00:00.46 | SteveTotaro | i pay a $.01/min except instate is $.05/min but that is easy to work around |
00:00.59 | SteveTotaro | six second rounding |
00:02.04 | SteveTotaro | ~tdm |
00:02.04 | jbot | it has been said that tdm is Time Division Multiplexing. It is a scheme in which numerous signals are combined for transmission on a single communications line or channel. Each signal is broken up into many segments, each having very short duration. |
00:02.18 | mitcheloc | SteveTotaro: mog wants to know how many minutes you push |
00:02.42 | SteveTotaro | not enough to seek out a better deal yet |
00:02.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
00:03.07 | SteveTotaro | plus i love to have a real t1 |
00:03.31 | mitcheloc | just got his first pri :) |
00:03.32 | SteveTotaro | not some "virtual T1", i guess i am old school like that |
00:03.40 | JT | virtual pris suck |
00:03.56 | JT | i just got a pri in the datacentre brought up on the weekend |
00:04.22 | SteveTotaro | once managed a t3 a year ago |
00:04.54 | JT | cabling was finally done last week |
00:04.57 | SteveTotaro | brought into a adtran 2800 m13 and broken into 28 t1 PRIs |
00:05.07 | JT | got the telco to run 2 * 16 pair screened cables to my rack for free |
00:05.18 | JT | hmm |
00:05.32 | SteveTotaro | i got the telco to run two coax past the demarc |
00:05.45 | SteveTotaro | for free |
00:06.14 | JT | in my case they had to run the cables vertically up 2 floors, then horizontally 30metres |
00:06.30 | JT | and the cables need to be tagged every 3 metres under datacentre rules |
00:06.38 | SteveTotaro | we already had fiber at the demarc |
00:06.54 | SteveTotaro | and a great big mux |
00:06.59 | mitcheloc | JT: is there a fee to run the cables between floors? |
00:07.08 | adeel | in the beginning of all my calls (zap -> sip, sip -> sip) there's a 2-3 second audio delay....is there anyway to get around that? or reduce it at the least? |
00:07.13 | JT | mitcheloc: usually whatever a cabling contractor charges |
00:07.22 | JT | mitcheloc: i got the telco to pay for the contractor though |
00:07.27 | SteveTotaro | between floors is the easy part |
00:07.39 | SteveTotaro | you just feed it down the conduit |
00:07.49 | JT | there is no conduit here |
00:07.54 | mitcheloc | yes, but some buildings charge monthly for that kind of a cross connect |
00:08.00 | SteveTotaro | cable contractors are for sissies |
00:08.05 | JT | just wiring cupboards and firestopper materials |
00:08.20 | JT | SteveTotaro: the datacentre won't let unlicensed people cable |
00:08.27 | JT | mitcheloc: those buildings suck ;) |
00:08.39 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
00:08.53 | SteveTotaro | not sure where you get a license for low voltage cabling around here..... |
00:09.10 | SteveTotaro | sometimes you need a permit but that's about it |
00:09.19 | JT | you need a licence to run telecommunications cables here |
00:09.32 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
00:09.32 | *** mode/#asterisk [+o mog] by ChanServ |
00:09.42 | SteveTotaro | what if you are just crossconnecting? |
00:09.44 | dansaz | can hard phones interface with asterisk through an ethernet switch? or do you need a channel bank? |
00:10.02 | SteveTotaro | if it is past the demarc then you shouldn't need a license, it's CPE |
00:10.04 | JT | you mean plugging in an RJ45 or punching down? |
00:10.09 | JT | i'm not in america. |
00:10.16 | SteveTotaro | i understand that |
00:10.21 | mitcheloc | dansaz: a channel bank or ata |
00:10.24 | SteveTotaro | just makes no sense |
00:10.38 | JT | if it's not involving plugging in premade stuff, you need a licence |
00:10.49 | SteveTotaro | wow, that's tough |
00:10.53 | JT | well they're worried about people screwing up telecommunications infrastructure, i guess |
00:11.06 | SteveTotaro | i am old school with my cabling skills too, has to perfect |
00:11.11 | JT | i wish i got the licence before they made it harder to get |
00:11.45 | SteveTotaro | i have nice long 200 pair in my trunk ;) |
00:13.08 | JT | i do my own telecomms, data and electrical cabling at home :) |
00:14.03 | SteveTotaro | well i have seen alot of telco guys including verizon yank someone else's existing pair with dialtone and cross connect it with new circuits just to close a ticket (historic Washington DC) |
00:14.29 | SteveTotaro | there just isn't enough copper |
00:14.36 | riddlebox | SteveTotaro, my experience has shown the old school guys were way sloppier with cabling than guys now |
00:15.08 | mipster | gotta love those wax covered strings |
00:15.11 | SteveTotaro | i would disagree with that but it is just from my personal experiences |
00:15.21 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
00:15.30 | fujin | Is there anyway (a channel variable) to get the name of the agent who answered a queue call for MONITOR_FILENAME? |
00:15.52 | fujin | I'm tryign to build a filename which looks like date_agent-${AGENTNUMBER}_caller-${CALLERID(num)} |
00:15.56 | riddlebox | SteveTotaro, I will take some pics on job sites of things guys did before us young guys came in |
00:15.59 | fujin | working out how to get agentnumber is the prob |
00:16.28 | SteveTotaro | well i have seen some rats nests too but usually in 100 year old buildings |
00:17.17 | xacatecas | how do I rewrite a number of the form 0ZX. to be of the form 27ZX. ? |
00:17.22 | SteveTotaro | fujin: you could look it up in your cdr or queue_log and rename it |
00:17.25 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
00:17.28 | xacatecas | basically replace the leading zero with 27 ? |
00:17.39 | riddlebox | in st. louis I guess it was just accepted to throw cabling everywhere, and the phone rooms whoa they are bad sometimes, we take before and after pics for customers |
00:18.02 | *** part/#asterisk dansaz (n=dan@c-68-58-81-102.hsd1.in.comcast.net) |
00:18.10 | *** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com) |
00:18.48 | mipster | exten => 0ZX.,n,Dial(Chan/Prov/27${EXTEN:1}) |
00:18.49 | fujin | SteveTotaro: ok, how about a real suggestion? |
00:18.50 | mipster | I think |
00:18.53 | fujin | a channel variable is what I'm after |
00:19.01 | SteveTotaro | that was a real suggestion |
00:19.39 | fujin | that was a fail suggestion |
00:19.50 | SteveTotaro | you could load your queue_log into mysql query it and rename the file |
00:19.57 | SteveTotaro | as a scheduled cron |
00:20.07 | *** join/#asterisk really_phukt (n=chatzill@209.216.64.44) |
00:20.55 | mipster | xacatecas, obviously you'd need to replace the n with a 1 if it's the first step in the dialplan for that extension |
00:20.59 | jblack | fujin: You can set variables in sip.conf |
00:21.09 | fujin | jblack: uh? |
00:21.20 | fujin | I want a variable to contain what agent the call was delivered to, for queueing |
00:21.22 | jblack | so, perhaps you could set FULLNAME for your authentication for your phones, and use that variable. |
00:21.34 | fujin | I'm doing hotdesking, Phones are irrelevant really |
00:21.40 | jblack | oh |
00:21.51 | xacatecas | exten => 0ZX.,n,Dial(Chan/Prov/27${EXTEN:1}) <-- ... i've got extensions of the form 27ZX., but a client may well dial 0ZX. instead, so I want to handle both. |
00:21.57 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
00:22.17 | xacatecas | the way I handled that until now was to have two exten lines, one for the 27 case and one for the 0 case ... i was hoping for a more elegant solution. |
00:22.28 | fujin | hm, setinterfacevar might do it. |
00:22.59 | SteveTotaro | my suggesting is totally workable unless you are in some kind of sweatshop outbound call center |
00:23.30 | SteveTotaro | hangups every couple of seconds |
00:23.58 | xacatecas | ok, i'm betting some sleep is going to get more done for me than trying to work more. |
00:24.01 | xacatecas | adios |
00:24.15 | fujin | hum |
00:24.17 | mipster | xacatecas, ok, hmm. in that case how about a exten=> 0ZX.,n,Goto(context-for-extenstions, 27${EXTEN:1}, 1) |
00:24.31 | SteveTotaro | he's gone in the wind |
00:25.09 | jblack | I would have told him I think he's missing a leading _ |
00:25.15 | mipster | oops |
00:25.16 | SteveTotaro | goto is the best thing evar invented in programming |
00:25.20 | mipster | good catch |
00:25.29 | mipster | I always forget the danged _ |
00:25.34 | fujin | heh. Stupid MONITOR_FILENAME is evaluated before a call is delivered. |
00:25.40 | fujin | what a failburger |
00:27.00 | SteveTotaro | obviously if you are using monitor in asterisk you cannot be doing more that ~70 simultaneous calls (without ramdisk or some hack) right? |
00:28.33 | SteveTotaro | you could do a little AMI magic to catch the event |
00:28.50 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
00:29.25 | fujin | mm |
00:29.37 | fujin | It'd be better to have monitor_format in the queues.conf, really |
00:29.44 | fujin | and it's evaluated @ mixdown time |
00:29.47 | fujin | not the case |
00:29.51 | fujin | will just have to ignore the agent id |
00:30.30 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
00:31.45 | SteveTotaro | sean has a way of making a room quiet |
00:35.27 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.135.38) |
00:38.11 | *** join/#asterisk lpmusic (n=dballeng@reddy.d-11.denetron.net) |
00:38.48 | Mavvie | .... silly users are reporting crossed calls again (where the RTP stream gets delivered to the wrong endpoint). |
00:38.52 | *** join/#asterisk draygon (n=lokbo@76.185.106.151) |
00:39.01 | Mavvie | Anybody here knowledgable with the way to troubleshoot this? |
00:39.11 | BobLutz | raises eyebrow |
00:40.07 | CCFL_Man2 | wctdm can be used with what hardware? |
00:40.38 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
00:41.06 | BobLutz | CCFL_Man2, TDM |
00:42.44 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
00:43.32 | SteveTotaro | Mavvie: this could very well be happening |
00:44.04 | CCFL_Man2 | BobLutz: does that include ant T1 cards? |
00:44.48 | SteveTotaro | i had the same complaint and thought "silly user" but then when all calls were recorded, it was proven that calls were being crossed |
00:44.53 | BobLutz | CCFL_Man2, Never touched a t1, If it has TDM in the model name, I would think it could work |
00:45.08 | Mavvie | SteveTotaro: I know it happens, that's for sure. But I don't know how to troubleshoot it. |
00:45.16 | Mavvie | Or even where to start with gathering information. |
00:45.29 | SteveTotaro | i never figured it out either Mavvie ;) |
00:45.37 | Mavvie | wrong answer :-P |
00:45.42 | CCFL_Man2 | BobLutz: ahh, i want to create a T1 from the asterisk box |
00:45.48 | SteveTotaro | it was so rare and everything looked fine in the logs |
00:46.02 | SteveTotaro | i could not reproduce it |
00:46.36 | SteveTotaro | wctdm works for pots lines |
00:47.22 | JT | yeah not TDM lines |
00:47.27 | JT | asterisk naming is weird like that |
00:47.38 | BobLutz | feels liek :-[ |
00:47.43 | CCFL_Man2 | ahh shit |
00:48.10 | robmac67 | SteveTotaro & Mavvie: I believe that this was identified as a bug and fixed in the latest 1.4.19 Release Candidate |
00:48.23 | CCFL_Man2 | i can't use a T1 card with solaris |
00:48.24 | Mavvie | robeph: oh! |
00:48.28 | Mavvie | robmac67: oh! |
00:48.45 | SteveTotaro | but there are 29 new bugs introduced |
00:49.12 | SteveTotaro | i don't think i could run an RC in production |
00:49.38 | SteveTotaro | i would live with the random crossed call |
00:49.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
00:49.50 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
00:49.55 | robmac67 | From what I recall it was introduced in 1.4.18 but you would have to check that out - I don't know what .17 was like |
00:50.05 | SteveTotaro | CCFL wctdm is really just for the pots cards |
00:50.15 | Mavvie | SteveTotaro: RCs are just for that. |
00:50.19 | lpmusic | with either 1.4 or 1.6 can you change the announcement to the agent answering a queue to make it play the announcement after you pickup but before you hit pound to accept the call? |
00:50.27 | SteveTotaro | other cards use wct4xxp and the like |
00:50.53 | SteveTotaro | maybe a release candidate is that to you but not to me |
00:50.53 | CCFL_Man2 | SteveTotaro: sucks, wonder if i can build those drivers for solaris |
00:51.32 | SteveTotaro | CCFL i bet google can tell you, i know you can run asterisk on solaris |
00:51.51 | SteveTotaro | i had a netra 100 running asterisk |
00:52.12 | *** join/#asterisk adeeln (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
00:52.17 | JT | SteveTotaro: zaptel is the issue |
00:52.27 | SteveTotaro | yes obviously |
00:52.40 | SteveTotaro | that is why i told him to google |
00:52.51 | Mavvie | robmac67: just gone through the changes, but it's as clear as mud which one it could be. |
00:52.59 | CCFL_Man2 | SteveTotaro: i'm trying to run it on a netra t1 200, solarisvoip.com has the stuff, but only tdm driver is tdm over ethernet |
00:53.31 | JT | CCFL_Man2: let me know how it goes :) |
00:53.40 | CCFL_Man2 | JT: lol! |
00:53.41 | JT | i have a pile of sun netra T1s lying about |
00:54.06 | CCFL_Man2 | sun claims they are "carrier grade" |
00:54.33 | JT | real carriers don't terminate to standard pci cards |
00:54.48 | SteveTotaro | http://www.voip-info.org/wiki/index.php?page=Asterisk+Solaris+Support |
00:55.28 | CCFL_Man2 | JT: i want to generate a T1 on the netra, not terminate |
00:55.33 | CCFL_Man2 | SteveTotaro: yeah |
00:55.50 | SteveTotaro | you are on the cutting edge my friend |
00:56.04 | JT | sure, but it is totally different aspects that get them the carrier grade certification :) |
00:56.19 | CCFL_Man2 | SteveTotaro: lol |
00:56.31 | CCFL_Man2 | JT: i know, wtf do telco's use them for? |
00:56.36 | SteveTotaro | any reason why you don't do it with other hardware? |
00:56.50 | CCFL_Man2 | i don't care for linux |
00:56.54 | JT | CCFL_Man2: running software on |
00:56.59 | SteveTotaro | ouch |
00:57.15 | JT | CCFL_Man2: they have DC models and all that |
00:57.22 | SteveTotaro | i am waiting for a VxWorks port myself, probably will be waiting a long time |
00:57.51 | CCFL_Man2 | JT: yeah, but what software? |
00:58.00 | CCFL_Man2 | SteveTotaro: ouch |
00:58.20 | JT | CCFL_Man2: expensive proprietary carrier software, that runs on solaris |
00:58.23 | JT | databases |
00:58.28 | JT | operations and maintenance |
00:58.36 | SteveTotaro | that is the OS running satellites and junk in space |
00:58.51 | rkeene | What's up with DUNDI.com redirecting to Digium ? |
00:59.06 | SteveTotaro | DNS FUBAR |
00:59.14 | rkeene | Oh |
00:59.16 | SteveTotaro | but they own dundi |
00:59.25 | SteveTotaro | (tm) |
00:59.27 | CCFL_Man2 | JT: in the CO? |
00:59.28 | BobLutz | Why? |
00:59.29 | rkeene | I was expecting to find useful information there :-P |
00:59.59 | SteveTotaro | probably find more useful info on someone's blog |
01:00.23 | SteveTotaro | or howto, i know there is a large group out there for dundi |
01:00.30 | SteveTotaro | forget the name though |
01:00.37 | riddlebox | rkeene, just wait till [TK]D-Fender comes around |
01:01.01 | JT | CCFL_Man2: sure |
01:01.01 | CCFL_Man2 | SteveTotaro: carrier access uses it on their adit 600 tdm card, motorola uses it on their satellite receivers too, both do run rock solid stable |
01:01.23 | outtolunc | 'dundi, so easy a caveman could do it' <G> |
01:01.23 | JT | riddlebox: you sure that would help? :P |
01:01.31 | SteveTotaro | 3com uses it on the NBX and V3000 PBXs |
01:01.57 | SteveTotaro | i think fender is part of that large group i was speaking of |
01:02.43 | CCFL_Man2 | JT: think they use any telecom software on it? |
01:03.22 | JT | CCFL_Man2: yes heaps of telecom software runs on solaris |
01:03.32 | JT | however not most of the stuff in the media path |
01:03.41 | JT | that's usually embedded with custom firmware |
01:03.55 | CCFL_Man2 | ahh |
01:04.00 | SteveTotaro | (busybox) |
01:04.16 | JT | lol |
01:04.27 | SteveTotaro | actually probably VxWorks |
01:04.36 | JT | or their own stuff |
01:04.52 | mipster | Anyone set up a Portech 370 GSM-SIP Gateway? |
01:05.05 | mipster | Seems it's delivering CIDNAME and CIDNUMBER backwards |
01:05.25 | mipster | IOW, Name is coming across as number and vice versa |
01:06.19 | SteveTotaro | mipster: that would be so low on my priority list as long as all the info is getting there |
01:06.24 | SteveTotaro | :) |
01:06.45 | mipster | Yeah, its only a minor hassle. Just wondering if it's an AT&TW thing or a Portech thing |
01:06.54 | JT | probably a setting or bug on the portech? |
01:07.20 | SteveTotaro | you could easily work around it if all you need to support is portech |
01:07.27 | mipster | I don't see a setting. I'm guessing a bug. Just wondered if anyone had used one with another carrier and/or SIM card |
01:07.35 | mipster | and if the NAM and NUM are reveresed |
01:07.58 | SteveTotaro | it is a feature |
01:08.02 | SteveTotaro | ;) |
01:08.03 | mipster | lol |
01:09.13 | draygon | does anyone need a server in Dallas, TX? I have way too much space that im not using |
01:09.38 | SteveTotaro | how much for how much |
01:09.40 | SteveTotaro | ? |
01:10.02 | draygon | i pmed you |
01:12.53 | rkeene | I have my Polycom phones with the minibrowser all setup now... what are some popular/useful things to do with that ? |
01:13.43 | SteveTotaro | display the company logo |
01:14.03 | rkeene | Useful ? :-P |
01:14.15 | JT | draygon: just space, or a server too? |
01:14.29 | SteveTotaro | to the ceo, heck yeah, they eat that stuff up |
01:14.40 | rkeene | We don't have a CEO |
01:14.45 | SteveTotaro | you could display weather info |
01:15.04 | SteveTotaro | i mean what is really useful in a minbrowser? |
01:15.05 | draygon | JT, colo or dedicated whichever you prefer. |
01:15.13 | JT | draygon: ah cool, how much? |
01:15.22 | SteveTotaro | when you have a pc right next to the phone? |
01:15.28 | rkeene | I dunno, I was thinking like an interface to change preferences or something |
01:15.38 | draygon | check pm |
01:16.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
01:16.12 | SteveTotaro | get FOP working on it with touchscreen |
01:16.33 | rkeene | I already set it up so they can change their SIP password through a special number :-P |
01:16.48 | CCFL_Man2 | can the wcte11xp driver use the new te205p? |
01:16.55 | SteveTotaro | cool, i don't give them options like that |
01:17.09 | rkeene | I want to be out of the business of dealing with phones |
01:17.22 | rkeene | So I want to give them as much control as possible :-P |
01:17.27 | SteveTotaro | then you will be more in the business when they blow it |
01:17.44 | SteveTotaro | what's my SIP password??? |
01:17.46 | rkeene | I already pull down all the user information from the LDAP server, hee hee |
01:18.07 | SteveTotaro | ah, you are ahead of the curve |
01:18.09 | rkeene | They can change it from someone else's phone, using their VM password |
01:18.28 | rkeene | And the documentation for this will be available... maybe even through the mini-browser ! :-P |
01:18.58 | *** part/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
01:19.05 | fujin | Is there anyway to 'automon' mixmonitor? |
01:19.13 | SteveTotaro | point them to sharepoint? |
01:19.24 | tainted_ | can someone help me with DTMF? |
01:19.35 | SteveTotaro | relaxdtmf=yes |
01:19.45 | SteveTotaro | dtmf=rfc2833 |
01:19.58 | rkeene | Sharepoint ? |
01:20.22 | tainted_ | SteveTotaro relaxdtmf is for tdm stuff |
01:20.36 | SteveTotaro | thought you were an M$ shop, sharepoint is like a intranet |
01:20.43 | rkeene | (All our documentation is generated using LyX, and available in HTML, PDF, or dead-tree format) |
01:21.02 | rkeene | No... We use Slackware for almost everything. |
01:21.11 | SteveTotaro | well you were not very specific with your question tainted |
01:21.12 | rkeene | (Well, and Cisco IOS and Foundry OS) |
01:21.25 | rkeene | (We're the network operations group) |
01:21.33 | tainted_ | SteveTotaro one user is complaining that dtmf doesn't work |
01:21.41 | tainted_ | i am using g729 / rfc2833 |
01:21.49 | tainted_ | and it works for everyone else |
01:22.12 | riddlebox | tainted_, I always have to use inband |
01:22.25 | tainted_ | but 729 doesn't support inband :( |
01:22.30 | fujin | uh, so, I found this bug http://bugs.digium.com/view.php?id=10185 re: automixmonitor |
01:22.33 | riddlebox | ahh |
01:22.35 | fujin | but I can't see that in my svn checkout |
01:22.37 | fujin | has it gone into 1.6 isntead? |
01:23.11 | tainted_ | riddlebox do you ever find out what causes rfc2833 to work for some but not for others? |
01:23.22 | *** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
01:23.39 | riddlebox | tainted_, nope, I just know with my phones inband always works |
01:23.56 | tainted_ | riddlebox hard or soft phones? |
01:23.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fd19466b33124673) |
01:24.04 | riddlebox | hard |
01:25.32 | SteveTotaro | hey riddlebox, ever get a chance to dig deeper into that project, just curious |
01:26.06 | mipster | what's the other dtmf option? notify? |
01:26.26 | riddlebox | SteveTotaro, nahh I have been real busy and havent had a chance to do much |
01:26.38 | mipster | info |
01:27.15 | mipster | tainted, any chance that users set has dtmfmode set to something wonky? |
01:27.26 | SteveTotaro | gotcha, well keep me informed, i plan on making it live to the public this month |
01:27.52 | riddlebox | cool |
01:27.56 | riddlebox | congrats |
01:28.57 | fujin | Anyone know if it's possible to use ODBC storage to write to one database, but read off another one, for the same thing? |
01:29.11 | fujin | e.g.; for voicemail, I want it to write to an upstream db which is replicated locally, but read off the local one |
01:29.36 | CCFL_Man2 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=290213819512 <--my new asterisk phone |
01:29.56 | mipster | does it support G.729a? |
01:30.12 | mipster | dtmfmode=rotary |
01:30.14 | CCFL_Man2 | no, just pots |
01:31.01 | SteveTotaro | they need a cover pump for their above ground |
01:31.06 | outtolunc | i'd like to see you *clip* that to your ear <G> |
01:31.26 | CCFL_Man2 | SteveTotaro: they do |
01:31.53 | outtolunc | that pool cover looks about ready to cave in also <G> |
01:31.56 | mipster | So will an FXS car recognize pulse dialing? |
01:32.08 | mipster | s/car/card/ |
01:32.09 | SteveTotaro | the silliest thing is the "air pillow" in the middle |
01:32.20 | SteveTotaro | pulse is supported |
01:32.39 | SteveTotaro | they sell those airpillow like it keeps water off the cover or something |
01:33.13 | SteveTotaro | at best it just pools the water more to the sides |
01:33.23 | SteveTotaro | i prefer looploc |
01:36.18 | outtolunc | ponders offloading the extra 40 pounds worth of stuff in my backpack so i do not have to hump it around at von <G> |
01:38.41 | mipster | good evenin' folks... |
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01:38.52 | *** part/#asterisk mipster (n=mipster@75.131.201.166) |
01:53.39 | CCFL_Man2 | i can't find the subset i want |
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02:03.31 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
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02:05.10 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
02:05.10 | *** mode/#asterisk [+o lmadsen] by ChanServ |
02:09.30 | really_phukt | hallo? |
02:10.45 | really_phukt | lmadsen: hey man, I was just reading section of your book about templates for conf files |
02:10.58 | really_phukt | does this apply to 1.2 version too? |
02:11.46 | lmadsen | really_phukt: maybe... I haven't used 1.2 in almost 2 years |
02:11.54 | lmadsen | I was an early adopter of 1.4, around 1.4.0 |
02:12.07 | lmadsen | you could try it and find out.... |
02:12.21 | *** mode/#asterisk [-o file] by lmadsen |
02:12.36 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
02:12.39 | outtolunc | aaaaattack |
02:12.46 | lmadsen | indeed :) |
02:13.05 | really_phukt | lmadsen: I could try it... mess it up and get fired... at least I will know ;) |
02:13.23 | lmadsen | really_phukt: so you have absolutely no test servers then... you should certainly be fired then |
02:13.48 | lmadsen | how appropriate of a name to be doing testing on your production servers |
02:13.58 | really_phukt | LOL |
02:14.13 | lmadsen | I don't remember adding a smiley face to the end of my sentences |
02:14.27 | jblack | No. The company he works for would be really_phukt. |
02:14.40 | lmadsen | ah I see |
02:14.58 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
02:15.05 | draygon | jblack, arent you also on DALnet? |
02:15.13 | drmessano | Must be a dot-com |
02:15.17 | jblack | No, I am not on dalnet |
02:15.20 | drmessano | Like overnightoothbrushes.com |
02:15.23 | drmessano | or |
02:15.36 | drmessano | electrickazoofantasticandthemonkey.com |
02:15.54 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
02:16.45 | jblack | I think dundi could use an accounting system like gnunet has. That could render dundi useful |
02:18.28 | JT | dundi seems like a lot of hype |
02:19.56 | jblack | well, it's kinda unusable in any sort of public manner. |
02:20.27 | jblack | But I think a trust accounting concept (which is poorly named), could make it rather usable |
02:20.45 | JT | enum seems to be where things are going |
02:21.08 | *** join/#asterisk chendy (n=chendy@121.35.51.33) |
02:21.40 | jblack | Yah. dundi hasn't been much of a competitor for enum (which has its own set of flaws) |
02:22.10 | jblack | anyone curious to hear more about trust accounting? |
02:22.52 | *** mode/#asterisk [-o lmadsen] by ChanServ |
02:27.19 | drmessano | Well |
02:27.26 | drmessano | One person deopped and two more quit |
02:27.31 | drmessano | I'll take that as a "no" |
02:28.20 | jblack | Yeah. That's why I kept my mouth shut. :) |
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03:07.15 | Iamnacho | test |
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03:09.20 | *** join/#asterisk ciphercast (n=cipherca@151.204.63.64) |
03:10.31 | ciphercast | hey guys |
03:10.49 | ciphercast | anyone set up a copy of 1.6b5? |
03:11.23 | ciphercast | i'm transitioning from 1.4, and i am trying to diagnose a problem |
03:12.02 | ciphercast | i want to make sure my problem is *not* a bug |
03:12.28 | ciphercast | for the Realtime configuration, is the Goto syntax still the same? |
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03:20.38 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
03:20.52 | ShadowHntr | got a question re: zaptel/digium card support |
03:21.08 | ShadowHntr | is the kernel code for x86 only, or would the module work (when compiled properly) on other architectures |
03:22.39 | ShadowHntr | on linux |
03:22.48 | ShadowHntr | i just acquired some old PPC hardware and thought about asterisk :) |
03:23.44 | ciphercast | I'm sure you can compile the driver for a different arch |
03:24.18 | ShadowHntr | cool cause last time i looked (i probably misread it) is that the support was for linux x86 only |
03:24.27 | ShadowHntr | i'd like to try it on linuxppc |
03:24.35 | ciphercast | digium uses the zaptel driver |
03:25.07 | ciphercast | yeah...hmm (digging thru ML's) |
03:25.43 | shido6 | good luck with that |
03:26.12 | ShadowHntr | thx |
03:26.13 | ciphercast | what I would try doing is compiling zaptel on the ppc unit |
03:26.26 | ShadowHntr | yeah i'm gonna probably try out yellow dog and compile a new kernel |
03:26.29 | *** join/#asterisk ccvp (n=axe@user-24-214-126-81.knology.net) |
03:26.34 | ShadowHntr | i just acquired a free Power Mac G3 300MHz unit |
03:27.20 | ciphercast | i got a g4 lying around with a busted psu |
03:27.46 | *** join/#asterisk ChrisTSIS (n=killa666@24.182.21.208) |
03:28.07 | ciphercast | its pretty quiet in here today |
03:29.57 | ShadowHntr | ciphercast: can i have it? ;) |
03:29.58 | ShadowHntr | j/k |
03:30.28 | ciphercast | location? :) |
03:31.20 | ciphercast | ShadowHntr: unless you're in the nyc metro area, its not worth you're trip ;-) |
03:31.26 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
03:31.47 | ShadowHntr | ciphercast: TN/Nashville here |
03:31.52 | ShadowHntr | though i do have a friend in DC... :() |
03:31.53 | ShadowHntr | er |
03:31.53 | ShadowHntr | :) |
03:31.58 | husimon | Hey does anyone know how to modify the voicemail so that if someone checks the voicemail but hangs up before they finish listening to it that the wmi won't go off? |
03:32.09 | husimon | I have a group of people that share a line and they need to see if a message is for them or not |
03:32.30 | husimon | the old system I had let you call in and listen to the message and if it wasn't yours hang up and it will count it as a new message |
03:32.57 | ciphercast | husimon: first off, good question. I'm not entirely sure, but can you give them diff vm boxes? |
03:33.06 | [TK]D-Fender | Hasn't ceased to amuse me how so many masochists drag out decrepit non-compatible gear and keep beating away at it in some vain attempt to redeem crap and save a buck. |
03:34.14 | husimon | [TK]D-Fender: ya I'm tossing all the old sccp phones in a few weeks |
03:34.24 | ciphercast | well, asterisk has been running fine @ home on my '97 era 600mhz ibm celeron |
03:34.30 | [TK]D-Fender | husimon, those HAVE a prayer.... you might not, but they do :) |
03:34.43 | [TK]D-Fender | ciphercast, See, its at least x86 ;) |
03:34.44 | ciphercast | laughs in amazemet |
03:34.51 | husimon | [TK]D-Fender: well I have a few where all the softkeys don't work |
03:35.25 | [TK]D-Fender | ciphercast, And I was buying C466's at the office in '99.... I think your calendar needs some adjusting :) |
03:36.09 | husimon | would be nice to be able to mark messages as new |
03:36.48 | husimon | [TK]D-Fender: any idea if that is possible for a voicemail message? or where I should look to find out |
03:37.32 | [TK]D-Fender | husimon, Have you actually LOOKED at the files? |
03:37.44 | husimon | the voicemail data files? |
03:38.09 | ciphercast | eh, this is true. must be the guinness talking ;-), prolly round '01 then. all i know is that box has been running asterisk since 1.0...purring in the basement |
03:38.10 | ccvp | well this is weird, I just got a job offer to go from $20/hour to $35 hour doing something |
03:38.15 | ccvp | i dont even have experience doing |
03:38.20 | husimon | ccvp can you learn it ? |
03:38.27 | ccvp | going from network engineering, to doing just backups in a unix environment |
03:38.27 | husimon | i think that's the real question :P |
03:38.29 | ccvp | at some huge defense company |
03:38.34 | ccvp | after they do missile firing simulations |
03:38.54 | husimon | do you have unix experience? |
03:38.56 | ccvp | all this will be is backing up that data using some unix stuff, premade scripts, no knowledge necessary except being "an all around generalist" |
03:38.56 | ccvp | heh |
03:39.17 | ccvp | just for recreation at home, 5 years |
03:39.17 | ccvp | im ccna, ccvp |
03:39.18 | ciphercast | thinks anything Un*x is never *kust* lol |
03:39.25 | ciphercast | ugh, just |
03:39.25 | husimon | i don't see why you wouldn't take it then |
03:39.34 | ccvp | well its something ive never done, its just weird |
03:39.39 | ccvp | how this friend of mien can get people in this easily |
03:39.41 | outtolunc | notes The previous reload command is slow as hell, and i wish it would hurry up! <G> |
03:39.50 | ChrisTSIS | Are there any known software/kernel version issues with the VPMADT032 that causes it to act more like echo suppression instead of cancellation? |
03:40.22 | ccvp | would you ditch yoru fortay of network engineering to go into a new area of IT, that is practically double the pay |
03:40.42 | ciphercast | ccvp: I would do it, as long as initally its not ttl |
03:41.07 | ccvp | i told him im not a engineer w/ EE degree or CS |
03:41.24 | ccvp | im just 3 years of MIS degree, with 1 year left, but aparently his boss is wanting people with business experience |
03:41.56 | ccvp | he was like so what, you'd be surprised how many we have here that are like that, because their all contract jobs |
03:41.57 | husimon | ccvp i'd sure consider it unless you see yourself getting a big raise soon |
03:42.04 | ccvp | government pays them massive amounts for the position |
03:42.11 | ccvp | then they hire someone like me cheaper, while the pay is still high in my eyes |
03:42.19 | ciphercast | yes |
03:42.49 | husimon | is it contract or yearly or how long will they hire you |
03:43.26 | ciphercast | every time they fire a missle...heh |
03:43.42 | ccvp | it's testing for for the GMD Program (Ground-Based Midcourse Defense), a DoD Missile Defense contract. |
03:43.54 | ccvp | they'll get me a secret clearance too |
03:43.59 | ccvp | so thats massively good for my future |
03:44.25 | ccvp | thats a paste from his email |
03:44.27 | husimon | if you are going to do more defense work |
03:44.50 | ccvp | well lets just say im sick of 20/hour |
03:45.03 | ccvp | for 4 years of what i do, this job got me at a cheap rate years ago |
03:45.07 | ccvp | and cant give a big jump |
03:45.42 | husimon | ccvp i'm sick of 27 |
03:45.44 | ccvp | so you say why im confused, going from network engineering to wtf? a unix backup specialist? |
03:45.51 | ccvp | s/say/see |
03:45.59 | ccvp | no clue what the title would be |
03:46.25 | husimon | ccvp you neec ccie to make the big bux with cisco |
03:46.30 | *** join/#asterisk _alex_df_ (n=Alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
03:46.36 | ccvp | well that is beyond my brain |
03:46.40 | ccvp | its to much thinking/studying |
03:46.49 | ciphercast | that's a hell of a title |
03:46.50 | husimon | at least that's what I would think would be the big bux |
03:46.59 | ciphercast | *so* much work |
03:47.03 | ccvp | screw it |
03:47.09 | ccvp | im already probably 3x underpaid what i do now |
03:47.10 | husimon | i'd say go for it |
03:47.12 | ccvp | thing is, i haven o degree |
03:47.23 | ccvp | famlyfriend got my current job, thats why im crazy low hourly rate |
03:47.24 | husimon | you can always use that degree again in a few years |
03:47.32 | *** part/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
03:47.35 | husimon | just take a refresher recourse |
03:47.41 | husimon | recourse = course |
03:47.42 | ccvp | im 20/hr, i should probably be at 45/hour |
03:47.47 | ccvp | doing call manager daily |
03:47.59 | ccvp | im told at this big company when degree is complete |
03:48.04 | ccvp | they give a substansial boost |
03:48.07 | ccvp | like 25-30% |
03:48.20 | ciphercast | and you have 1 yr left? |
03:48.23 | ccvp | 1.5 about |
03:48.33 | ccvp | MIS, its scaled down comp sci basically |
03:48.45 | ccvp | with 3 levels of stat classes, some accounting classes, databases, business |
03:49.04 | ccvp | mixed with programming here/there, and management theory classes |
03:49.08 | ciphercast | and the GMD program lasts how long? |
03:49.11 | husimon | i'd take the new job and complete the MIS degree |
03:49.24 | husimon | yeah that's what i was wondering, if it's only 6 mo |
03:49.27 | husimon | yuck |
03:49.37 | ciphercast | then go back after the contract expires :) |
03:49.47 | ccvp | doubt that'll happen, these are questions to ask |
03:50.20 | ciphercast | getting that security clearance is completely worth it |
03:50.40 | ccvp | he says if i get a offer and i accept |
03:50.40 | husimon | ya i'm pissed because all the jobs here that are worth a crap require you to already have a security clearence |
03:50.43 | ccvp | they get me that |
03:50.46 | husimon | so how the hell do you get one |
03:50.54 | ccvp | well u apply for a job |
03:50.55 | ciphercast | its hard |
03:50.56 | ccvp | that needs one |
03:51.04 | ciphercast | & | expensive |
03:51.07 | ccvp | and they agree to get you one |
03:51.07 | husimon | basically they probably mostly hire inside people |
03:51.09 | ccvp | costs about 10k |
03:51.13 | ccvp | for the company |
03:51.28 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-54-169.lns10.syd7.internode.on.net) |
03:51.30 | husimon | which is honestly not that much |
03:51.33 | husimon | compared to your salary |
03:51.38 | ccvp | well my friend told me, if i ever leave this potential one |
03:51.41 | ccvp | after i get the clearance |
03:51.48 | ccvp | the secret clearance = gold in this town |
03:51.53 | ciphercast | yeah |
03:52.19 | ciphercast | ...but they interview every inch of your life |
03:52.21 | ccvp | plus the graduate program for MIS at my school |
03:52.24 | ccvp | gets you an IA certificate |
03:52.35 | ccvp | and the training necessary for CISSP in place of a MIS masters |
03:52.49 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
03:52.59 | ccvp | ive already seen jobs on monster in town for a siprnet admin |
03:53.02 | ccvp | but needs SCI+cissp |
03:53.06 | ccvp | 135k+/yr |
03:53.14 | husimon | don't count your eggs heh |
03:53.21 | ciphercast | lol |
03:53.22 | rkeene | is a SIPRNET admin :-P |
03:53.29 | rkeene | (It's not very interesting) |
03:53.35 | JT | just a hint |
03:53.36 | ccvp | rkeene, so you got sci? |
03:53.54 | JT | if you're really interested in jobs with any sort of classification clearance |
03:54.00 | JT | don't talk about them on irc |
03:54.01 | JT | :) |
03:54.18 | ccvp | well i see, note taken |
03:54.26 | ciphercast | as long as you're bouncing thru tor, you're fine :) |
03:54.32 | rkeene | SIPRNET is Secret, not SCI. |
03:55.06 | JT | i know here, going for jobs with certain agencies, you are not even allowed to tell friends and family that you have *applied* for the job or it preclude you from being considered |
03:55.22 | ccvp | jt, why you trying to make me paranoid now |
03:55.23 | ccvp | lol |
03:55.28 | ccvp | like i axed my shit |
03:55.44 | ccvp | i see what your saying, but that spooked me |
03:55.55 | rkeene | has friends with many agencies and has known about it ahead of time for all of them... since I was notified as part of their background |
03:56.04 | JT | i'm not sure what sort of secrecy is involved with the job you're going for, but don't tell randoms what you're going to be doing :) |
03:56.43 | ccvp | well its all done then |
03:56.44 | ciphercast | essentially, JT's right... |
03:56.49 | ccvp | I know |
03:56.51 | rkeene | (Including NSA and IC) |
03:57.08 | ccvp | i should of thought about that,but who tells you this |
03:57.14 | ccvp | as a start? i wouldnt have thought about that |
03:58.18 | ciphercast | anyone mess around with 1.6 yet? |
03:58.20 | rkeene | Anyway, if you do get the job, don't put SCI on SIPR -- it's a spill and a PITA. |
03:59.10 | ccvp | so how far back to secrets/Ts's go |
03:59.15 | ccvp | in to your life? 5 years, 10,15? |
03:59.25 | mosty | rkeene |
03:59.26 | rkeene | 7, and 15, IIRC |
03:59.42 | rkeene | (with periodic refiles) |
03:59.45 | rkeene | mosty |
03:59.51 | ccvp | i remember about 8 years ago, some military officer of some sort called me one day |
03:59.55 | ccvp | asking personal questions about my neighbor |
04:00.07 | ccvp | mom told me he does intelligence stuff at some fort back home |
04:00.12 | ccvp | so that was part of his BG check? |
04:00.38 | rkeene | Usually the FBI does the background check |
04:06.10 | *** part/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
04:20.07 | _alex_df_ | hello, 1.4 noob but been using asterisk in production since pre-1.0. Today we put our first 1.4 server online. This is a SIP to PRI system. Past 110 or so active calls, core show channels was no longer able to show them all, and no summary at the end. A few minutes later, * I began loosing my SIP registrations and core show channels would show only a couple and still no summary. Any pointers where to start debugging this? |
04:21.15 | shido6 | get openser and leave asterisk setup as a feature server |
04:21.23 | shido6 | let openser handle your registrations |
04:21.43 | mosty | _alex_df_, probably a deadlock, i had similar issues with 1.4 |
04:22.48 | _alex_df_ | mosty, did you go back to 1.2? |
04:23.28 | _alex_df_ | shido6, had same setup with less hardware working with 1.2, but yeah openser might be the way to go |
04:24.07 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
04:25.03 | rkeene | Has anyone tried YXA. |
04:25.10 | rkeene | Err, has anyone tried YXA ? |
04:25.42 | Kumbang | hi guys, how can i get variable channel number , eg 23 from Zap/23-1 |
04:25.59 | mosty | _alex_df_, yes |
04:26.24 | _alex_df_ | mosty, was afraid you were going to say that :P |
04:27.46 | Kumbang | i want to get just channel number 23 not 23-1 from Zap/23-1 etc |
04:27.58 | Kumbang | number 1 from Zap/1-1 |
04:28.03 | Kumbang | how can i do this |
04:28.41 | _alex_df_ | Kumbang, might be an easier way to do it, but it can be done running it through cut twice |
04:30.45 | rkeene | echo Zap/23-1 | sed 's@^[^/]*/\([0-9]*\)-[0-9]*$@\1@' |
04:30.53 | ccvp | c:\documents&settings\kumbang\my documents\my-list-of-adult-sites-logins-pw's.txt |
04:31.00 | ccvp | waves |
04:36.28 | drmessano | fail |
04:37.29 | drmessano | C:\Documents and Settings\kumbang\Documents\my-list-of-adult-sites-logins-pw's.txt |
04:38.01 | drmessano | That's how you pwn someone |
04:38.17 | *** join/#asterisk sergey (n=chatzill@sergey.iks.ru) |
04:40.12 | lmadsen | for anyone who reads this... the way to do it this like this: Set(CHAN_NUMBER=${CUT(CUT(CHANNEL,/,2),-,1)}) |
04:40.23 | lmadsen | I do that all the time |
04:41.11 | lmadsen | too bad Kumbang didn't stay longer |
04:41.42 | rkeene | Oh, I didn't know you meant in Asterisk :-P |
04:41.56 | drmessano | ccvp ran him off |
04:41.57 | lmadsen | staying a whole 14 minutes will get you no where |
04:41.58 | ccvp | i get the feeling "kumbang" is a native phrase or word in indonesian |
04:42.04 | ccvp | and not what us american perverts think it means |
04:42.05 | lmadsen | aye |
04:42.05 | ccvp | hahaha |
04:42.09 | lmadsen | exactly :) |
04:42.35 | drmessano | kumbang is a providence of bangkok |
04:44.21 | *** part/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
04:45.18 | rkeene | Has anyone tried YXA the Erlang-based SIP router with Asterisk ? |
04:45.38 | JT | no, but let me know if you do |
04:45.40 | JT | sounds interesting |
04:45.59 | rkeene | I'm not sure I can justify it |
04:47.11 | rkeene | I'll setup a new test bed once I roll the current one into production, and test it then |
04:48.14 | lpmusic | with either 1.4 or 1.6 can you change the announcement to the agent answering a queue to make it play the announcement after you pickup but before you hit pound to accept the call? |
04:55.15 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
05:02.52 | BobLutz | LOL kumbang |
05:03.03 | BobLutz | o shoot I just woke up |
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05:11.05 | the_5th_wheel | hi. How can i forward all calls coming in on my zap interfaces with a certain context to a macro? |
05:12.01 | jql | [zap-channel] exten => s,1,Macro(foo) |
05:12.09 | jql | or rather, use i |
05:12.22 | jql | up to you |
05:13.38 | the_5th_wheel | in wich file would i do that? |
05:14.25 | jql | you modify extensions.conf to add the desired context, and perhaps you need to modify zaptel.conf to set the default context for incoming calls |
05:14.48 | jql | and by that, I mean zapata.conf |
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05:21.15 | mvdk | Does anyone know about what they intend to do instead of applying the codec negotiation patch (bugs.digium.com/view.php?id=4825)? |
05:23.52 | mvdk | Qwell: You closed issue 4825. Could you please elaborate on what's going to be implemented instead? |
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05:37.56 | mosty | mvdk, you can implement it in the dialplan manually in 1.4 |
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05:42.09 | CCFL_Man2 | who was it in here that collected vintage phones? |
05:43.19 | jameswf-home | hates phones |
05:43.33 | CCFL_Man2 | jameswf-home: why? |
05:44.18 | jameswf-home | they are awful little things that awful little people annoy you.. |
05:45.06 | jameswf-home | at home I dont answer the phone I let my wife or vm get it |
05:45.25 | jameswf-home | If you want to talk to me that bad you can email me |
05:47.15 | TJNII | methinks jameswf-home has had a little too much "Irish spirit," if you know what I mean. |
05:47.27 | jameswf-home | I wish :( |
05:47.36 | jameswf-home | too poor to drink |
05:48.30 | TJNII | You're not too poor to drink, you're just now willing to drink within your proce range. |
05:48.48 | TJNII | s/now willing/not willing/ |
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05:53.43 | mvdk | mosty: Suppose I have two offices, with an asterisk server at each running IAX to each other, and a SIP provider that uses g.729. Clients on the inside know how to use speex and how to use g.711. How do I get (a) A client in one office to call one in the other office using speex, (b) A client in an office calling a PSTN line to use g.711 so that it may be translated by the server to g.729 |
05:56.00 | mosty | i can't tell you how to make the client decide which codec to use, but in your asterisk dialplan you can examine the current codec for a call, then when you forward that call on you can use a sip or iax account that only supports the codec you want to use |
05:56.27 | TJNII | Uuh, isn't that the whole point of allow and disallow in both sip and iax.conf? |
05:57.04 | mvdk | TJNII: Not quite. The whole point, TJNII, is to use *different* codecs depending on the destination |
05:57.48 | mvdk | In other words, make the client use a different codec |
05:58.00 | mvdk | The way I found to do it was to use SER |
05:58.12 | mvdk | And make the clients register with it |
05:58.35 | mvdk | And when they try to dial, pass to the asterisk server only those codecs that apply |
05:59.07 | mvdk | So at call setup time, trim the offer list to just the ones that support the aim we're undertaking |
06:00.16 | mvdk | But the dial plan doesn't allow you to do what I'm talking about, because by the time the call enters the dial plan, the codec for that leg of the call has already been negotiated |
06:00.53 | mvdk | So the point of using SER in that context was to modify the INVITE request |
06:01.17 | mvdk | I consider this use of SER to be a hack at best, though |
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06:02.44 | jameswf-home | considers a giant time conditioned dialplan that sends people to a random place that changes every minute..... |
06:03.33 | mvdk | considers such a dialplan to be a gigantic waste of everyone's time, unless sending one's enemies there.... |
06:03.47 | lsf_work | Im having problems with quadBRI, I can get calls in, but can't call out. Im wondering if it has something to do with dialplan in zapata? |
06:03.51 | mosty | TJNII, no, allow/disallow doesn't allow you to minimise transcoding |
06:04.00 | lsf_work | errr... zaptel.conf |
06:04.19 | mvdk | lsf_work: How did you put your dialplan into zaptel.conf? |
06:04.29 | mvdk | That would've taken some skill :) |
06:06.02 | mvdk | lsf_work: You may wish to rephrase your question, it's somewhat hard to parse |
06:06.06 | lsf_work | mvdk: sorry, I meant zapata.conf hehe... in my extensions.conf I only have Dial(${TRUNK}/${EXTEN:2})... which gives Dial("Zap/73-1", "ZAP/g1/") in new stack. I have overlapdial=yes in zapata.conf but can't figure out whats going on. |
06:06.12 | jameswf-home | I am going to be touching up and rehosting telemarketer torture.... that will be fun |
06:06.23 | jameswf-home | ~rephrase |
06:06.23 | jbot | If you feel the urge to repeat your question, you'll get a much better response if you try rephrasing it in different terms, or preferably by providing more information about the problem. |
06:07.06 | lsf_work | mvdk: asterisk doesnt wait for the outgoing dial (manual dial from fax). It just hangs up before the fax even starts dialing :( |
06:07.30 | mvdk | That's what you expected to happen? |
06:07.42 | mvdk | Asterisk to wait for DTMF from a fax? |
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06:08.36 | lsf_work | mvdk: yes? overlapdial? |
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06:10.03 | op3r | hello |
06:10.04 | lsf_work | mvdk: I`ve tried changing immediate to no also, gives some more delay before it hangs up. But still the same problem. (this has been working on 1.2) |
06:10.19 | op3r | does anyone knows a good french voip provider that can handle calls made by a predictive dialer? |
06:11.43 | mvdk | lsf_work: I just looked at http://www.voip-info.org/wiki/index.php?page=zaptelBRI |
06:11.48 | jameswf-home | ~humor |
06:11.48 | jbot | [humor] Q: Why are the streets of Paris lined with trees? A: Because Germans like to march in the shade. |
06:12.05 | mvdk | lsf_work: I suggest you look into WaitExten |
06:12.17 | jameswf-home | ~ch6 |
06:12.18 | jbot | Read about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
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06:15.05 | lsf_work | mvdk: thank you, strange that it worked on 1.2 |
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06:16.33 | tengulre | where have asterisk gui (digium)? |
06:17.22 | jameswf-home | tengulre: could you repet that in the form of a question |
06:21.04 | mvdk | mosty: Anyway, a proper way of doing codec negotiation would delay deciding on a codec for both legs of the call until it knows what codecs are usable by both sides... |
06:22.58 | mosty | mvdk, i only bother with sip client -> my * box -> upstream provider, i optimise on the second leg |
06:23.23 | mosty | since i have a list of codecs the upstream supports |
06:23.38 | mvdk | Yeah, I do that too |
06:23.59 | mvdk | I mean, it's basically "always use g.729, as the upstream supports that" |
06:24.28 | mvdk | But I saw no reason to just waste CPU on the server when there's so much of it sitting on the client... |
06:25.03 | mosty | what i do is if the call comes on from the client using codec X, and the upstream supports X, i send it on using X |
06:25.39 | mvdk | I see, so you take the decision made a priori and run with it, right? |
06:27.39 | mvdk | But you don't optimise the choice made by the client |
06:28.10 | mosty | i restrict what the client can use, then i optimise for minimization of transcoding |
06:28.47 | mosty | i have very few calls client->client so i don't bother optimising that |
06:29.25 | mvdk | In other words, not a 2-office situation |
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06:31.52 | mosty | no. it sounds like you want to change the codec during a call, which i'm not sure how to do |
06:32.11 | mvdk | Hardly "during", the second leg isn't set up yet! |
06:35.03 | mvdk | oej: What is the team doing about codec negotiation? |
06:35.26 | oej | mvdk: I am not up to date, too little I would say. |
06:36.15 | mvdk | OK, so basically, my quasi-solution of using SER to modify the INVITE request based on whether the destination is inter-office or external is the right approach? |
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06:36.55 | mvdk | Or rather, the only approach to allowing inter-office calls to be made by clients to be encoded in speex? |
06:37.06 | mvdk | But the external provider calls to be made g.729? |
06:39.00 | mvdk | oej: Basically, the scenario is thus: Suppose I have two offices, with an asterisk server at each running IAX to each other, and a SIP provider that uses g.729. Clients on the inside know how to use speex and how to use g.711. How do I get (a) A client in one office to call one in the other office using speex, (b) A client in an office calling a PSTN line to use g.711 so that it may be translated by the server to g.729 |
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06:43.41 | oej | mvdk: Well, you know you can negotiate multiple codecs in one call? |
06:44.00 | mvdk | Really? |
06:44.13 | oej | yes, it's quite common in Asterisk |
06:44.50 | oej | So if your client sets up a call to Asterisk with speex AND g711 - check what codec will be used in the different call scenarious with ethereal. Asterisk may even change mid-call. |
06:45.50 | mvdk | So the client INVITEs the server with both speex AND g711 in the codec list? |
06:46.12 | mvdk | And the server replies "Go ahead, send me both"? |
06:47.32 | oej | Yes, that works |
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06:47.45 | oej | Then the client and server chooses what they believe is optimal |
06:47.55 | oej | You can experiment with the order to get things right |
06:48.20 | oej | And when you call *to* the client, you can use the SIP_CODEC dialplan setting to control what you want. |
06:48.43 | oej | It might not get exactly what you want, but it will help your situation |
06:49.02 | oej | And you might end up with the client sending speex and asterisk sending g.711 in the same call... :-) |
06:49.50 | mvdk | So when the client initiates the call, the server *can* decide to have the client send it speex or g.711, depending on the destination? |
06:50.22 | mvdk | I ended up using SER to make the server comply, by trimming the list depending on the destination. Very naughty, I know :) |
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06:52.43 | mvdk | Anyway, it seems to me that proper codec negotiation in asterisk would remove the need for these mind-bending tricks |
06:52.56 | mvdk | They tend to be hard to explain... |
06:53.16 | mvdk | I mean, not to people here, but to the people I work for |
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07:09.58 | mosty | oej, do you have any further feedback on my manager event patch? 11959 |
07:10.20 | oej | I haven't checked... |
07:10.54 | mosty | http://bugs.digium.com/view.php?id=11959 |
07:12.32 | oej | mosty: You forgot changes to sip.conf... Tss, tss ;-) |
07:13.16 | mosty | oej, i'm not sure what you mean? |
07:13.33 | oej | If you add a new config option, one has to update sip.conf.sample too |
07:13.38 | oej | I will fix |
07:13.43 | mosty | oh i see, sorry |
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07:20.26 | adorah | Hi where I can find jitterbufer definitions? |
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07:23.38 | oej | mosty: Committed. |
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07:37.22 | mosty | oej: cool- thanks |
07:40.28 | rkeene | Grr |
07:40.35 | oej | mosty: Thanks for reminding me |
07:40.38 | rkeene | kicks Erlang !@#$ing buggy software |
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07:52.44 | Nugget | what drugs would I have to take that would make erlang make sense? |
07:52.47 | Nugget | I really want to know. |
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07:57.24 | bjartis | chan_sip.c:1245 retrans_pkt: Hanging up call ZmNhOWI5MWI1ODg3ZmYyNjliMWEwOGFmNTNkOWMzYTc. - no reply to our critical packet |
07:57.31 | bjartis | what does that mean? |
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09:04.14 | zepmantra | hello there anyone using astribank,fxs+fxo modules? is it worth buying, can't seem to find reviews |
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09:07.18 | cy3o3 | What's a good softphone for linux these days? |
09:07.42 | tzafrir | SIP? twinkle and ekiga, basically |
09:08.26 | cy3o3 | Cool |
09:08.34 | cy3o3 | Thanks dood |
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09:14.00 | shital | hello all |
09:15.17 | agx | cy3o3, Zoiper |
09:16.06 | agx | i was testing chan_mobile but i get no audio on incoming calls and only 1 seconds of audio during outgoing calls; could it be the phone (Nokia 6210) the USB Dongle or some other stuff? "core set debug 255" didn't helped a lot... |
09:17.38 | cy3o3 | agx: nice, IAX |
09:17.40 | cy3o3 | thanks |
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09:19.26 | shital | <PROTECTED> |
09:22.45 | shital | can any body help, is thr any problem using this distro??? |
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09:29.18 | agx | cy3o3, well with IAX you don't get the pre-ring stuffs too; i suggest to use it on phones only if you have NAT problems |
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09:35.13 | tengulre | anybody here come from CHINA? |
09:36.02 | agx | no, but i'm from Tibet |
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10:02.04 | Diablus | hello all |
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10:03.35 | Diablus | i have a problem with my asterisk server. There is a 99 Sip-peers. Asterisk drop down every 10-15 minutes without any reason. This is a standard bug? |
10:06.41 | Diablus | can anybody help me? |
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10:56.52 | kombi | I'm experimenting with audio codecs, how could I make asterisk play a high quality audio file for testing? |
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11:04.13 | kombi | if I crank up audio quality of i.e. speex, will asterisk's stdout still be 8k/8bit? |
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11:10.23 | kombi | corydon-76-dig: you probably know the above, just read about your patch for wideband speex..;) |
11:12.30 | tzafrir | kombi, g722? |
11:12.58 | kombi | tzafrir: using speex at the moment |
11:13.54 | tzafrir | anybody here cross-builds Asterisk on ARM and cares to answer the question in asterisk-embedded? |
11:14.02 | kombi | tzafrir: I just wonder if, with any codec stdout of asterisk will be any different from 8k/8bit? |
11:14.16 | tzafrir | http://lists.digium.com/pipermail/asterisk-embedded/2008-March/thread.html |
11:14.22 | kombi | tzafrir: that is way beyond me..;) |
11:14.38 | wordzilla | is it possible to tell asterisk to auto-terminate a call if it exceeds, say, 20 mins? |
11:14.58 | tzafrir | kombi, what do you mean by "stdout of Asterisk"? |
11:15.27 | tzafrir | wordzilla, core show application dial |
11:15.32 | kombi | tzafrir: like when you pipe it ot another audio device such as a stream client |
11:15.46 | tzafrir | wordzilla, I forgot the name of the option, but it is one of them |
11:16.24 | wordzilla | yup looks like L |
11:16.24 | wordzilla | thx tzafrir :) |
11:18.05 | kombi | tzafrir: ..as you can do with app_ices. I wonder if that signal will vary depending on codecs |
11:18.44 | jblack | ohh man. wondershaper broke. |
11:19.55 | kombi | tzafrir: never mind, I'll just try it out after lunch..;) Would be great if it did.. |
11:20.07 | tzafrir | kombi, app_ices... I think that the format_mp3 of Asterisk is for 8kHz only |
11:20.58 | kombi | that's right, I patched that to go to ezstream instead. Just wonder if standard output of asterisk ever goes beyond 8k sample rate |
11:21.47 | kombi | anyway, off to lunch, cheers! |
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11:48.24 | SteveTotaro | what if the fed cuts the rate by a whole point? |
11:49.42 | cpm | what if? |
11:50.11 | SteveTotaro | depression |
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11:50.39 | cpm | SteveTotaro, I'm sure, by the nature of the question, that you are pretty aware that this is already the case. |
11:51.09 | cpm | we had a short period, where it was starting to make sense to finally start saving money again, that lasted for what? almost a year? |
11:51.12 | tzafrir | SteveTotaro, no need to get so depressed ;-) |
11:51.35 | SteveTotaro | the world is going to get really big again |
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11:52.30 | SteveTotaro | only thing that has made sense this last year is ownership of gold offshore, bullion |
11:52.49 | cpm | naw, gold in the mattress was fine too :) |
11:52.58 | SteveTotaro | bullionvault is my favorite |
11:53.14 | cpm | just can't bring himself to trust such outfits |
11:53.28 | tzafrir | SteveTotaro, but gold is much heavier than water. So you'd need a pretty big boat for that |
11:53.30 | cpm | paper gold != gold |
11:53.48 | *** join/#asterisk jblack (n=jblack@pool-71-181-194-75.sctnpa.east.verizon.net) |
11:53.50 | SteveTotaro | no gold in the US, history will tell you the US government seized all private gold and made it illegal to own gold |
11:54.17 | cpm | I was telling folks 3-4 years back, that gold would hit $1K per tz by 2010, guess I was overly conservative :) |
11:54.41 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
11:54.41 | SteveTotaro | i was all about gold a year ago :) |
11:54.42 | cpm | SteveTotaro, they only seized gold from folks who were willing to give it up :) |
11:55.04 | jblack | predicting the upward movement of commodities is a stupid pet trick at best. |
11:55.13 | cpm | jblack, noted |
11:55.47 | cpm | gold isn't exactly a commodity. It's whats known as real money. Folks like to diss that concept, but it has played out quite well thoughout history |
11:55.55 | SteveTotaro | but gold is and always has been the best form of money |
11:56.54 | cpm | if you were all about gold a year ago, you should be sitting good now. :) |
11:56.56 | SteveTotaro | until some alchemist comes up with a way to change atoms from one thing to gold it will continue to hold value over paper |
11:57.11 | SteveTotaro | i have a bit in Zurich |
11:57.31 | SteveTotaro | or so they say |
11:57.36 | jblack | I've never really subscribed to that rule of thought, because once you consider gold an exception, you have to consider other exceptions (diamonds, and particularly other precious metals for example) |
11:57.41 | cpm | well, some folks say, and I think they are not far wrong, it's a tail wagging the dog thing. Gold remains consistant, other values slide against it. Not the other way around |
11:57.53 | cpm | diamonds are a tightly controlled market. |
11:58.26 | jblack | heh. Gold isn't? |
11:58.31 | cpm | Nope. |
11:58.37 | SteveTotaro | there is a much more bountiful supply of diamonds, they are just withheld |
11:58.38 | cpm | Gold *is* the standard |
11:58.58 | cpm | it's the baseline currency |
11:59.15 | jblack | Gold _was_ the standard, in an on-again-off-again basis. And it hasn't been the only standard. |
11:59.30 | jblack | Ever hear of sterling? :) |
11:59.43 | SteveTotaro | any society that has printed money with no backing has failed |
11:59.47 | cpm | sure. it's of less value, hence more fungible. |
12:00.26 | cpm | This is all pretty fun, and ultimately, it doesn't matter. It's now about what you can borrow, not about what you have. Has been that way for some time now. |
12:00.50 | cpm | been a long time since anyone plunked down a nice sack of metal and bought a car |
12:01.16 | SteveTotaro | bartering is becoming a huge movement |
12:01.19 | cpm | but having a sack of metal or two, is a nice hedge. that has worked out well. I don't think I'd buy any right now. |
12:01.19 | jblack | I don't agree with you, for reasons that are non-intuitive and non-obvious, but I can understand you. |
12:01.28 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
12:01.42 | jblack | I suppose I was a little redundant there |
12:01.44 | cpm | jblack, that's cool, and your approach is probably quite likely more practical |
12:02.02 | cpm | oh yeah? how about 'probably quite likely' ? |
12:02.11 | SteveTotaro | just plot the values over the years |
12:02.21 | jblack | Oh, definitely certain. |
12:02.29 | cpm | chuckles |
12:02.33 | SteveTotaro | and then look at the dollar and inflation |
12:02.53 | SteveTotaro | look at the dollar pre-gold standard and then now |
12:03.19 | SteveTotaro | pre-no-goldstandard i should say |
12:03.20 | cpm | The dollar is a fiat currency, like *all* other currencies. Floating away from gold was the best thing that ever happened for the worlds economic prospects. |
12:03.37 | jblack | I agree! I thought you thought the opposite? |
12:03.40 | cpm | fiat currencies work quite well, allows for open ended gains |
12:03.59 | SteveTotaro | fiat currencies always fail, it is seen throughout history |
12:04.11 | SteveTotaro | i got it, let's print more paper |
12:04.24 | jblack | I definitely agree with you cpm. I'm particularly happy with the social effects. |
12:04.37 | SteveTotaro | will you be happy in a year? |
12:04.52 | cpm | the downside is, when the press picks up on reporting about 300 folks defaulting on home loans in a market, where a few hundred thousand *didn't* default as some kind of abberation, and folks panic, , , then it starts to get sketchy |
12:05.06 | SteveTotaro | we have seen the bright spot in human history, lived it |
12:05.53 | SteveTotaro | no it is the twilight, the sun is setting |
12:06.03 | jblack | cpm: Yeah, I mostly agree with you, yeah. I think there's enough blame to include over leveraging. |
12:06.34 | SteveTotaro | the fed is pushing for the amero like they euro |
12:06.44 | jblack | Which I credit for making the runs that you're implying possible in the first place. |
12:06.46 | SteveTotaro | or should i say the illuminati |
12:06.57 | cpm | SteveTotaro, perhaps. As long as the fed keeps trying to prop up failed institutions, like it's doing right now, and messing about in the market, like they always do, it's going to cause real consequences. It's like using a CC to pay the minimum on another CC |
12:07.00 | jblack | lol |
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12:07.07 | *** join/#asterisk tomfmason (n=tom@unaffiliated/tomfmason) |
12:07.54 | SteveTotaro | in the end, one central, private bank for the world |
12:08.34 | cpm | SteveTotaro, yeah, it's called the WTO/World Bank, and Americans hold all the paper. |
12:08.41 | SteveTotaro | the fed can only cut interest rates and it is bottoming out |
12:09.05 | jblack | The fed can, and has already done, much more than that, steve. |
12:09.15 | SteveTotaro | I argue China hold all the cards now |
12:10.01 | jblack | I'm gonna regret this, but based on one set of logic does "china hold all the cards" |
12:10.07 | jblack | on what set? |
12:10.13 | SteveTotaro | what has the fed done? |
12:10.38 | SteveTotaro | the fed is group of privately held banks, nothing fed about them |
12:10.41 | tomfmason | I am having trouble make installing zaptel. I have kernel-devel installed(2.6.18-53.1.14.el5) but it is complaining about 2.6.18-028stab053.4 . Any ideas? |
12:11.07 | jblack | They've worked with the two target rates, they've changed the rules for the discount window, they've set up TEF and TSLF, and just this past monday, they guaranteed 30 bill of debt that JP Morgan 'bought' from stearns. |
12:11.33 | jblack | Oh, and they've been printing a few dozen billion here, a couple hundred billion there. |
12:11.41 | SteveTotaro | JP Morgan was a traitor |
12:12.03 | SteveTotaro | yeah, that's what i am saying, fire up the presses |
12:12.14 | SteveTotaro | that will solve all of our problems, more paper |
12:12.44 | jblack | I must have misunderstood you when you said "the fed can only cut interest rates and is bottoming out" |
12:13.04 | SteveTotaro | yeah they can create inflation too |
12:13.12 | SteveTotaro | by printing paper |
12:13.55 | jblack | It's pretty well agreed that there will eventually be heavy inflationary costs in the future. |
12:13.59 | SteveTotaro | all unconstitutional |
12:13.59 | tzafrir | tomfmason, what do you mean by "complaining"? |
12:14.16 | tzafrir | tomfmason, what is the output of 'uname -r'? |
12:14.55 | SteveTotaro | any private organization printing money is guilty of counterfeiting |
12:15.07 | jblack | Right now we're looking at an s shaped curve. Short inflation spike, followed by mild-to-severe recession for 1-4+ years (which is deflationary), followed by moderate to high inflation. |
12:15.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
12:15.28 | SteveTotaro | how about straight depression |
12:15.34 | SteveTotaro | pottersville |
12:16.08 | jblack | That's my belief, but it's not the commonly stated one. Nobody wants to be the first person to start financial panics and bank runs. |
12:16.37 | SteveTotaro | why would the government cut a check to almost everyone for at least $300 and mostly $600 |
12:16.38 | tomfmason | tzafrir: it shows: You do not appear to have the sources for the 2.6.18-028stab053.4 kernel installed. and uname -r is 2.6.18-028stab053.4 |
12:17.12 | jblack | anyways, what the fed is trying to do right now is stave off deep deflation, because of the overleveraged hedge funds that are collapsing. |
12:17.16 | tzafrir | tomfmason, do you use distro kernel or your own built kernel? |
12:17.49 | jblack | If they ignored that problem, we'd certainly be in a depression today. That's what they're staving off right now as they create money. |
12:18.16 | tomfmason | I just installed it from yum last night- yum install kernel-devel |
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12:20.18 | SteveTotaro | that is what they want you to believe just as the great depression was intentional |
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12:27.20 | *** part/#asterisk gormux (n=julien@rei.genux.info) |
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12:32.36 | tzafrir | tomfmason, hmmm... that is for a newer kernel version than your running one |
12:33.29 | tomfmason | hmm |
12:33.32 | tzafrir | tomfmason, do consider upgrading the kernel. keyword for your searches: vmsplice |
12:34.09 | tomfmason | I really don't like idea of manually building a kernel on a remote box but it is not looking like I will have a choice |
12:34.20 | tzafrir | that said, maybe using './install_prereq install' will install the older kernel-devel package for your running kernel |
12:35.10 | tzafrir | tomfmason, it's not a matter of building a custom one. It's installing a newer one for your distro |
12:36.38 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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12:50.19 | *** join/#asterisk warlock_mza (n=warlock@200-112-143-197.bbt.net.ar) |
12:50.24 | warlock_mza | hi there |
12:51.02 | warlock_mza | I currently have a pbx that allows with one telefone line handling 4 incoming calls |
12:51.15 | warlock_mza | if I want to put asterisk before this pbx |
12:51.33 | warlock_mza | I mean line goes into asterisk and via an fxs -> old pbx |
12:51.48 | warlock_mza | can I still redirect 4 calls with one fxs card ? |
12:51.49 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
12:52.07 | warlock_mza | and let the old pbx handle the 4 of them |
12:52.08 | warlock_mza | ? |
12:52.16 | [TK]D-Fender | warlock_mza: If you have 4 analog lines going into your old system, you'll need 8 ports in your * server |
12:52.25 | [TK]D-Fender | warlock_mza: 4 FXO, 4 FXS |
12:52.40 | [TK]D-Fender | warlock_mza: What do you want * to do for you? |
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13:05.07 | warlock_mza | ok got it ... |
13:05.16 | warlock_mza | you gave me the explanation I needed |
13:10.37 | warlock_mza | [TK]D-Fender, does that kind of config come on a single pci ? |
13:10.48 | warlock_mza | do you recomend something particularly ? |
13:11.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:12.05 | [TK]D-Fender | warlock_mza: Digium TMD800P or Sangoma A200d |
13:13.29 | warlock_mza | cool |
13:13.54 | Katty | hewwoes. |
13:15.19 | [TK]D-Fender | Katty: Mew. |
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13:33.17 | jblack | I'm bored. |
13:33.23 | jblack | considers setting up a phone sex line for dogs |
13:35.14 | MatBoy | wonders why he doesn't have his 3,3V cards PCI boards yet :P |
13:35.36 | jblack | who is john galt? |
13:35.50 | MatBoy | dunno, ask him |
13:36.46 | jblack | Who is Bart Simpson? |
13:37.46 | jblack | is so glad you didn't answer that one |
13:38.14 | MatBoy | hehe, Bart Simpson is Bart Simpson :P |
13:38.25 | jblack | damn. |
13:38.31 | jblack | You need to read more, and watch TV less. |
13:39.10 | MatBoy | jblack is jblack |
13:39.14 | MatBoy | is MatBoy |
13:39.34 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137) |
13:39.56 | jblack | Not knowing who John Galt is, is like not knowing who Guy Montag is. |
13:40.20 | MatBoy | nah, I'm busy with pricing... I don't want to think that much about other things :) |
13:41.27 | *** join/#asterisk Newbie___ (n=Newbie__@213.8.50.60.kmr02-home.tm.net.my) |
13:41.59 | Newbie___ | morning all, how do i do a dial command after hangup ? |
13:44.24 | coppice | who is Guy Montag? |
13:45.18 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
13:46.29 | ccvp | - hello fellow internet addicts - are we all looking forward to another long & glorious day of irc/internet addiction :) |
13:49.29 | cpm | coppice, he's the hero in bradbury's fahrenheit 451 |
13:49.57 | coppice | well, its an awful long time since I read that :-\ |
13:50.09 | cpm | heh |
13:50.26 | cpm | took me a few moments |
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13:51.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:51.49 | coppice | I remember reading the bit about buildings being fire proof, and thinking they had the same idea bout the Crystal Palace |
13:52.00 | jameswf-home | whats the term for bridging to sip callers then dropping out... |
13:52.30 | ccvp | is ekiga worthless? |
13:52.35 | ccvp | whats a better client to get (free) |
13:52.42 | jameswf-home | ~ekiga |
13:52.43 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
13:52.54 | jameswf-home | welll |
13:53.05 | ccvp | I know im using it, i did an ekiga to ekiga this morning on a 1GB lan |
13:53.13 | ccvp | both with logitech quickcam pro9000 cams |
13:53.18 | ccvp | and the vidoe acceleration is garbage |
13:53.22 | ccvp | even though the camera is high end |
13:53.23 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
13:53.36 | ccvp | in windows, the video acceleration is near perfect in net meeting or skype |
13:53.44 | ccvp | wondering if ekiga uses crappy codecs by default for video |
13:54.36 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
13:54.41 | ccvp | skype video conference w/ dual quickcam pro 9000's gave high quality , high def, near 30fps...ekiga gives randomized blocks, and squarey smears |
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14:00.12 | jameswf-home | lmao http://www.bash.org/?117002 |
14:01.02 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
14:01.02 | cpm | chuckles |
14:01.46 | dandre | b |
14:01.53 | dandre | ~pb |
14:01.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
14:02.42 | ccvp | I dont think ive ever laughed |
14:02.47 | ccvp | at spam before, but just got this on efnet |
14:02.48 | ccvp | lol |
14:02.50 | dandre | Hello, |
14:02.52 | ccvp | <linuckzG> (SANDEEP PEHMDAYPEEZ) C1a_!5 TaB5 for $39.99.............www.pharma-ceuticals.com . . . . DONT THINK TWICE before YOU GO. (SANDEEP PEHMDAYPEEZ) |
14:03.12 | ccvp | what the hell is a sandeep |
14:03.49 | Katty | Anyone know how to make FOP show DID numbers? |
14:04.02 | Katty | they don't really have a set zap line they come in on. |
14:05.39 | *** join/#asterisk stoffell_h (n=stoffell@fw.catsanddogs.com) |
14:05.46 | dandre | I have put here part of log and dialplan: http://pastebin.ca/947402 |
14:05.46 | dandre | when 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16. |
14:05.46 | dandre | but I have a hungup just after 13. Why? |
14:09.07 | SteveTotaro | not why local channel and i do not see a timeout |
14:10.46 | SteveTotaro | trixbox? |
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14:13.05 | dandre | <PROTECTED> |
14:14.07 | dandre | I use Local because I want to make my dial not dependant on technology at this point |
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14:15.15 | Katty | anyone know how to use DID numbers with FOP? |
14:15.22 | *** join/#asterisk ciphercast (n=cipherca@pool-151-204-63-64.pskn.east.verizon.net) |
14:15.38 | dandre | it is not trixbox either |
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14:24.29 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
14:25.09 | UserReg_CL | hi.... what port use xlite whe connect to asterisk ? |
14:27.32 | BCS-Satori | How do you pass the originating CID/CNAME from an external caller to the transfered caller? When a receptionist answers the phone call they see the incoming callers cid/cname, but when the receptionist transfers the caller to extension 101 for example, the person on extension 101 has their callerid as the receptionist not the incoming caller for the rest of the call. This issue is on multiple asterisk's rev 1.4.11-1.4.18 w/ polycom and linksys. |
14:27.35 | lirakis | is away (leaving..."the internets" are safe ... for now) |
14:27.59 | n0cturn | Q: I'd like to hack around with Asterisk on a Virtual Machine. Would a pair of Zoom 5801's be appropriate for VoIP gateways, or is there a better product to use ? I would prefer lower cost hardware and I do not want to put in a PCI card. |
14:28.03 | [TK]D-Fender | BCS-Satori: Tell her to do BLIND transfers instead of ATTENDED |
14:28.17 | [TK]D-Fender | n0cturn: Zoom = bleh |
14:29.28 | BCS-Satori | [TK]D-Fender: that fine, but is there a way to send it with an attended ? |
14:30.55 | n0cturn | Can anyone suggest a low-cost 1-port FXS/1-port FXO VoIP gateway ? |
14:31.47 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
14:32.05 | [TK]D-Fender | n0cturn: Linksys SPA-3102 |
14:33.11 | n0cturn | Thanks, D-Fender |
14:33.41 | dandre | I have put here part of log and dialplan: http://pastebin.ca/947402 |
14:33.41 | dandre | when 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16. |
14:33.41 | dandre | but I have a hungup just after 13. Why? |
14:40.45 | shasta | [TK]D-Fender, by the way. does spa3102 handle sip register? |
14:41.20 | shasta | (i'd like asterisk to register on spa3102) |
14:43.52 | [TK]D-Fender | shasta: Yes. |
14:43.58 | [TK]D-Fender | shasta: IT registers |
14:44.40 | agx | with asterisk 1.4 is possibile to send a call out via a specific channell of an MISDN port ? I've a GSM box connected via ISDN with 2 SIM on it |
14:45.00 | shasta | [TK]D-Fender, I know that _it_ registers, but I'd like Asterisk to register at spa :) |
14:45.06 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
14:45.29 | [TK]D-Fender | shasta: don't think I saw an option for that... might be... go look. |
14:46.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
14:46.10 | *** mode/#asterisk [+o russellb] by ChanServ |
14:47.04 | BobLutz | What type of files am I able to stream with app_read.c ? |
14:47.18 | BCS-Satori | Is there any docs that explain how to block/route a call based on the caller id number |
14:47.19 | BobLutz | is looking at the API call to ast_app_getdata() on line 189 |
14:48.10 | dandre | is there a backport of the 1.6 hint() function for asterisk1.4? |
14:48.59 | [TK]D-Fender | BCS-Satori: "core show function CALLERID" |
14:51.25 | russellb | dandre: no, but give me one sec ... |
14:51.40 | [TK]D-Fender | ~devstate |
14:51.41 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
14:52.42 | russellb | ok, backport done .. |
14:53.11 | russellb | wait, that was the wrong module, lol |
14:53.19 | warlock_mza | does anyone know some kind of adaptor from rj45 -> rj11 voip->analog ? |
14:53.29 | warlock_mza | to avoid having so much fxs ? |
14:53.33 | Rienzilla | +ons |
14:53.42 | Rienzilla | I have a linksys app which does that |
14:53.46 | dandre | russellb: my problem is not directly related to the hint function: |
14:53.48 | warlock_mza | and dont have to throw away all my phones ? |
14:54.16 | dandre | I have put here part of log and dialplan: http://pastebin.ca/947402 |
14:54.16 | dandre | when 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16. |
14:54.16 | dandre | but I have a hungup just after 13. I don't understand why? |
14:55.21 | dandre | if I change the dial(...) by a stdexten macro call, this seem to work correctly so as I haven't found solution, I was trying something like |
14:55.39 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
14:55.59 | dandre | Macro(stdexten,13,${Hint(13@from-internal)}) |
14:56.11 | dandre | but I am in 1.4 |
14:56.50 | *** join/#asterisk bmg505 (n=leon@196-209-78-68-tbnb-esr-2.dynamic.isadsl.co.za) |
14:56.53 | *** join/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net) |
14:57.13 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
14:57.23 | twitchnln | good morning, how do I set the maximum channels on a per trunk basis? |
14:57.37 | russellb | what kind of trunk? |
14:57.42 | twitchnln | sip |
14:57.44 | russellb | and are you using freepbx? |
14:57.46 | twitchnln | no |
14:57.48 | russellb | ok |
14:57.53 | russellb | then you can set call limits in sip.conf |
14:58.04 | russellb | search for limit in configs/sip.conf.sample |
14:59.59 | tomfmason | finally compiled zaptel on centOS 4 |
15:00.41 | _ShrikE | russellb: where can I find that hint backport you just did :) |
15:01.10 | russellb | i didn't do the right module, heh |
15:01.17 | russellb | i did func_extstate |
15:01.27 | twitchnln | russellb: thanks |
15:01.28 | _ShrikE | Yeah I see that.. 2 minutes |
15:01.32 | russellb | but it's in ... svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4 |
15:01.35 | _ShrikE | 2 minutes old that is |
15:02.47 | *** join/#asterisk xacatecas (n=jkroon@dsl-241-143-116.telkomadsl.co.za) |
15:03.17 | xacatecas | hi, has anybody played with the linksys wip330 phones? specifically I'm unable to get WPA2 working. Any pointers would be appreciated. |
15:05.37 | dandre | russellb: do think a backport of the hint function could solve my problem? |
15:05.37 | dandre | I'd prefer understand why the Dial(Local/13@from-internal) doesn't return |
15:06.02 | [TK]D-Fender | dandre: Might be a thought to put the "/n" in there so it doesn't bridge them ..... |
15:06.09 | [TK]D-Fender | (hand-off) |
15:06.22 | russellb | dandre: i'm not sure .. i got busy with something else, sorry |
15:06.23 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
15:10.21 | xacatecas | ok, so i take it there are no-one using a wip330 here ... |
15:11.25 | *** join/#asterisk P4C0 (n=jannet@200.124.22.34) |
15:12.24 | dandre | I have put the "/n" but the result is the same |
15:12.30 | P4C0 | hello everyone, i bought two license for g729, it was working fine but now all of the suddent it doesn't and show translations shows - for all the g729... how can I check the status of the codec? |
15:13.32 | Newbie___ | morning all, how do i do a dial command after hangup ? |
15:14.11 | *** join/#asterisk op3r (n=Op3r@203.177.177.26) |
15:15.46 | P4C0 | is there anyway to check the status of g729 codec? |
15:15.54 | russellb | show g729 |
15:15.55 | russellb | i think |
15:15.57 | russellb | at the *CLI> |
15:17.42 | BobLutz | What type of files can app_read.c use ? |
15:17.58 | [TK]D-Fender | BobLutz: "CORE SHOW MODULES LIKE FORMAT" |
15:19.25 | russellb | O.O |
15:19.33 | russellb | [TK]D-Fender: i didn't know you could yell at the CLI! |
15:19.35 | russellb | that's cool |
15:19.35 | BobLutz | module show like format |
15:20.11 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
15:20.11 | *** mode/#asterisk [+o anthm] by ChanServ |
15:20.16 | BobLutz | so app_read can use anything that `module show like format` returns ? |
15:20.53 | BobLutz | app_read.c uses an API call to ast_app_getdata(), I assume the format type is transparent |
15:21.00 | Katty | anyone know how to use DID numbers with FOP? |
15:21.05 | twitchnln | question, will this fail over to the second trunk if i enable it? |
15:21.05 | twitchnln | exten => _X.,1,Dial(SIP/trunk1/${EXTEN}) |
15:21.05 | twitchnln | exten => _X.,2,Dial(SIP/trunk2/${EXTEN}) |
15:21.05 | twitchnln | exten => _X.,3,Macro(out-congestion) |
15:21.06 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
15:21.23 | ccvp | what's this caused by |
15:21.24 | ccvp | ryan@myubuntu:~/Desktop/xlite$ ./xtensoftphone |
15:21.25 | ccvp | ./xtensoftphone: error while loading shared libraries: libstdc++.so.5: cannot open shared object file: No such file or directory |
15:21.35 | ccvp | after i chmod +x'd the binary |
15:21.55 | shasta | install libstdc++ :-) |
15:22.05 | twitchnln | ccvp: looks like you need libstdc++ |
15:23.13 | ccvp | E: Couldn't find package libstdc |
15:23.14 | ccvp | ryan@myubuntu:~$ |
15:23.21 | ccvp | whts the package name? |
15:23.24 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
15:23.30 | BobLutz | ccvp: packages.ubuntu.com |
15:23.42 | twitchnln | ccvp: apt-get install libstdc++ |
15:23.44 | BobLutz | If I remember correctly.. |
15:23.51 | ccvp | thats what i did twitch |
15:23.58 | ccvp | must be named diff or osmething. |
15:24.06 | P4C0 | russellb, it says that show command is invalid... i will try to reinstall the codec |
15:24.17 | twitchnln | ccvp: apt-cache search libstdc++ |
15:24.20 | shasta | apt-cache search libstdc++ |
15:24.31 | shasta | but this is definitely NOT related to Asterisk |
15:25.29 | ccvp | i know shasta, but cant find the info on google, searched past 35min |
15:25.43 | ccvp | apt-cache search libstdc++ gave back like 40 results |
15:25.55 | BobLutz | ~ubuntu |
15:26.12 | km- | [TK]D-Fender: what's your opinion on the sangoma's hardware echo canceller? |
15:26.19 | [TK]D-Fender | km-: Great |
15:26.37 | shasta | ccvp, Results 1 - 10 of about 948 for ubuntu libstdc++.so.5: cannot open shared object file. (0.28 seconds) |
15:26.40 | km- | [TK]D-Fender: gonna try popping two of those cards into a dell 2850 chassis and see what asterisk does with it. |
15:26.41 | ccvp | http://pastebin.com/m95dd32 |
15:26.44 | ccvp | thats my output |
15:26.53 | [TK]D-Fender | km-: which card? |
15:29.28 | ccvp | twitch, it runs now |
15:29.33 | ccvp | apt-get install libstdc++5 worked |
15:30.19 | km- | [TK]D-Fender: the sangoma 8-port. a108d? |
15:30.31 | [TK]D-Fender | km-: ok. |
15:30.32 | km- | [TK]D-Fender: apparently we use them elsewhere in our company too |
15:32.15 | lirakis_away | is away (leaving..."the internets" are safe ... for now) |
15:32.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
15:36.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:36.27 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
15:37.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
15:38.09 | BobLutz | russellb: ast_write() writes a frame to a channel (i.e - you can essentially play a .wav over a channel), right ? |
15:38.34 | russellb | yeah, sort of |
15:38.40 | russellb | ast_write() is the lowest level API call |
15:38.46 | russellb | a frame is not a file |
15:38.49 | dandre | is there any reason why my Dial(Local/13@from-internal/n) doesn't return from 'from-internal' context? |
15:38.54 | russellb | a frame is like a packet of audio ... 20 ms usually |
15:40.00 | BobLutz | russellb: Is it possible to write a file to a channel? |
15:40.00 | russellb | yup |
15:40.00 | russellb | look in include/asterisk/app.h |
15:40.00 | BobLutz | :-D |
15:40.00 | BobLutz | danke |
15:40.00 | russellb | tons of examples of usage in apps/ |
15:40.05 | russellb | for example, app_voicemail.c plays a ton of files ... |
15:40.09 | russellb | though that's a pretty big app |
15:40.16 | russellb | app_playback is a simpler one |
15:40.17 | BobLutz | geusses app_playback.c might be good? |
15:40.19 | BobLutz | yea |
15:40.20 | russellb | nods |
15:40.22 | BobLutz | ok great, thanks |
15:40.24 | russellb | np |
15:45.56 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
15:50.13 | *** join/#asterisk patrick-- (i=patrick@devnull.biz) |
15:51.02 | patrick-- | Hey, im using asterisk with mISDN and Beronet ISDN cards. when 2 phones are connected to a port can i set an outgoing number for both of them? |
15:52.34 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
15:53.54 | *** join/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
15:55.44 | really_phukt | Looking for respectable and reliable IAX providers for small and large business. Who would you recommend? |
15:57.31 | dandre | if I put 'return' just after the stdexten macro call and use gosub instead of dial(Local...) this works. |
15:57.31 | dandre | is this a good idea? russellb, [TK]D-Fender ? |
15:58.45 | really_phukt | Is *any*body here using IAX providers or yall pretty much stuck with SIP? |
16:00.07 | really_phukt | Is there anybody awake this morning? (or whatever time applies to you) |
16:00.51 | cpm | uses IAX provider(s) |
16:01.07 | [TK]D-Fender | dandre: Perhaps you should do "/n" and not use "return".... |
16:01.26 | cpm | the values for respectable and reliable don't have the same meaning in telcom as they once may have. THerefore I don't feel qualified to give an answer |
16:01.55 | [TK]D-Fender | really_phukt: Only people using IAX for ITSP's are those in dire need of bandwidth or with firewall issues. |
16:02.17 | really_phukt | [TK]D-Fender: ...firewall... that would be me :( |
16:02.22 | dandre | [TK]D-Fender: I have put Dial(Local/13@from-internal/n) |
16:02.22 | dandre | is this ok? |
16:02.38 | dandre | because that doesn't works |
16:03.00 | [TK]D-Fender | dandre: pand the second part... |
16:03.14 | [TK]D-Fender | really_phukt: Is yours really screwed up? What are you running? |
16:03.18 | *** join/#asterisk svenna_ (n=svenna@p548D357E.dip0.t-ipconnect.de) |
16:04.27 | really_phukt | [TK]D-Fender: well, I am using SIP provider now (callcentric) and works OK. But IT guys is not happy with the amount of open holes in the firewall... |
16:04.49 | [TK]D-Fender | really_phukt: then downtune your RTP range and tell him to get over it. |
16:05.09 | dandre | [TK]D-Fender: I haven't understood |
16:05.23 | [TK]D-Fender | dandre: "remove the "return"" |
16:05.33 | [TK]D-Fender | dandre: jsut let "s" run out like normal. |
16:06.04 | dandre | that's what I have done |
16:06.32 | really_phukt | [TK]D-Fender: I wish I could. I've been fighting this battle for a while. Are there any major advantages of SIP over IAX that I can stress to try to convince him? |
16:07.44 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
16:07.49 | really_phukt | from what I read IAX seems to be better for NAT environments and limited bandwidth |
16:07.54 | [TK]D-Fender | dandre: and -- Local/13@from-internal-5273,1 answered Zap/2-1 <-- this is a real problem. its counting the call as "answered" |
16:08.11 | [TK]D-Fender | really_phukt: If you don't need the BW then don't do it. |
16:08.29 | [TK]D-Fender | really_phukt: IAX2 has quality/stability issues. |
16:09.59 | russellb | it should be good in the later 1.4 releases |
16:10.04 | russellb | we put a lot of work into it |
16:10.13 | x86 | works great here |
16:10.18 | russellb | see :-p |
16:10.46 | x86 | I've got 7 asterisk boxes sprinkled across most of my branch offices and HQ, and I do IAX2 trunking between all of them for intra-office calls |
16:10.53 | x86 | inter-office I mean |
16:10.56 | x86 | works great |
16:11.06 | x86 | 1.4.12.1 is what I'm running |
16:11.28 | russellb | you're lucky then, a lot of stuff has been fixed since then :) |
16:11.54 | russellb | well, 25 fixes to that module ... less than i thought |
16:13.32 | dandre | ok [TK]D-Fender but I don't why this channel answers . |
16:13.32 | dandre | Can this: |
16:13.32 | dandre | <PROTECTED> |
16:13.32 | dandre | be related to that? |
16:13.58 | x86 | russellb: guess so ;) |
16:14.06 | x86 | russellb: well "trunking" never worked for me |
16:14.22 | x86 | russellb: I say I'm trunking, but it's not setup that way in iax.conf :) |
16:17.38 | *** join/#asterisk esaym (n=user@72.183.198.134) |
16:20.06 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
16:21.24 | Qwell | ~hold |
16:21.24 | jbot | hold is probably a status flag to tell apt not to automatically upgrade a package. apt will place packages on hold if they require packages that are not currently installable; you can 'apt-get install pkgname' to explicitly install the package. To put a package on hold, 'echo pkgname hold | dpkg --set-selections' or use the '=' key on the package in dselect, or 'echo pkgname install | dpkg --set-selections' to remove the hold |
16:21.28 | Qwell | <3 |
16:25.22 | [TK]D-Fender | dandre: I'd also be looking at that voicemail line funny... |
16:25.32 | [TK]D-Fender | dandre: Comment it out. |
16:25.45 | [TK]D-Fender | dandre: its TRIED going in there.. that might have something to do with it. |
16:25.50 | dandre | [TK]D-Fender: that's it |
16:26.01 | dandre | I've just discored it now |
16:26.23 | [TK]D-Fender | dandre: Because it did say "stopping sounds", but you never saw a reall attempt to play audio. |
16:27.08 | dandre | how can I know wether a voicemail exists in the dialplan? |
16:27.52 | flush | what is said in the sound file "vm-login".. i dont understand the first 2 words.. "something mail" |
16:28.06 | Qwell | flush: you've already had like 3 people answer you |
16:28.08 | file | Comedian Mail. |
16:28.09 | [TK]D-Fender | dandre: You are calling it in your maco. Perhaps you should put some actual thought into your dialplan coding.... |
16:28.22 | flush | Comedian mail? hrm |
16:29.28 | flush | can i boost a phone volume in zapata.conf file cause when i use hands-free volume is so low its almost ridiculous.. |
16:29.58 | [TK]D-Fender | flush: "lookup rxgain" & "txgain" |
16:30.10 | flush | copy |
16:35.28 | UserReg_CL | hi... which is velue TOS for xlite ? |
16:37.40 | *** join/#asterisk adjohn (n=adjohn@p7221-ipad87marunouchi.tokyo.ocn.ne.jp) |
16:37.53 | *** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net) |
16:38.20 | nephfl | hello im trying to setup vtwhite for incoming and am having no success |
16:39.13 | flush | hey.. i have set voicemail its awesome i removed my bell message box and now saving money |
16:39.30 | flush | but id like to know.. can i put an option or something to say theres a message, like a busy tone when you pick up the phone or something ? |
16:39.43 | jblack | Now move to high speed internet and remove local phone service. That'll really save you |
16:40.08 | jblack | flush: Some ATAs can do that, like the linksys line. |
16:40.18 | flush | im powered by the POTS |
16:40.32 | jblack | * can probably do it on FXS ports too. Check indications.conf |
16:41.53 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
16:43.24 | BobLutz | russellb: file: Are you guys around / Would I be able to talk with via PM for a quick second ? |
16:43.56 | file | I don't accept private messages, anything said usually benefits all. |
16:44.15 | UserReg_CL | hi... wich is the value for bit TOS ? |
16:44.23 | russellb | same as file |
16:44.25 | nephfl | maybe he wants to talk dirty |
16:44.28 | russellb | and i'm quite busy right now |
16:44.58 | BobLutz | russellb: file: very specific question, I'll ask later |
16:45.00 | BobLutz | thanks |
16:45.06 | [TK]D-Fender | flush: Set the mailbox for your device. |
16:45.14 | russellb | BobLutz: if it's code related, just join #asterisk-dev and ask there |
16:45.20 | russellb | plenty of people around that can answer besides us |
16:45.37 | BobLutz | joins #asterisk-dev |
16:47.17 | *** join/#asterisk enjay5150 (n=chatzill@ip70-190-60-237.ph.ph.cox.net) |
16:48.24 | enjay5150 | Is it possible to use '#' within the same context "multiple times" for validation (i.e. press # to confirm) |
16:48.55 | enjay5150 | like in an IVR. Enter your name "press pound to continue" Enter your Address "press pound to continue" etc.. |
16:50.11 | [TK]D-Fender | enjay5150: "core show application read" |
16:51.10 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
16:51.19 | enjay5150 | thanks |
16:57.51 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
16:58.28 | enjay5150 | so exten => 1,n,Read(whatever,,#,,1) could work.. |
16:58.43 | enjay5150 | thanks D.. |
17:01.24 | rkeene | Man, Polycom's webpage sucks! |
17:02.27 | rkeene | All I want to do is download their crappy software, for the crappy phones which I have purchased |
17:02.41 | *** join/#asterisk bkw_ (n=brian@adsl-71-153-171-225.dsl.tul2ok.sbcglobal.net) |
17:02.50 | enjay5150 | lol what issues are you having with their phones? |
17:03.23 | rkeene | Well, mine locks up on boot 75%+ of the time |
17:03.30 | rkeene | And requires a hard-power-cycle to fix |
17:03.41 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
17:03.44 | enjay5150 | at the IP screen? |
17:04.03 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
17:04.10 | enjay5150 | what sip code are you currently running? |
17:04.27 | rkeene | SoundPoint IP 550; It locks up at a couple of different places. At the point where it tells me my IP/MAC and says "Please wait ..." and at the main screen |
17:04.46 | rkeene | 2.2.2.0084 |
17:05.59 | enjay5150 | I'll find out if theres an update sec.. |
17:06.00 | rkeene | (The latest version I could try without creating an account... then I created an account ... eventually, their web page wouldn't let me register my organization because it thought someone already registered for me (which could be, since we are huge military organization, but I still want support).. and now it won't give me access to the firmware) |
17:06.52 | [TK]D-Fender | rkeene: Just ask your reseller. |
17:07.04 | rkeene | [TK]D-Fender, Like I know who that is |
17:07.11 | rkeene | I don't control who we buy crap from |
17:07.15 | [TK]D-Fender | rkeene: You don' know where you bought them from? |
17:07.23 | rkeene | No, the orders get bid out |
17:08.10 | rkeene | It works like this, I tell someone in procurement what I want, they send it up to DC, who loses the paperwork, we resend, and they bid it out (even if it's on the GSA schedule) and 6 to 12 months later a product arrives |
17:08.41 | rkeene | That doesn't come to me, of course -- it goes to DC. They package it up and send it to me. |
17:08.52 | rkeene | (sometimes) |
17:09.23 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
17:09.38 | enjay5150 | 3.0.0 is out.. |
17:09.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
17:10.29 | rkeene | Yes, but their portal at this point won't let me access it |
17:10.42 | jblack | Is there a traditional extension to dial for pickups? |
17:11.09 | enjay5150 | depends on what you've defined in your features.conf |
17:11.34 | jblack | Oh, of course. I was about to mis-use the PickUp() app |
17:12.43 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-104-253.w86-203.abo.wanadoo.fr) |
17:13.01 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
17:13.10 | rkeene | Grr! |
17:13.14 | rkeene | hates Polycom's webpage |
17:13.40 | jblack | hmmm. actually, I can understand why I didn't look there. I've always thought of features.conf for already established calls. |
17:13.59 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
17:20.16 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
17:20.35 | RobH | Ok, time for a stupid question. I have Asterisk 1.4 compiled and running on ubuntu, kernel 2.6.22-14-generic. Also compiled are libpri-1.2.7 & zaptel-1.4.9.2. Asterisk loads up, and it will accept the simple dial of my IAX2 softphone, and reads that it is playing back a sound, but I hear no sound on my phone. They are on the exact same subnet, no firewall between them. Any ideas of what I should check first? (I have not had this |
17:20.51 | RobH | (Sorry for the long message, figured it best to throw out as much info as possible.) |
17:21.01 | Defraz | Hey all, all of the sudden my linksys pap2 and pap2t atas sound bubbly in one direction? I use a polycom or softphone and it is clear as day. |
17:21.28 | Defraz | but the 3 ATAs sound bubbly in one direction. I call from a land line and I hear it bad, but they hear me crystal clear. |
17:21.55 | Defraz | could be a codec issue |
17:22.07 | Defraz | that happens for me when my codec isn't right. |
17:22.28 | RobH | Interesting, ok, I am new at this, how would I troubleshoot the codecs? |
17:22.50 | Defraz | make sure all your devices have the same codec set. |
17:23.05 | RobH | Shouldnt Asterisk translate that for me automatically though? |
17:23.46 | RobH | Hmm, I just confirmed I can call hardphone to softphone, so its working somewhat. |
17:24.02 | Defraz | depends |
17:24.13 | Defraz | do you have canreinvite=no? |
17:24.25 | RobH | crap, that would do it huh? |
17:24.38 | RobH | i dont stop reinvites |
17:24.43 | Defraz | yep |
17:24.48 | RobH | but one is on iax2 and one is on sip, so wouldnt it have to stay in asterisk? |
17:24.48 | Defraz | could cause that. |
17:24.57 | Rico29 | hi |
17:25.29 | Rico29 | does anyone knows where I can find a softphone which supports h263 or h264 video ? (FOR LINUX) |
17:25.31 | [TK]D-Fender | RobH: Check your sound card, and fix your libpri while you're at it. |
17:25.44 | [TK]D-Fender | Rico29: |
17:25.46 | [TK]D-Fender | ~ekiga |
17:25.46 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
17:26.01 | Rico29 | thanks |
17:26.05 | eric2 | anyone have success getting the intercom functionality to work with snom phones? |
17:26.07 | RobH | check sound card on the server? It has no sound card. libpri compiled ok. |
17:26.15 | Rico29 | i thought that ekiga didn't support it |
17:26.24 | eric2 | linksys phones.. no problem, got it working.. but snom is a beyatch! |
17:27.02 | mpwizard | eric2: It worked for me, out of the box with Snom M3. |
17:27.50 | eric2 | so from one snom phone you can dial another snom phone and it automatically conferences? withouth having to pick up the phone or press any buttons on the receiving end? |
17:28.11 | eric2 | I have snom 300's |
17:33.15 | Defraz | Can't find an Ekiga binary for windows. |
17:34.10 | Juggie | http://snapshots.voxgratia.org/win32.php |
17:35.43 | RobH | Ok, my polycom is forced to ulaw, and my zoiper to gsm. They can call one another, reinvite is off, and they are staying bridged. Asterisk translates from ulaw to gsm just fine. |
17:35.52 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6 , 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
17:35.59 | RobH | but when I call a simple playback function, no audio is played. |
17:36.23 | [TK]D-Fender | RobH: thats often a zaptel issue. unload ztdummy. |
17:36.29 | Qwell | russellb: did you intentionally leave out 1.4.8.1? |
17:36.33 | russellb | no |
17:36.34 | Qwell | 1.4.18.1 rather |
17:36.39 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6 , 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
17:37.47 | RobH | [TK]D-Fender: That worked. So continuing my line of novice questions. How exactly can I troubleshoot the ztdummy issue. I require that it work so I can use meetme for my company. |
17:38.00 | RobH | and thank you for the answer =] |
17:38.12 | russellb | Qwell: and i'll have to update the web site after i get some food ... |
17:39.53 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6, 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
17:39.56 | Qwell | hmm |
17:40.13 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
17:40.16 | [TK]D-Fender | RobH: I don't know the fine points about this personally. I jsut know its usually the culprit. Manxpower has most often spoken about this in here to my memory. |
17:40.23 | b11d` | whats the dCAP all about? Is it worth getting>? |
17:40.41 | RobH | [TK]D-Fender: Thank you, your name came up in bootcamp last week =] |
17:40.57 | [TK]D-Fender | RobH: LOL... can't be good news :p |
17:41.04 | RobH | I noted that a few months ago you helped me, Jared made a point of saying that you are one of the regulars who help folks |
17:41.31 | [TK]D-Fender | RobH: Yeah, I'm something of a constant around here (others would likely add to that statement...) |
17:41.47 | RobH | Yes well, I am now an official lurker. |
17:42.07 | RobH | we pushed our asterisk server live, but its 1.2 and my dialplan looks like vomit =P |
17:43.03 | b11d` | anyone? dCAP worth it? |
17:43.05 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
17:43.26 | RobH | b11d`: It puts your name on Digium's website as an asterisk person to contact. |
17:43.47 | b11d` | im more about the learning and the content than the name on a website.. |
17:43.50 | RobH | So if there are no others in your area, or a limited amount, it may generate you some business. |
17:43.56 | b11d` | hmm.. thats cool |
17:44.01 | b11d` | my employer would pay for me to take it.. |
17:44.03 | RobH | bootcamp was good for the learning. |
17:44.05 | b11d` | so I was considering it |
17:44.08 | RobH | I just did it last week. |
17:44.10 | b11d` | cool |
17:44.13 | b11d` | where? |
17:44.21 | RobH | I went to the main Huntsville AL office |
17:44.24 | b11d` | I was thining Huntsville.. but would LOVE the Netherlands :) |
17:44.29 | b11d` | yeah Las Vegas sucks |
17:44.29 | b11d` | :) |
17:44.30 | RobH | met all the Digium folks, it was very neat. |
17:44.56 | RobH | the Vegas one is just in a training facility, at Huntsville it is in the main office, which is cool. |
17:45.08 | RobH | plus Huntsville has less distractions from study, heh. |
17:45.13 | b11d` | yeah.. i just hate vegas.. Huntsville sounded like the best place to go |
17:45.15 | Qwell | RobH: when did you do it? |
17:45.21 | Qwell | oh, last week. nevermind |
17:45.21 | RobH | Just last week =] |
17:45.36 | Qwell | ahh, yeah, I heard about you ;) |
17:45.39 | Rico29 | do you work with OPAL lib ? |
17:45.39 | RobH | Jared was an excellent trainer too. Everyone in our class took the dCAP |
17:45.47 | Qwell | RobH: yes, Jared is awesome |
17:45.48 | b11d` | how many people were there? |
17:45.52 | RobH | 10 folks total |
17:45.53 | *** part/#asterisk enjay5150 (n=chatzill@ip70-190-60-237.ph.ph.cox.net) |
17:45.55 | b11d` | good |
17:45.58 | RobH | plus jared as instructor |
17:46.27 | b11d` | im really in it for the Orange Ice Digium Pen |
17:46.32 | RobH | Now, my company paid for it, so my viewpoint is kind of skewed, but it was well worth my time and my company's money |
17:46.42 | Rico29 | do anyone work with OPAL libs here ? |
17:47.03 | Qwell | RobH: so, how extensively do you guys use asterisk there? |
17:47.30 | b11d` | I am a FreeBSD Man.. not linux.. will I have much trouble with the dCAP? |
17:47.44 | RobH | Our main office phone system is using 1.2 with an iax2 connection to our voip provider |
17:48.00 | Qwell | ahh.. gonna upgrade? |
17:48.03 | RobH | Yes. |
17:48.10 | Qwell | hence the training, I assume |
17:48.19 | RobH | I am trying to make my test system, which is an exact copy of the main one, into 1.4 deployment |
17:48.43 | RobH | training was more for the fact I struggled solo to get 1.2 up and running, and it works, but its basic and doesnt do all we want it to do |
17:49.11 | RobH | got the dCAP because it was only 300 bucks more and I was already there =] |
17:49.18 | RobH | dunno if I passed yet, heh. |
17:49.33 | Qwell | really? hmm..I thought they graded it while the people were still here |
17:49.42 | Qwell | maybe not.. *shrug* |
17:49.45 | file | not all of it. |
17:49.46 | RobH | We got the practical graded |
17:49.48 | Qwell | ahh |
17:49.50 | RobH | but not the written |
17:49.56 | Qwell | makes sense |
17:49.59 | b11d` | aye |
17:50.05 | RobH | I passed the practical easily, but the written has some obscure questions. |
17:50.39 | RobH | I think I passed, but I am often overconfident about my test taking abilities. |
17:50.54 | file | yarrrr it's harsh |
17:51.06 | RobH | now if i could figure out why ztdummy is being stupid... |
17:51.11 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
17:51.42 | RobH | I am almost to the point of cheating and just purchasing a hardware card to put in the server at the office for the timing... |
17:52.03 | Qwell | RobH: nothing wrong with that |
17:52.11 | alrs | RobH: you can pull timing off a POS x100p card |
17:52.36 | RobH | The problem is eventually we will have 4 servers deployed for this, I rather not have to put a card in all of them. I rather get ztdummy working =] |
17:52.42 | alrs | just don't try to load ztdummy and a real zaptel module at the same time? |
17:53.11 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
17:53.12 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
17:54.31 | Defraz | I still see bridging even though I have canreivite=no |
17:54.35 | Defraz | Don't' know what is up. |
17:54.58 | RobH | alrs: when zaptel compiled, it saw no hardware and stated it was loading only the ztdummy. |
17:55.21 | RobH | same thing when it starts the zaptel service |
17:55.28 | RobH | sees no hardware, loads ztdummy |
17:55.37 | RobH | Qwell: You said you heard about me? eh? |
17:58.05 | RobH | its ztdummy being stupid, i load the service and asterisk generated audio ceases to function |
17:59.55 | *** join/#asterisk pud (n=jkatka@sea01-v502-nat.marchex.com) |
18:09.01 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
18:10.20 | RobH | caves and installs the tdm411B |
18:12.06 | *** join/#asterisk svenna_ (n=svenna@p548D357E.dip0.t-ipconnect.de) |
18:16.00 | *** join/#asterisk implicit (n=implicit@200.12.227.181) |
18:17.29 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
18:18.17 | Rico29 | ~xlite |
18:18.18 | jbot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
18:19.48 | *** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br) |
18:19.50 | *** join/#asterisk kfb (n=kfb@c-76-116-250-235.hsd1.pa.comcast.net) |
18:20.45 | really_phukt | xlite is cool, but do u know how to add another account? |
18:21.38 | _ShrikE | ~mini-mall |
18:21.38 | jbot | It's just like, it's just like, a russell.. b |
18:21.43 | _ShrikE | ha |
18:22.38 | [TK]D-Fender | really_phukt: You don't. Its limited in that way |
18:23.01 | *** join/#asterisk hohum (n=dcorbe@68.26.108.244) |
18:23.26 | really_phukt | I guess I gotta upgrade to eyebeam or bria |
18:23.27 | *** join/#asterisk AJayMN (n=Me@75-134-29-194.dhcp.mdsn.wi.charter.com) |
18:24.25 | RobH | try zoiper, lets you have 2 for free. |
18:24.38 | RobH | but still limited. |
18:25.11 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
18:25.19 | really_phukt | I tried it some time ago, but it doesn't look as cool :) |
18:25.39 | RobH | True, x-lite is smooth looking. |
18:25.41 | [TK]D-Fender | Cool... yeah, because thats what really important.... </sarcasm> |
18:25.46 | RobH | heh |
18:25.48 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
18:25.58 | RobH | but iax2 connection via softphone > sip phone that looks cool. |
18:26.06 | RobH | atleast for me. |
18:26.18 | lmadsen | grrrr... so when I do a transfer from the destination extension, I get a <ZOMBIE> channel (something useful to parse on), but when the source does the transfer, I don't get that <ZOMBIE> channel, thus I can't tell if it was a transfer or just a regular call in my dialplan. Also wonder why my TRANSFER_CONTEXT isn't working too (I'm just thinking out loud) |
18:27.36 | really_phukt | what do u guys think about ekiga? |
18:27.53 | lmadsen | I think you should try it and form your own opinion |
18:28.20 | *** join/#asterisk stoffell (n=stoffell@d51A4DE51.access.telenet.be) |
18:29.06 | really_phukt | thank you lmadsen... not |
18:29.30 | lmadsen | that joke is so 1992 |
18:29.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
18:30.34 | really_phukt | I don't remember putting any smileys at the end of my statement |
18:30.36 | *** join/#asterisk esaym (n=user@72.183.198.134) |
18:31.02 | lmadsen | touche |
18:31.40 | lmadsen | if ekiga didn't work for someone, then there wouldn't be active development on it -- the only way to get a real opinion of something is to try it in your own environment and see if it is useful. Taking a poll is hardly useful. |
18:32.01 | lmadsen | might as well ask what distro is the best |
18:32.38 | [TK]D-Fender | really_phukt: No smiley? Good... that means you should be prepared for further berating and disappointment. |
18:34.04 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
18:34.47 | really_phukt | I just wanted ask a question, as I figured someone here has experience with it. What's wrong with that? |
18:35.42 | [TK]D-Fender | really_phukt: Well, you did ask. We answered and felt cheated. Just letting you know its all part of the experience. |
18:36.03 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
18:36.34 | lirakis | is away (leaving..."the internets" are safe ... for now) |
18:36.38 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
18:36.50 | [TK]D-Fender | really you* |
18:36.51 | really_phukt | [TK]D-Fender: ok, fair enough |
18:36.56 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
18:37.11 | [TK]D-Fender | really_phukt: Excellent. |
18:37.35 | jameswf-home | weeee. |
18:38.18 | [TK]D-Fender | ~whee |
18:38.18 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
18:38.58 | AJayMN | I need to rewrite the callerid ID from the person calling into asterisk.. I need to add a 1+callerid |
18:40.01 | ccvp | heh |
18:40.17 | ccvp | "really_phukt", is your name an irc indirectly reference to Really Fucked? |
18:40.18 | ccvp | :) |
18:40.20 | cpm | yeah, [TK]D-Fender, what the heck is wrong with you? |
18:40.22 | ccvp | brb, coffee |
18:41.18 | lmadsen | AJayMN: Set(CALLERID(num)=1${CALLERID(num)}) |
18:41.31 | ccvp | d-fender, i think i saw some guild last night in COD4 multiplayer all with {TK} clan tags |
18:41.32 | really_phukt | ccvp, not it refers to you |
18:41.48 | ccvp | i instantly thought of d-fender, and Tha killaZ :) |
18:42.03 | shido6 | can you use playback or background to play a blob from realtime? |
18:42.12 | lmadsen | shido6: sorry, no such luck |
18:42.32 | lmadsen | shido6: would be very cool though, Qwell and Corydon76-dig starting something for that, but didn't get a chance to finish it |
18:42.39 | lmadsen | afaik |
18:42.42 | Qwell | huh? |
18:42.47 | Qwell | oh |
18:42.55 | lmadsen | thought you guys has a branch for playing stuff from ODBC blobs |
18:43.01 | Qwell | ast_storage |
18:43.02 | lmadsen | or at least a framework started for it |
18:43.05 | lmadsen | ya, that was the one |
18:43.23 | RobH | Intersting, I record my voicemail name, it says it saves itm but gives an error |
18:43.32 | RobH | [Mar 18 14:41:27] WARNING[5030]: app_directory.c:197 retrieve_file: Failed to obtain database object for 'asterisk'! |
18:43.52 | RobH | ack ,thats the dir call |
18:43.52 | RobH | [Mar 18 14:41:06] WARNING[5025]: app_voicemail.c:1408 store_file: Failed to obtain database object for 'asterisk'! |
18:44.39 | ccvp | RobH |
18:44.40 | ccvp | http://www.google.com/search?hl=en&q=+app_voicemail.c%3A1408+store_file&btnG=Search |
18:45.06 | Rico29 | is it easy to set up an enum server ? |
18:45.22 | RobH | ccvp: saying dont paste in channel? My bad, sorry about that. |
18:46.33 | ccvp | huh? |
18:46.38 | ccvp | no heh |
18:46.42 | RobH | the link goes to a blank entry page |
18:46.43 | ccvp | its just my first result from a google |
18:46.46 | ccvp | yes, just noticed |
18:47.13 | *** join/#asterisk bdheeman (n=bsd@122.161.65.75) |
18:47.53 | Rico29 | nobody knows about seting up an ENUM server ? |
18:48.23 | ccvp | rico, we are in the darkness, only coming out at night , asteriskers |
18:48.27 | RobH | Give me a bit of credit, I tried googling a bit =] |
18:48.34 | Rico29 | huh ? |
18:48.35 | ccvp | can I lead you to the light, so you come out, and socialize in the daylight, ie: call manager express ? :) |
18:49.05 | RobH | I find lots of folks with the issue, but no answers, it looks like asterisk is trying to reach a non-existant relational database, rather than use the AstDB |
18:49.14 | Rico29 | sorry but I don't understand everything, particullary the " ie: call manager express " |
18:49.17 | BCS-Satori | If i am using the exten = s,n,GotoIf($[${CALLERID(num)} = 4105551234]?reject:allow) feature to reject the phone number 4105551234, how would i go about adding a second, or even a thrid number to reject |
18:49.22 | ccvp | men like the darkness, because their asterisk ways are kept secret, for if they come to call manager express, their asterisk sins will be exposed during the day |
18:49.36 | *** join/#asterisk ACiDV (n=joel@122-205-229.dr.cgocable.ca) |
18:49.54 | Rico29 | hu ? |
18:49.59 | *** join/#asterisk razu__ (n=razu@195.222.7.33) |
18:50.25 | Rico29 | in basic english ? |
18:50.29 | Rico29 | :) |
18:50.31 | *** part/#asterisk bdheeman (n=bsd@122.161.65.75) |
18:50.36 | ACiDV | What can explain, on CDR log, that a call has a duration/billsec in NEGative ? (ex. -21642 seconds) |
18:50.43 | ACiDV | date/time change ? |
18:50.43 | ccvp | n/m rico, your french |
18:50.53 | ccvp | you dont understand , is joking, i cannot help with your enum server issue. |
18:50.59 | Rico29 | ok |
18:50.59 | Rico29 | :)p |
18:51.12 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
18:51.14 | Rico29 | what means "m/n" ? |
18:51.14 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
18:51.19 | ccvp | "nevermind" |
18:51.22 | Rico29 | k |
18:51.24 | ccvp | i shall teach you the ways of american |
18:51.24 | Rico29 | ok |
18:51.27 | ccvp | internet addiction slang |
18:51.53 | Rico29 | hard for a frenchie to understang everything |
18:51.57 | Rico29 | :p |
18:52.15 | ACiDV | ah les francais :P |
18:52.17 | ccvp | <PROTECTED> |
18:53.11 | Rico29 | ACiDV > mé euh |
18:53.23 | Rico29 | huhu |
18:53.38 | ACiDV | ishh ca la ete long avant que le la catch la 'Parlay voo fron say' ... |
18:53.47 | ACiDV | sorry ;) |
18:53.48 | Rico29 | lol |
18:54.02 | ccvp | acid |
18:54.04 | ccvp | wtf did you say |
18:54.07 | ccvp | im being sarcastic |
18:54.11 | ccvp | with my american spelling heh |
18:54.13 | ccvp | ? |
18:54.32 | ACiDV | :D ccvp, just took me time to understand that ' Parlay voo fron say ' mean ' Parlez vous francais ' :P |
18:54.43 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
18:55.51 | RobH | well hell. I was getting the error because I had ODBC storage set for voicemail. |
18:56.01 | RobH | Not sure how to remove that setting, so I just recompiled. easy enough. |
18:57.26 | *** join/#asterisk hmm-home (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
18:59.47 | [TK]D-Fender | ccvp: Garde ta guele sinon tu l'perdrait bien-tot ;) |
19:00.42 | Rico29 | i'm french and I didn't understant a damn thing |
19:00.45 | Rico29 | :D |
19:00.50 | Rico29 | sorry [TK]D-Fender |
19:00.58 | rkeene | Grr. |
19:01.02 | rkeene | kills Polycom |
19:01.45 | rkeene | Their webpage sucks. |
19:02.30 | ACiDV | Rico29, j'ai mon traducteur francais -> anglais et francais -> quebecois a mes cote, tjrs utile |
19:02.45 | Rico29 | :p |
19:02.52 | Rico29 | ok huhu |
19:02.56 | ccvp | D-Fender, le garde-boue, quelle heure sont moi et vous allant avoir notre temps d'homme privé ensemble ce soir en arrière àvotre appartement ? :) |
19:03.14 | ACiDV | doh =) |
19:04.26 | [TK]D-Fender | ccvp: That is a positively HORRID translation... |
19:05.28 | [TK]D-Fender | ACiDV: Je parles bilingue pour me sauver du temps ostie! |
19:05.29 | Rico29 | ccvp > hahaha |
19:06.08 | ACiDV | [TK]D-Fender ... c'est pas Andrew ca ? |
19:06.36 | [TK]D-Fender | ACiDV: Salut toi-la :) |
19:07.00 | ACiDV | Just to be sure =) |
19:07.07 | [TK]D-Fender | ACiDV: Je t'ai pas reconnu de meme... tiens-donc une seule nom! |
19:07.53 | [TK]D-Fender | ACiDV: Just installed that framework you've been using at home with the basic kit.... cool stuff... |
19:08.18 | [TK]D-Fender | ACiDV: And getting close to that re-install. I have a who server rack to rebuild. |
19:08.46 | ACiDV | ok, np =) |
19:08.52 | ACiDV | must leave, ttyl here or on msn ;P |
19:08.56 | [TK]D-Fender | whole* |
19:09.02 | [TK]D-Fender | ACiDV: later. |
19:09.18 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
19:11.28 | *** join/#asterisk Hemos\ (n=cyberspa@80.104.208.187) |
19:12.23 | ccvp | wtf |
19:12.26 | ccvp | theonion.com is insane |
19:12.28 | ccvp | (BREAKING NEWS)(AP/REUTERS) - President Of The United States Barack Obama has been assassinated while in Manhattan, NY. Vice President Hillary Clinton assumes the Oath of Office as President of the United States. Details Soon. |
19:15.14 | bkruse | ccvp: lol |
19:15.23 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
19:23.26 | *** part/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
19:29.25 | *** join/#asterisk datachomper (n=russ@75.146.194.59) |
19:30.11 | datachomper | How can I tell what DTMF mode, a call being sent to my asterisk box is in? |
19:31.14 | datachomper | sipinfo, rfc2833, or inband? |
19:35.09 | lirakis | is away (leaving..."the internets" are safe ... for now) |
19:39.39 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
19:48.54 | Greek-Boy | will wav sounds sound better on ulaw and alaw calls? |
19:49.38 | Greek-Boy | if all sound file types are installed will asterisk playback the format most suitable for the channel? |
19:51.41 | Katty | anyone know sendmail enough to give me a hand? |
19:51.54 | Katty | i'm doing voicemail to email, and it isn't happy )= |
19:54.15 | *** join/#asterisk PepOSX (n=angeldav@190.72.154.247) |
19:54.26 | x86 | Katty: what ever happened with that PRI? |
19:55.01 | Katty | x86: it works! |
19:55.12 | Katty | x86: tho, FOP doesn't like DID numbers much. |
19:58.19 | datachomper | Hola Katty |
20:02.57 | *** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
20:03.51 | Katty | datachomper: hola! como estas? (= |
20:03.52 | *** join/#asterisk shtoom (n=godson@123.176.41.46) |
20:04.05 | jamessan | is it possible to get * to add a Route field to the Via header without having seen a Record-Route header first? |
20:04.22 | x86 | Katty: what was the problem with the PRI? |
20:04.48 | Katty | x86: i had a line in zaptel wrong... |
20:04.57 | Katty | x86: sangoma fixed me right up tho (= |
20:05.11 | Qwell | that'll teach ya |
20:05.13 | JunK-Y | Katty: which line? |
20:05.22 | Katty | sec, lemme me find my blog post |
20:05.51 | RobH | Greek-Boy: someone answer you? |
20:06.10 | x86 | Katty: yeah they are great :) |
20:06.19 | RobH | Greek-Boy: Asterisk will play back whatever soundfile it has that requires the least cpu power to transcode (so if the caller is on ulaw, it will choose the ulaw sound file, as it takes no cpu) |
20:06.29 | Qwell | Greek-Boy: no phones really support wav, so it'll have to be transcoded - and in some cases will sound "worse" |
20:07.08 | RobH | Yea, if you have a ton of HDD space to burn, just install all the possible codecs you will allow to connect, and remember to convert your custom recordings into the various formats. |
20:07.33 | Katty | x86: JunK-Y: http://angela.sleekgeek.org/2008/03/14/sangoma-pri-card-setup-a101/ <- see note from sangoma. |
20:08.35 | JunK-Y | Katty: so it was just a bug with ur config, right? |
20:08.40 | Katty | JunK-Y: ya |
20:09.38 | JayTee52 | Katty, I just read the blog |
20:09.42 | x86 | Katty: ah, see, I told you I've not done much with PRI's... |
20:11.16 | JayTee52 | Katty, I'm wondering if I can take a context that has all of our extensions in it and add it to my [from-pstn] context as an include and then have an additional line that would be executed if none of the extensions in the included context match |
20:14.10 | Greek-Boy | thanks RobH and Qwell |
20:14.17 | RobH | hope it helps =] |
20:15.14 | Greek-Boy | I'm sure it will, I'm just going to go ahead and install all the sound formats |
20:15.36 | RobH | if you have the space to kill, it cannot hurt |
20:16.06 | Greek-Boy | I noticed on the asterisk web site there is a format called sln16 available for download |
20:16.31 | RobH | I dunno that off the top of my head, I reference this: http://www.voip-info.org/wiki-Asterisk+codecs |
20:17.05 | RobH | I do not force many codecs on my system though, I just let folks use gsm, ulaw, and alaw in that order. |
20:17.42 | RobH | it all goes through a single voip provider which lets me trunk iax connections, so I am good as far as bandwidth. |
20:17.57 | JayTee52 | does Asterisk handle which codec to use automatically or does that have to be configured in the musiconhold.conf or another config file? |
20:18.26 | RobH | its automatic, in regards to cpu utilization, not bandwidth. |
20:18.38 | RobH | (if I am saying something incorrect, someone jump in and tell me.) |
20:19.16 | RobH | do a core show translation in the CLI, it shows you cpu loads to transcode codecs (in miliseconds) its just kinda nea.t |
20:19.18 | RobH | neat even |
20:19.51 | RobH | but then again, I am a giant freakin nerd who stares at a terminal for 10 hours a day ;_; |
20:22.21 | *** join/#asterisk harlequin516 (n=sham@stewart.styk.net) |
20:22.41 | harlequin516 | how can i find out fromthe asterisk console what is the soundfile path? |
20:25.09 | *** join/#asterisk mfedyk (n=mfedyk@adsl-71-134-153-204.dsl.irvnca.pacbell.net) |
20:25.44 | harlequin516 | Hello, anyone here? |
20:25.50 | harlequin516 | helo? |
20:25.58 | RobH | folks are here, but they are lurking and working, gotta be patient. |
20:26.58 | codefreeze | harlequin516: There's 265 lurkers in here. The walls are literally plastered with eyes! |
20:27.03 | harlequin516 | Okay, sorry. |
20:27.13 | mfedyk | anyone know how to use two or more subexpressions to extract data into variables? |
20:27.20 | mfedyk | in the dialplan |
20:27.34 | mfedyk | I have been able to get one expression and extract it |
20:27.42 | mfedyk | but two or more just return the first |
20:28.19 | codefreeze | harlequin516: I can't think of a cli command to show that; you may have to consult the source. Usually it's /var/lib/asterisk/sounds |
20:28.41 | mfedyk | ;extract dialed number |
20:28.42 | mfedyk | exten => s,n,Set(dialed_num=$[ "${MACRO_EXTEN}" =~ "(.*)\\*" ]) |
20:28.42 | mfedyk | ;extract user specified callerid |
20:28.42 | mfedyk | exten => s,n,Set(callerid_num_custom=$[ "${MACRO_EXTEN}" =~ "\\*(.*)" ]) |
20:28.55 | mfedyk | I currently use one line per part I want to extract |
20:29.11 | mfedyk | that's ok for two, but if I wanted to extract more, it'd be a lotta lines |
20:29.38 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
20:29.58 | *** join/#asterisk ManxPower (n=manxpowe@107.sub-70-221-153.myvzw.com) |
20:30.29 | harlequin516 | codefreeze: thanks |
20:31.40 | *** join/#asterisk implicit_ (n=implicit@200.12.227.181) |
20:33.22 | Greek-Boy | yeah Rob you're right, gotta be patient :) |
20:33.34 | Greek-Boy | RobH thanks for your time and help |
20:33.48 | RobH | anytime |
20:34.11 | RobH | when i answer questions i just end up understanding things better, like anyone else. |
20:39.41 | *** join/#asterisk adrin_ (n=adrin@chello083144070043.chello.pl) |
20:39.57 | adrin_ | hello |
20:40.17 | RobH | Ok, a question for me, whats the asterisk application to call to add to a number (may jut be expression) I have the pattern match if they dial a 7 digit number, it means its local to the office area code, so I want asterisk to slap the area code in front of the EXTEN before passing it to the outbound. |
20:40.57 | Greek-Boy | what is the app ivrdemo? |
20:43.52 | adrin_ | Hello, i have a question: is there a way to allow a calling user to select, after connecting, an extension (like 9x) and after a period of inactivity to redirect him (connect) to a default extension lets say 90? |
20:44.02 | *** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
20:44.08 | russellb | Greek-Boy: it's an example for an internal IVR API ... |
20:44.17 | russellb | which never has been used outside of that demo, heh |
20:46.09 | _ShrikE | Hey russelb, not to nag, but were you able to backport 1.6 hint? I was excited to hear you were doing it :) |
20:46.23 | russellb | no, i haven't done it, sorry |
20:46.29 | russellb | got very busy making releases |
20:46.41 | _ShrikE | thats OK. |
20:55.24 | *** join/#asterisk akira2014 (n=chatzill@173.Red-88-17-52.dynamicIP.rima-tde.net) |
20:55.30 | akira2014 | hello |
20:56.08 | akira2014 | can some one tell me how to dial a iax extension in the dialplan |
20:56.10 | akira2014 | ? |
20:56.15 | akira2014 | thk's |
20:56.40 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
20:56.56 | Maliuta | dial(IAX/etn) |
20:57.06 | Maliuta | RTFM FFS |
20:57.41 | *** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-32.myvzw.com) |
20:58.24 | *** join/#asterisk kamaji (n=kamaji@resnet-186224.resnet.bris.ac.uk) |
21:00.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
21:01.23 | akira2014 | Maliuta: i don't need to put IAX2 |
21:01.25 | akira2014 | ? |
21:01.59 | Maliuta | Read The F*@!ing Manual |
21:02.03 | *** join/#asterisk rpyne (n=richard@69.77.169.14) |
21:02.15 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
21:03.04 | rpyne | Does nayone know how to startup the tftp server on Asterisk 1.4. I'm running CentOS |
21:03.21 | Katty | wibbles |
21:03.24 | russellb | Maliuta: an RTFM attitude is not welcome here. |
21:03.37 | russellb | rpyne: asterisk has no tftp server ... |
21:03.43 | russellb | that's a completely different application |
21:03.54 | alrs | isn't there a tftp server going in to 1.6? |
21:04.01 | russellb | no |
21:04.09 | Qwell | alrs: there's a web server |
21:04.13 | Qwell | (in 1.4 too though) |
21:04.17 | russellb | a very minimal HTTP server, yes |
21:05.50 | kamaji | Can I run asterisk and a SIP cilent on the same computer or does SIP have to use a certain port |
21:05.53 | rpyne | How do I get a TFTP server up and running on my CentOS so that I can manage some 400 aastra sip phones |
21:06.00 | [TK]D-Fender | Qwell, What is the advantage of having an HTTP server directly as part of * again? |
21:06.26 | [TK]D-Fender | kamaji, Set your client to use another port so it doesn't interfere with * binding to the primery. |
21:06.38 | kamaji | [TK]D-Fender: mkay, thanks |
21:06.55 | [TK]D-Fender | rpyne, "man tftpd.conf" |
21:07.42 | rpyne | Thanks much [TK]D-Fender |
21:07.56 | Katty | how do i use asterisk?????????!!11 |
21:08.02 | Katty | oh, wait |
21:08.02 | rpyne | exit |
21:08.05 | Qwell | ~nowwhat |
21:08.06 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
21:08.07 | Qwell | Katty: ^^ |
21:08.09 | Katty | how to use asteriskk plzzzzzz |
21:08.11 | Katty | etc. |
21:08.13 | *** part/#asterisk rpyne (n=richard@69.77.169.14) |
21:08.26 | Katty | exim4 doesn't like me. |
21:08.28 | file | tickles Katty |
21:08.33 | Katty | i shall beat it with sticks until its attitude improves. |
21:08.43 | Katty | hugs file |
21:10.37 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
21:10.45 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
21:12.00 | lirakis_away | is away (leaving..."the internets" are safe ... for now) |
21:12.19 | Katty | file: i guess exim4 does love me afterall |
21:12.27 | Katty | file: i'm just not loving it back with my firewall :/ |
21:12.52 | file | :( |
21:13.06 | *** join/#asterisk WhipsMcGee (n=barney@c-71-199-49-168.hsd1.co.comcast.net) |
21:14.09 | WhipsMcGee | can someone walk me through get my a sip device working. my server is behind a firewall and so is the sip device. I've had it working before with opening a few ports and something with externip but I'm setting up a new server and can't remember what I did. |
21:14.53 | ManxPower | WhipsMcGee: externip/externhost and localnet |
21:15.40 | kamaji | does anyone know how to change Ekiga's default SIP port by any chance? people in #ekiga are asleep |
21:16.42 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
21:17.06 | Yourname` | Hello. I was wondering if some good soul would be kind enough to help me figure out why Asterisk isn't using all the cores.. or if it is and htop is fux0red (which looks like it may be, thanks to mvanbaak) how can i find out in top? |
21:17.13 | putnopvut | kamaji: I've done that before, and you don't actually change it in ekiga itself, oddly enough. I used gconf-editor to change it the time that I used it. |
21:17.13 | Yourname` | nods to russellb |
21:17.17 | WhipsMcGee | localnet should be 192.168.1.0/255.255.255.0 right? |
21:17.28 | putnopvut | kamaji: unfortunately, it's been a while so I don't know the exact steps involved. |
21:17.52 | WhipsMcGee | if my asterisk server is on 192.168.1.200 |
21:18.04 | kamaji | putnopvut: okay, that's bizarre ^^ thanks |
21:18.50 | jameswf-home | lmao I just copied the asterisk "critical update" email to the trix forums... now sit back and wait for an outcry of people demanding they update.... |
21:19.09 | *** join/#asterisk toddejohnson (n=toddejoh@63-252-82-2.ip.mcleodusa.net) |
21:19.46 | [TK]D-Fender | WhipsMcGee, Yes. And go read this : |
21:19.47 | [TK]D-Fender | ~sipnat |
21:19.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:19.49 | putnopvut | kamaji: inside gconf-editor, apps->ekiga->protocols->sip->listen_port |
21:20.25 | adrin_ | Hello, i have a question: is there a way to allow a calling user to select, after connecting, an extension (like 9x) and after a period of inactivity to redirect him (connect) to a default extension lets say 90? |
21:21.00 | kamaji | putnopvut: i haven't got gnome installed so I think i'll just switch to kiax |
21:21.07 | kamaji | putnopvut: thanks anyway though :> |
21:21.45 | putnopvut | kamaji: no prob. it took me forever to figure it out when I was using ekiga last year. |
21:21.57 | Greek-Boy | why is it that most of the formats core sound files in http://downloads.digium.com/pub/telephony/sounds/ have an abnormally small file size? |
21:22.19 | Qwell | Greek-Boy: because they are small |
21:22.21 | *** join/#asterisk jicksta_ (n=jicksta@dsl093-128-144.sfo4.dsl.speakeasy.net) |
21:22.27 | putnopvut | That was easy. |
21:22.34 | Yourname` | So easy. |
21:22.37 | Qwell | oh, those |
21:22.48 | Qwell | 55 bytes for the tar is abnormally small :D |
21:22.55 | file | symlinks? |
21:23.13 | Qwell | yeah...but some aren't |
21:23.45 | *** join/#asterisk Synoptic (i=Synoptic@modemcable105.136-203-24.mc.videotron.ca) |
21:24.04 | Synoptic | hi all |
21:24.11 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
21:24.37 | putnopvut | adrin_: the WaitExten application sounds like what you want. |
21:24.52 | Synoptic | i'mhaving trouble compiling the latest zaptel source. I'm using a base system running gentoo with kernel 2.6.24. I have configured my kernel myself, and maybe i forgot to include some option in it. can someone help me debug this please ? |
21:24.56 | putnopvut | adrin_: it will wait for the user to type an extension, and once it has been input, it will go to that extension. |
21:27.02 | adrin_ | ok thanks |
21:28.15 | adrin_ | i will see it |
21:28.26 | adrin_ | :beer: |
21:28.26 | *** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
21:29.06 | Synoptic | Hmm.. what is the kernel config to allow module loading regardless of kernel version ? |
21:29.31 | *** join/#asterisk Cazper (n=cazper@85.196.120.126) |
21:31.27 | Synoptic | I think I found it |
21:31.31 | Synoptic | recompiling |
21:33.30 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-183-106.dsl.tul2ok.sbcglobal.net) |
21:34.27 | Greek-Boy | Qwell so whats up with those sounds? |
21:35.09 | tzafrir | Synoptic, what errors do you get? |
21:35.21 | *** join/#asterisk warlock_mza (n=warlock@190.48.48.127) |
21:35.46 | *** part/#asterisk warlock_mza (n=warlock@190.48.48.127) |
21:36.48 | tzafrir | Synoptic, hmm.. why would you like to override module versions? |
21:37.30 | tzafrir | module versions can save you from nice panics when external modules are involved |
21:39.51 | Synoptic | tzafrir : Well, actually, i did not enable the options.. i was speaking out-loud. |
21:40.08 | Synoptic | here is the error when make module_install : wctc4xxp.ko needs unknown symbol release_firmware |
21:40.14 | Synoptic | this is one of the many error.. |
21:40.16 | WhipsMcGee | do I just do amportal restart after changing externip or do I need to restart other services |
21:40.21 | Synoptic | what can cause an unknown symbol ? |
21:41.25 | tzafrir | What version of Zaptel do you use? 1.4.9.2? |
21:41.32 | Synoptic | ya |
21:41.33 | Synoptic | but |
21:41.39 | Synoptic | I havent enabled osme kerbnel option |
21:41.43 | Synoptic | let me reboot hte new kernel |
21:41.48 | Synoptic | and try to recompile zaptel |
21:41.51 | *** join/#asterisk implicit (n=implicit@200.12.227.181) |
21:42.13 | tzafrir | hmmm... is the detection of firmware loading broken there? hmmm.... |
21:42.21 | tzafrir | do you have any Zaptel hardware? |
21:42.42 | Greek-Boy | I installed unixodb and myodbc packages on debian but menuselect still doesnt allow cdr obc and func_odbc? |
21:43.00 | tzafrir | Synoptic, to build vs. a specific kernel version or tree, use KVERS (and maybe also KSRC) |
21:43.03 | putnopvut | Greek-Boy: make distclean && ./configure |
21:43.14 | *** join/#asterisk RoyK (n=roy@ip-77-54-149-91.dialup.ice.no) |
21:43.39 | Synoptic | tzafrir: I have a TDM400P |
21:44.04 | tzafrir | so you don't actually need firmware. We can afford faking |
21:44.38 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
21:44.53 | Synoptic | ok |
21:46.18 | tzafrir | Synoptic, what is the value of CONFIG_FW_LOADER in your .config ? |
21:46.18 | Synoptic | tzafrir : WARNING: "crc_ccitt_table" [/root/download/zaptel-1.4.9.2/kernel/zaptel.ko] undefined! |
21:46.51 | Synoptic | let me check |
21:47.16 | Synoptic | NOT SET |
21:47.20 | tzafrir | http://zaptel.tzafrir.org.il/#_kernel_configuration |
21:47.24 | Greek-Boy | putnopvut: unfortunately, make distclean did not do the trick :( |
21:48.53 | Synoptic | tzafrir : readin your site, thanks |
21:49.31 | tzafrir | Synoptic, it's not "my site". It's the README file in the zaptel source tree, formatted with asciidoc |
21:49.47 | tzafrir | credit should go to the authors of asciidoc |
21:50.17 | putnopvut | Greek-Boy: did you also rerun configure? |
21:50.31 | tzafrir | Synoptic, try 'make README.html' |
21:51.05 | Synoptic | tzafrir : ok. btw, what is the name of the CONFIG_FW_LOADER in menuconfig ? |
21:51.34 | Synoptic | i know i dont need it since I'm not using firmware hardware |
21:52.38 | Greek-Boy | putnopvut: yes I did |
21:52.44 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
21:53.44 | putnopvut | Greek-Boy: hrmmm...what does menuselect say the dependencies for those modules are? |
21:53.52 | putnopvut | Greek-Boy: oh oh!! |
21:53.56 | putnopvut | unixodbc-dev! |
21:54.10 | putnopvut | Not just unixodbc. |
21:54.18 | lmadsen | indeed |
21:54.24 | *** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-32.myvzw.com) |
21:54.54 | Greek-Boy | brb |
21:55.50 | WhipsMcGee | OK, so I've read a few documents that all say I need to setup externip and localnet, forward the ports and that should be it. |
21:56.15 | Yourname` | A call is made by Asterisk, and when connected, it's sent via sip to another Asterisk install where there's a queue of agents. When agents rcv the call, the agent cannot hear the other side in about 10-12 calls in an hour out of the 60-70 calls they get in an hour. What could be the issue here? |
21:56.36 | Yourname` | I know NAT isn't. All agents have no firewalls on their XP PCs, and they are on direct IPs. |
21:57.30 | Yourname` | My guess, enroute to the queue box, the caller hangs up, but Asterisk still carrys on the call to the agent. |
22:01.31 | ManxPower | WhipsMcGee: Great, so it's working? |
22:01.44 | WhipsMcGee | no |
22:02.02 | ManxPower | what ports did you forward? |
22:02.22 | WhipsMcGee | 5060 and 10000-13000 and I set /etc/asterisk/rtp.conf to be the same and it's also the same on the phone. |
22:02.34 | ManxPower | you didn't do something silly like put a hostname in for the value of externip, did you? |
22:02.48 | ManxPower | WhipsMcGee: the phone is behind a different NAT, I assume. |
22:03.17 | WhipsMcGee | yeah, I've tried both externip= with my external ip and externhost= with a dynamic host that resolves to my IP. neither works |
22:03.26 | WhipsMcGee | yeah, the phone is behind my nat at home. |
22:03.29 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
22:07.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:08.49 | Synoptic | tzafrir : I'm getting nowhere, though errors are different now ! |
22:10.09 | Synoptic | tzafrir : no more module error.. which is good, but the executable file zaptel is nowhere to be found, and my /dev/ctl/zap is non existen. |
22:10.43 | Synoptic | tzafrir :a hh nevermind |
22:10.51 | Synoptic | it seems to work after a depmod -a |
22:10.55 | Synoptic | let me reboot to see |
22:12.31 | Synoptic | tzafrir : hmm, /dev/zap is non existent.. |
22:12.40 | Synoptic | i'musing udev, is that a problem ? |
22:13.32 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
22:13.41 | tzafrir | Synoptic, for starters, what is the output of: cat /proc/zaptel/* |
22:14.22 | Synoptic | tzafrir : befire I answer your question, I noticed that my zaptel modules are not loaded at boot, but they do work with a modprobe |
22:14.38 | Synoptic | modprobe zaptel and my /dev/zap appears |
22:14.49 | Synoptic | no I need them to load at boot |
22:15.23 | tzafrir | Synoptic, gentoo has a file (/etc/modules-2.6 ?) for a list of modules to load at startup, IIRC |
22:15.28 | Synoptic | yes |
22:15.30 | Synoptic | hold on |
22:15.37 | tzafrir | But I wonder why the device is not hotplugged |
22:15.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
22:18.16 | Synoptic | because it is absent for the moment |
22:18.41 | Synoptic | it's in my live tribox box (which I'm going to REMOVE cause I dislike it a lot) |
22:19.16 | Synoptic | now |
22:19.24 | Synoptic | let's assume them odule loading problem is solved |
22:19.35 | Synoptic | the make config isn't working |
22:20.00 | Synoptic | make: zaptel: Command not found |
22:21.13 | WhipsMcGee | Maybe it's something else. If I put both my asterisk server and my sip in DMZ I still get nothing |
22:22.17 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
22:22.31 | *** join/#asterisk droops (n=droops@74.193.237.138) |
22:24.10 | Synoptic | tzafrir : ok, i'm getting it to work |
22:24.21 | Synoptic | had to tweak small things in my systems |
22:24.23 | Synoptic | rebooting |
22:24.51 | tzafrir | <Synoptic> make: zaptel: Command not found --- where exactly? |
22:25.48 | tzafrir | any chance you have no perl on your system? |
22:25.50 | Synoptic | tzafrir : well, command is found, it's the exec file functions it calls that was not found. it was named functions.sh. it is fixed now |
22:26.32 | Synoptic | tzafrir : maybe i have no perl, but now I get the no telephony device message, which is good, and true. I have a X100p clone that I can throw-in for test purpose since I cannot remove my TDM400p from thelive system |
22:27.33 | tzafrir | Synoptic, kernel/xpp/utils/zaptel_hardware |
22:28.00 | tzafrir | (unless you have no perl) |
22:29.04 | r0d3nt | <SecNews> Title: (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released |
22:29.04 | r0d3nt | <SecNews> Link: http://www.asterisk.org/node/48466 |
22:29.04 | r0d3nt | <SecNews> Description: The Asterisk.org development team has released four new versions of Asterisk to address critical security vulnerabilities. |
22:29.07 | r0d3nt | opps |
22:29.18 | *** join/#asterisk CVirus (n=GoD@82.201.222.52) |
22:33.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
22:33.19 | Synoptic | tzafrir : i have no perl. my x100p is in, but zttool reports no x100p |
22:33.56 | Synoptic | tzafrir : i have to go play my game of badminton. I'll try to catch ya later, ok ? |
22:34.14 | tzafrir | Synoptic, modprobe wcfxo |
22:34.33 | tzafrir | or just install perl... |
22:34.37 | Synoptic | tzafrir : I do have perl.. |
22:34.45 | Synoptic | Module Size Used by |
22:34.45 | Synoptic | zttranscode 8840 0 |
22:34.46 | Synoptic | ztdynamic 11472 0 |
22:34.46 | Synoptic | ztdummy 5032 0 |
22:34.46 | Synoptic | wctdm 37952 0 |
22:34.46 | Synoptic | wcfxo 12576 0 |
22:34.48 | Synoptic | zaptel 193156 5 zttranscode,ztdynamic,ztdummy,wctdm,wcfxo |
22:34.50 | Synoptic | sorry for the flood |
22:34.58 | Synoptic | dev-lang/perl-5.8.8-r4 |
22:35.09 | Synoptic | wcfxo is loaded but not used |
22:35.57 | tzafrir | so what's the output of zaptel_hardware ? |
22:36.10 | Synoptic | i dont have that executable file. |
22:36.19 | Synoptic | btw my zaptel.conf is still untouched. |
22:36.26 | tzafrir | (a pastable lspci) kernel/xpp/utils/zaptel_hardware |
22:36.57 | Synoptic | 01:00.0 Communication controller: Motorola Wildcard X100P |
22:37.12 | Greek-Boy | putnopvut you were right. most dependencies required the dev package too. thank you |
22:37.15 | Synoptic | my clone card (which works btw.. poor quality, but it does work for the purpose of installing zxaptel) |
22:37.28 | putnopvut | Greek-Boy: glad to help |
22:37.40 | *** join/#asterisk mvicha (n=someaddr@201.234.95.162) |
22:37.44 | Synoptic | 01:00.0 Communication controller: Motorola Wildcard X100P |
22:37.44 | Synoptic | Kernel modules: wcfxo |
22:37.48 | Synoptic | it does use it |
22:39.01 | mvicha | hello everybody, can ne1 help me configure a digium te120p? |
22:39.54 | Nugget | digium can. |
22:40.22 | Greek-Boy | now I'm trying to find the debian package for imap_tk |
22:41.39 | Synoptic | asterisk kernel # zaptel_hardware |
22:41.39 | Synoptic | pci:0000:01:00.0 wcfxo- 1057:5608 Wildcard X100P |
22:41.46 | Synoptic | I just found how to compile the tool :) |
22:41.46 | Synoptic | ok |
22:41.47 | Synoptic | gtg |
22:42.04 | Synoptic | i'll catch ya later, thabnx for your patience tzafrir |
22:42.39 | seanbright | how do you cancel an attended transfer? |
22:44.32 | *** join/#asterisk mvicha (n=someaddr@201.234.95.162) |
22:44.38 | mvicha | hello everybody, can ne1 help me configure 01:01.0 Ethernet controller: Digium, Inc. Wildcard TE120P single-span T1/E1/J1 card (rev 11) |
22:45.40 | mvicha | or at least how to test if it's configured correctly? |
22:46.31 | russellb | mvicha: support@digium.com |
22:48.24 | mvicha | russellb, that's too much for just a simple test I think... I need just a basic configure :s |
22:48.52 | Greek-Boy | where do I get myself astersik core sounds since the ones on the asterisk site are only 55bytes in size and are obviously not intact? |
22:49.15 | Qwell | Greek-Boy: make menuselect, edit the sounds options, then make install |
22:49.50 | Greek-Boy | great |
22:53.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582453.dsl.bell.ca) |
22:59.48 | *** join/#asterisk djc123 (n=djc@65.209.147.253) |
23:00.22 | djc123 | hrm.. voicemail.conf... externnotify.. how can my script distinguish between the cases 'a voicemail was just left' and 'voicemail was checked'.. |
23:00.54 | djc123 | it would be more useful to have a script only called when a new vm is left, and pass it the # of the message, or even the filename to it |
23:01.31 | djc123 | i want to email new vm's to myself, but I want mp3.. since it doesnt support mp3 directly, i will have to run a script to use sox.. BUT.. I only want it to email me a message when a new one is left, not anytime I might check it manually |
23:02.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
23:06.17 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
23:06.19 | *** join/#asterisk qdk (n=qdk@93.160.126.34) |
23:07.00 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
23:07.18 | djc123 | to be honest, id almost like to suppress the entire normal voicemail system, and *just* have new voicemails sent as email attachments in mp3 format |
23:07.54 | djc123 | visual voicemail for iphones that dont have ATT = setup seperate pop account, have voicemails come to that mailbox as mp3.. |
23:11.46 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
23:19.43 | Rico29 | for realtime mode, if I add fields in the database, like "bandwith" or anything else which is not in http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip database, will they be ... used by asterisk ? |
23:20.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:21.02 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
23:21.22 | *** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
23:23.53 | *** join/#asterisk cm|laptop (n=gphreak@cpc1-cosh3-0-0-cust885.cos2.cable.ntl.com) |
23:28.04 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
23:40.52 | mfedyk | Rico29: some yes, some no. useragent, no. trustrpid, yes |
23:40.52 | mfedyk | Rico29: try it and see |
23:41.09 | Rico29 | ok |
23:41.10 | Rico29 | thx |
23:45.45 | Rico29 | looks like it works with bandwith |
23:45.49 | *** join/#asterisk talntid (n=swarm@66.208.251.174) |
23:45.53 | talntid | Hi guys! |
23:45.58 | jameswf-home | see it before its moderated off :)) http://www.trixbox.org/forums/trixbox-pro/trixbox-pro-help/paid-telephone-support-really-sucks |
23:46.07 | Rico29 | good night : 1am here. |
23:46.10 | Rico29 | :] |
23:47.24 | talntid | So I have a Nortel BCM 400 call box... and want to switch to Asterisk... possible? |
23:47.48 | talntid | I would like to eliminate the BCM 400, and use Asterisk to perform its duties.. |
23:48.02 | talntid | That is what Asterisk is for, right? |
23:48.21 | Rico29 | with realtime, y dont see my peers with a "sip show peers". How can I see them ? |
23:48.40 | Rico29 | tainted_ > dont know, sorry |
23:49.12 | Rico29 | talntid :] |
23:49.51 | fujin | mm, app_page and rick astley = win |
23:49.58 | jameswf-home | assuming Nortel BCM 400 is a PBX yes talntid |
23:50.12 | talntid | It is. |
23:50.18 | jameswf-home | ~rickroll |
23:50.18 | jbot | from memory, rickroll is http://www.internetisseriousbusiness.com, or http://www.xkcd.com/396/ |
23:50.33 | fujin | or http://www.mylazysunday.com |
23:51.02 | jameswf-home | dont forget the flea market |
23:51.06 | jameswf-home | ~fleamarket |
23:51.07 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
23:51.07 | talntid | Hmm... |
23:51.15 | talntid | So which one of you wants to come set me up on Asterisk? :) |
23:51.36 | jameswf-home | that would depend on your budget |
23:51.42 | jameswf-home | I am not a cheap date |
23:51.43 | Rico29 | sip show peer <peer-name> load |
23:51.58 | talntid | I'll pay ya $1700 to come do it :) |
23:52.05 | Rico29 | can I make the dynamic peers viewable by default ? |
23:52.07 | jameswf-home | + expenses |
23:52.08 | talntid | plus pizza ;P |
23:52.24 | Rico29 | pleae |
23:52.26 | Rico29 | please |
23:52.47 | talntid | you pretty good at Asterisk, james? :P |
23:53.08 | jblack | man. That Obama speech is incredible. I think it's one of the top 10 american speeches of all time. |
23:53.22 | jameswf-home | plays with it 10 hours a day |
23:53.26 | talntid | eep. |
23:53.41 | Rico29 | jameswf-home > any solution tu my little problem ? |
23:53.46 | Rico29 | to* |
23:53.51 | talntid | can ya answer some Q's for me? ones that I could probably answer myself by googling for a few hours... |
23:54.08 | talntid | don't want to impose, but you could probably answer them in 5 minutes |
23:54.14 | jameswf-home | [TK]D-Fender: is the google proxy |
23:54.19 | jblack | I'm not as good as jameswf-home, but I'll work for that |
23:54.41 | jameswf-home | ~[TK]D-Fender |
23:54.42 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
23:54.49 | jameswf-home | see |
23:54.52 | jameswf-home | :) |
23:55.23 | talntid | I run an outbound call center. I employ 30 sales reps who make primarily outbound calls. |
23:55.41 | jblack | ok |
23:55.48 | talntid | I currently have a BCM 400, without the ability to record calls ($13,000), and recently got the RSI telecost software to get call details |
23:55.57 | Rico29 | rtcachefriends=no in sip.conf doesn't change anything |
23:56.26 | talntid | I gotta say, I really dislike the BCM 400. I hate having to pay $13,000 to record calls. |
23:56.33 | jblack | That's silly. |
23:56.35 | talntid | so I won't pay that much. |
23:56.41 | jblack | I record every call I get |
23:56.43 | talntid | Does asterisk allow for that? |
23:56.48 | talntid | ok, so obviously it does |
23:56.48 | jameswf-home | Asterisk will record and log calls........ |
23:57.09 | talntid | Any downsides? |
23:57.22 | jameswf-home | talntid: you may wanna try elastix... give you all the features and easy for a novice to admin |
23:57.25 | jblack | and numerous things are logged by default, and many other things can be logged too. And if you want even more data than that, you can even dump more data into an sql server. |
23:57.27 | talntid | I am a very DIYer... |
23:57.34 | talntid | I can figure stuff out pretty well... :) |
23:57.53 | talntid | Will it work with my current phones? and How? |
23:58.05 | jameswf-home | what are your current phones |
23:58.13 | jblack | That's as far as I'm going until I get a deposit towards that promised 1700. :) |
23:58.36 | talntid | elastix only has support for 12 concurrent calls.. i need more ;) |
23:59.04 | jblack | that's no problem. |
23:59.07 | jameswf-home | says who? |
23:59.19 | talntid | says elastix |
23:59.27 | talntid | Elastix Appliance ELX-025 |
23:59.27 | jameswf-home | concurrent calls 100% hardware dependent up to like 400 |
23:59.42 | talntid | sorry, was just reading main page |
23:59.45 | jameswf-home | no no buy a server install elastix |
23:59.53 | talntid | ok. I have a server |