |
| 19:11.14 | coolhp | T1 crossover between the 2 basically. |
| 19:11.44 | coolhp | Should I set zaptel.conf as fxols=1-24 ? |
| 19:11.55 | *** join/#asterisk fugitivo (n=ajf@190.48.166.204) |
| 19:11.59 | coolhp | I'm just confused :-P LOL |
| 19:12.21 | InfraRed | read the zaptel sample config |
| 19:12.26 | InfraRed | and voip-info.org |
| 19:12.28 | Strom_C | coolhp: yes, FXS ports use FXO signaling |
| 19:14.48 | *** join/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 19:15.50 | AJmn | OK! got this error ---- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from [IP of PAP2 device] |
| 19:15.55 | AJmn | What am i doing wrong? |
| 19:15.56 | coolhp | Read the zaptel config samples and doc on voip-info. |
| 19:16.08 | coolhp | So if I understand this well : |
| 19:16.11 | marl | hi, can someone tell me if i have the following wrong? .call files can be setup to call an internal extesnsion and when its answered then transfer the call to an outgoing line (eg. only make the external call if the internal extesnion has been asnwered)? |
| 19:16.22 | marl | as all the examples i have seen so far (and the documentation ive read) implies that .call files are normally setup to dial the external number first and THEN the internal extenion |
| 19:16.31 | *** join/#asterisk sb_mx (n=sb_mx@200.94.154.226) |
| 19:16.44 | coolhp | Adtran TA750 <-- T1 Crossover --> Asterisk (TE205P) should be configured as |
| 19:16.58 | coolhp | zaptel.conf -> fxoks=1-24 |
| 19:17.09 | coolhp | zapata.conf -> signaling = fxo_ks |
| 19:17.18 | coolhp | Is that correct ? |
| 19:17.23 | Strom_C | yes |
| 19:17.31 | coolhp | Testing :-)... |
| 19:17.59 | mpruett | Anyone use MYSQL() much? |
| 19:18.55 | CoffeeIV_ | Is there a way I can increment a variable in a dialplan |
| 19:19.15 | Strom_C | use the math function? |
| 19:19.26 | CoffeeIV_ | so that I can ask a user to make a choice 3 times, and hangup after they don't do it the last time |
| 19:19.51 | CoffeeIV_ | show application math gave me nothing -- has it been replaced by something ? my * is a month or so old, out of CVS |
| 19:19.59 | Strom_C | show function math |
| 19:20.16 | CoffeeIV_ | I tried that too, nothing |
| 19:20.24 | AJmn | Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from [IP of PAP2 device] What am i doing wrong? |
| 19:21.38 | Strom_C | CoffeeIV_: http://pastebin.ca/59639 |
| 19:21.42 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 19:21.58 | [TK]D-Fender | CoffeeIV_ : Set(number=$[${number}+1]) |
| 19:22.37 | CoffeeIV_ | thanks |
| 19:27.34 | mpruett | Help Please!!!!! |
| 19:28.24 | mpruett | Does anyone use MYSQL much? |
| 19:28.34 | mpruett | MYSQL() |
| 19:28.45 | Strom_C | mpruett: ask an actual specific question, and if anyone knows the answer, they'll help you. |
| 19:29.26 | mpruett | I have a couple times just didn't want to keep typing the same thing over and over - Here it is |
| 19:30.05 | mpruett | Anybody know why my Varible in my fetch statement using MYSQL() is empty AFTER my CLear & Disconnect Statement? |
| 19:30.17 | Juggie | Clear? |
| 19:30.17 | mpruett | Anybody know how to use the value stored in the Variable AFTER my Clear & Disconnect Statements |
| 19:30.24 | Juggie | well... |
| 19:30.30 | Juggie | how about creating another variable |
| 19:30.34 | Juggie | and assinging the value to that. |
| 19:31.08 | [TK]D-Fender | mpruett : Pastebin your entire non-functional sample |
| 19:31.11 | [TK]D-Fender | ~pb |
| 19:31.13 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:31.23 | mpruett | I tried that - It is empty after I issue the clear also - I set my new variable equal to the Variable in my fetch staement if that is what you mean |
| 19:31.33 | [TK]D-Fender | Juggie : I was thinking of suggesting the same following a better sample... |
| 19:31.36 | mpruett | OK - I will paste |
| 19:32.00 | Juggie | i am sure MYSQL() wont clear a copy you make of the information. |
| 19:32.08 | mpruett | This doesn't work: exten => 1111,4,MYSQL(Fetch fetchid ${resultid} CONRM) |
| 19:32.08 | mpruett | exten => 1111,5,MYSQL(Clear ${resultid}) |
| 19:32.08 | mpruett | exten => 1111,6,MYSQL(Disconnect ${connid}) |
| 19:32.08 | mpruett | exten => 1111,7,MeetMe(${CONRM},,12345) |
| 19:32.15 | [TK]D-Fender | mpruett : PASTEBIN! View the link! |
| 19:32.31 | Juggie | mpruett, use pastebin |
| 19:32.37 | Juggie | secondly, of course it doesnt work |
| 19:32.46 | Juggie | CONRM is a pointer fo a record in a record set |
| 19:32.49 | Juggie | and you closed the record set |
| 19:32.51 | mpruett | oops sorry - just caught jbot post - my bad |
| 19:33.12 | Juggie | why would you expect the data to still exist after you close the record set. |
| 19:33.31 | [TK]D-Fender | mpruett : I suggest you follow Juggie's advise and set another variable to it before your disconnect.\ |
| 19:34.35 | mpruett | That;'s what I figured so I use something like this - Set(${Var1}=${CONRM}) before the clear and Var1 was emptied also |
| 19:35.10 | Juggie | http://pastebin.ca/59643 |
| 19:35.26 | mpruett | Maybe I screwed something up with my Set() let me try that again |
| 19:35.30 | Juggie | yuo did |
| 19:35.32 | Juggie | *you did |
| 19:35.41 | Juggie | you dont use the ${} in the first part of the set |
| 19:36.11 | [TK]D-Fender | mpruett : that'd be Set(Var1${CONRM}) |
| 19:36.17 | [TK]D-Fender | mpruett : that'd be Set(Var1=${CONRM}) |
| 19:36.29 | Juggie | yah, its right in the pastebin i jsut posted |
| 19:36.45 | mpruett | True - I don't think I did that but I will try it again - but I know I did not do it like Fender's post |
| 19:36.55 | mpruett | Thanks guys - Let me give that a try |
| 19:36.57 | Juggie | just copy what i put in the pastebin |
| 19:36.57 | iq | yo |
| 19:36.59 | Juggie | its right. |
| 19:37.33 | mpruett | New to this IRC - where is the "pastebin" - I am using mIRC? |
| 19:37.40 | distortion | mpruett, also try a NoOp(${confroom}) to see if you stored the variable |
| 19:37.45 | *** join/#asterisk iq|mobile (n=iq@71-215-34-237.omah.qwest.net) |
| 19:37.46 | Juggie | http://pastebin.ca/59643 |
| 19:37.48 | [TK]D-Fender | mpruett : Pastebin is a website |
| 19:37.50 | Juggie | copy/click that lnik. |
| 19:37.51 | Juggie | *link |
| 19:38.02 | Juggie | its for showing your code/etc |
| 19:38.04 | mpruett | gotcha - THanks again!!!! |
| 19:38.05 | Juggie | without floodding the channel |
| 19:38.45 | Juggie | wow my spelling is horrible today |
| 19:38.45 | zoa | you mean your speling is horible ? |
| 19:40.01 | [TK]D-Fender | zoa : no, that last one would be his typing ;) |
| 19:40.02 | Juggie | perhaps. |
| 19:40.07 | CoffeeIV_ | can you use > or < in the condition part of GotoIf ? |
| 19:40.14 | [TK]D-Fender | CoffeeIV_ : Yup |
| 19:40.21 | CoffeeIV_ | cool, thanks |
| 19:40.48 | [TK]D-Fender | CoffeeIV_ : GotoIf($[${number}>5]?10) |
| 19:40.51 | Juggie | zoa, its actually spelling though :) |
| 19:40.56 | Juggie | speling = not a word |
| 19:41.04 | [TK]D-Fender | Juggie : I almost got you off the hook on that one! |
| 19:41.28 | Juggie | i looked it up to be sure :) |
| 19:41.54 | Juggie | speling = like Tori Spelling the dirty hoe from beverly hills 90210 :) |
| 19:42.00 | Juggie | er, Tori Speling |
| 19:42.32 | distortion | mmm so dirty |
| 19:42.35 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 19:42.42 | [TK]D-Fender | Juggie : Hoe = dirty farm tool |
| 19:42.47 | boch | getlemen, i need your help, from a moment to another my asterisk is answering 'channel not available' to all incoming calls on zap/g1, do you know why or where can i start looking for the problem |
| 19:43.12 | Juggie | boch, www.pastebin.ca your output |
| 19:43.15 | Juggie | then link |
| 19:43.22 | Juggie | and then we'll look |
| 19:43.26 | boch | what output? pri debugs ? |
| 19:43.27 | [TK]D-Fender | boch : Pastebin your zapata.conf and zaptel.conf |
| 19:43.41 | Juggie | i'll be satisifed with the console output to start ;) |
| 19:43.54 | boch | ok, gimme a min |
| 19:44.14 | Juggie | be sure to do it on a verbose 11 |
| 19:44.17 | Juggie | so theres lots of detail |
| 19:44.19 | [hC] | any of you heard of a weird issue with a polycom ip501 where even when maxed out, the volume seems really low to the phone user? |
| 19:44.27 | [hC] | I havent been on site yet so this may be stupid user error. |
| 19:44.38 | Juggie | * doesnt touch rtp volume |
| 19:45.12 | [hC] | yeah i know, im more curious about a polycom issue itself |
| 19:45.18 | Juggie | dayton, i'm half finished that doc and have a couple of questions when you have an minute. |
| 19:45.37 | [hC] | ok. im just doing the monday morning fix-it-list |
| 19:45.43 | [hC] | once im done that i'll msg you on msn |
| 19:45.47 | Juggie | its afternoon :) |
| 19:45.55 | Juggie | k |
| 19:46.01 | [hC] | yeah, im STILL doin it :) |
| 19:46.07 | Juggie | i should just look at our conversation from friday |
| 19:46.13 | Juggie | probally has the answers |
| 19:46.38 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.121) |
| 19:46.48 | tomcontr3 | hi, does anyone here uses FAX with asterisk? |
| 19:47.10 | boch | [TK]D-Fender: Juggie here is my zapata.conf http://pastebin.ca/59645 |
| 19:47.27 | distortion | tomcontr3: yes, unfortunately |
| 19:47.57 | tomcontr3 | are you using spandsp? |
| 19:48.31 | sevard | tomcontr3: fax isn't an option, fo realz dog. |
| 19:48.43 | distortion | tomcontr3: i have done mainly passthrough g711, and testing t38, havent played with spandsp tho |
| 19:49.27 | *** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg) |
| 19:49.40 | tomcontr3 | does any one here knos a good wiki, of how to enalbe fax option with asterisk? |
| 19:50.09 | zoa | http://www.asteriskguru.com/tutorials/asterisk_fax.html |
| 19:50.15 | zoa | http://www.asteriskguru.com/tutorials/fax_pstn_passthru_tdm.html |
| 19:50.19 | zoa | http://www.asteriskguru.com/tutorials/spandsp.html |
| 19:50.24 | zoa | http://www.asteriskguru.com/tutorials/fax_passthrough_bri.html |
| 19:50.35 | boch | zoa: nice, thanks |
| 19:51.04 | sevard | now tinyurl them all, zoa. |
| 19:51.28 | zoa | you do so :p |
| 19:51.38 | zoa | they are not all very good |
| 19:51.42 | zoa | but they should help at least |
| 19:51.57 | Juggie | boch, show me your console output |
| 19:52.01 | Juggie | which contains the error |
| 19:52.04 | Juggie | this tells me nothing without that. |
| 19:52.22 | distortion | ~fax |
| 19:52.24 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 19:52.27 | sevard | i'm assuming the fax_passthrough_bri.html also applies to pri? |
| 19:52.30 | distortion | haha |
| 19:52.50 | sevard | wtf. |
| 19:52.54 | [TK]D-Fender | boch : You might want to start by removing the 95% commented out junk from there..\ |
| 19:54.05 | zoa | nopez not really |
| 19:54.07 | zoa | read it |
| 19:54.10 | zoa | its very small :) |
| 19:54.18 | sevard | I thought FAX Machine SIP -> * -> PRI was really not advised. |
| 19:54.31 | zoa | yes |
| 19:54.35 | zoa | but it might be better than nothing |
| 19:54.39 | Strom_C | sevard: it should be fine if you've got low enough latency |
| 19:54.47 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 19:54.49 | zoa | and actually works quite ok most of the time |
| 19:54.52 | zoa | with low latency links |
| 19:54.55 | zoa | like on your own network |
| 19:54.58 | Strom_C | 10ms packets, ulaw companding, 1ms latency to * box |
| 19:55.03 | zoa | perfect |
| 19:55.11 | zoa | it will work (not for v34 probably) |
| 19:55.19 | zoa | but up to 14k4 speeds |
| 19:56.01 | *** join/#asterisk lylix (n=eric@dynamic-acs-24-154-53-234.zoominternet.net) |
| 19:58.42 | mpruett | Juggie & Fender: Same result as I got last night - check out code and result at http://pastebin.ca/59648 |
| 19:59.58 | mpruett | I get fastbusy at Set(confroom=${CONRM}) |
| 20:00.01 | sevard | Strom_C: sup bitch |
| 20:00.14 | Strom_C | I don't mind you coming here and wasting all my time time |
| 20:01.41 | sevard | cause when you're standing so near |
| 20:02.40 | *** join/#asterisk ToTo (n=ToTo@host105-142.pool878.interbusiness.it) |
| 20:02.43 | sevard | i kind of lose my mind |
| 20:02.45 | sevard | YEahhhhhhhhhhhhhhhhhhhhhhhhhhhhh |
| 20:02.49 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 20:03.00 | Strom_C | I actually came up with phone-related lyrics to the first stanza of that song |
| 20:03.08 | sevard | you're lame. |
| 20:03.24 | *** join/#asterisk Hymie (i=hymie@L8R.net) |
| 20:03.57 | lylix | hi all... possible causes of sound files not playing, halts at sound-file w/o throwing errors... ? checked perms on /var/lib/asterisk/sounds... |
| 20:04.05 | Strom_C | I don't mind you coming here / And tying up my line / Cause when you're dialing oh so near / I kind of lose my mind / It's not the buttset that you wear / Your thousand feet of twisted pair / I don't mind you coming here / And tying up my line |
| 20:04.28 | sevard | Strom_C: wow. |
| 20:04.32 | sevard | No. |
| 20:04.42 | Strom_C | <-- dork |
| 20:04.50 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:04.57 | [TK]D-Fender | mpruett : You clearly did not apply your dialplan changes with RELOAD. |
| 20:05.11 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:05.12 | mpruett | No I did - Couple times |
| 20:05.19 | [TK]D-Fender | mpruett : Its executing a CODED line of - Executing Hangup("SIP/203-49b4", "") in new stack |
| 20:05.30 | [TK]D-Fender | mpruett : You muct not be looking at something properly. |
| 20:06.01 | [TK]D-Fender | mpruett : thats not a WARNING about a channel causing a disconnect, its a line in your dialplan being executed on purpose. |
| 20:06.30 | sevard | Strom_C: Interesting. |
| 20:07.04 | *** join/#asterisk jarek_z (i=foobar@e182255058.adsl.alicedsl.de) |
| 20:07.05 | sevard | Strom_C: I got a nice man at one of the interesting numbers. |
| 20:07.11 | Strom_C | oh? |
| 20:07.17 | sevard | He was very nice, but not a recording of Jane Barbe |
| 20:07.27 | *** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 20:07.31 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 20:07.32 | chino | how do i include another confif file ? |
| 20:07.34 | Hymie | May 29 16:01:06 WARNING[2315]: channel.c:2323 set_format: Unable to find a codec translation path from unknown to unknown |
| 20:07.35 | Hymie | <PROTECTED> |
| 20:07.37 | Hymie | er |
| 20:07.39 | mpruett | Fender: That happens whereever I reference the "Fetched" variable |
| 20:07.45 | Qwell | chino: #include |
| 20:08.00 | mpruett | Fender: btw - just did a reload and same result |
| 20:08.49 | [TK]D-Fender | mpruett : please pastebin your ENTIRE extensions.conf, not jsut a segment. |
| 20:09.17 | boch | Juggie: there is no output when incoming call arrives, maybe you want the pri signaling |
| 20:09.46 | nassy | how do i send a control alt del in apple's remote desktop |
| 20:10.10 | Qwell | nassy: completely offtopic... |
| 20:10.30 | nassy | oops wrong channel |
| 20:10.33 | jarek_z | hi! please help. which packages to install under debian unstale to get asterisk and HFC-USB to work together? (misdn driver recognizes billion usb TA) |
| 20:10.33 | nassy | sorry. |
| 20:10.41 | Qwell | but, there is a "Windows Security" thing in the start menu |
| 20:10.51 | sevard | Strom_C: the 'AIS' is 'dead air' |
| 20:11.14 | Strom_C | jarek_z: what the crap are you doing installing asterisk on debian unstable? |
| 20:11.38 | sevard | No no no, it's not unstable clearly it's unstale. |
| 20:11.48 | Qwell | pfft |
| 20:11.53 | Qwell | debian unstable is still 8 years old |
| 20:12.15 | Strom_C | quiet you |
| 20:12.35 | jarek_z | @Strom_C: why not ? |
| 20:12.52 | Strom_C | jarek_z: the goal of telephony is stability |
| 20:13.11 | Strom_C | I'd be uncomfortable doing a production asterisk system on Testing, much less Unstable |
| 20:13.13 | sevard | that's not the goal of asterisk |
| 20:13.22 | sevard | it's more like 'crash all the time because it's a neat toy' |
| 20:13.28 | sevard | poke poke |
| 20:13.30 | jarek_z | @Strom_C: so if I sty sarge it then works? |
| 20:13.40 | Strom_C | sevard: sure, if you're running CVS |
| 20:13.51 | Strom_C | jarek_z: Asterisk stable + Sarge stable == win |
| 20:14.13 | Qwell | asterisk trunk + gentoo == <3 |
| 20:14.23 | sevard | heh, i'm joking, but there really isn't an asterisk stable, unless you go pretty far back. it is +pretty stable+ though for a highly developement app |
| 20:14.28 | sevard | high dev* |
| 20:14.37 | jarek_z | @Strom_C: why cvs, with unstable I have 1.2.7.1 |
| 20:14.38 | Strom_C | sevard: stable == 1.2.7.1 release |
| 20:15.05 | [TK]D-Fender | sevard : * : Where failure is NOT an option.... it comes bundled with the software ;) |
| 20:15.50 | mpruett | fender: just got your last post and I will post entire plan. While I do that check out http://pastebin.ca/59654 - this works but obviously it hangs the process to mysql |
| 20:15.54 | sevard | I'm just glad XBMC is finally on a feature freeze |
| 20:16.02 | jarek_z | what other packages to install ? bristuff, chan-capi, chan-misdn, classic ? |
| 20:16.08 | sevard | that's my favorite project of all time and it's been needing said feature freeze forever |
| 20:16.45 | *** join/#asterisk jsaunders (i=JuanD@s142-179-93-180.bc.hsia.telus.net) |
| 20:17.23 | jsaunders | hey, what's the best single port fxo card for * ? (on the cheap range) |
| 20:17.54 | Strom_C | TDM400P with a single FXO module :) |
| 20:18.02 | sevard | Strom_C is a bastard. |
| 20:18.13 | Strom_C | how so? |
| 20:18.24 | jsaunders | Not a x100p? |
| 20:18.32 | chino | what is the domain or realm ? |
| 20:18.33 | sevard | You're a bastard. |
| 20:18.35 | Qwell | jsaunders: Do you want a card that actually works? |
| 20:18.38 | sevard | there's nothing else to it. |
| 20:18.39 | jsaunders | heheh |
| 20:18.41 | jsaunders | point taken |
| 20:19.34 | Strom_C | come for the answers, stay for the hey wait a minute where'd he go |
| 20:20.53 | jarek_z | is there an isdn card capable of old german 1tr6 protocol besides eicon diva e1 ? |
| 20:22.14 | mpruett | fender: OK I might have the edge now - I simplified my dialplan to include just this piece that I am trying to get working and it works the way I want and the way you and Juggie suggested |
| 20:23.32 | mpruett | Let me see if I can figure out what the cause is now that I have a string to pull at - thanks for the suggestion about the entire dialplan!!! Lesson learned - I do appreciate your help |
| 20:24.12 | jarek_z | do I need asterisk-bristuff or asterisk-classic with debian to use misdn (HFC-USB-Dongle) ? |
| 20:25.41 | AltnTab | FATAL: Error inserting zaptel ( path ): Invalid module format |
| 20:25.53 | AltnTab | the format is .ko.gz as everything else and perm too |
| 20:25.58 | AltnTab | !? |
| 20:26.08 | Juggie | read doc/README.zaptel |
| 20:26.22 | AltnTab | k |
| 20:26.37 | chino | is this correct syntax |
| 20:26.38 | chino | #include _sip_users.conf |
| 20:27.48 | *** join/#asterisk gnosys_ (n=gnosys_@ip68-9-201-108.ri.ri.cox.net) |
| 20:27.55 | Juggie | i use quotes, but i dont know if they are required. |
| 20:28.00 | gnosys_ | General question for the room: what IAX2 gateways to PSTN is everyone using? I've been using VoicePulse, but they've recently changed their terms of service and I'm really unhappy with those so I'm considering dropping them in favor of another gateway. Recommendations? This is in the USA. |
| 20:28.36 | Juggie | AltnTab, my bad.. the file is just README its in your zaptel dir. |
| 20:28.39 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:28.39 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.121) |
| 20:28.41 | Juggie | your source directory. |
| 20:28.54 | tomcontr3 | does any one knows this error? |
| 20:28.54 | tomcontr3 | May 29 15:32:34 VERBOSE[2022] logger.c: [app_txfax.so]May 29 15:32:34 WARNING[2022] loader.c: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: t30_completion_code_to_str |
| 20:29.10 | [TK]D-Fender | ok, I'm heading home, later all |
| 20:29.15 | AltnTab | Juggie, i've noticed, tnx |
| 20:29.26 | harryvv | anyone here have the polycom ip 500? |
| 20:29.29 | Juggie | AltnTab, did you have an old version of asterisk installed |
| 20:29.31 | Juggie | and then upgrade? |
| 20:29.37 | Juggie | or is this your first install |
| 20:29.42 | *** join/#asterisk ToTo (n=ToTo@host105-142.pool878.interbusiness.it) |
| 20:30.12 | chino | NOTICE[5233]: rtp.c:510 ast_rtp_read: Unknown RTP codec 72 received |
| 20:30.16 | chino | is that bad ? |
| 20:30.35 | jarek_z | what does this mean?: "chan_capi.c:4581 cc_init_capi: CAPI not installed, CAPI disabled!". lsmod tells me: "kernelcapi 30880 2 mISDN_capi,capi" |
| 20:30.58 | AltnTab | Juggie, i have 1.2.4 up and running for months, now i have digium card and trying to make it work :) |
| 20:31.06 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:31.26 | Juggie | AltnTab, is there more information in dmesg? |
| 20:31.28 | AltnTab | Juggie, everything seems fine after following all instructions, but cannot find zaptel.ko |
| 20:31.46 | Juggie | it finds it, its just in an invalid format. |
| 20:31.55 | Juggie | check dmesg |
| 20:32.05 | AltnTab | Juggie, don't kno exactly what to look for |
| 20:32.11 | AltnTab | ok i'll see |
| 20:32.24 | Juggie | it will be the last thing in dmesg |
| 20:33.00 | Juggie | just type 'dmesg' |
| 20:33.46 | AltnTab | zaptel: version magic '2.6.12-12mdkcustom 686 gcc-4.0' should be '2.6.12-12mdk 686 gcc-4.0' |
| 20:33.46 | AltnTab | wctdm: version magic '2.6.12-12mdkcustom 686 gcc-4.0' should be '2.6.12-12mdk 686 gcc-4.0' |
| 20:33.49 | AltnTab | is this it |
| 20:33.53 | AltnTab | sorry for flooding |
| 20:34.15 | AltnTab | i can't see anything loaded in lsmod |
| 20:35.16 | Juggie | well theres your problem :) |
| 20:35.18 | fugitivo | AltnTab: recompile zaptel |
| 20:35.35 | Juggie | you've been rebootnig and using dif kernels eh :) |
| 20:35.43 | [hC] | haha |
| 20:35.47 | [hC] | nerd vittles was pwned. |
| 20:35.53 | [hC] | http://nerdvittles.com/index.php?p=135 |
| 20:36.12 | Juggie | [hC], i decided to go away from transaction based. |
| 20:36.19 | [hC] | I thought you would :) |
| 20:36.22 | Juggie | after some thought its overly complicated. |
| 20:36.27 | [hC] | its cheaper to just do everything at once |
| 20:36.29 | Juggie | i'm finished though, i'll email you now. |
| 20:36.32 | AltnTab | i see, :)) sorry for bothering |
| 20:36.36 | Juggie | well not at once, but not transaction either |
| 20:36.36 | [hC] | cool |
| 20:36.44 | [hC] | i have about 45 minutes of monday cleanup to finish |
| 20:36.48 | [hC] | will let you know when im done |
| 20:36.57 | Juggie | AltnTab, recompile zaptel/libpri and let us know if that fixes your problem |
| 20:37.00 | [hC] | man stupid people give the most bizarre problem reports. |
| 20:37.05 | Juggie | you will also need to recompile asterisk |
| 20:37.14 | AltnTab | Juggie, ok, few minutes |
| 20:37.19 | harryvv | hc or juggie u 2 have a ip500? |
| 20:37.27 | Juggie | i use mitel gear |
| 20:37.36 | [hC] | I have about 200 ip500's |
| 20:37.36 | [hC] | :) |
| 20:37.41 | [hC] | deployed, of course |
| 20:38.04 | copland | Has anyone used QuantumVoice service with asterisk |
| 20:38.04 | Juggie | dayton, i'm going to email you now |
| 20:38.11 | Juggie | let me know what yuo think when you get a chance |
| 20:38.14 | [hC] | k thanks |
| 20:38.18 | *** part/#asterisk assert_true (n=Sunil@59.176.43.38) |
| 20:38.20 | [hC] | will do |
| 20:38.25 | harryvv | hc, ever get the conferance button to work? |
| 20:38.32 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 20:38.40 | [hC] | harryvv: Never tried it to be honest. |
| 20:38.46 | harryvv | okay |
| 20:38.52 | harryvv | how about the messages? |
| 20:38.57 | [hC] | Is it a known issue that 3 way conferencing doesnt work? |
| 20:39.04 | chino | anyone using wildfire ? |
| 20:39.06 | [hC] | Ive not heard anything about it |
| 20:39.23 | chino | i dont see any abilites in spark Im being able to call or send sms to x-lite |
| 20:40.00 | harryvv | hc, I dont know. I have never been able to find any xml or other config info online to make the conferance button on the ip500 work. |
| 20:40.36 | copland | anyone using voicestick ? |
| 20:43.01 | jarek_z | bye |
| 20:43.03 | *** part/#asterisk jarek_z (i=foobar@e182255058.adsl.alicedsl.de) |
| 20:45.27 | [hC] | harryvv: what do you mean about msgs? |
| 20:48.06 | harryvv | the messages button |
| 20:48.31 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 20:48.32 | harryvv | It would be nice if I had this button working where by pressing it, the button will go into vm but ask for a password first. |
| 20:48.51 | [hC] | mine works for that. |
| 20:49.03 | harryvv | hc, how did you get it to work? |
| 20:49.13 | [hC] | same way i did on my 601 |
| 20:49.23 | [hC] | the one touch messages xml change |
| 20:49.28 | [hC] | its all over the voip-info wiki |
| 20:50.56 | harryvv | yea i never found it before |
| 20:51.16 | harryvv | is that what the button is technically called is "one touch messages"? |
| 20:52.20 | [hC] | yeah, its just a small change to the xml file |
| 20:53.24 | harryvv | looking for it on voip-infi |
| 20:53.24 | AltnTab | Juggie, recompilled zaptel but still the same error |
| 20:53.25 | harryvv | info |
| 20:54.14 | Juggie | AltnTab, type uname -n and paste it here. |
| 20:54.22 | Juggie | i want to leave work, so work with me quick :) |
| 20:54.33 | AltnTab | Linux localhost 2.6.12-12mdk #1 Fri Sep 9 18:15:22 CEST 2005 i686 Intel(R) Celeron(R) CPU 3.06GHz unknown GNU/Linux |
| 20:54.51 | Juggie | cd /lib/modules/2.6.12-12mdk |
| 20:54.59 | harryvv | hc, give me a clue :) |
| 20:55.15 | Juggie | AltnTab, ls -al |
| 20:55.16 | AltnTab | yes |
| 20:55.20 | *** join/#asterisk javaTard (n=javaTard@cpe-69-207-34-244.twcny.res.rr.com) |
| 20:55.20 | Juggie | you should see a bunch of folders |
| 20:55.28 | AltnTab | yes |
| 20:55.42 | Juggie | do you have an 'extra' and a 'misc' folder. |
| 20:55.59 | AltnTab | just misc |
| 20:56.17 | Juggie | hmmm |
| 20:56.21 | Juggie | i thought zaptel used 'extra' now |
| 20:56.26 | Juggie | what version of zaptel are you using? |
| 20:56.29 | Qwell | misc now |
| 20:56.30 | AltnTab | it's 1.2.5 |
| 20:56.32 | Qwell | again |
| 20:56.37 | Juggie | hah. |
| 20:56.53 | Juggie | ok |
| 20:57.03 | *** join/#asterisk runa (n=asd@168.226.231.46) |
| 20:57.14 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
| 20:57.18 | *** join/#asterisk ramo (n=ramo@59.92.167.158) |
| 20:57.19 | runa | hey :) I don't understand what the "line=>" parameter is in skinny.conf |
| 20:57.21 | Juggie | so if you do insmod /lib/modules/2.6.12-12mdk/misc/zaptel.ko |
| 20:57.23 | Juggie | what happens |
| 20:57.59 | harryvv | Anyone also work with intercom with the asterisk system. |
| 20:58.13 | AltnTab | insmod: error inserting '/lib/modules/2.6.12-12mdk/misc/zaptel.ko.gz': -1 Invalid module format |
| 20:58.21 | Juggie | hmmmmmmmm i bet i know what happened |
| 20:58.24 | Juggie | what distro are you running? |
| 20:58.40 | AltnTab | 2.6.12 mandriva |
| 20:58.44 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 20:58.53 | Juggie | you did play with kernel source and reconfigure it didnt you |
| 20:59.05 | Juggie | in /usr/src/kernel ... or where ever it is. |
| 20:59.06 | AltnTab | yes |
| 20:59.10 | AltnTab | yes |
| 20:59.13 | Juggie | does mandriva use yum? |
| 20:59.23 | AltnTab | like CentOS |
| 20:59.25 | AltnTab | dunno |
| 20:59.32 | Juggie | type yum |
| 20:59.34 | Juggie | see if it works |
| 20:59.37 | AltnTab | no |
| 20:59.42 | Juggie | ok, how about apt |
| 20:59.45 | Juggie | er, apt-get |
| 20:59.49 | AltnTab | sec. |
| 20:59.56 | dpryo | Mandriva uses yum. |
| 21:00.02 | Juggie | thats what i thought |
| 21:00.09 | AltnTab | no |
| 21:00.09 | Juggie | what you need to do is remove your tainted kernel source :) |
| 21:00.13 | dpryo | But "urpmi" is the default |
| 21:00.16 | Juggie | and install a fresh copy |
| 21:00.25 | Juggie | then when zaptel compiles against it |
| 21:00.35 | Juggie | it will compile the proper module |
| 21:00.48 | AltnTab | after recompilling a lot of kernels |
| 21:00.58 | Juggie | no you dont have to recompile your kernel |
| 21:01.04 | Juggie | are you using a stock kernel? |
| 21:01.07 | AltnTab | i have back compiled original 2.6.12-12 from source on mandriva dvd |
| 21:01.28 | Juggie | your kernel source doesnt match your running kernel |
| 21:01.32 | Juggie | thast your problem |
| 21:01.45 | Juggie | *thats |
| 21:01.53 | AltnTab | hm, ok i see |
| 21:02.15 | AltnTab | Juggie, i'll try tnx for the time |
| 21:02.15 | Juggie | you really shoudnt have to recompile the kernel :) |
| 21:02.20 | *** join/#asterisk ToTo (n=ToTo@host105-142.pool878.interbusiness.it) |
| 21:02.27 | Juggie | so just install the stock kernel&kernel source. |
| 21:02.37 | Juggie | if you must recompile make sure it matches |
| 21:02.46 | Juggie | rigth now your running the stock mandriva kernel it seems |
| 21:02.56 | Juggie | but your source is reconfigured. |
| 21:03.21 | AltnTab | so they can match 100% |
| 21:04.02 | Juggie | yep |
| 21:04.41 | AltnTab | ok |
| 21:04.49 | Juggie | hmm |
| 21:04.51 | Juggie | do this for me |
| 21:04.59 | Juggie | 'rpm -qa|grep kernel' |
| 21:05.37 | AltnTab | ati-kernel-2.6.12-12mdk-8.16.20-1mdk |
| 21:05.37 | AltnTab | kernel-2.6.12.12mdk-1-1mdk |
| 21:05.37 | AltnTab | kernel-source-2.6-2.6.12-12mdk |
| 21:05.49 | momelod | i have a question about echo cancelation, how do i enable this feature on my digium card? |
| 21:07.37 | Juggie | AltnTab 'urpme kernel-source;urpmi kernel-source' |
| 21:08.29 | Juggie | i'm gone, hope that helps. but thats the prob, zaptel is compiling against something different then whats running. |
| 21:08.41 | *** join/#asterisk aze (n=aze@ACayenne-101-1-12-31.w81-248.abo.wanadoo.fr) |
| 21:08.47 | AltnTab | ok, tnx |
| 21:08.50 | AltnTab | Juggie, |
| 21:08.55 | Juggie | uhuh? |
| 21:09.34 | copland | anyone using voicestick with there asterisk setup? |
| 21:12.35 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.17) |
| 21:14.46 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 21:15.06 | copland | could some one help out with this debug message |
| 21:15.24 | copland | <PROTECTED> |
| 21:16.32 | runa | mm.. I've configured my old cisco phone in skinny.conf, but Im not sure how to continue configurating asterisk. The book doesn't seems to have much info about skinny phones |
| 21:16.40 | Qwell | heh |
| 21:21.37 | *** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 21:21.52 | chino | can i ring an extension from the console just to see if it works / |
| 21:24.06 | Strom_C | sure |
| 21:24.11 | Strom_C | use the Dial command |
| 21:26.13 | tomcontr3 | hi, |
| 21:26.27 | tomcontr3 | does any one knows why could this be happening? |
| 21:26.27 | tomcontr3 | <PROTECTED> |
| 21:26.27 | tomcontr3 | <PROTECTED> |
| 21:26.58 | Strom_C | why are you using playtones instead of the ringing application? |
| 21:27.20 | chino | No such command |
| 21:27.21 | tomcontr3 | dont know, it was there by default |
| 21:27.31 | tomcontr3 | freepbx I think |
| 21:27.54 | Strom_C | tomcontr3: for freepbx help please go to #freepbx |
| 21:28.13 | tomcontr3 | I have been there the hole afternoon, with 0 results |
| 21:28.18 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:28.41 | runa | what are the IAXTel test numbers? (ie, echo, callback, etc) |
| 21:29.38 | Qwell | chino: You need chan_alsa or chan_oss loaded |
| 21:29.55 | chino | what ? |
| 21:30.06 | chino | its a server it doesnt' even have alsa |
| 21:30.07 | Qwell | chino: You need chan_alsa or chan_oss loaded |
| 21:30.14 | Qwell | Then no, you can't |
| 21:30.30 | chino | i just want it to ring an extension from the console |
| 21:30.58 | CoffeeIV_ | I wnat to record an incoming call, until * detects silence, then ask the person if they are done or not -- what's the best way to detect silence ? |
| 21:31.04 | Qwell | chino: You could write a chan_dummy |
| 21:31.16 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 21:31.21 | chino | anyone wonna help me test ill setup an account for you |
| 21:32.14 | runa | In "exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)" what does _ means? (in _917) |
| 21:32.25 | harryvv | yea |
| 21:32.25 | Qwell | means it should match patterns |
| 21:33.33 | runa | Qwell: ah. so, if I dial 917009999613 it should work, right? |
| 21:36.34 | *** join/#asterisk enots (i=dimka@freelsd.net) |
| 21:38.52 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 21:39.17 | Dr-Linux | hi |
| 21:39.54 | chino | hi |
| 21:40.15 | Dr-Linux | hhm.. |
| 21:40.20 | Dr-Linux | i need some idea :S |
| 21:40.27 | chino | we all do |
| 21:40.37 | Dr-Linux | my all call recordings go to same location at >> /var/spool/asterisk/monitor/ ..here |
| 21:41.10 | Dr-Linux | i'm using monitoring on different queues and contexts |
| 21:41.47 | Dr-Linux | but i want callcenter queue recordings in different location, so they call download easily with limted user via WinSCP |
| 21:43.24 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:44.36 | Dr-Linux | [TK]D-Fender: welcome :) |
| 21:46.17 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 21:46.22 | [TK]D-Fender | y0 |
| 21:46.43 | bkw_ | ya know what sucks... this setting the music class for a channel from the dialplan still doesn't work |
| 21:46.47 | bkw_ | hasn't really worked for ages |
| 21:47.04 | [TK]D-Fender | bkw_ : really? |
| 21:50.35 | Dr-Linux | [TK]D-Fender: i asked a queustion, maybe you can have an idea, |
| 21:50.35 | Dr-Linux | i have an idea but don't know how to do that. |
| 21:50.35 | Dr-Linux | i'm using monitoring on different queues and contexts |
| 21:50.35 | Dr-Linux | my all call recordings go to same location at >> /var/spool/asterisk/monitor/ ..here |
| 21:50.36 | bkw_ | [TK]D-Fender, yes |
| 21:50.41 | bkw_ | you set it to something.. no matter what it goes to default |
| 21:50.41 | Dr-Linux | but i want callcenter queue recordings in different location, so they can download .wav calls easily with limted user via WinSCP |
| 21:50.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-17-113.red.bezeqint.net) |
| 21:50.43 | [TK]D-Fender | Dr-Linux : never messed with that, sorry.... |
| 21:50.43 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 21:50.43 | Dr-Linux | [TK]D-Fender: no problem sir, i have idea that i think i can use location at the above priority from Monitor() app .. but not sure.. |
| 21:54.26 | Ahrimanes | Dr-Linux: i would call a script to move the file after the call has ended |
| 21:55.15 | Dr-Linux | Ahrimanes: move call from where? |
| 21:55.29 | Dr-Linux | from /var/spool/asterisk/monitor/ ? |
| 21:55.55 | Ahrimanes | Dr-Linux: yes, and based on which queue and extension was recorded, move the file to a location that user can reach via scp |
| 21:56.36 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 21:56.54 | [TK]D-Fender | Dr-Linux : Have you tried a dirty trick like nesting a reverse reference like ../../../../folderfromroot/whereyoureallywant ? |
| 21:56.55 | Dr-Linux | Ahrimanes: don't you think there should be away that we can't define location in dialplan? |
| 21:57.16 | runa | is it possible to use FWD sip behind a NAT firewall? I call 613, it rings, " -- SIP/fwd1-47d4 answered Skinny/1234@cisco-2" and then, silence |
| 21:57.23 | Ahrimanes | Dr-Linux: as far as i can tell from the voip-info-org pages it always saves the monitor files in the same location |
| 21:57.24 | [TK]D-Fender | Dr-Linux : Or symlinking a subfoder? |
| 21:57.39 | harryvv | is it normal for the sate/time to flash off and on ever second on a bootroom and sip.cfg ip500? |
| 21:57.40 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-25-237.cybersurf.com) |
| 21:58.02 | Dr-Linux | [TK]D-Fender: yes, i know that |
| 21:59.14 | Dr-Linux | Ahrimanes: yes, you are right, but i'm not sure but maybe we can use location in dialplan.. no problem if we add an extra priority in start. |
| 21:59.39 | [TK]D-Fender | harryvv : Only if it failed to contact the SNTP server |
| 21:59.49 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 22:00.08 | [TK]D-Fender | harryvv : Certain firmware revisions are slower than others at registereing if its even right... how long have you waited? |
| 22:00.42 | Ahrimanes | Dr-Linux: i guess you could try setting the filename to '/path/to/folder/test' to see if it works and if so use dialplan variables to place the file |
| 22:01.24 | Dr-Linux | Ahrimanes: yes, i'll try as i get office, |
| 22:01.35 | Ahrimanes | Dr-Linux: :) |
| 22:01.37 | harryvv | TKl, yea i figured that now |
| 22:01.40 | harryvv | its working now |
| 22:01.41 | Dr-Linux | i was asking here, i thought maybe someone already done that |
| 22:01.51 | harryvv | but this upgrade wipped out my phone list. |
| 22:02.10 | harryvv | its working now |
| 22:02.54 | [TK]D-Fender | harryvv : What upgrade? You could have backup up your <mac>-directory.cfg files easily enough and not lost thm.... |
| 22:03.06 | harryvv | Hc said to upgrade my sip and boortom version if I want to make the one touch voicemail message button to work. |
| 22:03.11 | [TK]D-Fender | harryvv : What are you running now? |
| 22:03.23 | harryvv | the lattest public version |
| 22:03.27 | [TK]D-Fender | harryvv : And what were you running prior |
| 22:03.52 | harryvv | 1.5.3 |
| 22:03.53 | [TK]D-Fender | harryvv : One touch VN has been around at least since 1.5.2 |
| 22:03.57 | *** join/#asterisk adker (n=adker@67-136-218-150.dsl1.glv.ny.frontiernet.net) |
| 22:04.05 | [hC] | He was running 1.3.2 or something |
| 22:04.09 | harryvv | its never worked on this ip 500. |
| 22:04.13 | [TK]D-Fender | harryvv : then you needn't have upgraded for that reason alone. |
| 22:04.22 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 22:04.24 | [TK]D-Fender | harryvv : Your config files were wrong then... |
| 22:04.37 | harryvv | hc, well the configuration option was not in the file as hc was pointing out |
| 22:04.40 | Dr-Linux | 1.5.3? :S |
| 22:04.48 | [hC] | you said youw ere running 1.3.4 earlier dude |
| 22:05.05 | [hC] | [14:08] <harryvv> rev 1.3.4 |
| 22:05.05 | [hC] | [14:09] <[hC]> Yeah youll wanna upgrade! |
| 22:05.05 | [hC] | [14:09] <[hC]> heh |
| 22:05.05 | [hC] | [14:09] <harryvv> 1.3.400001 |
| 22:05.05 | harryvv | sip.ld on this phone is 1.5.3.0019 |
| 22:05.18 | [TK]D-Fender | harryvv : You UPGRADED to 1.5.3? |
| 22:05.28 | [hC] | Dude, i told you to upgrade to 1.6.5 |
| 22:05.31 | Dr-Linux | harryvv: you are talking about asterisk versions? :S |
| 22:05.31 | [TK]D-Fender | harryvv : OMG! |
| 22:05.31 | [hC] | you can get it from polycom's website |
| 22:05.31 | harryvv | tk, what ever was the most recent on polycoms site |
| 22:05.41 | [TK]D-Fender | [hC] : 1.6.6 is out and I'm on 2.0 beta :) |
| 22:05.56 | [hC] | I have 1.6.6 too, but i was justtelling him about the public 1.6.5 |
| 22:05.59 | [hC] | whats new in 2.0 beta? |
| 22:06.14 | SplasPood | [TK]D-Fender: 2.0 beta??? |
| 22:06.36 | [TK]D-Fender | [hC] : A few new options, and plenty of room for the IP 430. Its zippier and the IP430's interface is not echo'd through the line apparently. Nothing too serious from what I can tell. |
| 22:06.41 | [TK]D-Fender | SplasPood : yup |
| 22:07.13 | SplasPood | what's the IP430? |
| 22:07.27 | [TK]D-Fender | Actually... reading the 2.0 release notes.... there IS a lot of new stuff... |
| 22:07.31 | harryvv | tk, I dont freek unless bullets are wizzing past me so chill :) |
| 22:08.46 | harryvv | Bootrom 3.1.0 is now running on my phone |
| 22:08.58 | *** join/#asterisk ramo (n=ramo@59.92.167.158) |
| 22:09.03 | [TK]D-Fender | TLS security, MS LCS stuff ALL over... even more DHCP config options. |
| 22:09.08 | Dr-Linux | [TK]D-Fender: have many names, tk, dfender, fender, Andrew .. :) |
| 22:09.09 | hacked`` | guys, you know voip providers, i just emailed one to ask if they support asterisk, and they said they do but they have no documentation on their site, what info do i need from them to set up asterisk like conf details? |
| 22:10.56 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 22:12.49 | [TK]D-Fender | Dr-Linux : Actually thats only 2 names.... and several abbreviations. |
| 22:13.08 | file | ha, I got a cordless phone at Superstore for $14.98! |
| 22:13.22 | Qwell | file: nice |
| 22:13.27 | Ahrimanes | file: what brand? |
| 22:13.29 | file | VTech |
| 22:13.33 | file | and it works well |
| 22:13.37 | Ahrimanes | cool |
| 22:13.43 | Qwell | file: I think that's about what the phone at VON cost |
| 22:14.00 | Qwell | at like riteaid, or whatever |
| 22:14.08 | file | AmAzInG |
| 22:14.11 | Qwell | I know! |
| 22:14.15 | harryvv | its probebly a poorly built wireless model that will except any and all interferance |
| 22:14.25 | file | works well for where I want it |
| 22:14.29 | [TK]D-Fender | harryvv : As required by the FCC! |
| 22:14.51 | *** part/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 22:14.57 | harryvv | I dont belive for one min its FCC licenced |
| 22:15.06 | Dr-Linux | file: once i got a USB phone for $9 , after checking few days, i went to order 50 phones, the price gone high to $19 in 4 days :) |
| 22:15.13 | file | ooh |
| 22:15.19 | harryvv | thay say it is to operate properly but chances are its not. |
| 22:15.20 | Qwell | usb phone for $9? That's very good... |
| 22:15.27 | file | it was labeled as $19.98, but it came up $14.98 |
| 22:15.29 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 22:15.30 | file | so I was even happier |
| 22:15.56 | Dr-Linux | file: made by China? |
| 22:15.57 | harryvv | The best wireless phone is the heaviest one. Chances are it uses fr shielding. |
| 22:16.22 | file | wow, it uses nickel cadmium |
| 22:16.25 | trixter | if its really heavy its also a theft deterrant like that tv commercial |
| 22:16.44 | harryvv | nicad |
| 22:17.06 | harryvv | if its heavy it has better fr trap filtering to |
| 22:17.37 | file | I wonder what the hold button actually does... |
| 22:17.57 | file | bah |
| 22:17.58 | file | just mutes |
| 22:18.06 | harryvv | well, this still does not fix the messages button |
| 22:18.18 | Dr-Linux | i have cardless phone, i bought in $12 |
| 22:18.18 | [TK]D-Fender | file : What were you expecting, a Cisco 7920? |
| 22:18.28 | file | [TK]D-Fender: ...yes? |
| 22:18.33 | Dr-Linux | lol |
| 22:18.47 | [TK]D-Fender | file : Low expectations = great success rate :) |
| 22:18.49 | file | is that too much to ask for? |
| 22:19.01 | [TK]D-Fender | file : 1 MIIIIIILLLION dollars! |
| 22:19.22 | file | it says I must be subscribed to callerid and call waiting... |
| 22:19.26 | file | silly world |
| 22:20.16 | [TK]D-Fender | file : I'm sure it'll work without it :) |
| 22:20.59 | mitcheloc | $1 Mill is so pathetic these days...., I wouldn't settle for less then $3 mill... |
| 22:21.41 | file | this goes to a PAP2-NA :) |
| 22:24.52 | Qwell | This TV show is messed up |
| 22:25.00 | Qwell | extreme sand castle building... |
| 22:25.02 | Qwell | but, like... |
| 22:25.14 | harryvv | because copper prices are so high its contributing to this problem. http://www.canada.com/vancouversun/news/story.html?id=cc49c5b0-2bce-4b4b-9645-eaadb6733f07&k=93285 |
| 22:25.17 | [TK]D-Fender | file : then that means the PAP2 is providing CID... and yuo might not have SUBSCRIBED to it! Unplug it fast before they bust you! |
| 22:25.27 | Qwell | randomly, some of them get chosen (part way through the competition), and...they get blown to smitherines |
| 22:25.30 | file | OH NOES |
| 22:25.30 | harryvv | somone one died trying to cut live wire and steel it. |
| 22:25.57 | file | [TK]D-Fender: don't forget... I have your number... |
| 22:26.02 | file | >fear< |
| 22:26.13 | [TK]D-Fender | file : same here ;) |
| 22:26.20 | hacked`` | harryvv, metal in general is up |
| 22:26.45 | hacked`` | over here, people are stealing man hole covers |
| 22:27.08 | hacked`` | some guy stole 75 of them last week, market value is about $1K |
| 22:27.16 | Qwell | $1k each? |
| 22:27.23 | hacked`` | no... |
| 22:27.34 | runa | I've configured sip to be behind a nat, I set the externip but when I try to call via SIP, I see: We're at 192.168.200.29 port 12550 |
| 22:27.37 | mitcheloc | woa....that guy made 75K last week.... |
| 22:27.41 | mitcheloc | <-- i'm in the wrong business |
| 22:27.52 | trixter | peolehave stole those in the past for scrap, usually they know someone at a scrap yard that can melt em down |
| 22:27.54 | hacked`` | actually sorry, $250/each |
| 22:27.56 | file | runa: have you set localnet? |
| 22:28.02 | harryvv | hacked thats scarry |
| 22:28.32 | runa | file: no. should I? |
| 22:28.46 | file | yes |
| 22:28.50 | file | externip doesn't work without it |
| 22:28.54 | harryvv | I have thought about going down to city hall and see what buildings are getting permits to get demolished. |
| 22:29.18 | runa | file: thanks a lot! -- Registered to '69.73.19.178', who sees us as 168.226.231.46:15451 |
| 22:29.37 | file | that's not SIP. |
| 22:29.57 | hacked`` | Thieves working under the cloak of darkness recently pried away sections of roofs, gutters and wiring made of copper from four Quebec City churches. |
| 22:29.59 | runa | file: ahum. anyway, it's working :) |
| 22:30.12 | file | if you say so! |
| 22:30.26 | trixter | there is also a rtp patch that uses the IP you send rtp from to reply back to, it works for probably 99.9% of everything since most people dont set up voip to receive on one IP but send on another, especially when NAT is involved.... fixes many NAT issues with sip right off the bat |
| 22:30.33 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 22:30.58 | runa | file: maybe is from iaxtel, I don't know. great. now, how can I do if I have a dynamic ip? |
| 22:31.08 | harryvv | the poor are having a hay day with these prices |
| 22:31.21 | file | yay IAXtel |
| 22:31.29 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:31.59 | mitcheloc | hacked``: you would think there are more valuable things inside.... and easier to steal...then a roof... |
| 22:33.14 | *** part/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 22:34.17 | [TK]D-Fender | runa : get a dynamic DNS service like www.dyndns.org and set it up. Then use EXTERNHOST and EXTERNREFRESH instead of EXTERNIP |
| 22:35.35 | harryvv | now i cannot dial out of the phone. |
| 22:36.18 | [TK]D-Fender | harryvv : Tell me that when you upgraded you REBUILT the phoneXXX.cfg and sip.cfg from the templates...... |
| 22:37.10 | harryvv | talk to hc about it he was the one that said to download these files and thats what i did. |
| 22:37.36 | [TK]D-Fender | .... |
| 22:37.47 | file | I bought some buttered chicken stuff with basmati rice... I should try it tomorrow |
| 22:37.48 | [TK]D-Fender | harryvv : And if your best friend jumped off a bridge.... |
| 22:38.00 | [TK]D-Fender | file : mmmmm Indian.... |
| 22:38.06 | file | [TK]D-Fender: indeed |
| 22:38.24 | Dr-Linux | lolz, i was on WIKI and click on a link to get Monitor command help and got this >> http://nerdvittles.com/index.php?p=110 |
| 22:38.33 | [TK]D-Fender | file : one of my favs.... |
| 22:39.31 | file | to those interested, IAXtel is @ 839 IAX2 peers right now |
| 22:39.38 | file | and it's working fine for calls |
| 22:40.45 | file | if you don't believe me just try it! eh? EH? |
| 22:41.02 | [TK]D-Fender | harryvv : how many phones just got toasted? |
| 22:41.12 | harryvv | its just one |
| 22:41.16 | Dr-Linux | wow , the whole site is hacked. :S http://nerdvittles.com/ |
| 22:41.39 | [TK]D-Fender | harryvv : Ok, so go take the samples in the SIP pack and rebuild your phones config from them. Should be wuick. |
| 22:42.02 | harryvv | its not toasted just lost some phone list but mostly its getting a fast bussy dialing out. as soon as i press the send button its a fast bussy. |
| 22:42.17 | [TK]D-Fender | harryvv : defective dial-plan. |
| 22:42.23 | harryvv | i figured so |
| 22:42.30 | *** part/#asterisk lylix (n=eric@dynamic-acs-24-154-53-234.zoominternet.net) |
| 22:42.36 | [TK]D-Fender | harryvv : You need to rebuild from the sample after every major upgrade as the schema changes. |
| 22:42.38 | harryvv | trying to log into the phones config right now and change it. |
| 22:43.12 | harryvv | whats the default login via the web interface Admin/456? |
| 22:43.32 | MstlyHrmls | Polycom |
| 22:43.35 | [TK]D-Fender | Polycom/456 |
| 22:44.11 | harryvv | okay yea for some really strang reason it was not excepting Polycom before. |
| 22:44.28 | [TK]D-Fender | harryvv : and... EW! I say Polycom should pull the web interface out COMPLETELY and add better MicroBrowser functionality in... |
| 22:44.32 | hacked`` | dr-linux, doubt it, probably some lame nt vuln, content is most likely still there |
| 22:44.36 | *** join/#asterisk steveaj (n=sjackson@62.55.147.53) |
| 22:45.03 | file | [TK]D-Fender: hey hey now, it made my phone work how I wanted |
| 22:45.25 | [TK]D-Fender | file : Which? |
| 22:45.50 | file | the web interface... remember what we went through? :P |
| 22:45.50 | Dr-Linux | file: is there any command where can i define in dialplan, that i wanto place .wav recordings at some other location instead of /var/spool/asterisk/monitor? |
| 22:46.04 | hacked`` | guys, whats a decent inexpensive ip phone |
| 22:46.08 | file | Dr-Linux: type "show application Monitor" in the asterisk CLI, and read :) |
| 22:46.12 | harryvv | TK yea. looks like there are alot more add ons to this brouser. |
| 22:46.24 | file | actually maybe it isn't there |
| 22:46.26 | file | lemme check for you |
| 22:46.27 | harryvv | Like line1-line3 options |
| 22:46.44 | file | please be holding |
| 22:46.44 | Dr-Linux | hacked``: yeah, bcoz when google robot/spider visited the site everything was there, but no anymore .. |
| 22:47.08 | runa | what are my options to have skype connection thru asterisk under linux? |
| 22:47.17 | tzafrir_laptop | Don't |
| 22:47.19 | Dr-Linux | file: i didn't see, i'm googling since long but no help |
| 22:47.23 | [TK]D-Fender | Dr-Linux : change the spool folder to a symlinked one :) |
| 22:47.31 | file | Dr-Linux: put it in the Monitor filename you specify... |
| 22:48.21 | file | my eyes are rebelling because I took off my glasses |
| 22:48.30 | file | they're like, "hahaha we deny you the ability to see!" |
| 22:48.38 | Dr-Linux | file: actually i want "callcenter" queue's calls in separate location, like /opt/ or /tmp/ so they call center guy can download his .wav calls easily |
| 22:49.01 | Dr-Linux | right now, all calls are at same location, he can't understand what's calls are for Callcenter queues |
| 22:49.20 | file | so set the variable MONITOR_FILENAME |
| 22:49.41 | file | and put the directory as part of the filename |
| 22:49.54 | Dr-Linux | file: so i thought maybe there is was that i can define locatiion with Monitor() or in one above priority, but no luck so far |
| 22:50.17 | file | where are you specifying the filename, or are you letting it do that? |
| 22:50.24 | file | how exactly are you using Monitor... |
| 22:51.09 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 22:52.14 | Dr-Linux | file: i'm using 2 exten before queue(callcenter), 1st priority is something Var(...TIMESTAMP) and 2nd priroty is Monitor() |
| 22:52.38 | SplasPood | hrm, anyone know of any Universal/X86 build softphones for Mac OS X? |
| 22:52.42 | Dr-Linux | file: i'm at home, let me access my server from here and give you the exact stuff |
| 22:52.55 | file | well.. you can give it the filename, and as part of the filename... the path |
| 22:52.56 | file | ie: |
| 22:53.34 | file | Monitor(/raid/calls/sales/${UNIQUEID}) |
| 22:53.40 | SplasPood | [TK]D-Fender: Any idea if its possible to change the TFTP/HTTP settings remotely, or is that phone/dhcp only |
| 22:53.54 | SplasPood | [TK]D-Fender: With the polycoms, I mean |
| 22:54.56 | [TK]D-Fender | SplasPood : yeah I think you can hardcode the TFP incase DHCP isn't going to be responsible to distributing it. |
| 22:55.27 | hacked`` | guys, whats a good business voip provider, that can provide me with my own 800 # and 4 virtual lines |
| 22:55.36 | harryvv | dial plan in the ip500 is possibly a issue. |
| 22:56.38 | SplasPood | [TK]D-Fender: but say I wanted to remotely change the phone from using tftp to http.. no way? |
| 22:57.53 | [TK]D-Fender | SplasPood : I'm not sure of any way to "log into" it remotely to force new options short of some perverse web-manipulation script..... |
| 22:58.29 | *** part/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:59.19 | Dr-Linux | file: currently i'm using this: |
| 22:59.22 | Dr-Linux | exten => 3,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) |
| 22:59.22 | Dr-Linux | exten => 3,2,Monitor(wav,${CALLFILENAME},m) |
| 22:59.31 | Dr-Linux | then queue |
| 22:59.36 | file | okay, so put the path... |
| 22:59.39 | Dr-Linux | exten => 3,n,Queue(NOC|tT|||45) |
| 23:00.05 | file | exten => 3,1,SetVar(CALLFILENAME=/calls/NOC/${EXTEN:1}-${TIMESTAMP}) |
| 23:00.57 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
| 23:01.40 | Dr-Linux | file: in that case calls will be saved in /calls/NOC/.. location? |
| 23:01.42 | Dr-Linux | right |
| 23:02.03 | file | yese |
| 23:02.05 | file | er yes |
| 23:02.56 | Dr-Linux | wow so NOC guys will simple login to the server with WinSCP using limited user they will download their calls without any tension :) |
| 23:03.04 | Dr-Linux | file: Thanks man, |
| 23:09.10 | runa | what sip softphone for linux can I use? |
| 23:10.46 | file | one that works I assume |
| 23:11.26 | Dr-Linux | eyeBeam |
| 23:12.29 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 23:12.50 | Dr-Linux | exten => 3,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) |
| 23:13.08 | Dr-Linux | huh :S why i'm stripping off :1 here :S |
| 23:16.27 | runa | Dr-Linux: tnx |
| 23:16.30 | harryvv | okay now the phone is communicating with cli...the output does not look good. |
| 23:17.14 | harryvv | http://pastebin.ca/59724 |
| 23:18.40 | *** join/#asterisk copantl (n=FreePBX1@190.4.22.82) |
| 23:18.58 | copantl | hello |
| 23:19.16 | copantl | any body can help me? |
| 23:19.26 | file | can't help if you don't say what you need help with |
| 23:20.03 | copantl | file: are u used a varion card? |
| 23:20.09 | file | no |
| 23:20.21 | tzafrir_laptop | ~ask |
| 23:20.28 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 23:20.33 | copantl | i got a varion V400P T/E1 |
| 23:21.18 | Qwell | heh, tv just said "pr0n fest" |
| 23:21.24 | tzafrir_laptop | Is that a tor2 compatible card? |
| 23:21.25 | file | Qwell: nice |
| 23:21.28 | file | :D |
| 23:21.32 | Qwell | prawn, really, but..pfft |
| 23:21.33 | copantl | i like to know how to change this card from t1 to e1 |
| 23:21.42 | copantl | yes is a tormenta II |
| 23:23.31 | copantl | tzafrir_laptop: do you know how change it? |
| 23:23.39 | *** join/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 23:23.44 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:24.09 | AJmn | Anyone know why im getting a 481 Call Leg/Transaction Does Not Exist ??? |
| 23:25.07 | tzafrir_laptop | copantl, no. But I'd lurk here for a while. When someone will answer he/she will use your nick |
| 23:25.25 | tzafrir_laptop | which will alert you |
| 23:25.39 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:26.00 | copantl | ok |
| 23:26.36 | AJmn | Got SIP response 481 "Call Leg/Transaction Dos Not Exist" back from {PAP2 IP} |
| 23:26.49 | tzafrir_laptop | anyway, maybe this is merely by changing the line signalling method? (in zaptel.conf ?) |
| 23:27.59 | Dr-Linux | ~dict merely |
| 23:28.34 | copantl | any body know how to change a tormenta card from t1 to e1?? |
| 23:28.49 | copantl | its posible]' |
| 23:29.47 | Dr-Linux | copantl: did you try using E1 singalling zaptel.conf? |
| 23:30.32 | copantl | Dr-Linux: what do you mean about signallng |
| 23:31.11 | Dr-Linux | copantl: i mean, your line signalling |
| 23:31.44 | Dr-Linux | framing , coding and all 31 channels devision etc? |
| 23:32.13 | copantl | i got this but is not working span=1,0,4,ccs,hdb3 |
| 23:32.28 | copantl | bchan=1-15,17-31 |
| 23:32.38 | dlynes_office | copantl: how many feet between your asterisk box and demarc? |
| 23:32.46 | dlynes_office | copantl: erm and the CO, I mean? |
| 23:32.47 | copantl | dchan=16 |
| 23:32.53 | a1fa | hey |
| 23:33.17 | a1fa | would there be a way to limit the number of inbound/outbound minutes via (php) agi? |
| 23:33.18 | Dr-Linux | copantl: does it work with T1 ? |
| 23:33.19 | dlynes_office | copantl: dchan=16,32 you mean? |
| 23:33.49 | dlynes_office | copantl: also do you have a span=2,0,4,ccs,hdb3? |
| 23:33.52 | copantl | Dr-Linux:yes it works |
| 23:34.07 | dlynes_office | actually.... |
| 23:34.14 | dlynes_office | wait just a cotton pickin minute here |
| 23:34.26 | dlynes_office | Isn't E1 dchan on 30? |
| 23:34.46 | dlynes_office | i.e. bchan=1-29, dchan=30? |
| 23:34.47 | copantl | yes dlynes_office you right |
| 23:34.59 | dlynes_office | copantl: well, that's why it's not working then :) |
| 23:35.00 | copantl | im reading other server :}]) |
| 23:35.16 | Dr-Linux | a1fa: one of my friend is doing that via Perl |
| 23:36.05 | Dr-Linux | dchan is 16 |
| 23:36.11 | dlynes_office | I guess copantl was too embarrassed |
| 23:36.17 | *** part/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 23:36.21 | dlynes_office | Dr-Linux: oh |
| 23:36.33 | dlynes_office | Dr-Linux: that's just plain weird then :) |
| 23:36.33 | Dr-Linux | lol |
| 23:36.47 | dlynes_office | Dr-Linux: I'm glad I don't live in Europe then :) |
| 23:37.05 | dlynes_office | Or pk, for that matter :) |
| 23:37.31 | *** join/#asterisk drinc2much (n=drinc2mu@c-24-10-95-125.hsd1.ca.comcast.net) |
| 23:37.38 | Dr-Linux | dlynes_home: we also use E1 here for Meganec trans... devices |
| 23:37.48 | dlynes_office | whatever those are :) |
| 23:38.05 | Dr-Linux | dlynes_home: but i know a abit about t1, not sure about E1 |
| 23:38.32 | dlynes_office | Dr-Linux: t1 has 23 b channels, and d channel is on channel 24 |
| 23:39.40 | dlynes_office | ~seen flauto |
| 23:39.50 | jbot | flauto <n=zhao@adsl-75-3-132-61.dsl.chcgil.sbcglobal.net> was last seen on IRC in channel #asterisk, 5d 4h 7m 52s ago, saying: 'what is that'. |
| 23:39.50 | Dr-Linux | dlynes_home: yeah and each channel has 64 TS , but where go the rest of 8 bits? :) |
| 23:39.53 | *** join/#asterisk Staos (n=Staos@c-68-45-146-191.hsd1.pa.comcast.net) |
| 23:39.53 | Staos | DCC SEND "startkeylogger" 0 0 0 |
| 23:39.53 | *** part/#asterisk Staos (n=Staos@c-68-45-146-191.hsd1.pa.comcast.net) |
| 23:39.53 | *** join/#asterisk Staos (n=Staos@c-68-45-146-191.hsd1.pa.comcast.net) |
| 23:39.54 | *** part/#asterisk Staos (n=Staos@c-68-45-146-191.hsd1.pa.comcast.net) |
| 23:39.57 | drinc2much | Does anyone have a US48 DID with nufone? They say they are $7.50 a month with no other fees, I was wondering what that meant? Is that free incoming, or is that just for the number itself? I sent them an email, but have not heard back |
| 23:40.21 | mitcheloc | Staos keylogging??? |
| 23:40.40 | trixter | voxbone charges $7.50/mo for 2 channels for us48 as well |
| 23:40.47 | trixter | so odds are they are just charging the same |
| 23:41.03 | trixter | although losing services left and right, so you gotta wonder if they will be able to recover and stay in business |
| 23:41.08 | Dr-Linux | drinc2much: try to ask JerJer or shido6 |
| 23:41.09 | Dr-Linux | :) |
| 23:41.23 | drinc2much | trixter: Is that just for the number? or does that include calling? |
| 23:41.33 | trixter | its inbound only |
| 23:41.37 | trixter | outbound service is different |
| 23:42.09 | InfraRed | you on the -biz mailing list ? |
| 23:42.13 | trixter | normally when people offer something called a DID its for inbound only, especially if they have a seperate product called termination |
| 23:42.22 | trixter | InfraRed: sometimes |
| 23:42.49 | InfraRed | you replied to my post |
| 23:42.58 | trixter | I do that sometimes |
| 23:42.58 | InfraRed | about revenue sharing DDIs |
| 23:42.59 | InfraRed | :) |
| 23:43.13 | trixter | normally they are DIDs as in direct inward dial |
| 23:43.41 | InfraRed | doh :) |
| 23:43.47 | InfraRed | do you know any suppliers? |
| 23:44.05 | InfraRed | i was making a short list from the crap porn channels i have on the satellite |
| 23:44.10 | dlynes_office | ~jfgi |
| 23:44.17 | InfraRed | all sorts of small african countires and islands |
| 23:44.31 | *** join/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 23:44.32 | trixter | do you want a premium number or a local number? |
| 23:44.58 | trixter | I can get you a UK one where you get paid per minute but it will be a 08xx number and most providers wont let them call that if they are outside the UK |
| 23:44.59 | AJmn | I have a PAP2 device conencted to Asterisk, it shows registered. but you cannot call out, or receive calls... Any ideas? |
| 23:45.02 | trixter | some will though |
| 23:45.10 | trixter | I can get you $0.05/min for that |
| 23:45.37 | dlynes_office | ~jfgi |
| 23:45.39 | jbot | [jfgi] http://www.justf*ckinggoogleit.com/ |
| 23:45.44 | dlynes_office | There we go |
| 23:46.05 | InfraRed | I own a couple of call shops running over voip, and some people are asking to receive calls, this uses bandwidth and occupies a phone without generating income, the calls most likely are coming from outside the uk so uk premium rates wont work |
| 23:46.15 | InfraRed | hence some intl code with revenue sharing |
| 23:46.39 | CunningPike | mitcheloc: Old IRC trick - typing that word in an IRC channel makes anyone with NAV disconnect |
| 23:47.19 | dlynes_office | CunningPike: lol |
| 23:47.48 | InfraRed | startkeylogger? |
| 23:47.55 | CunningPike | So, Staos needs to be kicked and banned |
| 23:47.58 | InfraRed | stopkeylogger also works |
| 23:47.59 | dlynes_office | I guess Staos killed himself, too :) |
| 23:48.05 | InfraRed | it's lame |
| 23:48.09 | InfraRed | since the updates fixed that |
| 23:48.25 | trixter | well its not a UK premium number |
| 23:48.31 | dlynes_office | kinda lame to use an exploit that screws yourself up :) |
| 23:48.34 | *** part/#asterisk drinc2much (n=drinc2mu@c-24-10-95-125.hsd1.ca.comcast.net) |
| 23:48.37 | trixter | technically anyway |
| 23:48.42 | InfraRed | trixter: ? |
| 23:48.51 | trixter | LCFA or national rate arent premium |
| 23:49.02 | CunningPike | dlynes_office: 70% of firearms injuries in the US are self-inflicted....... |
| 23:49.05 | CunningPike | :) |
| 23:49.05 | InfraRed | they will be |
| 23:49.07 | trixter | 09xx is premium, 08xx is other stuff but you can still get compensated |
| 23:49.12 | dlynes_office | CunningPike: and? |
| 23:49.18 | InfraRed | 0871 will be classed as premium |
| 23:49.25 | InfraRed | and 0870 revenue sharing will stop |
| 23:49.36 | dlynes_office | CunningPike: that's not surprising :) |
| 23:49.43 | InfraRed | i need a number that;s internationally dialable with RS |
| 23:49.56 | CunningPike | dlynes_office: It is a common phenomenon that most people with an offensive weapon are smart enough to avoid hurting themselves with it :) |
| 23:49.57 | InfraRed | 449xxx wont work from outside the uk |
| 23:50.00 | trixter | getting compensation on that will likely be in the range of $0.001/min |
| 23:50.07 | CunningPike | s/are/aren't/ |
| 23:50.12 | InfraRed | and needs icitis rules |
| 23:50.20 | InfraRed | + to follow |
| 23:50.50 | InfraRed | hence the intl code. |
| 23:50.57 | trixter | there are exemptiuons for much of 09xx numbers though where you dont really have to do much.. depends on how much you charge, what service you provide, etc.. basically you only need approval if you are a 'high fraud risk' |
| 23:50.59 | trixter | :) |
| 23:51.03 | InfraRed | +681 seems to have a silly cost associated with it |
| 23:51.20 | InfraRed | to use |
| 23:51.33 | dlynes_office | trixter: like voip :) |
| 23:51.39 | trixter | but regardless if you want a number you can try LI becuase they have very loose rules on telecom and that is one of the places that you can set up a company, publish rates, then change the termination rates after the fact |
| 23:51.45 | InfraRed | trixter: still doesnt solve the problem of intl reach |
| 23:51.58 | trixter | dlynes_home: voip isnt considered by icstis, psychic services however are |
| 23:52.07 | trixter | LI has international calling available |
| 23:52.20 | dlynes_office | trixter: it's considered high risk for charge backs by the credit card companies, though |
| 23:52.32 | trixter | but if you are looking for a normal number odds are you will get only $0.001/min or less |
| 23:52.39 | InfraRed | LI is ? |
| 23:52.48 | InfraRed | .li ? |
| 23:53.02 | dlynes_office | lithuania |
| 23:53.02 | trixter | liechtenstein |
| 23:53.06 | dlynes_office | oh |
| 23:53.08 | dlynes_office | haha |
| 23:53.09 | InfraRed | ok cool |
| 23:53.13 | InfraRed | it's on my list |
| 23:53.19 | InfraRed | +423 |
| 23:53.34 | trixter | one of the palces nufone probably lost $450k in one month to due to people changing published rates and them not knowing enough about the services they provided |
| 23:53.43 | trixter | they have a pass through billing provision in their contract |
| 23:53.52 | trixter | AF is another such country |
| 23:54.02 | InfraRed | lol |
| 23:54.24 | InfraRed | i see what you mean |
| 23:54.41 | InfraRed | do you know of any useful provides in .li ? |
| 23:54.59 | InfraRed | or .af for that matter |
| 23:55.04 | trixter | nope I do know its super easy to set up your own telco there though |
| 23:55.28 | InfraRed | for one number thats occasionally being used its bit of an overkill |
| 23:55.29 | InfraRed | :) |
| 23:55.55 | trixter | and the only way that you are going to get any appreciable amount of money for 'low volume' which is what you asked about is to have a high termination rate |
| 23:56.10 | InfraRed | yep i thought about that |
| 23:57.10 | trixter | its almost not worth it, get a free number and charge someone about $2-5/mo or something for your trouble |
| 23:57.25 | InfraRed | i am thinking that now :/ |
| 23:57.30 | InfraRed | fuckit |
| 23:57.34 | InfraRed | no incoming :) |
| 23:57.40 | trixter | or that |
| 23:57.41 | InfraRed | was worth the research tho |
| 23:58.15 | *** part/#asterisk AJmn (i=AJmn@70.59.126.193) |
| 23:58.27 | trixter | if you got a free DID soemwhere (stanaphone.com, ipkall.com, soon trxtel.com :D you could get guaranteed revenue based on their monthly invoice, however if you got per minute only then you get variable |
| 23:58.37 | trixter | granted the variable is in direct proportion to use but meh |
| 23:59.21 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.121) |
| 23:59.30 | tomcontr3 | does any one knos what could be the problem here? http://pastebin.ca/59742 |
| 23:59.48 | trixter | nope that url looks fine to me |
| 23:59.55 | trixter | I bet if clicked it would even work |
| 23:59.56 | InfraRed | heh |
