| 00:00.42 | rvhi | in sip.conf, what's the difference between callgroup and pickupgroup? |
| 00:01.06 | dijungal | fujin: thanks |
| 00:01.16 | dijungal | i just need to figure out how to get the agent id on the file name |
| 00:01.26 | fujin | 1) |
| 00:01.29 | fujin | stop using agentcallbacklogin |
| 00:01.30 | fujin | 2) |
| 00:01.36 | fujin | use aqm/rqm for call delivery |
| 00:01.40 | fujin | err, for agent logins |
| 00:01.45 | fujin | 3) use a local channel for call delivery |
| 00:01.48 | fujin | =win win situation |
| 00:02.05 | dijungal | why use a local channel.. |
| 00:02.22 | fujin | local channels are awesome :) |
| 00:02.37 | dijungal | most docs recommend u use agents then queues setup with agents as members |
| 00:02.45 | dijungal | and use agentcallbacklogin |
| 00:02.58 | denon | rvhi: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 00:03.08 | dijungal | although i don't think that setup works perfectly |
| 00:03.16 | rvhi | denon: i read that, but still don't get it |
| 00:04.13 | rvhi | i can understand pickupgroup, if you are in the group, you can pickup other people in the same pickupgroup |
| 00:04.19 | rvhi | but what is callgroup? |
| 00:04.53 | dijungal | fujin: what is someone is not as the local channel |
| 00:04.56 | fujin | dijungal: those docs are wrong |
| 00:06.45 | dijungal | but when using local channels, someone may getup from their seat and go the the restroom |
| 00:06.53 | dijungal | that channel will still ring |
| 00:09.17 | fas3r | good night |
| 00:13.32 | *** join/#asterisk digimania (n=digimani@24-119-242-39.cpe.cableone.net) |
| 00:16.08 | JerJer | has anyone else noticed a 2 second delay when using the A dial modifier ? |
| 00:17.50 | Qwell | What's A supposed to do? |
| 00:18.30 | _charly_ | A(x) - Play an announcement to the called party, using 'x' as the file. |
| 00:18.56 | Qwell | it may be waiting to setup the audio path |
| 00:19.32 | digimania | can someone help me figure out how to manually copy text files to my box for tts? What I really need to figure out is how to create the INBOX .txt file. I can transfer the file and create the wav from flite (for now), but I don't understand the txt the Asterisk creates |
| 00:21.00 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 00:26.49 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 00:31.14 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:47.47 | *** join/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 00:47.48 | dijungal | does anyone know of a variable that stores the agent extension dialed |
| 00:50.28 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 00:50.59 | linagee | does anyone know about grandstream firmwares? is it best to stay with 1.1.1.14 instead of 1.1.4.18 on a GXP-2000? |
| 00:51.21 | JT | no idea |
| 00:52.59 | linagee | JT: the firmware page on voipinfo is very long and disturbing, but i've never had one problem on 1.1.1.14 in 2 years of use |
| 00:53.43 | [TK]D-Fender | linagee: GRANDSTREAM is very disturbing, not just their firmware :D |
| 00:54.25 | JT | ~gs |
| 00:54.26 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 00:55.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 00:55.30 | asterisknerds | <PROTECTED> |
| 00:55.33 | coppice | I've come across several issues where people are complaining about Grandstream, when it turns out to be the end that's doign the right thing. |
| 00:57.15 | [TK]D-Fender | coppice: True, and there has historically speaking been the odd flying pig or too :) |
| 00:57.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:57.49 | linagee | [TK]D-Fender: is there an official grandstream religious wars flame outfit? |
| 00:58.05 | linagee | flame/flame proof |
| 00:58.18 | [TK]D-Fender | linagee: there is a wide selection on the "ready to wear" rack over there ;) |
| 00:59.26 | linagee | i think one of the most interesting things about setting up a call center with PCs instead of "hard phones" is that the volume levels can always be fscked up in the most annoying of ways. (real life experience w/ that. :( ) |
| 00:59.33 | coppice | Their cases seem like something for a giveaway item in cornflakes, but their firmware is excessively maligned. its too easy to blame them for everything. a number of things won't talk T.38 to a grandstream. this is mostly granstream doing the right thing |
| 01:00.13 | linagee | i don't know what T.38 even is |
| 01:02.28 | coppice | ah, if only everyone were like that :-) |
| 01:03.23 | linagee | coppice: fax over IP? that sounds so dumb |
| 01:04.58 | s0ck | linagee: any particular reason for upgrading? |
| 01:05.13 | coppice | voice over IP sounds dumb, too, but it doesn't stop people trying it |
| 01:05.48 | linagee | coppice: let's do FC over IP! (j/k. you'd want the vise versa. ;) ) |
| 01:07.35 | JT | fax over ip sounds smarter than fax over VoIP |
| 01:08.59 | linagee | fax over voip?? |
| 01:09.24 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.45.45) |
| 01:09.30 | JT | linagee: yes |
| 01:09.46 | linagee | JT: faxes in general suck. who likes seeing something at 100dpi? :( |
| 01:10.10 | linagee | JT: faxes are for PHBs who are too dumb to use a scanner |
| 01:11.45 | JT | yes and no |
| 01:12.05 | JT | faxes look a lot better when good fax machines are involved |
| 01:12.32 | *** join/#asterisk batterymann (n=batterym@netblock-208-127-19-114.dslextreme.com) |
| 01:13.07 | JT | they have more accountability than random emails |
| 01:13.10 | JT | and let's face it |
| 01:13.16 | JT | scanners are really annoying to use |
| 01:16.18 | *** join/#asterisk whhheeEEEeeee (n=nathan@adsl-68-74-132-52.dsl.klmzmi.ameritech.net) |
| 01:17.20 | linagee | JT: make something better than a fax and call it a completely new buzz word |
| 01:17.29 | linagee | and in color too |
| 01:17.38 | linagee | it requires a color laserjet at both sides. :) |
| 01:18.14 | whhheeEEEeeee | hey guys! Ok, so I just compiled 1.4.10.1 and signed up for Teliax. I seem to be having some default configuration error stuff going on and I'm not sure where to start on the asterisk side. I talked with teliax and everything is going through to my box, but its refusing connection? |
| 01:19.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:19.44 | JT | linagee: and scanner |
| 01:20.20 | linagee | JT: yes. 600dpi color scanner will be part of the standard |
| 01:20.34 | linagee | JT: you just plug it into an ethernet jack (also required to have high speed interweb) |
| 01:21.21 | linagee | JT: i bet you could actually sell such a device to PHBs |
| 01:22.12 | JT | it would need to be accountable |
| 01:22.29 | JT | and need to be easy to tell is the transmission was sent to the other side |
| 01:29.59 | *** part/#asterisk batterymann (n=batterym@netblock-208-127-19-114.dslextreme.com) |
| 01:30.26 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 01:34.49 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:42.36 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 01:44.25 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 01:44.48 | yidiyuehan | <yidiyuehan> hi, anyone knows how I can install both asterisk-core-sounds-en-gsm-current.tar.gz and asterisk-extra-sounds-en-gsm-current.tar.gz? Do I still need to install asterisk-sounds-1.2.1.tar.gz |
| 01:45.47 | CCFL_Man2 | yes |
| 01:46.27 | whhheeEEEeeee | could anyone let me know what some basic things I need to set in order to get teliax (or any voip provider) to work with asterisk? |
| 01:46.59 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 01:47.26 | yidiyuehan | hi CCFL_Man2 |
| 01:47.50 | yidiyuehan | then after I install the asterisk-sounds-1.2.1.tar.gz, how i can install the core sounds and extra sounds ? |
| 01:48.01 | yidiyuehan | also perhaps the asterisk-moh-freeplay? |
| 01:48.08 | *** join/#asterisk saftsack (n=saftsack@p57A7578E.dip.t-dialin.net) |
| 01:49.00 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:51.07 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 01:53.04 | [TK]D-Fender | whhheeEEEeeee: You need to understand the dialplan, sip or IAX peers, SIP + NAT settings where applicable. |
| 01:54.12 | yidiyuehan | can any one direct me a link how to install the additional sound tarball like asterisk-core-sounds and asterisk-extra-sounds? |
| 01:54.16 | whhheeEEEeeee | [TK]D-Fender: I'm just trying to get incoming calls to the demo setup using IAX. |
| 01:54.49 | [TK]D-Fender | whhheeEEEeeee: take the demo config fiiles and TRASH THEM |
| 01:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:55.49 | fujin | I keep em around so that I can see the options they use, occasionally |
| 01:55.52 | [TK]D-Fender | whhheeEEEeeee: Go set up 2 phones (soft or otherwise) and get them talking to each other. Once you can do that then add in your ITSP |
| 01:56.03 | asterisknerds | <PROTECTED> |
| 01:56.05 | fujin | and it's generally more up-to-date than voip-info |
| 01:56.09 | [TK]D-Fender | fujin: They're ALWAYS around... |
| 01:56.45 | [TK]D-Fender | fujin: /usr/src/[asterisk-ver]/ <- folder for samples & docs |
| 01:56.54 | fujin | yea, I mean |
| 01:57.03 | fujin | 'trash them' |
| 01:57.10 | fujin | :P |
| 01:57.18 | whhheeEEEeeee | [TK]D-Fender: hmm. alright |
| 01:57.44 | whhheeEEEeeee | [TK]D-Fender: am I doing it backwards or something? |
| 01:58.22 | fujin | you're doing it wrong! |
| 01:58.24 | fujin | :) |
| 01:58.31 | fujin | Make a basic setup :0 |
| 01:58.41 | *** join/#asterisk TheNewAndy (n=TheNewAn@60-242-149-188.tpgi.com.au) |
| 01:58.42 | fujin | get two phones talking together, get the phones talking to the ITSP |
| 01:59.17 | [TK]D-Fender | :) |
| 01:59.20 | whhheeEEEeeee | lol |
| 01:59.33 | fujin | ^^ |
| 01:59.35 | fujin | SOCK PUPPPET |
| 01:59.45 | fujin | I'm a winner |
| 02:00.05 | whhheeEEEeeee | so, uh, why? seems like it'd be easy to get people to an automated prompt even if you didn't have extensions set up? |
| 02:00.18 | fujin | don't ask stupid question |
| 02:00.23 | JT | why? |
| 02:00.24 | fujin | you need to learn the concepts |
| 02:00.28 | JT | run before you fly |
| 02:00.31 | fujin | ^^. |
| 02:00.32 | [TK]D-Fender | whhheeEEEeeee: the samples show you all sorts of command and crap in a completely psycho-sized mess that will take you forever to dig through. A decent dialplan can fit on a page |
| 02:00.34 | JT | and walk before you run |
| 02:00.40 | fujin | less than a page, even |
| 02:00.44 | [TK]D-Fender | walk before you run! |
| 02:00.52 | [TK]D-Fender | Check both ways before crossing the street! |
| 02:00.55 | JT | i like my [TK]D-Fender sock puppet |
| 02:01.18 | [TK]D-Fender | Stay in milk! Don't do school! Drink your drugs! |
| 02:01.20 | [TK]D-Fender | ... |
| 02:01.21 | [TK]D-Fender | no wwait... |
| 02:01.35 | [TK]D-Fender | something seems.. I dunno.... OFF... |
| 02:01.42 | whhheeEEEeeee | lol |
| 02:01.50 | whhheeEEEeeee | I drink my school. |
| 02:02.33 | whhheeEEEeeee | ok, so I shouldn't have complied the demo stuff, how do I blow it away, just re-compile and install? |
| 02:02.37 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:02.43 | whhheeEEEeeee | (after deleting all config) |
| 02:02.46 | whhheeEEEeeee | or is there an easier way |
| 02:02.57 | fujin | jjust dont' make samples |
| 02:03.11 | [TK]D-Fender | whhheeEEEeeee: just grab sip.conf, extensions.conf, and iax.con in your favourite text editor and DELET the contents and start over |
| 02:03.35 | [TK]D-Fender | Actually you DO want to "make samples", just not for THOSE 3. |
| 02:03.58 | fujin | ooh dunno about that |
| 02:04.03 | fujin | I don't, but then again, I know what I need |
| 02:04.09 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 02:04.18 | [TK]D-Fender | You DO want defaults like voicemail.conf which is 2s to clear the sample entries, but the CORE is correct. Modules.conf <- important and 100% safe as default, etc. |
| 02:04.44 | whhheeEEEeeee | bleh |
| 02:04.51 | whhheeEEEeeee | so just blow away estensions, iax, and sip |
| 02:05.17 | [TK]D-Fender | whhheeEEEeeee: so just trash the 3 I told you to and start them over. You can always look at the samples for tiny bits of inpiration after. |
| 02:05.23 | [TK]D-Fender | whhheeEEEeeee: yup |
| 02:06.12 | whhheeEEEeeee | [TK]D-Fender: yeah, thats was the main reason I kept the samples. kinda as refference. |
| 02:06.36 | [TK]D-Fender | whhheeEEEeeee: Well they are still in your source folder if you want to look later. |
| 02:06.42 | whhheeEEEeeee | oh duh! |
| 02:06.44 | whhheeEEEeeee | bleh |
| 02:08.11 | fujin | mm. chanspy. fun. |
| 02:08.15 | fujin | I feel kind of dirty |
| 02:08.17 | fujin | like a pervert |
| 02:08.19 | fujin | ;x |
| 02:08.21 | whhheeEEEeeee | [TK]: so if I'm connecting via iax, I don't need to put anything in sip, do I. |
| 02:09.42 | whhheeEEEeeee | secondly, do I want to blow away extensions.ael as well. |
| 02:10.31 | fujin | learn AEL! |
| 02:10.31 | fujin | :D |
| 02:10.37 | fujin | it's incredibly awesome |
| 02:10.52 | whhheeEEEeeee | doesn't answer my question. :P |
| 02:11.45 | fujin | yes |
| 02:11.47 | fujin | probably |
| 02:11.59 | whhheeEEEeeee | lol |
| 02:12.38 | fujin | extensions.ael is just an ael version of extensions.conf |
| 02:12.43 | fujin | you can happily toast both of em |
| 02:13.26 | whhheeEEEeeee | awesome, ty |
| 02:16.16 | [TK]D-Fender | whhheeEEEeeee: no, you don't really need sip.conf if you're not going to be using it. empty it out. Forget AEL for now as its nowhere near documented enough to suggest learning from scratch on, not will you see the benifit at the earliest stages |
| 02:17.53 | whhheeEEEeeee | [TK]: dude, I really appreciate your help here man. I'd be so lost. |
| 02:18.51 | tengulre | hi,all |
| 02:19.27 | [TK]D-Fender | whhheeEEEeeee: well these are just ideas to START with. |
| 02:19.33 | [TK]D-Fender | ~book |
| 02:19.34 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:20.18 | whhheeEEEeeee | [TK]: ok, threw the info they gave me into the iax.conf file, I assume I'm going to need things for both the extensions and aix file. oh yeah, that thing. |
| 02:20.25 | whhheeEEEeeee | gos thats going to be a pain in the ass |
| 02:20.28 | [TK]D-Fender | whhheeEEEeeee: work your way through the book and learn dialplan basics. I'd suggest 2 soft-phones. As to avoid SIP, get KIAX or Zoiper soft-phones for your early learning |
| 02:20.53 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:20.58 | [TK]D-Fender | *nix/windows respectively |
| 02:21.40 | whhheeEEEeeee | [TK]: I like that kiax can encrypt too. |
| 02:22.35 | whhheeEEEeeee | score! I can see the attempt coming through with debug |
| 02:22.45 | whhheeEEEeeee | yeah, I know I'm doing it backwards, but hey. I'm crazy. |
| 02:23.35 | [TK]D-Fender | whhheeEEEeeee: Actually, thats not "backwards", thats "bottom-up programming". |
| 02:24.07 | [TK]D-Fender | whhheeEEEeeee: You see the call and are building the system up around what it complains about so it won't. makes for a very minimal system. |
| 02:24.25 | whhheeEEEeeee | yeah, I'm a computer scientist trying to not go through the 101's and learn a whole new topic from scratch. |
| 02:24.32 | [TK]D-Fender | whhheeEEEeeee: When all you're worried about is the next 3 lines, you don't have 500 useless ones in your wya |
| 02:24.42 | whhheeEEEeeee | indeed |
| 02:24.48 | whhheeEEEeeee | I thought it was complex to set up though. |
| 02:25.23 | [TK]D-Fender | whhheeEEEeeee: Bus seriously, stop for a moment to understand dialplan patters, and CONTEXTS. This is SACRED. Go read how the dialplan is divided up. |
| 02:26.05 | whhheeEEEeeee | [TK]: very true, but its critical that I have a line up immediately. |
| 02:26.09 | *** join/#asterisk marcan (i=1337@host214-220.cvd.fit.edu) |
| 02:26.18 | [TK]D-Fender | whhheeEEEeeee: Because when you get down to it, the minimal setup for your ITSP will be REALLY petty (20 lines maybe?) your DIALPLAN will be far larger.... |
| 02:26.36 | *** join/#asterisk Mikhalych (n=mich@Lexus-sc300.vlad.ru) |
| 02:26.40 | [TK]D-Fender | whhheeEEEeeee: Whats the rush? |
| 02:27.05 | whhheeEEEeeee | [TK]: clients need to get in touch with me in an area where I don't have cellphone or any kind of line. |
| 02:27.08 | Mavvie | oh fsck... asterisk 1.4 has screwed up the quoting in master.csv |
| 02:27.09 | Mikhalych | Greetings to all guru :-) |
| 02:28.14 | Mikhalych | can I ask for help ? ;-) |
| 02:28.30 | fujin | no |
| 02:28.34 | fujin | We don't do help in here |
| 02:29.11 | whhheeEEEeeee | its anti-help |
| 02:29.14 | *** join/#asterisk minkus (n=minkus@doc-24-206-212-154.bh.wv.cebridge.net) |
| 02:29.16 | whhheeEEEeeee | or torture. pick one. |
| 02:29.21 | Mavvie | we don't need help neither. |
| 02:29.28 | Mikhalych | that is not trivial problem with sip outbound calls :-) |
| 02:29.51 | [TK]D-Fender | It's time to get things started on the Muppet Show tonight! |
| 02:30.14 | Mikhalych | Pigs in space ? ;-) |
| 02:30.39 | whhheeEEEeeee | [TK]: Heading out for some food, thanks so much for your starting help. afk for a bit. |
| 02:32.32 | Mikhalych | where I can get little help about problems with sip outgoing calls ? |
| 02:34.16 | kiscokid | what is the problem Mikhalych? |
| 02:34.50 | fujin | heh |
| 02:34.54 | fujin | what's everyone using for queue statistics? |
| 02:35.01 | fujin | or a realtime wallboard |
| 02:35.07 | JT | ~question |
| 02:35.08 | jbot | i heard question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 02:37.21 | fujin | anyone? no one? |
| 02:38.04 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 02:39.05 | fujin | like, not crappy queuemetrics |
| 02:39.08 | fujin | and not orderlyq |
| 02:39.23 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 02:44.44 | *** join/#asterisk Tako-san (n=Tako-san@154.5.212.245) |
| 02:48.11 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 02:48.27 | CCFL_Man2 | in the world of telephony, what is ear & mouth? |
| 02:50.23 | [TK]D-Fender | CCFL_Man2: the analog interfaces you couple to a decent phone :) |
| 02:52.20 | [TK]D-Fender | fujin: I get wall-board detail ON my CSR's phones :) |
| 02:52.44 | fujin | oh really? |
| 02:52.51 | fujin | what phones? |
| 02:53.00 | fujin | what product is this? :P |
| 02:53.39 | russellb | phones are so lammmmme |
| 02:54.22 | fujin | come on, there must be *someone* with a wallboard written that isn't queuemetrics |
| 02:54.23 | Qwell | russellb: homeownership sucks |
| 02:54.53 | [TK]D-Fender | fujin: I get live queue stats for 2 queues & 4 agents on my CSR's Polycom IP 600's |
| 02:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:55.42 | asterisknerds | <PROTECTED> |
| 02:56.30 | russellb | there is a little app in svn ... |
| 02:56.32 | russellb | qview |
| 02:56.43 | russellb | it's used around digium |
| 02:57.14 | fujin | qview |
| 02:57.17 | fujin | got a url handy mate? |
| 02:57.50 | *** join/#asterisk jmacz (n=jmacz@190.25.32.48) |
| 02:58.35 | russellb | svn co http://svn.digium.com/svn/qview/trunk qview |
| 02:58.41 | russellb | then there is a sample config file in there |
| 02:58.49 | fujin | gracias senor |
| 02:58.50 | fujin | appreciate it |
| 02:58.52 | russellb | not terribly documented, but easy to set up |
| 02:59.05 | russellb | i haven't touched it in ... years probably |
| 02:59.17 | fujin | did you write it? |
| 02:59.20 | russellb | nope |
| 02:59.44 | russellb | just used it before playing around |
| 02:59.54 | fujin | ah ok |
| 03:00.00 | fujin | ah, I see, it uses a manager connection |
| 03:00.05 | russellb | yeah |
| 03:00.15 | fujin | Are you able to log manager stuff out to a database? |
| 03:00.19 | russellb | all would have to do that to get live stuff |
| 03:00.20 | fujin | I may look into designing an in-house solution. |
| 03:00.35 | russellb | i don't quite understand the question ... |
| 03:00.36 | fujin | something which is like asterisk manager -> database <- php parser |
| 03:00.40 | russellb | manager is just a network connection |
| 03:00.44 | fujin | oh right |
| 03:00.45 | fujin | of course |
| 03:00.47 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:00.48 | fujin | hrm |
| 03:00.53 | fujin | perhaps what I'm thinking of wouldn't work then |
| 03:01.06 | fujin | I'd have to write an API around the manager interface |
| 03:01.13 | russellb | i've never heard of someone putting events through a database, doesn't make much sense to me :) |
| 03:01.21 | russellb | plenty of people have already written manager APIs |
| 03:01.27 | russellb | there is one for most common languages |
| 03:01.32 | Qwell | I dunno.. persistent caching, heh |
| 03:01.50 | russellb | you mentioned PHP, there is phpagi |
| 03:01.54 | russellb | err, not agi ... |
| 03:02.27 | Juggie | russellb, i assume what he means is that if you just listened to the manager interface you would have to listen for a while to find out the states of all the devices/agents/etc. |
| 03:02.50 | russellb | well, you can query that to get initial state |
| 03:02.51 | Juggie | so i presume he means some way to cache all that so you can just ask for it and get the event we most recentally received |
| 03:03.02 | rudholm | any teliax users around? |
| 03:03.15 | russellb | i have a framework in trunk that does exactly that, though :) |
| 03:03.20 | fujin | ;o |
| 03:03.23 | russellb | manager hasn't been converted to use it though ... |
| 03:03.31 | russellb | it's used for other internal event handling .. |
| 03:03.59 | fujin | That's probably the only way I can do it though, right, is to connect up to manager and send/receive commands through a socket |
| 03:07.14 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:08.30 | fujin | ho-hum. |
| 03:08.34 | fujin | qview.c doesn't build |
| 03:08.50 | russellb | it does when you have the right libraries installed |
| 03:09.05 | russellb | libdsl-dev, libsdl-ttf-dev, libsdl-net-dev |
| 03:09.30 | russellb | i should just add a configure script here that will check for these for you ... |
| 03:09.31 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:09.39 | Juggie | russellb, thats true, converting the manager to that would be awesome, but converting the manager would mean converting almost every part of * to your event system |
| 03:09.46 | Juggie | as the manager gets events from all o ver |
| 03:10.00 | russellb | yup ... |
| 03:10.04 | russellb | it's not a small task, heh |
| 03:11.19 | *** join/#asterisk minkus (n=minkus@pool-72-84-43-24.clrkwv.east.verizon.net) |
| 03:11.43 | Shadowfire_ | Has anyone here worked with SPA942 phones with Asterisk? |
| 03:12.06 | fujin | yes, I have 50 of them |
| 03:12.25 | Shadowfire_ | would you mind if I asked you some questions? |
| 03:12.39 | fujin | sure |
| 03:13.11 | Mikhalych | Mine question is stayed unanswered ;-( |
| 03:13.23 | Shadowfire_ | I am fairly new to Asterisk.. I had some privious experience with Cisco, but am loving what Asterisk can do... |
| 03:13.26 | Shadowfire_ | it is amazing |
| 03:14.04 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 03:14.22 | Mikhalych | Anyone can help with sip otbound calls via 2 sip accounts ? |
| 03:14.23 | Shadowfire_ | On the SPA942... what method to I go to get an SPA942 to call another SPA942 |
| 03:14.23 | russellb | Shadowfire_: awesome :) |
| 03:14.51 | Shadowfire_ | russellb: :) thanks sir.... |
| 03:17.32 | Mavvie | only seems to be cdr-custom which is screwed up... |
| 03:18.04 | Mavvie | the normal cdr-csv is still okay. |
| 03:20.28 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 03:21.03 | [TK]D-Fender | Shadowfire_: Time to learn how to write your dialplan. Thats chapter 5 in THE BOOK. Get reading |
| 03:21.05 | [TK]D-Fender | ~book |
| 03:21.06 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:21.15 | Mavvie | Mikhalych: not via private messages no. |
| 03:21.49 | Mikhalych | sorry |
| 03:21.51 | Mikhalych | gtg |
| 03:21.53 | Mikhalych | bb all |
| 03:23.39 | Shadowfire_ | [TK]D-Fender: I have a dailplan... not that I haven't read enough books, but would you have a specific book you would personally recommend instead of saying read a book. I have been over a number of books... so a specific would be great ... thnx |
| 03:24.24 | [TK]D-Fender | jbot>somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:24.25 | jbot | [TK]D-Fender: I think you lost me on that one |
| 03:24.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 03:24.43 | [TK]D-Fender | Shadowfire_: Watch the big print :) |
| 03:24.51 | [TK]D-Fender | Shadowfire_: And i even gave you the CHAPTER! |
| 03:25.13 | [TK]D-Fender | Shadowfire_: not "a" book, "THE" book. |
| 03:25.16 | [TK]D-Fender | ~thebook |
| 03:25.17 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:25.59 | techman97_andy | evening all - anyone know of a good software package that I can use to monitor agent calls (monitoring and barging)? I used to use HUDLite, but they removed this functionality in the latest release. |
| 03:28.14 | sysreq | anyone knows what's going on with AFOT's 2nd edition? amazon said it'd be available by august 1st so i pre-ordered it, but now they've removed it from their listing and i received several e-mails warning me that it would be delayed.. |
| 03:29.01 | Shadowfire_ | [TK]D-Fender: I have read parts of that one, I will look closer at Chapter 5 ... thanks for pointing me in that direction. |
| 03:29.10 | sysreq | i know there's a 'rough cuts' copy on o'reilly, but then.. |
| 03:29.29 | [TK]D-Fender | Shadowfire_: this is the basics of a simple exten that will dial your other phone |
| 03:46.39 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 03:51.00 | linagee | ack. i upgraded my grandstream and now the quality sucks. :( :( |
| 03:51.06 | linagee | d'oh |
| 03:51.09 | linagee | like really really sucks |
| 03:54.21 | *** join/#asterisk bmg505 (n=leon@196.209.176.131) |
| 03:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:56.03 | asterisknerds | <PROTECTED> |
| 04:01.28 | x86 | linagee: you expected grandstream to be good? or even decent?? |
| 04:01.34 | x86 | that's what really sucks ;) |
| 04:01.43 | linagee | x86: ?!?!?!?!?!! |
| 04:01.46 | x86 | polycom man... that's all you need to know :) |
| 04:01.48 | linagee | x86: IT WAS! :( |
| 04:01.52 | linagee | x86: it was awesome |
| 04:01.57 | x86 | no, it never was awesome |
| 04:01.58 | linagee | x86: then i upgraded the firmware. :( |
| 04:02.06 | x86 | grandstream != awesome.... ever |
| 04:02.15 | linagee | x86: yes. in the old firmware it works just fine |
| 04:02.24 | x86 | if you ever thought grandstream == awesome, then you were sadly mistaken |
| 04:02.38 | x86 | dude, seriously, don't use grandstream |
| 04:02.38 | linagee | x86: by awesome, i mean functional. |
| 04:02.47 | linagee | x86: now that i upgraded the firmware, now it's just crap. :( |
| 04:02.50 | x86 | buy real hardware |
| 04:03.18 | x86 | polycom for IP phones, linksys PAP2 for ATAs |
| 04:03.29 | x86 | adtran for channel banks |
| 04:03.35 | x86 | you can't go wrong :) |
| 04:05.57 | fujin | ~grandstream |
| 04:05.58 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 04:06.07 | fujin | ~phones |
| 04:06.08 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 04:07.09 | raidenz | Does anyone have app_pppd working with the latest asterisk 1.4? |
| 04:10.34 | fujin | what's app_pppd do? |
| 04:13.14 | russellb | fujin: you get qview going? |
| 04:13.35 | fujin | nah, cbf with it |
| 04:13.39 | fujin | going to write a daemon |
| 04:13.52 | fujin | or something which parses queue_log |
| 04:13.54 | fujin | for uber laziness |
| 04:17.13 | *** join/#asterisk jmacz (n=jmacz@190.25.32.48) |
| 04:17.38 | CCFL_Man2 | [TK]D-Fender: i never heard of ear and mouth interfaces |
| 04:18.01 | CCFL_Man2 | they a speaker and microphone, basically? |
| 04:19.43 | [TK]D-Fender | CCFL_Man2: Get googling if you want the specifics. Its a crappy signalling standard you are best not to have to encounter |
| 04:22.07 | dlynes | [TK]D-Fender: having a good chuckle, i see :) |
| 04:22.22 | CCFL_Man2 | [TK]D-Fender: ahh, i see |
| 04:22.47 | CCFL_Man2 | [TK]D-Fender: i kind of knew it was crappy because very few cisco products support it |
| 04:22.58 | JT | it's inband signalling |
| 04:23.11 | JT | one type of inband signalling, anyway |
| 04:25.41 | CCFL_Man2 | ahh |
| 04:25.50 | CCFL_Man2 | actually, it does suck |
| 04:26.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:27.48 | CCFL_Man2 | so anyone here use an adit 600? |
| 04:28.48 | x86 | heh, you're still trying to fuck with that thing? |
| 04:28.54 | Shadowfire_ | I have used an adit 600 |
| 04:29.05 | Shadowfire_ | voice router.. |
| 04:29.08 | Shadowfire_ | nice :) |
| 04:29.14 | Shadowfire_ | but a pain to setup |
| 04:29.51 | CCFL_Man2 | x86: they are one of the most reliable channel banks ever |
| 04:30.01 | CCFL_Man2 | Shadowfire_: you use it with asterisk? |
| 04:30.10 | Shadowfire_ | nah... LEC |
| 04:30.51 | Shadowfire_ | I would definately love to help you, but unfortunately I was working with a LEC at the time... T1/E1 connection... |
| 04:32.08 | Shadowfire_ | bet asterisk would be a beautiful setup with the adit 600 |
| 04:32.18 | Tako-san | CCFL_Man2: I am using an Adit 600 with Asterisk |
| 04:32.28 | Tako-san | but unfortunately slightly busy at the moment |
| 04:32.44 | Tako-san | will you be around again tomorrow? if so catch me online then |
| 04:32.46 | CCFL_Man2 | Shadowfire_: ahh |
| 04:32.59 | CCFL_Man2 | Tako-san: i just have one quicj question |
| 04:33.03 | Tako-san | or if you are around a little later tonight perhaps then |
| 04:33.16 | Tako-san | CCFL_Man2: ok, go for it..no promises though |
| 04:38.06 | CCFL_Man2 | Shadowfire_: i hear that adtran and carrier access channel banks are the best in the business |
| 04:38.22 | [TK]D-Fender | dlynes: Channel banks.... meh |
| 04:39.04 | CCFL_Man2 | channel banks are great with asterisk, if you can figure out a way to interface them |
| 04:39.28 | JT | what do you mean a way to interface them? |
| 04:39.30 | JT | it's easy |
| 04:39.42 | Tako-san | t1 card->channel bank |
| 04:39.48 | CCFL_Man2 | JT: i know it's easy, but it's expensive |
| 04:40.15 | JT | that's not figuring out a way to interface with them, that's called "getting funds" |
| 04:40.28 | CCFL_Man2 | you know what i mean :P |
| 04:40.45 | JT | you made it sound like it was hard to connect them |
| 04:40.57 | CCFL_Man2 | well, it is if you have no funds :P |
| 04:41.12 | JT | not a technical issue |
| 04:41.18 | CCFL_Man2 | no, not at all |
| 04:41.59 | CCFL_Man2 | i don't think adtran and carrier access offer any easy sip solutions |
| 04:42.09 | JT | easy |
| 04:42.10 | CCFL_Man2 | and i'm not sure why |
| 04:42.15 | JT | going your definition of easy |
| 04:42.19 | JT | that means cheap |
| 04:42.32 | JT | and sip gateways are never cheap |
| 04:42.43 | JT | because they make telco gear |
| 04:42.51 | JT | and telcos don't use this sort of stuff? |
| 04:42.52 | CCFL_Man2 | it's not really a gateway if it's fxo though |
| 04:43.02 | CCFL_Man2 | err |
| 04:43.04 | CCFL_Man2 | fxs |
| 04:43.06 | JT | it is a gateway |
| 04:43.16 | [TK]D-Fender | Unless you're talking faxing or other modem-like things, then channel banks are overpriced as a total solution, less flexible and compounds points of failure |
| 04:43.29 | x86 | CCFL_Man2: a "gateway" is a device that translates between one media and another... by definition |
| 04:43.31 | JT | any sip device that allows the interconnect of a non-sip technology is usually a gateway |
| 04:43.46 | CCFL_Man2 | JT: oh, yeah, you're right |
| 04:43.52 | x86 | CCFL_Man2: so something translating between just SIP and IAX2 would be a gateway |
| 04:44.01 | CCFL_Man2 | right |
| 04:44.19 | CCFL_Man2 | even an ATA is a gateway, the linkshit ones are pretty cheap |
| 04:44.23 | CCFL_Man2 | but they suck |
| 04:44.37 | CCFL_Man2 | well, they are not bad |
| 04:44.54 | x86 | [TK]D-Fender: unfortunately, my company is too cheap to put an IP phone on each desk, so we use these tiny analog phones for sales people (since we're a call center), which need channel banks to interface with asterisk |
| 04:44.54 | CCFL_Man2 | the 1st generation ones don't work well with old phones |
| 04:45.11 | [TK]D-Fender | x86: WRONG! |
| 04:45.20 | CCFL_Man2 | x86: sweet, each desk has a regular pots phone? |
| 04:45.21 | [TK]D-Fender | x86: SIP gateways <------ |
| 04:45.25 | x86 | [TK]D-Fender: either that or use a TDM2400 |
| 04:45.31 | [TK]D-Fender | x86: WORSE!!!! |
| 04:45.46 | x86 | [TK]D-Fender: how else do you get 24 analog stations into asterisk? |
| 04:46.00 | [TK]D-Fender | x86: SIP gateways <------ |
| 04:46.04 | CCFL_Man2 | lol |
| 04:46.27 | [TK]D-Fender | x86: Audiocodes MP-112 , Mediatrix 1124, etc |
| 04:46.28 | x86 | [TK]D-Fender: ok, so a channel bank with an IP/SIP interface instead of T1/TDM ;-) |
| 04:46.43 | x86 | same thing just more complicated |
| 04:46.46 | CCFL_Man2 | x86: old cisco 7912 phones kick ass and are cheap, but their sip firmware sucks balls for them |
| 04:47.00 | x86 | CCFL_Man2: i would never buy cisco phones... ever |
| 04:47.03 | JT | they kick arse and suck ballas at the same time |
| 04:47.15 | x86 | cisco phones sound tinny as hell |
| 04:47.17 | [TK]D-Fender | x86: Exactly. You save the multi-t1 overhead, PCI complications, add built-in IP *AND* analog failover, reduce server load, etc |
| 04:47.43 | x86 | [TK]D-Fender: hmm... true... |
| 04:47.47 | CCFL_Man2 | most, if not all, channel banks don't have any SIP interfaces |
| 04:47.51 | [TK]D-Fender | x86: and net out CHEAPER. |
| 04:47.55 | x86 | [TK]D-Fender: can i tell these devices that if server A is dead, use server B? |
| 04:48.01 | [TK]D-Fender | x86 : Yes <- |
| 04:48.11 | x86 | interesting... |
| 04:48.13 | CCFL_Man2 | x86: i use one and they don't sound tinny, we use em at work with ccm and they sound great |
| 04:48.17 | [TK]D-Fender | x86: And barring that even fail out an analog port on the unit itself |
| 04:48.48 | CCFL_Man2 | JT: cisco phones can be great if they had good software |
| 04:49.15 | JT | sorry |
| 04:49.21 | JT | what's great about them again? |
| 04:49.22 | CCFL_Man2 | [TK]D-Fender: failure of an analog port is unlikely in a call center |
| 04:49.30 | CCFL_Man2 | JT: lol |
| 04:49.37 | JT | that's like saying a grandstream could be good if it doesn't suck balls |
| 04:49.57 | [TK]D-Fender | CCFL_Man2: thats why its considered a theoretical bonus. Nobody said its entire functioning was with YOU in mind :) |
| 04:50.28 | CCFL_Man2 | JT: i like the 7912 interface, the display, and the sound quality, i never had any bad experiences with them |
| 04:50.32 | CCFL_Man2 | [TK]D-Fender: hehe |
| 04:51.12 | CCFL_Man2 | lol |
| 04:51.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:51.13 | x86 | the only thing cisco phones have going for them is that they have nice displays... |
| 04:51.16 | [TK]D-Fender | Cisco = completely pointless compared to Polycom with is CHEAPER |
| 04:51.31 | fujin | I found a polycom reseller in NZ |
| 04:51.31 | CCFL_Man2 | x86: their displays i like |
| 04:51.37 | fujin | shame I had already bought 50x spa942's |
| 04:51.45 | CCFL_Man2 | i hear polycom phones are the king |
| 04:51.52 | x86 | [TK]D-Fender: well the "my penis is THIS BIG" types, the 7971G is always a big eye catcher |
| 04:52.15 | x86 | s/the/to the/; |
| 04:52.21 | CCFL_Man2 | the 7912G is the basic of basics, i like it's interface |
| 04:52.22 | x86 | CCFL_Man2: indeed |
| 04:52.25 | [TK]D-Fender | x86: A decision clearly in proportion to the idiocy advcating it :) |
| 04:52.46 | flenders | I had a couple of ciscos and they are shit |
| 04:52.59 | x86 | fujin: spa942's aren't terrible.... polycom is definitely better, but at least the spa942's are better than any grandstream product |
| 04:53.00 | CCFL_Man2 | flenders: sip? |
| 04:53.03 | flenders | yeah |
| 04:53.13 | fujin | haven't had any issue with centrally/automatically provisioning my spa942s |
| 04:53.16 | flenders | their software is shit |
| 04:53.17 | fujin | they've been stable as hell |
| 04:53.26 | fujin | only way I've managed to break the 942 is delivering like 500 calls to it at 20cps |
| 04:53.30 | fujin | which causes it to reboot |
| 04:53.31 | flenders | I still think they look great, but usability on them is stupid |
| 04:53.33 | x86 | fujin: a lot less configurable than polycom's though |
| 04:53.41 | CCFL_Man2 | flenders: the sccp firmware is great though |
| 04:53.52 | [TK]D-Fender | x86: "Better that GS" shouldn't be the sole measure of good.. thats like lowering the bar incrementally :) |
| 04:53.56 | x86 | CCFL_Man2: go back to #cisco, nub ;) |
| 04:54.02 | flenders | even linksys 94x are far better than ciscos |
| 04:54.06 | x86 | [TK]D-Fender: haha |
| 04:54.15 | x86 | flenders: *nod* |
| 04:54.33 | [TK]D-Fender | flenders: Debatable...... |
| 04:54.35 | x86 | cisco should stay in the networking gear market... routers and switches... end of story ;) |
| 04:54.36 | CCFL_Man2 | x86: hehe, granted, cisco sucks, they purposely make their sip firmware shit |
| 04:54.43 | [TK]D-Fender | flenders: but not a stretch |
| 04:54.51 | CCFL_Man2 | what about telephony routers? |
| 04:54.58 | x86 | no way |
| 04:55.01 | flenders | ciscos sound quality is better than linksys |
| 04:55.08 | flenders | but usability is terrible |
| 04:55.15 | CCFL_Man2 | flenders: i agree |
| 04:55.17 | x86 | i've got an 1841 loaded with CCME... what a piece of shit |
| 04:55.20 | fujin | configuration on the linksys is nice |
| 04:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 04:55.34 | fujin | can do everything i've required, apart from configuring the 'line keys' to do something apart from be a line |
| 04:55.41 | CCFL_Man2 | x86: i'll gladly take your 1841 |
| 04:55.48 | fujin | like I wanted to change it to a toggle button that dialed 600 or something |
| 04:55.51 | x86 | CCFL_Man2: i'm using it for data ;) |
| 04:56.03 | asterisknerds | <PROTECTED> |
| 04:56.16 | flenders | fujin: transferring calls is one feature that's not very good on the linksys |
| 04:56.23 | fujin | how so? |
| 04:56.29 | CCFL_Man2 | x86: i use a 1721 for my routing, with adsl wic, i love it |
| 04:56.30 | fujin | bxfer, dial number, dial |
| 04:56.50 | [TK]D-Fender | fujin: other caveeats : having to PAY to use all the regs, only being able to handle a single call per key. |
| 04:56.54 | flenders | yeah, but you have to press the arrow key (right) and then bxfer |
| 04:57.04 | CCFL_Man2 | x86: how much did the 1841 cost you? |
| 04:57.10 | fujin | [TK]D-Fender: nah, you don't have to pay to open all 4 chans now |
| 04:57.11 | fujin | they took that out |
| 04:57.12 | [TK]D-Fender | On SPA's when a call is disconnected you get REORDER TONE! Bleh!! |
| 04:57.22 | fujin | 5.1.10 enabled spa942 4 lines |
| 04:57.26 | [TK]D-Fender | Its like a dumb analog phone bolted onto an ATA! |
| 04:57.28 | flenders | firmware 5.1.8 has 4 lines |
| 04:57.30 | fujin | but yeah, the single call per key thing is dumb |
| 04:57.33 | flenders | from 5.1.8 I mean |
| 04:57.37 | fujin | flenders: there ya go :P |
| 04:57.44 | fujin | but yeah, on a spa942 |
| 04:57.47 | fujin | line=call |
| 04:57.48 | x86 | CCFL_Man2: which one? :) |
| 04:57.49 | fujin | which is dumb ;P |
| 04:58.03 | [TK]D-Fender | I can handle 3 regs w/ 8 calls each on my IP 501! |
| 04:58.05 | CCFL_Man2 | x86: you have more than one 1841!? |
| 04:58.08 | flenders | you can have 2 calls per 'line' |
| 04:58.08 | x86 | CCFL_Man2: i've been buying the CISCO-1841T1 bundles lately... come with a WIC-1DSU-T1 |
| 04:58.19 | flenders | but it's pretty hard to switch between them |
| 04:58.33 | CCFL_Man2 | x86: if you have any extra i'll gladly take it out of your hands :P |
| 04:58.41 | x86 | CCFL_Man2: i've got (4) 1841's, each with (1) WIC-1DSU-T1 and (1) VWIC2-2MFT-T1/E1 |
| 04:58.54 | fujin | meh, more than 4 concurrent calls = hard to manage |
| 04:58.59 | x86 | total of (3) T1 interfaces per 1841 router |
| 04:59.07 | [TK]D-Fender | SPA's screen is too small and the pixelization is crappy... pporly utiized. Aastra is possibly worse. |
| 04:59.08 | CCFL_Man2 | ahh VWIC2-2MFT-T1/E1 is for voice trunks? |
| 04:59.25 | [TK]D-Fender | fujin: Easy on my 501 :) |
| 04:59.28 | x86 | CCFL_Man2: it can do TDM, IP, and DSX |
| 04:59.48 | x86 | CCFL_Man2: only way I could get 2 T1 interfaces on a WIC with integral CSU |
| 05:00.24 | CCFL_Man2 | x86: i thought DSX was an electrical standard for the short haul between the csu/dsu and the smart jack? |
| 05:00.31 | x86 | CCFL_Man2: I can get the WIC-1DSU-T1's all day for like $350... the VWIC2-2MFT-T1/E1's are about $1700/ea |
| 05:00.48 | x86 | CCFL_Man2: no, DSX in this case means drop and insert |
| 05:01.11 | x86 | CCFL_Man2: demuxing DS0's from your DS1 into your PBX |
| 05:01.14 | CCFL_Man2 | ahh |
| 05:01.28 | CCFL_Man2 | or your channel bank? |
| 05:01.34 | x86 | sure |
| 05:01.41 | x86 | I only use the shit for data... |
| 05:01.55 | x86 | i'm NOT putting a wannabe PBX on my data routers... no thanks ;) |
| 05:02.00 | CCFL_Man2 | what did cisco voice ever do to you? :P |
| 05:02.11 | fujin | my as5400 has been great |
| 05:02.20 | fujin | that's about as far as my cisco voip experience goes |
| 05:02.25 | x86 | and I don't need DSX anyway, as all of my voice rides across full TDM T1's |
| 05:02.37 | x86 | i've got about 12 voice T1's nwo |
| 05:02.39 | x86 | now* |
| 05:03.23 | CCFL_Man2 | x86: isn't a DSX still TDM? |
| 05:04.32 | x86 | CCFL_Man2: you dont use TDM on DS0's |
| 05:05.16 | x86 | CCFL_Man2: TDM is a time division method that lets you time-share a single medium (say, a DS1) with many circuits (24 DS0's per DS1, for example) |
| 05:05.22 | CCFL_Man2 | on DS0s you use CAS? |
| 05:05.30 | CCFL_Man2 | right |
| 05:05.42 | x86 | CAS is signalling... not related to TDM :) |
| 05:05.46 | CCFL_Man2 | oh |
| 05:05.57 | x86 | just like PRI is signalling |
| 05:06.04 | CCFL_Man2 | ahh, right |
| 05:06.26 | CCFL_Man2 | what would DSX be then? |
| 05:06.34 | x86 | without TDM, you would only be able to use 1 big channel on the DS1 |
| 05:06.41 | x86 | DSX is drop-and-insert... |
| 05:06.43 | CCFL_Man2 | right |
| 05:07.13 | JT | time division multiplexing |
| 05:07.16 | x86 | think fractional T1, where you have 24 total channels... you use 12 channels for data (768kbps) and 12 channels for voice (12 DS0's) |
| 05:07.20 | fujin | blasted polycom reseller is only offering me 25% off the 'audio' products. |
| 05:07.22 | fujin | gaysex. |
| 05:07.22 | x86 | JT: yup |
| 05:07.24 | JT | basically you just have timeslots |
| 05:07.32 | x86 | exactly |
| 05:07.47 | CCFL_Man2 | x86: oh, ok |
| 05:08.06 | x86 | CCFL_Man2: DSX is where you split off the 12 DS0's on the other end... you drop them off the DS1, and insert them elsewhere :) |
| 05:08.10 | [TK]D-Fender | fujin: 25% = ? |
| 05:08.23 | fujin | [TK]D-Fender: 'audio' products doesn't apply to IP phones |
| 05:08.32 | fujin | that's 25% discount (wholesale) |
| 05:08.55 | x86 | how does 'audio products' not include IP phones? |
| 05:09.13 | CCFL_Man2 | x86: i never heard of DSX being used much |
| 05:09.26 | x86 | to me, 'audio products' would include just about every polycom product except the video stuff |
| 05:09.42 | x86 | CCFL_Man2: it's used all the time... |
| 05:10.02 | fujin | maybe, i'll clarify with him |
| 05:10.10 | fujin | perhaps he means only accessories, or the non-IP phone stuff |
| 05:10.22 | fujin | like the standard conference units |
| 05:10.26 | x86 | CCFL_Man2: a small company that doesn't need 24 phone lines at all times, nor a full T1 of data bandwidth... they can run a single T1 in and get a happy medium ;) |
| 05:10.47 | JT | modern polycoms have no audio, they have a direct telepathic link into the user's brain |
| 05:11.10 | x86 | CCFL_Man2: and it doesn't have to be proportional by any means... you can use 22 channels for data and 2 channels for voice, if you wanted |
| 05:11.22 | x86 | CCFL_Man2: so it's really flexible... saves you money on the local loop |
| 05:11.25 | CCFL_Man2 | x86: kind of like with DIDs, i suppose? |
| 05:11.42 | JT | not really |
| 05:11.43 | x86 | CCFL_Man2: now you're talking about phone numbers :) |
| 05:11.56 | x86 | which is totally different from what our conversation is about |
| 05:12.14 | JT | CCFL_Man2: http://en.wikipedia.org/wiki/Time-division_multiplexing |
| 05:12.22 | CCFL_Man2 | please be patient with me, i'm sort of new to telephony :P |
| 05:12.33 | fujin | now, to talk the accounts manager into upgrading 'specific' peoples phones :) |
| 05:12.34 | CCFL_Man2 | JT: oh i understand TDM |
| 05:15.04 | x86 | CCFL_Man2: no wonder you still like cisco... you don't know a whole lot about telephony ;) |
| 05:15.16 | x86 | CCFL_Man2: cisco put the 'phony' in telephony ;) |
| 05:15.17 | JT | ;) |
| 05:15.21 | JT | haha |
| 05:15.23 | JT | top quote |
| 05:16.00 | CCFL_Man2 | lol, pwntage |
| 05:17.13 | x86 | CCFL_Man2: did you just get done reading the new book "From Counterstrike to TDM in 24 hours" ? |
| 05:17.38 | CCFL_Man2 | lol |
| 05:18.15 | [TK]D-Fender | CCFL_Man2: Damn man... better go goole up an AimBot! |
| 05:18.46 | CCFL_Man2 | i wanted a nix based megahal aim bot, but couldn't find one |
| 05:20.38 | CCFL_Man2 | never cared for counterstike |
| 05:20.49 | CCFL_Man2 | that where the famous typo came from? |
| 05:21.00 | [TK]D-Fender | Action:Half-Life foreva! |
| 05:22.49 | CCFL_Man2 | i only liked the single player versions |
| 05:26.17 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:31.53 | jarod14 | hi there |
| 05:32.13 | *** join/#asterisk lbow (n=lbow@41-195-77-196.access.uunet.co.za) |
| 05:36.18 | CCFL_Man2 | the old click and bang switching systems required constant maintanence |
| 05:41.03 | JT | click and bang |
| 05:41.16 | JT | you mean crossbar, or step by step, or relay |
| 05:46.20 | Strom_M | don't forget XY and rotary |
| 05:46.36 | rudholm | hehe, I was just gonna mention XY |
| 05:46.56 | rudholm | and panel |
| 05:47.09 | rudholm | although panels aren't that bangy |
| 05:50.42 | Strom_M | panels are more clickity |
| 05:51.00 | rudholm | less clackety |
| 05:51.27 | rudholm | more zipity |
| 05:53.08 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 05:53.17 | CCFL_Man2 | XY |
| 05:53.23 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 05:53.35 | rudholm | XY? |
| 05:53.37 | rudholm | XY! |
| 05:53.38 | CCFL_Man2 | step by step actually |
| 05:53.43 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 05:53.55 | rudholm | step by step? what about inch by inch? |
| 05:54.13 | rudholm | my CO had a WECo inch by inch in the 70s |
| 05:55.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:55.30 | asterisknerds | <PROTECTED> |
| 05:56.32 | CCFL_Man2 | ahh, i'm not sure what my CO had |
| 05:57.21 | rudholm | actually, by about 1976 mine had a three step prefixes, two xbar prefixes, and four 1AESS prefixes |
| 05:57.44 | rudholm | make that four step prefixes |
| 06:01.32 | tako-san|afk | CCFL_Man2: still around? |
| 06:04.07 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:08.26 | *** join/#asterisk OuterSpace (n=OuterSpa@201.230.161.178) |
| 06:09.21 | OuterSpace | hi, i have a 2 fxo card conected to pstn, i can use one port but if i try to use both i get: No translator path exists for channel type ZAP (native 76) to 256 |
| 06:09.50 | OuterSpace | show g729 -> 0/0 encoders/decoders of 1 licensed channels are currently in use |
| 06:11.20 | OuterSpace | dont suppose to show all of my channels ? |
| 06:13.00 | JT | rephrase? |
| 06:14.26 | OuterSpace | i can use 1 line, if i use 2 lines at same time it complains about: No translator path exists for channel type ZAP (native 76) to 256 |
| 06:14.58 | JT | <PROTECTED> |
| 06:15.23 | CCFL_Man2 | http://www.alleged.com/telephone/5ESS/ <---damn |
| 06:15.45 | CCFL_Man2 | no wonder why CO jobs are so hard to get |
| 06:16.12 | OuterSpace | i dont know how that licence works... but i suppose it have to say 4 licenced channels as i have a 4fxo card |
| 06:17.16 | CCFL_Man2 | operating a 5ess is pretty damn complicated |
| 06:19.51 | JT | CCFL_Man2: most telco switches are pretty complicated |
| 06:19.55 | OuterSpace | ok, my license say 1 channel... im wondering how was this server working before... they had @home. |
| 06:20.19 | JT | OuterSpace: is english your first language? |
| 06:21.42 | OuterSpace | peru here |
| 06:23.54 | CCFL_Man2 | JT: i never realized the software was so complicated |
| 06:25.09 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:25.22 | JT | it can do a lot of things |
| 06:25.33 | OuterSpace | nvm, old server says 4 registered channels |
| 06:25.38 | CCFL_Man2 | it can |
| 06:25.53 | JT | a lot of software appears to be complicated if you don't understand it |
| 06:25.59 | CCFL_Man2 | JT: i'd love if i can take a tour at my CO, but i doubt it |
| 06:26.06 | CCFL_Man2 | well, true |
| 06:27.25 | J4k3 | my telco has always had an open-door policy at the CO |
| 06:27.45 | J4k3 | of course the local head CO guy takes a lot of pride in their work... I wish the linesmen did. |
| 06:27.52 | J4k3 | err guys |
| 06:28.41 | CCFL_Man2 | your co isn't vzn? |
| 06:28.47 | CCFL_Man2 | my RBOC is |
| 06:28.56 | J4k3 | it was GTE back when I was in and out of it regularly. |
| 06:29.11 | J4k3 | during the merger we got spun off to Valor, which turned into Windstream |
| 06:29.35 | J4k3 | but they hadn't been GTE very long, everyone working there was ex-contel employees. |
| 06:29.35 | JT | CCFL_Man2: you can get tours of datacentres... close |
| 06:29.38 | J4k3 | and acted accordingly. |
| 06:31.54 | CCFL_Man2 | JT: CO is way more interesting though :P |
| 06:32.13 | CCFL_Man2 | it's amazing how old the switches are |
| 06:32.26 | J4k3 | haha we have a DMS-10 here |
| 06:32.34 | J4k3 | the friggin console |
| 06:32.41 | CCFL_Man2 | 5ESS and still uses magnetic tape, line printers, and various channels for console access |
| 06:32.42 | J4k3 | is a typewriter |
| 06:32.45 | J4k3 | no bs. |
| 06:32.47 | CCFL_Man2 | lol |
| 06:33.27 | *** join/#asterisk lbow (n=lbow@falcon.clarotech.co.za) |
| 06:33.27 | CCFL_Man2 | those decwriter type thing? |
| 06:33.31 | JT | what's wrong with magnetic tape? |
| 06:33.44 | CCFL_Man2 | nothing |
| 06:33.59 | J4k3 | a lot. |
| 06:34.16 | J4k3 | CCFL_Man2: I forget... it was huge.... wide format typewriter. |
| 06:34.25 | J4k3 | and quirky |
| 06:34.33 | *** join/#asterisk webman (n=chatzill@200.179.233.220.exetel.com.au) |
| 06:34.40 | J4k3 | the engineer mentioned he needed to recap it again |
| 06:34.47 | J4k3 | power supply going to crap. |
| 06:35.09 | J4k3 | it had to be at least 25 years old. |
| 06:35.11 | *** join/#asterisk KaiHanari (n=KaiHanar@CPE001478ec4f47-CM0011e6c7e1cf.cpe.net.cable.rogers.com) |
| 06:35.52 | CCFL_Man2 | J4k3: hah, recap, thats sweet |
| 06:36.37 | CCFL_Man2 | why couldn't he just use a laster printer and a cheap pc? |
| 06:36.40 | webman | I'm trying to get h323 to work with asterisk, can anyone suggest the best channel driver to use? |
| 06:37.42 | J4k3 | CCFL_Man2: why didn't dude just use his laptop and a decent terminal app? :P |
| 06:37.54 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 06:37.54 | KaiHanari | is there a way to execute a dial command in an extension path, so after the dialed extension/user/phone picks up, the context continues for instance after the call is connected, both users are played a sound? |
| 06:40.50 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:41.20 | CCFL_Man2 | J4k3: also true |
| 06:41.37 | CCFL_Man2 | J4k3: i want a 3b2 computer, you got one? |
| 06:47.27 | *** join/#asterisk putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:51.32 | CCFL_Man2 | http://cgi.ebay.com/AT-T-6300-PERSONAL-COMPUTER-WITH-EXTRAS-AT-T-UNIX_W0QQitemZ230163258560QQihZ013QQcategoryZ4193QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:51.40 | CCFL_Man2 | that makes me hard |
| 06:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 06:56.03 | asterisknerds | <PROTECTED> |
| 06:56.14 | KaiHanari | ... |
| 06:58.58 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 07:05.19 | *** join/#asterisk OuterSpace (n=OuterSpa@200.89.20.3) |
| 07:05.55 | OuterSpace | hi, how can i add a g729 licence ? (i added one already and i donde have g729 option anymore) |
| 07:08.02 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 07:08.17 | RsaMan | Hi |
| 07:08.18 | RsaMan | :) |
| 07:08.30 | RsaMan | i have one last issue with my bloody blind call transfer |
| 07:08.47 | RsaMan | http://www.asterisk.org/forum/viewtopic.php?t=17557&sid=17c959a74bd6964d55ec9b6a7a2f8597 |
| 07:09.00 | RsaMan | its summed up in that post |
| 07:12.41 | kaldemar | RsaMan: you don't have 102 in your dialplan. it's commented. |
| 07:14.16 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:16.43 | *** part/#asterisk TheNewAndy (n=TheNewAn@60-242-149-188.tpgi.com.au) |
| 07:18.06 | *** join/#asterisk dds (n=dds@41.98.156.220.st.bbexcite.jp) |
| 07:18.59 | dds | Hi, I'm trying to engineer a HA system for pure IAX2 traffic, basically over a grid where peers are added and removed dynamically. |
| 07:19.35 | RsaMan | kaldemar: its covered by the [office] context |
| 07:19.54 | dds | There is one frontend node to the system that all incoming calls first go to. It then determines, given the state of the grid at that point in time, where to route the call. |
| 07:20.22 | dds | Actually, I shouldn't have said HA, sorry, I was just reading the wiki page on HA since a lot of the information overlaps, but this is clearly load balancing. |
| 07:21.01 | kaldemar | RsaMan: yes, but you're not including office anywhere. as you can see the call to 102 lands into [incoming] and you only include internal there. |
| 07:21.08 | dds | My first question is I can't find an API call in the manager for adding a DUNDi peer at runtime. Does such a function exist? |
| 07:21.29 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:22.58 | RsaMan | kaldemar : but internal includes office |
| 07:25.19 | RsaMan | kaldemar : or am i missing something ? |
| 07:26.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:29.34 | RsaMan | can i set a single context for all blind transfers ? |
| 07:30.48 | RsaMan | Transferring IAX2/102-5 to '500' (context incoming) priority 1 |
| 07:30.48 | RsaMan | <PROTECTED> |
| 07:30.48 | RsaMan | <PROTECTED> |
| 07:30.57 | RsaMan | i get that output on my cli |
| 07:33.17 | CCFL_Man2 | so i think i know how the layers in vzn adsl work |
| 07:34.49 | CCFL_Man2 | the adsl physical layer, the two wire interface, then the atm layer, within that the vp/vc, within that the ethernet layer, within that the ppp layer, within that the tcp/ip layer |
| 07:37.39 | J4k3 | I thought vz went pppoa? |
| 07:38.05 | CCFL_Man2 | no, pppoe |
| 07:38.14 | J4k3 | ahh |
| 07:39.02 | CCFL_Man2 | but they use atm to transport ethernet |
| 07:41.14 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 07:52.46 | *** join/#asterisk jkrueger (n=jkrueger@ip66-104-41-250.z41-104-66.customer.algx.net) |
| 07:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 07:56.03 | asterisknerds | <PROTECTED> |
| 08:02.34 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:04.19 | *** join/#asterisk kkn088 (n=kikoun@88-136-177-151.adslgp.cegetel.net) |
| 08:05.39 | *** join/#asterisk shinao1 (n=shinao1@41.219.193.41) |
| 08:08.04 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 08:11.38 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:12.20 | RsaMan | i cannot find a solution to my silly blind transfer issue |
| 08:13.25 | RsaMan | :( |
| 08:13.29 | RsaMan | i want to cry |
| 08:13.48 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 08:14.18 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 08:14.32 | jkrueger | reboot? ^___^ |
| 08:15.02 | RsaMan | tried |
| 08:15.03 | RsaMan | :( |
| 08:15.07 | jkrueger | whats it doin? |
| 08:16.30 | RsaMan | dropping the call after the transfer |
| 08:16.32 | RsaMan | both calls |
| 08:17.00 | jkrueger | well, both calls would make sense as it would kinda see a hangup by one party |
| 08:17.27 | jkrueger | regular calls work fine to the extension its being transfered to? and this is an issue for any extension/number? |
| 08:18.07 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:18.57 | tzafrir | RsaMan, from zaptel alone? |
| 08:19.17 | tzafrir | ah, blind transfer |
| 08:19.41 | RsaMan | http://www.asterisk.org/forum/viewtopic.php?t=17557&sid=21578e6861f057ab22f48366cf8fde8f |
| 08:19.44 | RsaMan | i summed it up here |
| 08:21.24 | tzafrir | what is extension 102 on context incoming? |
| 08:21.29 | tzafrir | show dialplan incoming |
| 08:22.00 | RsaMan | kk |
| 08:22.55 | RsaMan | [incoming] |
| 08:22.56 | RsaMan | ;exten => _X!,1,NoOp(Got a call to ${EXTEN}) |
| 08:22.56 | RsaMan | include => internal |
| 08:22.56 | RsaMan | exten => 444,1,Playback(tt-monkeys) |
| 08:24.30 | RsaMan | 102 falls under office |
| 08:24.44 | jkrueger | RsaMan: can you increase the verbosity level and do the blind transfer again? |
| 08:25.13 | tzafrir | RsaMan, so is there an extension 102 there? |
| 08:25.31 | tzafrir | you say that there isn't |
| 08:25.52 | tzafrir | and thus the problem is that you transfer in the wrong context |
| 08:26.49 | RsaMan | if i try transfer to 444 |
| 08:26.53 | RsaMan | i get this cli output |
| 08:27.03 | *** join/#asterisk vlt (n=dm@suez.musketa.de) |
| 08:27.28 | RsaMan | -- Stopped music on hold on IAX2/102-6 |
| 08:27.28 | RsaMan | <PROTECTED> |
| 08:27.28 | RsaMan | <PROTECTED> |
| 08:27.28 | RsaMan | <PROTECTED> |
| 08:27.28 | RsaMan | <PROTECTED> |
| 08:28.40 | RsaMan | :( |
| 08:28.56 | jkrueger | RsaMan: what is your verbosity level? |
| 08:29.01 | vlt | Hello. Is there a global variable the original (dialled) extension of an incoming call is available even after ${EXTEN} has changed after jumping between several contexts or do I have to store it in a separate var my own? |
| 08:29.03 | RsaMan | 10 |
| 08:29.49 | *** join/#asterisk kv0s (n=kv0s@p4FD24C3A.dip.t-dialin.net) |
| 08:29.59 | kv0s | Houston, some problems here ... :-( |
| 08:30.01 | kv0s | Hi! |
| 08:30.28 | jkrueger | RsaMan: IAX2 calls Zap/1 (FXS), Zap/1 tries to blind transfer the IAX call to another extension? |
| 08:30.31 | kv0s | My asterisk works well, but one issue doesn't look really beautygull ... :-( |
| 08:30.45 | RsaMan | jkrueger : correct |
| 08:31.20 | jkrueger | have you checked your zap configs to make sure transfer=yes or whatever that setting is? |
| 08:31.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 08:31.57 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 08:32.00 | jkrueger | and have you tried a call in the opposite direction and tried a blind transfer from the IAX side? |
| 08:32.01 | tzafrir | jkrueger, transfer=yes is unrelated |
| 08:32.12 | tzafrir | this is for zaptel flash transfer |
| 08:32.37 | RsaMan | sec |
| 08:32.40 | RsaMan | i will try that |
| 08:32.42 | jkrueger | tzafrir: i figured i was wrong on having him check that anyways. long day :| |
| 08:32.58 | kv0s | At my box i've isdn (hfc-s) card and my clients use sip-softphones such as x-lite. if i call someone thru my sip-softphone on my outgoing zap/isdn trunk ... the first second i can't hear my called party ... |
| 08:33.04 | RsaMan | arg |
| 08:33.14 | kv0s | .... i don't hear their names or company ... |
| 08:33.23 | RsaMan | blind transfer does not work from the iax side |
| 08:34.38 | RsaMan | bleh |
| 08:34.44 | RsaMan | do i need to enable it ? |
| 08:36.15 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 08:36.24 | RsaMan | :( |
| 08:36.35 | jkrueger | nah, it should work via the tT options in your dial command |
| 08:36.46 | jkrueger | and crap i just rubbed cigarette ash all over my keyboard |
| 08:37.37 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:37.39 | RsaMan | hmm |
| 08:38.57 | jkrueger | RsaMan: have you tried attended transfers? |
| 08:40.39 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 08:43.33 | RsaMan | jkrueger: no i havent |
| 08:44.13 | jkrueger | RsaMan: and calling those extensions from the zap line work fine? |
| 08:44.30 | RsaMan | jkrueger: nope |
| 08:45.22 | jkrueger | so you cant even call the extensions? no xfers. just calling? |
| 08:45.47 | RsaMan | jkrueger: oops |
| 08:45.52 | RsaMan | jkrueger: i meant no problems |
| 08:45.56 | jkrueger | ahh ok |
| 08:46.05 | jkrueger | i was gonna say, bigger problems than xfers |
| 08:46.26 | tzafrir | kv0s, hmm, echo canceller training? |
| 08:46.35 | RsaMan | jkrueger: the blind hotkey works in iax now |
| 08:46.44 | RsaMan | jkrueger: but it still doesnt transfer |
| 08:46.51 | tzafrir | what happens if you disable the echo canceller altogether in zapata.conf ? |
| 08:47.25 | kv0s | tzafrir: Mhm. Ok. I have a look at zapata.conf ... mom. |
| 08:49.07 | kv0s | tzafrir: echotraining = 100 |
| 08:49.25 | kv0s | tzafrir: does it make sense to increase these value? 800 for example? |
| 08:49.46 | RsaMan | :( |
| 08:50.23 | jkrueger | RsaMan: just curious. what version of asterisk are you running? |
| 08:51.11 | RsaMan | 1.4 |
| 08:51.25 | RsaMan | how do i check the exact? |
| 08:52.18 | jkrueger | show version |
| 08:52.25 | jkrueger | err, core show version |
| 08:52.31 | jkrueger | both should work |
| 08:52.31 | RsaMan | 1.4.10.1 |
| 08:52.38 | jkrueger | ok, so you are fairly current |
| 08:53.59 | RsaMan | :( |
| 08:55.12 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:55.15 | kv0s | tzafrir: echotraining set 800 - now it wokrs ... perfect! |
| 08:55.31 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 08:55.34 | RsaMan | can i rather use flash to transfer? |
| 08:55.42 | asterisknerds | <PROTECTED> |
| 08:55.51 | RsaMan | if i use the transfer function on my iax client it transfers |
| 08:56.39 | jkrueger | i may be wrong, but i believe the transfer function on sip/iax phones transfers differently than blindxfer from features.conf |
| 08:57.50 | RsaMan | i dont mind using that |
| 08:58.22 | RsaMan | but i want to transfer with zap |
| 08:58.23 | RsaMan | as well |
| 08:58.29 | RsaMan | so mayhbe use flash() |
| 08:59.12 | RsaMan | instead to transfer |
| 08:59.30 | jkrueger | try it out |
| 09:03.17 | s0ck | tzafrir: last time we spoke 'zap show channels' wasn't showing my '1' channel; now it does but still can't dial in/out. any ideas? :D |
| 09:04.14 | tzafrir | what dial command do you use? |
| 09:04.24 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 09:05.03 | vlt | Hello. Is there a similar function like Background() that waits for the caller to enter a digit w/o playing a file? I tried Wait() but that didn't work. |
| 09:05.26 | dds | The documentation claims that the EID '*' can be used in DUNDi for matching arbitrary hosts, but after reading the source code, that does not appear to be the case. |
| 09:06.02 | dds | dundi_eid_cmp does not check for it, |
| 09:06.02 | YonahW-Work | vlt: i think it might be waitexten |
| 09:06.05 | dds | build_peer doesn't check for it... |
| 09:06.11 | vlt | YonahW-Work: Aah, that easy ... thank you. |
| 09:06.25 | YonahW-Work | :) |
| 09:06.37 | dds | dundi_str_to_eid doesn't do it.. |
| 09:07.19 | dds | and set_config doesn't seem to have any special handling for it |
| 09:07.24 | dds | am I missing something? |
| 09:11.01 | s0ck | tzafrir: when i dial in, even with verbose @ 99 nothing appears on the cli and the call is never answered |
| 09:11.04 | jeremy_g | ok girls time is up |
| 09:11.22 | jeremy_g | now you better go to your homes |
| 09:11.55 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 09:12.20 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:12.32 | tzafrir | can you pastebin the dialplan context you dial into? the one you see in zap show channels |
| 09:12.49 | *** join/#asterisk hot_wheelz (i=hotwheel@124-168-132-199.dyn.iinet.net.au) |
| 09:14.45 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 09:15.19 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 09:15.46 | hot_wheelz | how do i set the cvsroot variable for asterisk pls |
| 09:19.02 | tzafrir | hot_wheelz, svn, not cvs |
| 09:19.22 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 09:19.30 | tzafrir | http://asterisk.org/ ->developers |
| 09:19.54 | tzafrir | http://asterisk.org/developers -> Get the latest Source |
| 09:19.57 | tzafrir | http://asterisk.org/developers/get-source |
| 09:20.50 | tzafrir | what branch of asterisk do you want , exactly? trunk (bleeding edge) or 1.4 (current stable)? |
| 09:21.08 | hot_wheelz | stable |
| 09:21.16 | jhiver | hey guys, do you know if IAX trunking works OK with freebsd's asterisk port on a box without any hardware timing device? |
| 09:21.32 | tzafrir | svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 09:21.53 | tzafrir | also try visiting that http:// link in your browser |
| 09:22.04 | hot_wheelz | so hat command do I type to download and install source then |
| 09:22.06 | hot_wheelz | thx |
| 09:22.48 | jhiver | i need to get an IAX trunk quick between a rented data center box and a local site |
| 09:23.38 | jhiver | i'm wondering if freebsd's asterisk will do the job or if i need to get debian (i think debian comes with ztdummy by default if i remember) |
| 09:23.40 | tzafrir | hot_wheelz, just a standard building from source: ./configure; make; make install |
| 09:24.15 | tzafrir | jhiver, with Debian you generally need m-a a-i zaptel |
| 09:24.21 | tzafrir | but that's basically it |
| 09:25.20 | hot_wheelz | i neeed to downlaod source first rigt |
| 09:25.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:29.10 | jhiver | tzafrir, what do you mean? |
| 09:29.51 | jhiver | i'm more on freebsd so i haven't looked at debian for quite a while =) |
| 09:29.55 | tzafrir | getting zaptel modules (and ztdummy specifically) on Debian. Though your interest was FreeBSD, I gather |
| 09:29.58 | Bladerunner05 | With TDM400p and asterisk 1.4.9 when I hang up a call the caller don't receive the hang up signal |
| 09:30.05 | jhiver | ok |
| 09:30.16 | jhiver | so i can just use debian packages or do i need to recompile? |
| 09:30.37 | tzafrir | the recompile is just for the zaptel modules themselves |
| 09:30.56 | tzafrir | zaptel itself is from binaries |
| 09:31.00 | *** join/#asterisk HarryR (n=Harry@77.240.48.173) |
| 09:31.11 | jhiver | ok so as i understand i need to recompile the ztdummy module |
| 09:31.39 | rati | in asterisk Mangment portal when i am adding trunk with outbond id 7237570239 its giving error |
| 09:32.09 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 09:33.04 | jhiver | is there a way i can check with asterisk if a timing device is loaded? |
| 09:33.13 | jhiver | on the CLI for example? |
| 09:34.46 | rati | in asterisk mangment portl --> add trunk for outbond caller id which number i should give |
| 09:35.30 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:38.40 | tengulre | hi,all |
| 09:45.09 | tzafrir | jhiver, zttest |
| 09:47.00 | Zefk | Hi, does any one know how to install B410P on * 1.4? The kb on digium site is old and I guess is for * 1.2. Thx |
| 09:47.29 | tzafrir | jhiver, a simpler test: head -c 0 /dev/zap/pseudo |
| 09:48.08 | tzafrir | this should test that you can open a pseudo device |
| 09:48.19 | tzafrir | if you get an error, there's no timing source |
| 09:51.24 | *** join/#asterisk HarryR (n=Harry@77.240.48.173) |
| 09:56.56 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:57.43 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 10:00.16 | *** join/#asterisk shtoom (n=shtoom@221.128.190.26) |
| 10:01.57 | *** join/#asterisk saftsack (n=saftsack@p57A76CD7.dip.t-dialin.net) |
| 10:03.14 | zeeesh | i just download "app_cbmysql.c" in "asterisk-1.4.8/apps" for meetme webcontrol.. how to Edit the Makefile in that folder using the patch ? |
| 10:04.28 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 10:13.50 | Yoe | Is there a piece of hardware that allows me to connect DECT phones to an asterisk box? |
| 10:14.04 | Yoe | preferably in such a way that I can transfer calls between DECT handsets and other phones |
| 10:15.01 | J4k3 | yes |
| 10:15.06 | J4k3 | there are dect voip phones available |
| 10:15.33 | Yoe | can you give an example? |
| 10:15.53 | J4k3 | personally I'm just getting a wm5 cellphone with wifi, and use it exclusively on wifi with a sip client. |
| 10:16.12 | Yoe | well, I already have four DECT handsets, and I'm not going to throw them away |
| 10:17.09 | HarryR | you could try buying some cheap atas |
| 10:17.26 | J4k3 | atas are sometimes acceptable |
| 10:17.34 | HarryR | or try exchanging them with a reseller for S450IP models :\ |
| 10:17.36 | Yoe | yes, but then I can't transfer a call from a DECT handset to an asterisk system |
| 10:17.58 | HarryR | put them on hold and use asterisk extensions to do it.. |
| 10:18.07 | J4k3 | it seems like somebody even sells a USB DECT base for * |
| 10:18.11 | Yoe | yeah, but that sucks, doesn't it? |
| 10:18.16 | HarryR | not really |
| 10:18.19 | Yoe | J4k3: ah, that sounds interesting |
| 10:18.39 | Yoe | J4k3: any pointers? |
| 10:18.48 | J4k3 | I can't remember any brand or model info offhand |
| 10:18.51 | Yoe | darn |
| 10:18.55 | J4k3 | I'm in the US |
| 10:19.05 | J4k3 | dect isn't here yet afaik |
| 10:19.06 | Yoe | heh |
| 10:19.10 | Yoe | IC |
| 10:19.14 | Yoe | didn't even know that |
| 10:20.08 | HarryR | http://www.broad-tel.com/products/wireless.php - DVG-100 |
| 10:20.20 | HarryR | DECT to VoIP gateway |
| 10:20.54 | *** join/#asterisk ajohnstone (n=ajohnsto@85.189.117.98) |
| 10:21.46 | Yoe | "Call transfer between handsets" -- doesn't look like it actually does transfer over VoIP, too. |
| 10:21.56 | HarryR | call them up and ask :) |
| 10:22.08 | Yoe | yeah, I'll have to do that |
| 10:24.46 | Yoe | ah -- there's a "Siemens Gigaset M34 USB" which seems to be a USB DECT thing |
| 10:25.03 | Yoe | now the only hope is that asterisk supports it |
| 10:25.39 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 10:30.33 | JT | usb, stuff that |
| 10:31.25 | RsaMan | eeep |
| 10:32.40 | RsaMan | i am still stuck with my blind transfer issue |
| 10:32.41 | RsaMan | :( |
| 10:32.55 | RsaMan | both callers are hung up on |
| 10:33.00 | RsaMan | after i enter the new number |
| 10:33.26 | RsaMan | http://www.asterisk.org/forum/viewtopic.php?t=17557&sid=21578e6861f057ab22f48366cf8fde8f |
| 10:33.32 | RsaMan | those are my settings |
| 10:33.33 | RsaMan | :( |
| 10:34.29 | RsaMan | when i transfer the call |
| 10:34.33 | RsaMan | to 500 |
| 10:34.38 | RsaMan | cli output =Stopped music on hold on IAX2/102-2 |
| 10:34.38 | RsaMan | <PROTECTED> |
| 10:34.38 | RsaMan | <PROTECTED> |
| 10:34.38 | RsaMan | <PROTECTED> |
| 10:34.38 | RsaMan | <PROTECTED> |
| 10:34.39 | RsaMan | <PROTECTED> |
| 10:34.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:34.41 | RsaMan | so |
| 10:34.43 | RsaMan | z |
| 10:34.52 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 10:35.50 | *** join/#asterisk kkn088 (n=kikoun@88-136-177-151.adslgp.cegetel.net) |
| 10:36.15 | kombi | I am quite sure there was this feature that allowed callers to leave a queue and drop into another defined context instead, I just can't remember it.. I believe it worked with some standard extension. Does anyone remember of the top of her head? |
| 10:36.22 | Yoe | hmm |
| 10:36.46 | Yoe | okay, so it seems that what's available out there either is way too expensive, or does not allow transfers of calls between DECT and asterisk seamlessly |
| 10:36.49 | Yoe | morons |
| 10:37.07 | Yoe | or simply doesn't work with asterisk -- that too |
| 10:39.23 | J4k3 | it'll work |
| 10:39.47 | Yoe | J4k3: no, what I mean is, there is hardware out there that doesn't have Linux drivers |
| 10:39.53 | Yoe | no chance of getting those to work, obviously |
| 10:41.02 | J4k3 | yeah, and thats always a possibility |
| 10:41.29 | Yoe | exactly |
| 10:41.50 | Yoe | one device was very nice, but cost over 1100EUR |
| 10:41.57 | J4k3 | weow |
| 10:42.00 | Yoe | no chance in hell I'm going to buy that for 4 handsets :) |
| 10:42.08 | Yoe | http://www.voipsolutions.be/dect/kirk-wireless-server-ip600v3-sipdect.html |
| 10:42.37 | Yoe | it even mentions asterisk in its description |
| 10:42.41 | J4k3 | yeah |
| 10:44.42 | J4k3 | I dunno, I'm paying ~$180/ea for PDA phones... that kinda hurts |
| 10:44.58 | J4k3 | but not as bad as 1100 euro |
| 10:45.26 | Yoe | yeah, but then that kirk thing allows me to connect up to 35 DECT phones to one device, which is way overkill for what I need |
| 10:46.01 | J4k3 | http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-c450-ip-voip-phone.html |
| 10:46.02 | Yoe | there's other systems on that site that allow me to connect a reasonable number of DECT handsets to one base station, but most of those only allow me one VoIP line |
| 10:46.05 | Yoe | which isn't enough |
| 10:46.29 | J4k3 | http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-c455-ip-6-providers-voip-phone.html |
| 10:46.31 | *** join/#asterisk trwunna (n=trwunna@203.81.71.91) |
| 10:46.48 | trwunna | hello all |
| 10:46.51 | Yoe | that one, for example, only allows one VoIP connection |
| 10:46.58 | trwunna | hi Yoe |
| 10:47.05 | trwunna | can i get any help for question? |
| 10:47.22 | Yoe | trwunna: likely, but perhaps not from me |
| 10:47.31 | Yoe | J4k3: the C455 sounds more interesting... |
| 10:47.35 | trwunna | yes, thanks, |
| 10:48.44 | Yoe | trwunna: what I mean is, don't ask to ask, just ask |
| 10:50.15 | trwunna | Yoe, you mean, one time asking? |
| 10:51.40 | Yoe | no, I mean, just ask your fucking question, rather than bothering everyone here with "can i get any help for question" |
| 10:51.59 | trwunna | ok, i see it |
| 10:52.02 | trwunna | thanks Yoe |
| 10:54.52 | *** part/#asterisk RsaMan (n=aa@196.210.154.3) |
| 10:57.39 | Yoe | is an iax2 call encrypted? |
| 10:57.41 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.209) |
| 10:58.34 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.209) |
| 11:01.26 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 11:04.43 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.209) |
| 11:05.24 | jkrueger | not that im aware of |
| 11:06.44 | jkrueger | the only encrypted VoIP that I know of is with a Snom phone via a Snom SIP gateway. I have heard no news of Asterisk supporting this encryption in the last few years. could have changed *shrugs* |
| 11:06.45 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 11:08.00 | trwunna | can i get any help for question ? |
| 11:08.29 | *** join/#asterisk saftsack (n=oliver@p54A7BD8F.dip.t-dialin.net) |
| 11:08.44 | Yoe | jkrueger: okay. It's just that I'm wondering what security the password in an iax2 configuration brings |
| 11:08.47 | Yoe | not that much then, I guess |
| 11:08.59 | jkrueger | well, there are a few ways to do it |
| 11:09.07 | jkrueger | plain text secret is the most popular |
| 11:09.11 | jkrueger | MD5 |
| 11:09.19 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-17-42.w81-251.abo.wanadoo.fr) |
| 11:09.26 | jkrueger | and me thinks Public/Private key auth |
| 11:09.36 | jkrueger | i remember it being in one of the channel drivers |
| 11:09.38 | Yoe | I do have a plaintext password in my configuration; but if that's also sent in plaintext over the wire, that isn't very helpful |
| 11:09.47 | Yoe | if there is a public/private thing, that might be worth looking into |
| 11:09.48 | trwunna | can i get any help for question ? |
| 11:10.06 | jkrueger | trwunna: just ask the question instead of asking if you can ask |
| 11:10.09 | Yoe | trwunna: you would have gotten help ages ago if you'd just asked your question, stupid |
| 11:10.23 | tzafrir | asterisk supports IAX encription, or even simple RSA private/public key authentication. |
| 11:10.38 | tzafrir | Not sure which other IAX software supports that |
| 11:10.55 | trwunna | Yoe, i write down what you said before,, |
| 11:10.58 | Yoe | okay, thanks |
| 11:11.02 | trwunna | if that is mistake , sorry |
| 11:12.04 | jkrueger | tzafrir: it may be stupid to do so, but i would assume any other IAX software based off digiums libiax has the potential to support it. providing that digium put the authentication in libiax and not based on 3rd party support. |
| 11:12.13 | trwunna | I am facing with that problem in my SIP , i can not call to sip phone and at SIP client Micronet error show like that,,,, "SIP/2.0 500 Internal Server Error" |
| 11:12.14 | jkrueger | but i never recall seeing any iax software having that option :( |
| 11:13.34 | tzafrir | jkrueger, two years ago I searched and found nothing |
| 11:14.38 | jkrueger | its open source. write your own :) or pay some talented asterisk consultant to write one for you ^___^ |
| 11:15.22 | HarryR | tzafrir, most asterisk IAX clients are based on libiax, so if libiax supports it most clients will |
| 11:16.04 | jkrueger | HarryR: that is assuming those clients actually decided to use the functionality |
| 11:16.32 | jkrueger | developers are inherently lazy and will most of the time only code in most features people will need until complained at |
| 11:17.27 | JunK-Y | you certainly not a dev. :) |
| 11:17.41 | HarryR | I was presuming that it'd support it transparently |
| 11:18.04 | *** join/#asterisk aikanaro79 (n=noone@89.181.72.154) |
| 11:18.43 | aikanaro79 | hi..using meetme is it possible to define conference rooms with pattern matching? (i.e. the same way it's possible to do inside the dialplan) |
| 11:18.55 | jkrueger | JunK-Y: I am a dev. I'm speaking from experience. Not what I actually do. But what I have seen many co-workers and friends co-workers do. |
| 11:19.55 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 11:22.47 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 11:22.50 | jeremy_g | aikanaro79:yeah its possible. |
| 11:23.05 | aikanaro79 | jeremy_g: thanks |
| 11:23.15 | jeremy_g | aikanaro79:the pleasure is all yours |
| 11:23.19 | jeremy_g | aikanaro79:i mean mine |
| 11:23.20 | jeremy_g | :p |
| 11:23.25 | jkrueger | Anyone play with a Pika Technologies analog board? I just got a free eval board from them yesterday. |
| 11:23.40 | jeremy_g | jkrueger:yeah send to us and we ll play then |
| 11:23.41 | aikanaro79 | jeremy_g: by the way...do you know where I can find more information about meetme? I've been having problems finding it |
| 11:23.53 | jkrueger | jeremy_g: haha no. |
| 11:24.07 | jeremy_g | aikanaro79:usually voip-wiki is the best resource... its quite simple actually |
| 11:24.44 | jeremy_g | jkrueger:cmon, dont be a __, its freee after all you got |
| 11:24.48 | jeremy_g | :p |
| 11:24.54 | aikanaro79 | jeremy_g: ok...thanks...in that case I'll just start experimenting...do you know of any SIP (free) client I can use to test it? |
| 11:25.14 | jkrueger | jeremy_g: sure. you can have it. need to pay me $550 shipping + handling :p |
| 11:25.47 | Strom_M | handling is done by little old ladies in Bavaria; shipping is by luxury liner |
| 11:26.25 | jkrueger | luxery liner to coast, then 3 private helicopters. |
| 11:26.38 | jeremy_g | jkrueger:550$ for shipping n handling...are you an * user from artic |
| 11:26.44 | tzafrir | "how do I install asterisk"? |
| 11:26.54 | jeremy_g | lol |
| 11:26.57 | jkrueger | jeremy_g: $550 is aprox retail for the board :p |
| 11:27.06 | tzafrir | Part of the anstawer: http://bugs.digium.com/10523 |
| 11:27.10 | jeremy_g | jkrueger:you got it free |
| 11:27.17 | tzafrir | A script to install prerequerments |
| 11:27.24 | tzafrir | Please provide feedback |
| 11:27.47 | jeremy_g | jkrueger:we dont charge each other for stuff we get free |
| 11:28.02 | jeremy_g | tzafrir:looking at it... |
| 11:28.16 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:28.28 | tzafrir | A similar build_tools/install_prereq is already in Zaptel |
| 11:29.10 | jkrueger | I WAS JOKING |
| 11:29.12 | jkrueger | jeez |
| 11:29.14 | jkrueger | ball buster |
| 11:36.18 | *** join/#asterisk ToyMan (n=Stuart@user-160uamh.cable.mindspring.com) |
| 11:37.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:42.28 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:43.59 | aikanaro79 | has anyone used iaxclient instead of meetme to create conference calls with asterisk? (only now have I found it) |
| 11:45.42 | JunK-Y | you probably means app_conference. |
| 11:45.45 | *** join/#asterisk Yoe (n=wouter@samba.grep.be) |
| 11:47.03 | *** join/#asterisk OuterSpace (n=OuterSpa@200.89.20.3) |
| 11:49.02 | aikanaro79 | JunK-Y: sorry, that's it |
| 11:49.32 | OuterSpace | hi, i have problems with music on hold (using mode=files and default moh .wav files), when play the files sounds like weird noises, but if i use mixmonitor to save the call when i hear the file, the music on hold sounds ok. |
| 11:49.35 | JunK-Y | search the wiki for this, the info is pretty clear. |
| 11:50.15 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 11:50.16 | OuterSpace | btw, i have no sound card |
| 11:53.23 | Yoe | is there a variable available in the dialplan that contains the current Unix time? |
| 11:55.11 | JunK-Y | Yoe: with 1.4 ^ |
| 11:55.12 | JunK-Y | ? |
| 11:55.13 | kaldemar | Yoe: use function STRFTIME |
| 11:55.14 | JunK-Y | core show function STRFTIME |
| 11:55.20 | Yoe | JunK-Y: no, 1.2 |
| 11:55.32 | kaldemar | 1.2 has that function too. |
| 11:55.37 | Yoe | okay, thanks |
| 11:55.46 | JunK-Y | Yoe: or use ${EPOCH} if i remember correctly |
| 11:56.09 | JunK-Y | but thats deprecated, use function which will make things easier for you in the future. |
| 11:56.46 | *** join/#asterisk zippytech (n=zippytec@71.155.129.242) |
| 11:56.51 | zippytech | any one here |
| 11:57.09 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:57.35 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 12:02.55 | lirakis | zippytech: yep |
| 12:03.06 | zippytech | any idea way i seam to loose the parking feature on asterisk boxes, no logs or nothing |
| 12:03.16 | lirakis | zippytech: .. no |
| 12:03.19 | lirakis | ha ha |
| 12:03.26 | zippytech | great |
| 12:03.28 | lirakis | ... you have to be a lot more specific than that |
| 12:04.14 | zippytech | well we was placing calls in the parking lot last week now without any changes nothing happend when you press #700 |
| 12:04.16 | zippytech | or 700 |
| 12:04.25 | *** join/#asterisk oej_ (n=olle@213.212.1.52) |
| 12:04.35 | zippytech | don't work on 2 diferent boxes |
| 12:16.18 | *** join/#asterisk shinao1 (n=shinao1@41.219.201.185) |
| 12:17.11 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 12:21.12 | jeremy_g | ok girls time is up |
| 12:21.24 | jeremy_g | dont do this anymore |
| 12:23.33 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 12:24.52 | *** join/#asterisk ManxPower (n=manxpowe@032-457-509.area7.spcsdns.net) |
| 12:25.41 | *** join/#asterisk guillote_GNU (n=bancaria@host98.190-31-248.telecom.net.ar) |
| 12:26.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:35.24 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:36.31 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 12:38.34 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 12:44.15 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 12:45.12 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
| 12:45.14 | Penggu | hi all |
| 12:45.43 | Penggu | is there any way to check an extension (specifically SIP/<something>) *before* dialling it, for any active channels ? |
| 12:46.18 | Penggu | i wanted to warn the caller that the person they're calling *are* on the phone, and may or may not answer them |
| 12:46.19 | JunK-Y | Penggu: ischanavail() |
| 12:46.47 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 12:46.53 | Penggu | JunK-Y: will that work if the person hasn't got max-calls (or was it call-limit) configured for their sip channel? |
| 12:47.06 | Penggu | (meaining will it work even if it will otherwise normally trigger a call waiting signal?) |
| 12:48.00 | *** join/#asterisk Strom_M (n=strom@160.79.29.94) |
| 12:49.08 | JunK-Y | try it, u will see :) |
| 12:49.22 | x86 | Penggu: what happened when you tried it? |
| 12:49.36 | Penggu | i havent yet.. |
| 12:49.42 | Penggu | im just trying to find doco on it now |
| 12:50.28 | Penggu | btw, any reports on app_rxfax/app_txfax compiling on 1.2.24 ? |
| 12:53.52 | Penggu | chanisavail |
| 12:54.06 | Penggu | not ischanavail... thre me off a bit |
| 12:54.24 | kippi | can anyone help me configure FOP? need to know how to see my extenstions etc |
| 12:56.05 | [TK]D-Fender | Penggu: "show application chanisavail" |
| 12:56.56 | Penggu | yep, it's available |
| 12:57.02 | Penggu | nice coloured help there btw |
| 12:57.05 | Penggu | never knew about that |
| 12:57.37 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 12:58.07 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:03.00 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:04.28 | Pagautas | hi all |
| 13:04.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:04.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:04.42 | Pagautas | any developer there? |
| 13:04.59 | Pagautas | im trying new feature on 1.4.11 |
| 13:05.02 | Pagautas | endbeforehexten=yes |
| 13:05.39 | Pagautas | but cdr is written after exiting h extension |
| 13:06.00 | Pagautas | i'm using mysql cdr |
| 13:06.05 | Pagautas | maybe there is a prob? |
| 13:08.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.41 | Mavvie | """?? ???? :)) ??"" <502132>","502132","911","barnet-from-internet","SIP/sip.barnet.com.au-090cb518","","NoOp","BYE", |
| 13:09.53 | Mavvie | at least try to dial the emergency number of the country where the server is located in... |
| 13:09.56 | Mavvie | dumb script kiddies. |
| 13:10.56 | *** join/#asterisk ManxPower (n=manxpowe@032-457-509.area7.spcsdns.net) |
| 13:11.37 | ManxPower | BTW, if anyone has recommendtations for a corporate quality (500 users) firewall with a idiot..er..GUI config please /msg me |
| 13:12.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:14.00 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:17.21 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:17.55 | Penggu | i've looked the asterisk commands, but can't find this: is there an IVR function to take dialled input from a caller and store it into a variable (not translate it into an extension as WaitExten does...) ? |
| 13:19.05 | Penggu | hmm |
| 13:19.07 | Penggu | Read() |
| 13:22.45 | *** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com) |
| 13:23.43 | Dr-Linux | ManxPower: free? :P |
| 13:24.08 | ManxPower | Dr-Linux: every free firewall I've seen sucked. |
| 13:24.52 | Dr-Linux | ManxPower: my servers are behind the PIX also using local iptables |
| 13:25.13 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 13:25.32 | *** join/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-38-168.w82-123.abo.wanadoo.fr) |
| 13:25.38 | Dr-Linux | aww wtf, |
| 13:26.01 | Dr-Linux | ManxPower: i always see manager msgs loop at CLI: |
| 13:26.16 | Dr-Linux | <PROTECTED> |
| 13:26.16 | Dr-Linux | <PROTECTED> |
| 13:26.29 | bertrand^ | hello |
| 13:26.31 | Dr-Linux | don't know how to stop it |
| 13:26.39 | ManxPower | Dr-Linux: stop using asterisk -c |
| 13:26.44 | ManxPower | ..er.. |
| 13:26.44 | ManxPower | -r |
| 13:26.48 | ManxPower | that is -r |
| 13:27.08 | bertrand^ | anyone has an idea about chasing a SIP registration bug? |
| 13:27.10 | tzafrir | bertrand^, testing, 1, 2, 3 |
| 13:27.17 | Dr-Linux | what i did is enabled = no in manager.conf |
| 13:27.21 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:27.35 | bertrand^ | i've Polycom SoundPoint 300/301, registering for 3600 seconds |
| 13:27.48 | Dr-Linux | but now when i did "reload manager" it shows me: |
| 13:27.58 | Dr-Linux | <PROTECTED> |
| 13:27.59 | Dr-Linux | Aug 22 06:20:46 WARNING[6159]: manager.c:1737 init_manager: Unable to change management port / enabled |
| 13:28.03 | robl^ | BUG ALERT -- Zaptel 1.4.5 has a bug -- won't "make install": make: *** No rule to make target `install-inlcude', needed by `install-programs'. Stop. |
| 13:28.03 | Dr-Linux | and still getting same loop |
| 13:28.10 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-a996f0eb4a581f98) |
| 13:28.11 | Dr-Linux | ManxPower: then what i can use to get CLI>? |
| 13:28.23 | bertrand^ | and sometimes, the phone randomly disappear from the "sip show peers" list |
| 13:28.35 | bertrand^ | 3003/3003 (Unspecified) D 0 Unmonitored |
| 13:28.49 | ManxPower | Dr-Linux: when asterisk -r is run, it generates a unix connection to the existing Asterisk process and prints that message. |
| 13:28.57 | robl^ | Easy fix -- "inlcude" is mis-spelled. search and replace with the correct spelling and it works |
| 13:29.54 | Dr-Linux | ManxPower: but i've alot of other servers, i can't see those messages there but only on this server, how can i stop these msgs? |
| 13:30.06 | x86 | <PROTECTED> |
| 13:30.15 | Dr-Linux | as it annoying when i do debug and trace the call |
| 13:30.18 | *** join/#asterisk sasch (n=sasch@host117-234-static.4-79-b.business.telecomitalia.it) |
| 13:30.20 | sasch | hi all |
| 13:30.25 | *** part/#asterisk saftsack (n=oliver@p54A7BD8F.dip.t-dialin.net) |
| 13:30.58 | sasch | i install in ubuntu-server asterisk 1.2.x ... where i can download for music on hold ?? |
| 13:31.36 | Dr-Linux | sasch: use native |
| 13:31.58 | ManxPower | Dr-Linux: you CANNOT STOP THEM. |
| 13:32.17 | *** join/#asterisk SgtDitt (n=SgtDitt@38.103.84.36) |
| 13:32.17 | ManxPower | the message is printed when an instance of asterisk connects to an existing instance of asterisk. |
| 13:32.17 | Dr-Linux | ManxPower: ok, but why only on this asterisk server |
| 13:32.30 | ManxPower | Dr-Linux: SOMETHING is connecting |
| 13:32.53 | sasch | <Dr-Linux> but in /usr/share/asterisk/mohmp3 i don't found any mp3 files |
| 13:33.40 | Dr-Linux | ManxPower: yeah, that's why i'm trying to stop manager interface |
| 13:33.45 | Dr-Linux | that also i can't stop |
| 13:33.57 | Dr-Linux | i2c-pbx*CLI> reload manager |
| 13:33.58 | Dr-Linux | <PROTECTED> |
| 13:33.58 | Dr-Linux | Aug 22 06:20:46 WARNING[6159]: manager.c:1737 init_manager: Unable to change management port / enabled |
| 13:34.15 | bertrand^ | sip show peer 3003 |
| 13:34.16 | bertrand^ | <PROTECTED> |
| 13:34.42 | ManxPower | Dr-Linux: rm /etc/asterisk/manager.conf |
| 13:35.59 | Dr-Linux | ManxPower: opss no, maybe i'd need it later |
| 13:36.07 | Dr-Linux | for my monitoring system |
| 13:36.21 | Dr-Linux | ManxPower: maybe i can rename it for time being |
| 13:36.28 | ManxPower | Dr-Linux: the rename it |
| 13:37.15 | Dr-Linux | ManxPower: hhm... |
| 13:37.15 | Dr-Linux | [root@i2c-pbx root]# netstat -anp|grep :5038 |
| 13:37.16 | Dr-Linux | tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 1135/asterisk |
| 13:38.21 | ManxPower | Dr-Linux: you realize that a TCP connection is not a "Unix Connection", right? |
| 13:39.33 | Dr-Linux | ManxPower: yes |
| 13:39.41 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:39.55 | ManxPower | so screwing with the manager connection will do nothing about that message. |
| 13:40.05 | [TK]D-Fender | sasch: go to www.asterisk.org and check out the downloads section for their HTTP server. You will find free MoH files there. |
| 13:40.19 | ManxPower | that message happens when a second asterisk instance connects to a running asterisk instance. |
| 13:41.12 | Dr-Linux | ManxPower: second asterisk? |
| 13:41.23 | ManxPower | YES! ARE YOU NOT LISTENING!!!!!!!! |
| 13:41.34 | sasch | <[TK]D-Fender> thanks !! |
| 13:41.57 | Dr-Linux | 1133 ? 00:00:00 safe_asterisk |
| 13:41.57 | Dr-Linux | <PROTECTED> |
| 13:42.05 | Dr-Linux | hhm.. yeah :) |
| 13:42.11 | *** join/#asterisk jtexter3 (n=jtexter3@adsl-70-234-146-58.dsl.tul2ok.sbcglobal.net) |
| 13:42.47 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.210) |
| 13:43.08 | jtexter3 | I'm trying to sort out some audio quality issues between 2 asterisk servers using IAX on 1.4.10. I have the output of "iax2 show netstats", but not sure what all of the columns mean |
| 13:44.15 | jtexter3 | ah, finally found the magic combo to get it to show up on Google |
| 13:44.32 | Dr-Linux | opss |
| 13:44.39 | Dr-Linux | ManxPower: thanks, i understand what is the issue |
| 13:45.48 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 13:45.53 | Dr-Linux | actually, i put an asterisk bootup starting script in /etc/rc.local and also i'm using init script for the same purpose |
| 13:46.44 | jkrueger | jtexter3: i could never find it on google. you are lucky. i gave up and just read the source code. crappy comments ftw |
| 13:48.07 | jtexter3 | jkrueger: http://www.voip-info.org/wiki/view/Asterisk+cli+iax2+show+netstats |
| 13:49.04 | jkrueger | jtexter3: this was like a year and a half ago |
| 13:49.05 | jkrueger | :) |
| 13:49.16 | jtexter3 | ah, makes sense then :D |
| 13:56.48 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 13:58.16 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-191-178.dhcp.embarqhsd.net) |
| 13:59.05 | Yoe | does asterisk have something that will call you when someone whom was previously on the hook is now available? |
| 13:59.38 | sasch | i reboot my system and zaptel don't start |
| 13:59.40 | sasch | http://pastebin.ca/666814 |
| 13:59.41 | sasch | why ?? |
| 14:00.16 | sasch | if I look my /dev/ i can found only zapctl not zap/ctl |
| 14:00.17 | Yoe | looks like some modules haven't been loaded |
| 14:00.39 | Yoe | oh, in that case you can just 'mkdir /dev/zap/; ln -s /dev/zapctl /dev/zap/ctl' |
| 14:00.54 | ManxPower | Yoe: that is pretty useless. |
| 14:01.10 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:01.13 | ManxPower | sasch: as always are the expected kernel modules listed in "lsmod" |
| 14:02.02 | tzafrir | on which system is it? which distro? what kernel? |
| 14:03.21 | [TK]D-Fender | sasch: Modprobe your card(s) again. |
| 14:03.34 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 14:04.02 | ManxPower | what sasch needs to do is learn how to make the modules load on boot for his distro |
| 14:04.30 | jarod14 | <PROTECTED> |
| 14:05.06 | sasch | <[TK]D-Fender> ok wait the system start up and i try it |
| 14:05.51 | ManxPower | sasch: the zaptel and card modules do not get magically loaded my the Kernel Module Fairy. You need to make the system load them on boot. |
| 14:06.03 | *** join/#asterisk zoa (n=k2@pirus.securax.be) |
| 14:06.05 | zoa | YOOOOOOOOOOOo |
| 14:06.11 | zoa | i finally reinstalled irc :) |
| 14:07.25 | sasch | http://pastebin.ca/666820 |
| 14:07.52 | sasch | I must ln -s /dev/zapchannel /dev/zap/channel ?? |
| 14:08.10 | ManxPower | sasch: no. |
| 14:08.15 | *** join/#asterisk lbow (n=lbow@41-195-71-105.access.uunet.co.za) |
| 14:08.18 | ManxPower | you must look at "lsmod" |
| 14:08.35 | ManxPower | the stuff under /dev is AUTOMATICALLY CREATED when the drivers load. |
| 14:09.02 | [zoa] | you shouldnt touch any of that stuff sasch |
| 14:09.09 | [zoa] | hey ho manxy |
| 14:09.22 | ManxPower | [zoa]: Hello! |
| 14:09.30 | [zoa] | i was not here for ages |
| 14:09.37 | sasch | ok ... i look this http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu |
| 14:09.48 | *** join/#asterisk zapp-branigan (n=zapp-bra@85.230.217.87.dynamic.jazztel.es) |
| 14:09.53 | sasch | and i found in poit 4 this Check /etc/udev/rules.d/udev.rules to see if the necessary |
| 14:10.28 | ManxPower | sasch: If you continue to ignore advice on this channel people will stop helping you. |
| 14:10.39 | ManxPower | now WHAT THE HELL IS THE OUTPUT OF "lsmod" |
| 14:10.42 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:10.44 | [zoa] | :) |
| 14:11.24 | *** join/#asterisk mog (i=mog@nat/digium/x-f69e3fef6aeec9f2) |
| 14:11.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:11.28 | ManxPower | [zoa]: as you can see nothing has changed. People are still using the Wiki to get wrong information, people still don't know how to make zaptel load at boot and people are still ignoring what they are told to do. |
| 14:11.31 | sasch | <ManxPower> i run lsmod and i found wctdm and zaptel |
| 14:11.52 | ManxPower | sasch: what exact card do you have? |
| 14:12.51 | *** join/#asterisk bkruse_home (n=kruz@thuroros.wca-hsv.org) |
| 14:14.08 | *** join/#asterisk zeppelin_ (n=zeppelin@189-30-1-246.paemt700.dsl.brasiltelecom.net.br) |
| 14:14.13 | ManxPower | I give up |
| 14:14.21 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:14.29 | sasch | <ManxPower> tdm400p |
| 14:14.32 | zapp-branigan | hi i have a problem loading cdr_addon_mysql.so loader.c:375 load_dynamic_module : Module cdr_addon_mysql.so did not register itself during load i have compiled the library in a cross compiler and asterisk work fine and i have compiled only the mysql client library who compiled ok and link without errors thanks for all |
| 14:14.40 | sasch | ok now all work |
| 14:14.41 | sasch | i add |
| 14:14.49 | sasch | 51-zaptel.rules |
| 14:14.52 | ManxPower | this is like dealing with support people in austrailia. ask a question, wait a day for an answer. |
| 14:14.54 | sasch | in /etc/udev/rules.d |
| 14:15.01 | sasch | and all work fine |
| 14:15.11 | sasch | but now if i run asterisk manually start |
| 14:15.20 | sasch | if i use init.d script don't run |
| 14:15.26 | sasch | where i can ound asterisk log ?? |
| 14:21.30 | *** join/#asterisk kwame (n=kwame@209.213.194.7) |
| 14:23.33 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:23.34 | Dr-Linux | anybody using cisco 7960? |
| 14:27.50 | sasch | but the blf and hint features is only in asterisk 1.4 ?? |
| 14:30.29 | ManxPower | sasch: it works just fine in 1.2 |
| 14:30.49 | ManxPower | SLA is only in 1.4 IIRC. |
| 14:31.51 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 14:32.15 | blitzrage | correct |
| 14:32.50 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:32.58 | ManxPower | I never could figure out what Service Level Agreements have to do with it. |
| 14:33.39 | [zoa] | :) |
| 14:37.38 | Mercestes | <PROTECTED> |
| 14:38.08 | sasch | <ManxPower> for hint function i must edit a conf file or is load on asterisk startup ?? |
| 14:38.52 | ManxPower | sasch: I cannot help you further |
| 14:39.12 | Seldon75 | Hi all, I have altered my Zapata.conf (http://georgef.hopto.org:8080/zapata.conf) to have different ports in G1 and G5; now I need to change my dialplan (http://georgef.hopto.org:8080/extensions.conf) to try to use trunks from G5 first, and then G1 once G5 is all used up. can someone please help me with this because I havent worked on dialplans for ages? |
| 14:39.13 | Mercestes | sasch: You have to turn on hinting or presence in your phone... |
| 14:39.22 | [TK]D-Fender | * doesn't HAVE real SLA. |
| 14:39.23 | Mercestes | sasch, for the asterisk side, you just add hint extensions pre the wiki |
| 14:39.25 | NOT_guru | yes Dr. Linux I have used 7940's and 7960's |
| 14:39.40 | ManxPower | Seldon75: a channel can be in more than one group. example group=1,5 |
| 14:39.45 | [TK]D-Fender | * 1.4's SLA is a marginally usably hack in only a few cases. |
| 14:39.55 | *** join/#asterisk mog (i=mog@nat/digium/x-48f49464da9dbb7c) |
| 14:39.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:40.05 | Seldon75 | ManxPower: ok, but how is that relevent |
| 14:40.46 | Mercestes | Seldon75, just use two dial lines, Zap/g1 then zap/g5 |
| 14:40.51 | Mercestes | Seldon75, or in your case, vice verse |
| 14:40.58 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 14:41.03 | [TK]D-Fender | Seldon75: Just dial G1 first, and the G5 immediately after. |
| 14:41.12 | [TK]D-Fender | Seldon75: Reverse taht, oops. |
| 14:41.22 | Seldon75 | can you please hint me the syntax |
| 14:42.01 | Seldon75 | e.g. in my dialplan I see: |
| 14:42.03 | Seldon75 | [trunks_local] |
| 14:42.04 | Seldon75 | exten => _8NXXNXXXXXX,1,Set(CALLERID(name)=${MAIN_CALLER_NM}) |
| 14:42.06 | Seldon75 | exten => _8NXXNXXXXXX,n,Set(CALLERID(num)=${MAIN_CALLER_ID}) |
| 14:42.07 | Seldon75 | exten => _8NXXNXXXXXX,n,Dial(${POTS}/${EXTEN:1}|${forward_bit}) |
| 14:42.09 | Seldon75 | exten => _8NXXNXXXXXX,n,Hangup() |
| 14:42.34 | Seldon75 | POTS = Zap/G1 |
| 14:43.02 | Seldon75 | whats the conditional syntax to use to detect that g1 (for example) is all busy |
| 14:44.14 | ManxPower | Seldon75: flood the channel again and we will kill you |
| 14:44.14 | Mercestes | Seldon75, Just dial g5 first, then g1. |
| 14:44.16 | ManxPower | ~pb |
| 14:44.16 | jbot | A Pastebin is a place to paste your stuff 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 |
| 14:44.24 | [TK]D-Fender | Seldon75: if it is anything except answered it will flow to the next priority. |
| 14:44.30 | Mercestes | IF you want me to feed you the exact syntax to type in, I require paypal. |
| 14:44.36 | [TK]D-Fender | Seldon75: so just shove them back to back. |
| 14:44.47 | Mercestes | [TK]D-Fender, that sounds so dirty |
| 14:44.53 | ManxPower | Mercestes: it is. |
| 14:45.11 | Mercestes | ManxPower, he should atleast take a shower break. =/ |
| 14:45.13 | ManxPower | make a new group, put all the channels in it |
| 14:45.29 | ManxPower | or dial the first group, check the value of dialstatus, then do your 2nd dial |
| 14:45.35 | [TK]D-Fender | no, front/front and front/back is where the action is ;) |
| 14:45.51 | Seldon75 | sorry to be dense but what do you mean by "back-to-back"? |
| 14:45.51 | ManxPower | otherwise if the called number is BUSY, the end Dial will call that busy number again. Same with anything except for answer. |
| 14:46.54 | Seldon75 | do you mean like this http://pastebin.ca/666858 |
| 14:47.25 | Mercestes | Seldon75, on yes, exactly like that. |
| 14:47.29 | Mercestes | Zap/g5 being a variable and all |
| 14:47.42 | Seldon75 | ok thanks all |
| 14:47.43 | Mercestes | Zap/g1 too. |
| 14:47.51 | Mercestes | Oh, your welcome. |
| 14:47.55 | Mercestes | Good luck. |
| 14:48.14 | Mercestes | Um...before you go.... |
| 14:48.17 | Seldon75 | yes? |
| 14:48.18 | Mercestes | Zap/g5 is *not* a variable. |
| 14:48.32 | Seldon75 | doesnt it refer to what I have in Zapata.conf? |
| 14:48.43 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:48.43 | Mercestes | Unless you happen to have a Set(Zap/g5="Zap/g5" somewhere. |
| 14:49.02 | Seldon75 | ie: Zap/gX refers to group 'X' as nominated in zapata? |
| 14:49.07 | Mercestes | Yes, but how is that relevent? |
| 14:49.07 | ManxPower | "POTS" is a variable "Zap/g5" is not a variable |
| 14:49.25 | Seldon75 | oh i see |
| 14:49.26 | Mercestes | In short, Loose the ${} around your zap commands. |
| 14:49.32 | Seldon75 | because I have it enclosed in ${} |
| 14:49.35 | Seldon75 | got you |
| 14:49.37 | Seldon75 | thx |
| 14:50.13 | Mercestes | np |
| 14:51.00 | Seldon75 | just to confirm; I can do this for ALL occurrences of "POTS" in my dialplan; just put the attempt to use G5 before it... |
| 14:51.20 | bertrand^ | incoming calls coming from my zaptel card have first MSD stripped |
| 14:51.39 | bertrand^ | not sure where i can chase the culprit: zapata.conf or extensions.conf? |
| 14:51.58 | ManxPower | bertrand^: usually it has to do with your PRI dialplan settings |
| 14:52.14 | [TK]D-Fender | bertrand^: You'd know if it were extensions.conf. It could be Zaptel, or it could be the way it comes from your telco |
| 14:52.40 | bertrand^ | it's 0 in extensions.conf, not set in zaptel |
| 14:53.17 | bertrand^ | maybe the telco, any way to sniff it? |
| 14:53.29 | ManxPower | pri debug span 1 |
| 14:54.22 | Seldon75 | another related question: both G1 and G5 have context=zap_incoming in Zapata.conf; (how) can I make incoming calls use G5 before G1 |
| 14:54.46 | *** join/#asterisk xFlux (n=xFlux@69.61.136.254) |
| 14:54.51 | bertrand^ | Presentation: Presentation allowed of network provided number (3) '663347974' ] |
| 14:55.11 | bertrand^ | first '0' missing, i guess the telco's the culprit then |
| 14:55.12 | [TK]D-Fender | bertrand^: Telco's fault then. Just add it yourself in your dialplan. |
| 14:55.15 | xFlux | does anyone have any articles or forum links that could describe the creation and use of a .call file? |
| 14:55.26 | ManxPower | so you do not have prodialplan= localprofix= nationalprfix=, etc |
| 14:55.32 | [TK]D-Fender | xFlux: Go check the WIKI for "asterisk auto dialout" |
| 14:55.38 | xFlux | im trying to create an automated monitoring script for my network, and need to play an alert to a phone when its triggered |
| 14:55.51 | ManxPower | bertrand^: put your /etc/asterisk/zapata.conf on pastebin.ca |
| 14:55.56 | xFlux | hehe, thanks [TK] |
| 14:56.18 | lbow | seldon75: for incoming calls your telco needs to arrange the order calls uses your spans |
| 14:57.15 | Seldon75 | lbow: got you; thx |
| 14:57.45 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 14:58.41 | *** join/#asterisk Renacor (n=kvirc@62.157.211.194) |
| 14:58.46 | bertrand^ | ManxPower, http://pastebin.ca/666874 |
| 14:58.47 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:59.02 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:59.27 | flujan | hi all. :D |
| 14:59.46 | Renacor | k so I set a variable like INTLINE=SIP/internal_phone and try to use it in the same context with exten => 222,1,Dial(${INTLINE}) but the var is empty when I look in the console |
| 15:00.54 | ManxPower | Renacor: then you are not setting it correctly. |
| 15:01.07 | Renacor | how am I doing it wrong? |
| 15:01.22 | ManxPower | you CANNOT do INTLINE=SIP/internal_phone ANYWHERE except for [globals] |
| 15:01.35 | ManxPower | you can Set(INTLINE=SIP/internal_phone) most anywhere. |
| 15:01.41 | ManxPower | as part of an exten => line, of course. |
| 15:01.51 | [TK]D-Fender | Renacor: And clarify what you mean "at console". Pastebin your dialplan |
| 15:01.53 | [TK]D-Fender | ~pb |
| 15:01.53 | jbot | A Pastebin is a place to paste your stuff 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 |
| 15:01.53 | Renacor | it should work without that also |
| 15:01.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 15:01.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 15:02.02 | Renacor | [TK]D-Fender: asterisk -r |
| 15:02.18 | [TK]D-Fender | Renacor: SHOW us. Dialplan AND its execution |
| 15:02.27 | Renacor | sure one sec |
| 15:02.31 | [TK]D-Fender | ManxPower: And that wouldn't be a VARIABLE :) |
| 15:02.43 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:03.36 | Renacor | http://pastebin.ca/666877 |
| 15:03.40 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:05.21 | bertrand^ | [TK]D-Fender, you mean 'nationalprefix = 0' in zapata.conf? |
| 15:05.23 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 15:06.31 | [TK]D-Fender | bertrand^: No, that wasn't what I said at all. |
| 15:06.40 | Renacor | [TK]D-Fender: It works when I define the variable in the [global] context |
| 15:07.30 | [TK]D-Fender | Renacor: PLSINT=SIP/arndt_int <- NOT how to declare a variable. That is how you would declare a CONTANT under [globals] You cannot make contants within any other context. |
| 15:07.48 | xFlux | [TK]: bare with me, I just started reading up on this last night, but are there utils to convert a wav to a msg? |
| 15:07.58 | Renacor | k thanks now I understand |
| 15:08.04 | [TK]D-Fender | xFlux: and what is a "msg"? |
| 15:08.40 | xFlux | im just reading this example configuration, and I see the outbound message is played using exten => 2051,4,Playback(outboundmsgs/msg1) |
| 15:09.15 | xFlux | unless that is a raw wav? |
| 15:09.29 | zpertee | is there a way to have asterisk recognize when someone starts talking |
| 15:10.03 | coppice | outsource it to cheap labour |
| 15:10.46 | xFlux | ....cheap labor is why im trying to make a system to begin with, good thought though :) |
| 15:11.19 | [TK]D-Fender | xFlux: * supports many sound formats. WAV should be in 8khz mono. MP3 is supported with the asterisk-addons package as well |
| 15:11.30 | xFlux | ahhh nice |
| 15:11.32 | [TK]D-Fender | xFlux: You can use "sox" to convert files as needed to a compatible format |
| 15:11.48 | xFlux | you guys are making me all tingly inside |
| 15:11.51 | xFlux | fun stuff |
| 15:12.04 | [TK]D-Fender | zpertee: WHEN do you expect to do this "detection", and to what end? |
| 15:12.51 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 15:12.52 | a1fa | hi |
| 15:13.05 | xFlux | so off the subject kind of, have any of you ever messed with an open source speech synthesis that doesn't sound like a 1950's Robot? |
| 15:13.07 | a1fa | i am getting strange behaviour on one of my linksys phones |
| 15:13.13 | a1fa | they can hear me, but i cant hear them |
| 15:13.14 | a1fa | no nat |
| 15:13.19 | a1fa | [Aug 22 15:10:45] WARNING[21394]: chan_sip.c:5653 set_destination: Can't find address for host '450' |
| 15:13.47 | [TK]D-Fender | a1fa: pastebin your sip.conf. I'm betting you set a fixed IP to talk BACK to where it should be dynamic. |
| 15:14.36 | a1fa | host=dynamic |
| 15:14.45 | a1fa | i have nat=yes, just in case |
| 15:14.56 | [TK]D-Fender | a1fa: pastebin a call with SIP debug, verbose 10, and your phone entries |
| 15:14.59 | ManxPower | and you have a [450] entry in sip.conf? |
| 15:15.08 | a1fa | ManxPower : of course |
| 15:18.22 | zpertee | [TK]D-Fender: never mind figured out what I needed |
| 15:18.45 | a1fa | data is almost ready |
| 15:19.08 | a1fa | http://www.pastebin.ca/666896 |
| 15:20.56 | [TK]D-Fender | a1fa: * has a public & pricate IP, and SPA = local to * on LAN? |
| 15:21.09 | [TK]D-Fender | private* |
| 15:21.11 | a1fa | * public IP, and SPA = public IP |
| 15:21.14 | a1fa | via PPPOE |
| 15:21.54 | a1fa | same bahvious with nat=no |
| 15:22.00 | [TK]D-Fender | a1fa: each is unique? Could be that its trying to reinvite and doesn't like it. set "canreinvite=no for [general] and all your entries and test |
| 15:22.10 | [TK]D-Fender | a1fa: reinvites = evil |
| 15:22.17 | *** join/#asterisk BrokenNoze (n=root@62.253.194.107) |
| 15:22.37 | a1fa | thats set |
| 15:23.04 | a1fa | in general and [450] |
| 15:23.28 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 15:23.50 | BrokenNoze | Hi, I'm having problems detecting DTMF with asterisk 1.2 and polycom handsets, is anyone aware of any other settings I can try other than dtmfmode= ? |
| 15:23.52 | [TK]D-Fender | a1fa: pastebin configs & the DIALS <------- |
| 15:24.08 | [TK]D-Fender | a1fa: You always leave stuff out from pastebins we request. |
| 15:24.40 | [TK]D-Fender | BrokenNoze: Describe the scenario. dtmfmode=rfc2833 should be it. |
| 15:24.46 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:25.15 | a1fa | hehe |
| 15:25.16 | a1fa | hold up |
| 15:25.18 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:25.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:25.29 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 15:26.34 | a1fa | http://www.pastebin.ca/666903 |
| 15:26.37 | Renacor | would _499X. match 499222 ? |
| 15:27.01 | BrokenNoze | Fender: I'm attempting to make an outbound call to a remote number , though asterisk. from a polycom600 . for the sake of testing simply a bank number with a simple DTMF auto attendant. I've tried rfc2833, info, inband, auto.. none of them work |
| 15:27.04 | JunK-Y | it should. |
| 15:28.02 | BrokenNoze | perhaps the gateway is incorrectly configured? would it make a difference? |
| 15:28.47 | [TK]D-Fender | BrokenNoze: Forget outbound, that may be its OWN problem. veryfy between the phone & * first |
| 15:28.51 | BrokenNoze | sorry, I should have added asterisk passes the call out to a Vegastream gateway and onto the TDM network over ISDN |
| 15:28.59 | BrokenNoze | ok |
| 15:29.37 | [TK]D-Fender | a1fa: * version? And you still didn't show the DIAL <-- |
| 15:30.23 | a1fa | [TK]D-Fender : 1.4.10 |
| 15:30.25 | *** join/#asterisk jhiver (n=jhiver@161-242.206-83.static-ip.oleane.fr) |
| 15:30.31 | a1fa | [TK]D-Fender : BT102 phone works fine |
| 15:30.42 | [TK]D-Fender | a1fa: First, you should upgrade, and then change to type=peer |
| 15:30.47 | a1fa | [TK]D-Fender : dial is Dial(SIP/450) |
| 15:30.57 | jhiver | hey all, i'm trying to set up an IAX2 trunk however I get Rejected connect attempt from 88.191.37.213, who was trying to reach '1234@', any ideas what it could be? |
| 15:31.07 | a1fa | 1.4.10.1 |
| 15:31.12 | [TK]D-Fender | a1fa: and "sip show peer 450 shows....? PB |
| 15:31.27 | [TK]D-Fender | jhiver: Yeah, a CONTEXT wasn't specified. |
| 15:31.38 | jhiver | let me pastebin the relevant section |
| 15:31.39 | a1fa | ok hold on |
| 15:31.40 | [TK]D-Fender | a1fa: UPGRADE <- |
| 15:31.53 | [TK]D-Fender | a1fa: and dits "usernam=" |
| 15:32.17 | jhiver | http://pastebin.ca/666912 |
| 15:32.20 | [TK]D-Fender | ditch |
| 15:32.22 | jhiver | there is a context |
| 15:32.36 | jhiver | this section of the conf is on the receiving end |
| 15:32.41 | [TK]D-Fender | jhiver: the inbound DIAL needs one. |
| 15:33.01 | a1fa | upgrade to what? |
| 15:33.13 | jhiver | surely you mean outbound? |
| 15:33.23 | a1fa | eeek 1.4.11 is out |
| 15:33.25 | a1fa | eeek |
| 15:33.32 | jhiver | exten => _X.,1,Dial(IAX2/etang-sale/${EXTEN}) |
| 15:33.32 | [TK]D-Fender | a1fa: Read the damn channel topic, will you! |
| 15:33.37 | a1fa | :P |
| 15:33.41 | a1fa | bitchx sucks |
| 15:33.42 | jhiver | is what i do to dial the trunk |
| 15:33.47 | a1fa | brb |
| 15:33.49 | jhiver | any problem with this syntax? |
| 15:33.50 | a1fa | quick upgrade |
| 15:34.06 | [TK]D-Fender | exten => _X.,1,Dial(IAX2/etang-sale/${EXTEN}@route) <- Could I have been any MORE explicit? |
| 15:34.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 15:34.44 | jhiver | ok but... but... then why define 'context' in the receiving end IAX2 section? I'm not sure to understand the logic of this... |
| 15:34.54 | jhiver | thanks for the tip anyway, i'm going to give it a shot |
| 15:34.56 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 15:35.38 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 15:35.39 | ManxPower | you put the context in the receiving end so you don't have to specify the context on the Dial line of the sending system |
| 15:35.54 | jhiver | <PROTECTED> |
| 15:36.13 | jhiver | same error |
| 15:36.25 | jhiver | and i did define context=route on the receiving end |
| 15:36.42 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:36.49 | *** join/#asterisk aikanaro79 (n=noone@89.181.182.127) |
| 15:36.53 | [TK]D-Fender | jhiver: where on the RECEIVING end is the call supposed to go? |
| 15:37.01 | [TK]D-Fender | jhiver: Thats the context you need to provide |
| 15:37.05 | jhiver | in the [route] context |
| 15:37.18 | jhiver | where i do an Answer() and then Echo() for testing purposes |
| 15:37.20 | [TK]D-Fender | jhiver: pastebin BOTH sides clearly please |
| 15:37.25 | jhiver | ok |
| 15:38.16 | aikanaro79 | using app_conference is there a way to establish a conference with another user directly? i.e. without him having to dial the conference room...I'd call him and the call would be a conference call |
| 15:38.40 | *** join/#asterisk HarryR (n=Harry@77.240.56.17) |
| 15:39.25 | jhiver | http://pastebin.ca/666922 |
| 15:39.32 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:39.42 | kippi | <PROTECTED> |
| 15:41.45 | jhiver | [TK]D-Fender, what do you think? |
| 15:43.50 | [TK]D-Fender | aikanaro79: How many people are we talking about? |
| 15:44.06 | aikanaro79 | 2+MORE |
| 15:44.17 | aikanaro79 | sorry...I meant me+MORE |
| 15:44.18 | [TK]D-Fender | aikanaro79: Call him and transfer him to the conference |
| 15:44.50 | aikanaro79 | [TK]D-Fender: so I dial him and then add him to the conference is that it? |
| 15:45.15 | *** join/#asterisk Sweeper (i=sweeper@softcheese.net) |
| 15:45.25 | [TK]D-Fender | aikanaro79: transfer him to your conference exten and thats it |
| 15:45.46 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:45.46 | aikanaro79 | [TK]D-Fender: thanks |
| 15:45.56 | [TK]D-Fender | jhiver: First, you have 2 PEERS set up, where you should have a USER on the receiving end. Next the NAMES don't match. |
| 15:46.36 | jhiver | ok, so it's not the same as with SIP where a peer can be defined to send and receive? |
| 15:46.44 | [TK]D-Fender | aikanaro79: Or if you're already loggend into the conference you start a 3-way from your phone to drag him in |
| 15:47.01 | Sweeper | [TK]D-Fender: should qualify be literally set to qualify=xxx, or should I just keep decrementing until the phones stay alive? |
| 15:47.47 | [TK]D-Fender | Sweeper: typically start with "yes", and work your way UP as needed. |
| 15:47.57 | Sweeper | o.O |
| 15:48.06 | [TK]D-Fender | Sweeper: IF you REDUCE your system with waste a lot of time and bandwidth for nothing |
| 15:48.27 | jhiver | and also, why should the name match? [etang-sale] is the name of the receiving end, [dedibox] the name of the sending end |
| 15:48.29 | [TK]D-Fender | Sweeper: And device needing to respond that fast is just nuts |
| 15:48.35 | Sweeper | [TK]D-Fender: but it doesn't work at 2s :/ |
| 15:48.42 | jhiver | that makes sense (well it does to me =) |
| 15:49.04 | [TK]D-Fender | jihiYou are dialing a remote end that doesnt' KNOW YOU. You have no account to match what you're dialing out with |
| 15:49.22 | [TK]D-Fender | Sweeper: what kind of "doesn't work"? |
| 15:49.28 | jhiver | how do you mean it doesn't know me? that's why i specify host=<ip_address> |
| 15:49.47 | jhiver | it (can) work that way with SIP, isn't it the same with IAX |
| 15:49.48 | jhiver | ? |
| 15:50.06 | [TK]D-Fender | jhiver: You need to be AUTHORIZED on the remote system and you don't have a MATCHING account and you are getting REFUSED. |
| 15:50.16 | Sweeper | [TK]D-Fender: 206.176.131.27 D N 53493 UNREACHABLE <-- that kind |
| 15:50.30 | jhiver | ok so you're telling me that you cannot do ip based auth with IAX |
| 15:50.38 | jhiver | fine |
| 15:50.46 | *** join/#asterisk ccvp (n=axe@66.0.46.210) |
| 15:50.59 | [TK]D-Fender | jhiver: "Hi, I'm Joe and want to talk to 123!" , "I don't know any #$^ing Joe, get lost!" |
| 15:51.09 | a1fa | [TK]D-Fender : i upgraded.. same issue :p |
| 15:51.14 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 15:51.20 | aikanaro79 | [TK]D-Fender: call him and transfer him means that my dialplan will have a priority with dial followed by another with conference right? |
| 15:51.24 | [TK]D-Fender | a1fa: show me the rest. |
| 15:51.26 | jhiver | "Hi, I'm joe my ip address is FOO and i want to talk to 123" |
| 15:51.35 | a1fa | [Aug 22 15:49:45] WARNING[31965]: chan_sip.c:5674 set_destination: Can't find address for host '450' |
| 15:51.36 | jhiver | "aha, i know ip address FOO" |
| 15:51.42 | jhiver | "-> authorized" =) |
| 15:51.51 | [TK]D-Fender | aikanaro79: mean whatever exten you use to dial the conference. This has nothing to do with the DIAL dialplan app. |
| 15:52.13 | jhiver | anyway, all i'm saying is that i'm surprised it's not consistent with SIP config, i'll do some research and try to do it the way you describe |
| 15:52.14 | *** join/#asterisk mog (i=mog@nat/digium/x-7a5eebf27e39a612) |
| 15:52.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:52.29 | *** part/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 15:52.31 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:52.58 | a1fa | [TK]D-Fender : still same issue |
| 15:53.12 | [TK]D-Fender | a1fa: ............ |
| 15:53.17 | aikanaro79 | [TK]D-Fender: ok...but with the conference app I can't include a destination to call...I can only configure a conference room |
| 15:53.27 | aikanaro79 | sorry for the mess I'm making |
| 15:53.40 | a1fa | [TK]D-Fender : you wanted to see sip show peer, right? |
| 15:53.42 | [TK]D-Fender | aikanaro79: He is already a channel on your system. Transfer him to the EXTEN withere you call Confernce! |
| 15:53.48 | [TK]D-Fender | aikanaro79: There is no "Dial"! |
| 15:54.43 | a1fa | [TK]D-Fender : http://www.pastebin.ca/666936 |
| 15:54.54 | Sweeper | [TK]D-Fender: any ideas on the NAT? |
| 15:55.21 | datachomper | Has anybody ever made a "soundboard" type application, where you can connect to a current channel and drop sounds into it by clicking on buttons and whatnot? |
| 15:55.22 | [TK]D-Fender | Sweeper: pastebin the phone's entry |
| 15:55.27 | aikanaro79 | [TK]D-Fender: right...I got the dial part...but what I mean is I call him (I dial an extension that supposedly gets me a channel to him) but nothing happens on the other side |
| 15:55.38 | aikanaro79 | only if I dial the same extension do I have both users in the same conference |
| 15:55.47 | [TK]D-Fender | aikanaro79: Dial him like NORMAL with your PHONE. |
| 15:56.04 | Sweeper | [TK]D-Fender: http://pastebin.ca/666937 |
| 15:56.27 | [TK]D-Fender | aikanaro79: Then AFTER he answers and you are TALKING, you TRANSFER his ass to the same exten you would use to log into the conference. AFTER that go join the conference YOURSELF. |
| 15:57.48 | a1fa | [TK]D-Fender : hey.. that phone can intitate connections |
| 15:57.56 | [TK]D-Fender | Sweeper: qualify is for the phone's entry, not [general]. Second, after reloading you need to make the phone RERIGSTER to initiate the new qualify because * doesn't know WHERE to qualify too right now because he has ALREADY dropped off the map. |
| 15:58.21 | a1fa | [TK]D-Fender : i hear audio when user calls me |
| 15:58.32 | Sweeper | [TK]D-Fender: I'll retry, I suppose, but the entries originally had qualify=yes in the individual entries |
| 15:58.33 | [TK]D-Fender | a1fa: that is SCREAMING NAT you know.... |
| 15:59.29 | a1fa | yes |
| 15:59.35 | a1fa | nat=yes |
| 16:00.02 | [TK]D-Fender | a1fa: You said he was PUBLIC |
| 16:00.02 | tzanger | will SER proxy video calls too? |
| 16:00.12 | [TK]D-Fender | tzanger: RTP = RTP :) |
| 16:00.21 | a1fa | [TK]D-Fender : he is |
| 16:00.23 | [TK]D-Fender | tzanger: I'm pretty sure it can... |
| 16:00.23 | tzanger | [TK]D-Fender: fair enough :-) |
| 16:00.25 | a1fa | [TK]D-Fender : via PPOE |
| 16:00.46 | a1fa | [TK]D-Fender : the phone is attached via PPPOE |
| 16:00.52 | [TK]D-Fender | a1fa: And you're 100% sure his modem isn't giving the phone a private address? |
| 16:00.57 | a1fa | yes |
| 16:01.03 | a1fa | because i can connect to his phone |
| 16:01.07 | a1fa | on his public IP |
| 16:01.17 | Sweeper | [TK]D-Fender: there's something interfering with calls routing through.... |
| 16:01.39 | Sweeper | [Aug 22 11:00:47] NOTICE[31001]: chan_sip.c:15418 sip_poke_noanswer: Peer '110' is now UNREACHABLE! Last qualify: 17 <-- like so, when I dial |
| 16:01.40 | [TK]D-Fender | Sweeper: if they drop off, you don't want to REDUCE the qualify. Set to 4000 |
| 16:01.47 | Sweeper | ok |
| 16:02.03 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 16:02.03 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 16:02.21 | [TK]D-Fender | Sweeper: And you could be experiencing inconvenient packet loss. as in only the qualify ones... |
| 16:02.30 | [TK]D-Fender | Sweeper: Or at the moment. |
| 16:03.00 | [TK]D-Fender | a1fa: and you double checked the phone's stats? |
| 16:03.05 | a1fa | [TK]D-Fender : yup |
| 16:03.13 | [TK]D-Fender | a1fa: ok, not sure then... |
| 16:03.31 | a1fa | nat=yes, type=friend or peer |
| 16:03.34 | a1fa | it doesnt matter |
| 16:03.45 | a1fa | this used to work before the fall-out |
| 16:03.47 | a1fa | :( |
| 16:03.53 | Sweeper | [TK]D-Fender: and also the pokes? |
| 16:03.54 | a1fa | his settings did not change at all |
| 16:04.29 | [TK]D-Fender | a1fa: get him to re-register and see if anything looks funny |
| 16:04.50 | [TK]D-Fender | Sweeper: when 1 fails, it drops out and has nothing to retry |
| 16:05.12 | jhiver | ok... now i have both boxes registered with each other, and i still get a rejected connect attempt error |
| 16:05.16 | [TK]D-Fender | Sweeper: until the phone re-registers. You could try changing the reg frequency as well |
| 16:05.27 | jhiver | should i pastebin my new config files to see wether they make any sense? |
| 16:05.29 | [TK]D-Fender | jhiver: PASTEBIN |
| 16:05.34 | jhiver | ok |
| 16:05.57 | [TK]D-Fender | jhiver: Of course you should, we're neither psyhcih (except on Tuesdays), nor trusting here, with good reason! |
| 16:06.09 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 16:06.09 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 16:06.23 | [TK]D-Fender | a1fa: And I told you to drop the "username=", did you? |
| 16:06.52 | a1fa | no |
| 16:07.35 | a1fa | i did now |
| 16:07.45 | a1fa | [Aug 22 16:07:35] WARNING[31956]: chan_sip.c:5674 set_destination: Can't find address for host '450' |
| 16:07.58 | jhiver | [TK]D-Fender: http://pastebin.ca/666950 |
| 16:08.40 | a1fa | other people have this problem |
| 16:09.06 | jhiver | now both boxes are registred with each other |
| 16:09.11 | jhiver | and they are both 'friends' |
| 16:09.49 | a1fa | and google ha snothing on this |
| 16:10.10 | jhiver | my dialstring on the sending box is exten => _X.,1,Dial(IAX2/etang-sale/${EXTEN}@route) |
| 16:10.18 | [TK]D-Fender | a1fa: have him re-reg |
| 16:10.38 | a1fa | power on-off? |
| 16:11.48 | [TK]D-Fender | jhiver: You have missed the point. You are using a peer named [etang-sale]. There is no [etang-sale] on the receiving end! |
| 16:11.56 | [TK]D-Fender | a1fa: That'd do |
| 16:12.10 | [TK]D-Fender | jhiver: And you have NO need for registering |
| 16:12.18 | [TK]D-Fender | jhiver: You are running fixed IP's |
| 16:12.29 | [TK]D-Fender | jhiver: No? |
| 16:12.37 | jhiver | but that is just plain silly! why should the RECEIVING and then SENDING box share the same name? |
| 16:12.43 | jhiver | i really don't understand! |
| 16:13.26 | *** join/#asterisk davixx (n=davixx@ASt-Lambert-151-1-63-4.w83-199.abo.wanadoo.fr) |
| 16:13.27 | jhiver | one is called etang-sale, the other is called dedibox, so what should this 'friend' be called? 'etang-sale-dedibox'? |
| 16:13.47 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.169.196) |
| 16:13.51 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 16:13.57 | jhiver | and share this silly name with both boxes? |
| 16:14.07 | [TK]D-Fender | your SENDING end says "Hi, I'm etang-sale", The receiving side says "I don't know any etang-sale, I only know dedibox! Get lost! |
| 16:14.41 | jhiver | ok |
| 16:14.45 | jhiver | so it's silly =) |
| 16:14.51 | jhiver | then i will put the same name |
| 16:14.55 | a1fa | same issue on power/on/off |
| 16:15.02 | [TK]D-Fender | jhiver: No, its not silly, YOU are :p |
| 16:15.12 | jhiver | no, it's totally dumb |
| 16:15.33 | [TK]D-Fender | jhiver: You don't set up an account and then have an NON-MATCHING entry to receive it! |
| 16:15.52 | jhiver | dedibox calls etang sale, so it should say 'hi, i'm dedibox', but than Dial() something called etang-sale |
| 16:15.54 | *** part/#asterisk xFlux (n=xFlux@69.61.136.254) |
| 16:15.58 | jhiver | don't you think? |
| 16:16.03 | [TK]D-Fender | jhiver: You don't tell one side to say "Hi I'm Joe", and only tell the receiving end about BOB <--------- |
| 16:16.23 | *** part/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:16.35 | jhiver | doesn't work that way with SIP =) |
| 16:16.46 | jhiver | but i understood already it's totally different |
| 16:16.47 | jhiver | look |
| 16:16.57 | jhiver | i'll define dedibox-etang-sale |
| 16:17.01 | jhiver | and see how it goes |
| 16:17.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:17.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:17.33 | [TK]D-Fender | jhiver: the name has to match, EXACTLY |
| 16:18.03 | jhiver | yes that's why i say it's silly... imagine you have a customer, call him 'bob', and you define him as a 'friend' |
| 16:18.07 | jhiver | when he dials you |
| 16:18.09 | [TK]D-Fender | jhiver: Don't invent different shit that lokks CLOSER. I you say you're "Joe" there had better be a "Joe" entry on the other side. |
| 16:18.12 | jhiver | he's got to dial... 'bob' ! |
| 16:18.29 | jhiver | and you say that's not silly! |
| 16:18.32 | jhiver | but it is! |
| 16:18.34 | jhiver | very much! |
| 16:18.41 | [TK]D-Fender | jhiver: [etang-sale] doesn't mean SHIT to the other side. What patr don't you get? |
| 16:19.00 | jhiver | if i do ip based auth, it should |
| 16:19.01 | jhiver | anyway |
| 16:19.10 | [TK]D-Fender | jhiver: guess what, usernames matter! |
| 16:19.11 | jhiver | i don't care as long as i get it to work =) |
| 16:19.19 | jhiver | so i'll do what you recommend |
| 16:19.40 | [TK]D-Fender | jhiver: You're problem is you've got this "usernames don't matter, I think it shoulod only be IP" part stuck in your head :p |
| 16:19.52 | jhiver | no, no |
| 16:20.16 | jhiver | with sip i can define a username called [foo] and on the otherside define a provider called [bar] |
| 16:20.24 | jhiver | then foo can dial bar |
| 16:20.26 | jhiver | and it works |
| 16:20.31 | jhiver | and it's perfecly logical |
| 16:20.36 | *** join/#asterisk chino (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 16:20.47 | rickross | hi all, we have enable g722 on our asterisk 1.4.1.10 installation, and now we cannot record voicemail message - any ideas? |
| 16:20.55 | chino | where is the askterisk wiki site ? |
| 16:21.00 | [TK]D-Fender | jhiver: only if [foo] does "username=bar" to OVERRIDE the fact he'd identify himself as "foo" otherwise! |
| 16:21.18 | rickross | voicemail recording was working before we allowed g722 |
| 16:21.32 | [TK]D-Fender | jhiver: Otherwise [foo] will call the other system, and [bar] won't care. |
| 16:21.42 | rickross | it gets to the point where it says "record your message" then hangs up |
| 16:22.21 | rickross | <PROTECTED> |
| 16:22.22 | rickross | <PROTECTED> |
| 16:22.22 | rickross | <PROTECTED> |
| 16:22.29 | rickross | and it hangs up |
| 16:23.11 | jhiver | [TK]D-Fender, i see |
| 16:23.45 | jhiver | so on dedibox side inside [etang-sale] i can have username=dedibox, say |
| 16:23.57 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 16:24.01 | jhiver | and make that a 'peer' |
| 16:24.20 | jhiver | and on etang sale i can have [dedibox] with username = dedibox, and make that a 'user' |
| 16:24.33 | jhiver | and pehaps then it will work :-/ |
| 16:24.50 | jhiver | do i still need to have username / pass or is host=<ip_address> enough? |
| 16:25.08 | [TK]D-Fender | jhiver: Well you wouldn't need the 2nd one, because the [] matches what you WANT it to be |
| 16:25.22 | jhiver | ok |
| 16:25.24 | *** join/#asterisk hydra__ (n=hydra@bl8-165-19.dsl.telepac.pt) |
| 16:25.25 | jhiver | that's great |
| 16:25.27 | jhiver | i'll do this |
| 16:25.38 | [TK]D-Fender | jhiver: put the secret, on both ends. |
| 16:25.41 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:25.44 | [TK]D-Fender | jhiver: And forget the host. |
| 16:25.58 | [TK]D-Fender | jhiver: just have "host=dynamic" on the receiving end. |
| 16:26.00 | hydra__ | Someone uses iptables to nat sip clients? |
| 16:26.22 | hydra__ | I'm having really weird issues |
| 16:26.29 | jhiver | well, then i do need a login / password |
| 16:26.48 | jhiver | otherwise anybody will be able to make calls by just pretending they are 'dedibox' |
| 16:27.31 | *** part/#asterisk chino (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 16:28.11 | jhiver | ok, KEWL, it works with host=<ip_address> |
| 16:28.15 | [TK]D-Fender | jhiver: Ditch your registeres, they aren't needed yet. This is a 1-way link right now. |
| 16:28.28 | jhiver | and no password whatsoever |
| 16:28.42 | [TK]D-Fender | jhiver: host w/o pass should work. |
| 16:28.48 | jhiver | cool |
| 16:28.50 | [TK]D-Fender | jhiver: as long as the name matches |
| 16:28.55 | jhiver | great |
| 16:28.58 | jhiver | thanks for your help |
| 16:28.59 | [TK]D-Fender | jhiver: Which was ALWAYS your prolem :p |
| 16:29.04 | [TK]D-Fender | heh |
| 16:29.11 | [TK]D-Fender | jhiver: Ok, you're on your way.... |
| 16:29.27 | jhiver | yes i'm trying to save some bandwith with IAX2 trunking |
| 16:29.36 | jhiver | i set up both systems today |
| 16:29.42 | jhiver | and have trunkfreq=40 |
| 16:29.59 | jhiver | since sometimes i have some 40ms ptimes (on some calls) |
| 16:31.14 | datachomper | Wierd, the iax channel on one of my asterisk boxes stopped trunking upstream for about 60seconds, throwing up this message : chan_iax2.c:1885 __attempt_transmit: Max retries exceeded to host <ip address> |
| 16:31.24 | jhiver | echo test works fine, it's all setup, thank you |
| 16:32.48 | [TK]D-Fender | jhiver: You are missing 1 thing though. |
| 16:33.03 | [TK]D-Fender | jhiver: Oh, there it is... never mind :) |
| 16:33.10 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:33.11 | [TK]D-Fender | jhiver: Got Zaptel installed, right? |
| 16:37.56 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 16:38.49 | *** part/#asterisk kwame (n=kwame@209.213.194.7) |
| 16:41.13 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-94-138.dsl.peoril.sbcglobal.net) |
| 16:41.42 | MACscr | Anyone have any ideas why a server would completely crash when i try to start asterisk? |
| 16:41.46 | MACscr | It was working fine |
| 16:43.20 | [TK]D-Fender | MACscr: How about doing something completely silly like running as "asterisk -gvvvvc" and SHOWING us! :) |
| 16:44.23 | MACscr | Well, because i didnt know about running that. I always just do: service asterisk start |
| 16:44.45 | MACscr | So now that i know that, i will try that |
| 16:44.57 | Yoe | can I somehow know where a call is coming from? |
| 16:45.36 | [TK]D-Fender | Yoe: "a call"? Wow, can you be a little less specific please... the deatil is over-whelming... |
| 16:46.23 | Yoe | sorry. There's ${EXTEN} to know who I'm trying to reach, but I'd like to know in my dialplan who the other end is |
| 16:46.28 | syzygyBSD | when my polycom 501 starts, it says "could not contact boot server, using existing configuration" is that normal if I don't specify a boot server in setup, but only with dhcp? |
| 16:47.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@64.149.29.208) |
| 16:47.12 | [TK]D-Fender | Yoe: First guess, the CHANNEL NAME, second, the CONTEXT the call is coming in from, 3rd : CALLER ID <----------- |
| 16:47.33 | [TK]D-Fender | syzygyBSD: dhcp=irrelevent, no bott server = relevant |
| 16:47.37 | [TK]D-Fender | boot* |
| 16:47.48 | Strom_M | irrelevant* |
| 16:48.16 | syzygyBSD | can't I specify what the boot server is with dhcp? |
| 16:48.25 | Yoe | syzygyBSD: next-server in dhcpd.conf |
| 16:48.33 | [TK]D-Fender | Strom_M: if I got 1/2 right, then its already corrected. I fixed the worst of the 2 blunders :p |
| 16:48.41 | syzygyBSD | ya, I have it, works great for all my computers |
| 16:48.56 | [TK]D-Fender | syzygyBSD: sure you can. |
| 16:49.07 | Yoe | hmm, but my callerid isn't being set for some obscure reason |
| 16:49.13 | Yoe | oh well, sod it, I'll look at it tomorrow |
| 16:49.31 | [TK]D-Fender | syzygyBSD: option tftp-server-name "iphere"; |
| 16:50.04 | [TK]D-Fender | syzygyBSD: note this sets the IP, but not the TYPE. That has to match the phone. |
| 16:50.59 | Strom_M | je ne sais boner |
| 16:52.32 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 16:52.32 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 16:52.59 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.77) |
| 16:53.39 | jhiver | are there any tricks to do cyclic load balancing with asterisk (load balance across multiple gateways) |
| 16:53.53 | Daejeo1 | guys!!!!! is it ok to go with "Polycom SoundPoint IP 400"? |
| 16:53.58 | [TK]D-Fender | Daejeo1: NO |
| 16:54.02 | jhiver | i know i could use random() but it's not quite as nice as cyclic load balancing |
| 16:54.10 | [TK]D-Fender | Daejeo1: Ancient model that supports only OLD firmware/. |
| 16:54.18 | Daejeo1 | oh |
| 16:54.35 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 16:54.46 | Daejeo1 | someone is selling it for 49$ on ebay |
| 16:58.34 | hmmhesays | there is about 100 ways you could load balance across gateways |
| 16:58.36 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 16:59.29 | jhiver | hmmhesays, i'm listening to your science =) |
| 16:59.38 | jhiver | i used to use Random() |
| 16:59.52 | jhiver | but i'd prefer to have a cyclic ascending algorithm |
| 17:03.06 | JunK-Y | jhiver: use RAND() :) |
| 17:03.49 | jhiver | heh |
| 17:03.53 | jhiver | that's what i do already |
| 17:03.56 | jhiver | not so nice =) |
| 17:04.14 | jhiver | or what would be even better would be to load balance on least utilization |
| 17:06.02 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:06.17 | [TK]D-Fender | jhiver: then you need to count usage across the points to distribute |
| 17:06.47 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 17:07.13 | Daejeo1 | [TK]D-Fender:how about " Polycom SoundPoint IP 600". is it ok to buy? |
| 17:07.48 | [TK]D-Fender | Daejeo1: While I normally stick with CURRENT models, the 600 is at least workable |
| 17:07.53 | [TK]D-Fender | Daejeo1: ho much? |
| 17:07.56 | [TK]D-Fender | how* |
| 17:08.14 | [TK]D-Fender | Daejeo1: price makes the choice. |
| 17:08.41 | Daejeo1 | i am bidding on it. current price 85$ +15$ shipping |
| 17:08.44 | [TK]D-Fender | Daejeo1: Because the 600 can't take the expansion model |
| 17:09.05 | [TK]D-Fender | Daejeo1: Lets say I'd value it at the price of an IP 501 new (sold as used) |
| 17:11.16 | Daejeo1 | I never used polycom. so I want try |
| 17:12.04 | a1fa | aww' |
| 17:12.17 | [TK]D-Fender | Daejeo1: 600 is a great phone, but value-wise I gave you its realtive place. 600 supports PoE as-is AND brick which it comes with. |
| 17:13.48 | Daejeo1 | 501 (used ) around 65-75$ |
| 17:13.59 | Daejeo1 | on ebay |
| 17:14.25 | Daejeo1 | I am saving 25$ only |
| 17:16.03 | Daejeo1 | <PROTECTED> |
| 17:17.07 | jhiver | counting usage has nothing to do with the number of channels being in use... but actually thinking about it, least utilisation probably isn't a good plan either, since if a gateway has terrible ACDs it would keep sending lots and lots of calls to it |
| 17:17.48 | [TK]D-Fender | Daejeo1: No, IP 501 = $165 USD new best places I've seen. |
| 17:18.02 | jhiver | i guess rand() coupled with call-limit and fallback will have to do the trick |
| 17:18.20 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:18.20 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:18.32 | [TK]D-Fender | Daejeo1: And Polycom won't have Cisco PoE issues, firmware legality, and then the raw lack of Scall-handling & sip support that Cisco's have. |
| 17:18.40 | syzygyBSD | does anyone have a dhcpd.conf file that works with off the shelf polycoms? |
| 17:18.52 | syzygyBSD | 501's* |
| 17:20.58 | [TK]D-Fender | syzygyBSD: http://pastebin.com/m41f73207 |
| 17:21.14 | syzygyBSD | [TK]D-Fender: you rule |
| 17:21.23 | [TK]D-Fender | syzygyBSD: You're welcome |
| 17:22.04 | [TK]D-Fender | syzygyBSD: change routers if your * box isn't your gateway, etc.... should be rather evident. |
| 17:22.24 | mintee | Hey all |
| 17:23.40 | mintee | I'm looking to do a project using asterisk... as a voicemail only box. The box will hold a T1 (of some sort) and will hold only incoming calls for voicemail... |
| 17:23.52 | Daejeo1 | [TK]D-Fender: did you ever play with cisco? |
| 17:24.03 | mintee | basically a number that anyone can call, and leave a voice mail, and the user can call in and check all his voicemails |
| 17:24.07 | *** join/#asterisk fas3r (n=chatzill@haz95-2-82-243-75-212.fbx.proxad.net) |
| 17:24.13 | fas3r | hello everybody |
| 17:24.13 | syzygyBSD | ya, I had one that was working, but I messed it up trying to get ntp working as well. didn't have a backup |
| 17:24.27 | mintee | I'm looking to roll out about 100 voicemails to start with all on a single t1. |
| 17:24.44 | mintee | I'm just not sure what hardware card to be looking at |
| 17:24.47 | a1fa | [TK]D-Fender : i am running out of ideas :( |
| 17:24.57 | *** join/#asterisk pat2man (n=pat2man@c-24-19-34-19.hsd1.wa.comcast.net) |
| 17:25.20 | [TK]D-Fender | a1fa: And you still haven't shown me everything I asked for. |
| 17:25.46 | [TK]D-Fender | a1fa: pastebin everything you've got up, show the new register, the dropout, EVERYTHING |
| 17:25.57 | [TK]D-Fender | Daejeo1: Nope... I won't buy one personally... |
| 17:26.16 | a1fa | [TK]D-Fender : let me do a factory reset again |
| 17:26.19 | a1fa | and start over |
| 17:26.31 | fas3r | asterisk need option in kernel for work ? |
| 17:26.31 | [TK]D-Fender | mintee: Sangoma A102d (2-port is you're thinking of expanding) |
| 17:27.06 | mintee | [TK]D-Fender, thanks... I will be looking to expand. |
| 17:27.09 | Nivex | The more I read about these Polycoms, the more I want to sell off my cicsco 7960 and replace it with an IP501 |
| 17:27.41 | mintee | [TK]D-Fender are there any Digium cards in comparison |
| 17:27.43 | syzygyBSD | Nivex: I have both, keep your 7960 |
| 17:27.58 | [TK]D-Fender | mintee: Nothing I would suggest |
| 17:28.00 | Nivex | syzygyBSD: oh? what gripes you have? |
| 17:28.01 | Sweeper | syzygyBSD: rather expensive doorstop |
| 17:28.11 | Daejeo1 | Nivex: give one to me |
| 17:28.22 | syzygyBSD | Nivex: none, but they are about equal.. so why spend the extra $300? |
| 17:28.23 | Daejeo1 | i need polycom |
| 17:28.42 | mintee | [TK]D-Fender, ok, guess your not a fan of digium... you'd know better than I. |
| 17:29.14 | Nivex | Daejeo1: ??? |
| 17:29.15 | [TK]D-Fender | mintee: Correct |
| 17:29.39 | [TK]D-Fender | syzygyBSD: Whats a doorstop? |
| 17:29.47 | Daejeo1 | [TK]D-Fender: why do you hate Mark S |
| 17:30.07 | mintee | [TK]D-Fender, we're looking at getting a PRI, and porting toll free numbers to point to the voicemailbox.. Is there another way to do this than having a DID number for each toll-free voicemail box? |
| 17:30.08 | syzygyBSD | [TK]D-Fender: if he got rid of a 7960 for a 501 |
| 17:30.44 | Daejeo1 | Nivex: I am going to get polycom 600 |
| 17:31.07 | Daejeo1 | you can replace your cisco shit with my polycom |
| 17:31.14 | [TK]D-Fender | syzygyBSD: I don't follow you. Which would become a "doorstop", and why? |
| 17:31.15 | Yoe | am I correct in assuming that if I have a line with multiple DID numbers, asterisk will jump to a different extension based on what DID number I'm getting? |
| 17:31.33 | syzygyBSD | [TK]D-Fender: I never made the doorstop comment |
| 17:32.07 | Daejeo1 | Nivex: :) |
| 17:32.14 | [TK]D-Fender | syzygyBSD: OOps, bad aim :) |
| 17:32.29 | syzygyBSD | :) |
| 17:32.35 | [TK]D-Fender | Sweeper: rather expensive doorstop? which/why? |
| 17:33.00 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 17:33.28 | Sweeper | [TK]D-Fender: syzygyBSD said he had both the cisco and the pcom, and I was referring to the cisco being a rather expensive doorstop : |
| 17:33.58 | [TK]D-Fender | Sweeper: Well it IS expensive, but not a "doorstop", just not as good as the Polycom... |
| 17:34.05 | mintee | Yoe, that sounds about right, I'm trying to do nearlty the oppisite |
| 17:34.20 | zpertee | is there any professional naturally sounding text to speech software for asterisk. Festival(Flite) and Cepstral both are horrid. |
| 17:34.22 | syzygyBSD | ahh, ok it makes sense now.. |
| 17:34.24 | mintee | opposite even, |
| 17:34.41 | Yoe | mintee: eh, create a separate context, and use a _X. pattern in there? |
| 17:35.01 | syzygyBSD | I only got the 7960 to duiplicate a client's setup, they had all cisco phones |
| 17:35.13 | Sweeper | [TK]D-Fender: yea, but if he's got a pcom, and doesn't sell the cisco, he's gonna be using the pcom, and the cisco does doorstop duty :D |
| 17:35.15 | Nivex | I could probably sell the 7960 and the cisco PoE injector I scrounged for >= the cost of an IP501 |
| 17:35.30 | mintee | Yoe, I have no idea what you mean by that... i was speaking of, |
| 17:35.30 | mintee | we're looking at getting a PRI, and porting toll free numbers to point to the voicemailbox.. Is there another way to do this than having a DID number for each toll-free voicemail box? |
| 17:36.15 | Yoe | well, I don't know that much about all this, anyway, so don't ask me :) |
| 17:36.20 | [TK]D-Fender | mintee: Well yes you need a way for each to land on your PRI. How it comes in is up to you and your telco. |
| 17:36.25 | Daejeo1 | Nivex: why do you want to replace it with polycom? |
| 17:37.01 | Nivex | Daejeo1: some of my friends have polycoms, and the cisco firmware is kind of a PITA |
| 17:37.28 | [TK]D-Fender | Polycom call handling & voice quality = superior |
| 17:37.42 | Daejeo1 | what firmware did you load on it? |
| 17:37.44 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:38.00 | Sweeper | and green/silver doesn't match many colour schemes, black/silver goes with everything :D |
| 17:38.06 | Daejeo1 | Nivex: |
| 17:38.07 | Nivex | Daejeo1: P0S3-07-4-00 |
| 17:38.14 | Daejeo1 | oh man |
| 17:38.56 | a1fa | [TK]D-Fender : fixed |
| 17:39.27 | a1fa | [TK]D-Fender : factory reset+firmware upgrade == issue resolved |
| 17:39.30 | *** join/#asterisk pourriture (n=pourritu@mail.cshorecomputing.com) |
| 17:39.43 | mintee | Does Asterisk recognize ANI digits if we could convience the Telco to provide specifics. |
| 17:39.46 | syzygyBSD | ahhh, oh.. I hate when I have orphaned services |
| 17:40.02 | Daejeo1 | Nivex: how about trying 8.6? |
| 17:40.11 | Nivex | Daejeo1: send it over and I'll try it |
| 17:40.13 | syzygyBSD | spent the last 2 hours trying to reconfigure my dhcp server when an orphaned was running so configuration was never reloaded |
| 17:40.21 | Daejeo1 | ok |
| 17:40.52 | [TK]D-Fender | a1fa: Anal POS phone :p |
| 17:41.04 | mintee | Basically we're getting a block of toll-free numbers but would like to prevent doing a 1-to-1 relationship from 800 to DID |
| 17:41.17 | [TK]D-Fender | mintee: Ok... and? |
| 17:41.21 | Daejeo1 | Nivex:do you want to use my TFTP ? |
| 17:41.41 | Daejeo1 | anyway I send you the file |
| 17:41.43 | Daejeo1 | use your |
| 17:42.33 | mintee | [TK]D-Fender, so your saying that's possible? |
| 17:42.42 | [TK]D-Fender | mintee: Sure |
| 17:43.00 | mintee | can you briefly describe, or link me to HOWTO? |
| 17:43.04 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 17:43.16 | mintee | sorry to be a pain, but once i figure out how, i will be able to DO it :P |
| 17:43.19 | [TK]D-Fender | mintee: This is * 101. I take it this means you have never USED * before. |
| 17:43.32 | Daejeo1 | Nivex: |
| 17:43.42 | Daejeo1 | got it? |
| 17:43.42 | mintee | correct.. i'm just getting info before I start spending the money. |
| 17:43.52 | [TK]D-Fender | mintee: telling you means DOING IT for you. And for that I am contractable at a very accessable rate. |
| 17:44.07 | Nivex | Daejeo1: receiving |
| 17:44.13 | Daejeo1 | good |
| 17:44.19 | mintee | no worries.. . you've provided enough... however, it's good to know that you'll help at a rate |
| 17:44.32 | Nivex | Daejeo1: I'm a little concerned that the file size is 666kB :) |
| 17:45.17 | Daejeo1 | it is zipped |
| 17:46.15 | mintee | [TK]D-Fender, can ya give me a guess onto what page of the docs it will be describing how to route individual 800's to individual VMboxes without using DID's (one to many relationship... |
| 17:46.26 | Daejeo1 | I guess you know how to deal with cisco shit |
| 17:46.38 | [TK]D-Fender | mintee: "help" is usually free, "do for you" = very accessably priced :) |
| 17:47.03 | Nivex | Daejeo1: I was referring to the number. 666 == mark of the beast |
| 17:47.15 | Nivex | Daejeo1: file's done. thanks. I'll take a look when I get home this evening |
| 17:47.21 | mintee | [TK]D-Fender, yes, i understand that... I just cannot grasp the concept on how it's done... |
| 17:47.52 | [TK]D-Fender | mintee: Then download * play around, and then when you're a few odd inches away we'll be here. |
| 17:48.02 | mintee | you got it... |
| 17:48.03 | mintee | danka |
| 17:48.07 | Daejeo1 | Nivex: ur welcome |
| 17:49.11 | Daejeo1 | Nivex: send me if you get 8.7 |
| 17:49.25 | Daejeo1 | 8.7 has been released |
| 17:50.22 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:50.54 | Daejeo1 | [TK]D-Fender: do you work for any firm? |
| 17:50.55 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:51.02 | mintee | [TK]D-Fender, oi, but regardless, you still agree on the Sangoma A102d for this type of project? |
| 17:51.11 | EmleyMoor | Is there a way I can set up a "join call" function? I can do listening in easily but sometimes it would be helpful if I could join and talk |
| 17:51.21 | Tako-san | hey Daejeo1 |
| 17:51.27 | [TK]D-Fender | mintee: Highly. |
| 17:51.32 | mintee | cool |
| 17:51.33 | [TK]D-Fender | Daejeo1: I'm freelance. |
| 17:51.37 | EmleyMoor | (on the whole call, not just whisper to the party this side) |
| 17:51.49 | coppice | cheaplance |
| 17:52.05 | Daejeo1 | Tako-san: wow! good to hear from you |
| 17:52.12 | Tako-san | :) |
| 17:52.18 | Daejeo1 | how are you doing? |
| 17:52.26 | Tako-san | You still in south Korea? |
| 17:52.27 | [TK]D-Fender | coppice: Free as in un-attached, not as in beer or speech ;) |
| 17:52.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-93-103.red.bezeqint.net) |
| 17:52.46 | Daejeo1 | yes. I could not finish my thesis so far |
| 17:53.26 | coppice | its best not to finish a thesis. i'd hate to be doctored :-\ |
| 17:53.53 | Nivex | Daejeo1: I was staying at 7.4 since 7.5 had a reported bug with SIP |
| 17:54.03 | Nivex | Daejeo1: going by the wiki article: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 17:54.13 | a1fa | [TK]D-Fender : Linksys SPA921 |
| 17:54.15 | a1fa | I love it |
| 17:54.17 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:54.22 | a1fa | ...not |
| 17:55.06 | Daejeo1 | Nivex: i installed 8.6 on my phone |
| 17:55.18 | Daejeo1 | no issues |
| 17:55.21 | Daejeo1 | so far |
| 17:55.21 | [TK]D-Fender | a1fa: IP320/330 KILLS Linksys's entrie line through SPA-942 :) |
| 17:55.29 | a1fa | IP320? |
| 17:55.36 | a1fa | Polycom? |
| 17:56.11 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 17:56.37 | a1fa | wow |
| 17:56.42 | a1fa | thats a sexy feature phone |
| 17:56.56 | a1fa | IP650 and I jizz1 |
| 17:57.22 | coppice | the SPA941/2 look like they were thrown together in someone's garage |
| 17:57.28 | a1fa | its ok |
| 17:57.29 | a1fa | :P |
| 17:57.34 | EmleyMoor | Is there a way I can "enter into" a call in progress on my system without having to have the user put that call on hold? |
| 17:57.35 | styelz | i like the linksys handset, you dont punch yuourself in the face when you hold it to your ear |
| 17:57.43 | EmleyMoor | (to talk as well as listen) |
| 17:57.43 | jingles | I find I like the SPA941/942 as well. |
| 17:57.48 | a1fa | how does IP320 handle those two ethernet ports? |
| 17:57.51 | jingles | they're really easy to administrate. |
| 17:58.01 | a1fa | Polycom IP 650 REFURB $250 |
| 17:58.02 | a1fa | no |
| 17:58.07 | a1fa | is it nat or bridge? |
| 17:58.16 | Dr-Linux | my asterisk is consuming 99.9% CPU, what could be check? |
| 17:58.30 | a1fa | ulaw -> gsm |
| 17:58.32 | EmleyMoor | Dr-Linux: Is this with or without calls on it? |
| 17:58.49 | Dr-Linux | EmleyMoor: no calls |
| 17:59.03 | [TK]D-Fender | EmleyMoor: Since 3-way calling is done on the phone your only option to force your way in is to AMI redirect the 2 legs to a conference and join it yourself. |
| 17:59.14 | Dr-Linux | EmleyMoor: it's testing server, but very important |
| 17:59.28 | [TK]D-Fender | EmleyMoor: If force isn't required just have the inside end bridge you in on their phone. |
| 17:59.38 | zpertee | does anyone know how to integrate neospeech with asterisk |
| 17:59.52 | EmleyMoor | [TK]D-Fender: No real problem |
| 18:00.05 | [TK]D-Fender | a1fa: IP320 = 1 port = $87.50. IP 330 = 2 port = $111 |
| 18:00.16 | [TK]D-Fender | EmleyMoor: So which scenario? |
| 18:00.22 | tzanger | fun |
| 18:00.37 | tzanger | instead of synchronizing tdm clocks I've been tasked to create an elastic buffer to do it |
| 18:00.52 | EmleyMoor | [TK]D-Fender: Just wondered if there was a way I could initiate me getting in withot too much hassle - but I can easily train the user to get me in. |
| 18:00.54 | a1fa | [TK]D-Fender : you selling at that rate? |
| 18:02.22 | [TK]D-Fender | a1fa: Thats just a store I refer poeple to : www.telephonydepot.com |
| 18:02.36 | [TK]D-Fender | a1fa: pure retail, and they don't know me at all. |
| 18:02.57 | [TK]D-Fender | EmleyMoor: Sorry, nothing convenient unless ZapBarge will do. |
| 18:03.10 | [TK]D-Fender | EmleyMoor: And not sure exactly HOW convenient that is. |
| 18:03.11 | a1fa | hehe |
| 18:03.23 | [TK]D-Fender | a1fa: thats assuming PoE |
| 18:03.40 | [TK]D-Fender | a1fa: You'll need to buy power bricks otherwise |
| 18:03.46 | masterisk | [TK]D-Fender what do you think about digiumcards.com |
| 18:03.51 | masterisk | ever ordered from them ? |
| 18:03.54 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:03.54 | EmleyMoor | I have ChanSpy available for me to listen/whisper - but that only partly joins me in |
| 18:04.12 | [TK]D-Fender | masterisk: No solid opinion, but the pricing is right and their lineup decent. Could be just fine. |
| 18:04.30 | masterisk | I use them all the time, they're located in MI(like me) and they've been awesome so far |
| 18:04.38 | masterisk | you can always haggle on the price |
| 18:04.49 | [TK]D-Fender | masterisk: Ok, I'll take that as a vote for them en. |
| 18:05.42 | a1fa | interesting |
| 18:06.04 | EmleyMoor | Shame I've had to "squelch" the R button... still, it's a consequence of supporting a dial |
| 18:06.40 | rickross | anyone know if there exists a real G.722 codec for asterisk (not the limited pass-throught) |
| 18:07.48 | coppice | * has a real G.722 codec |
| 18:08.12 | [TK]D-Fender | rickross: in TRUNK, not FTp release |
| 18:08.24 | tzanger | coppice: ever had to "synchronize" using elastic buffers before? |
| 18:08.33 | rickross | maybe we just didn't build correctly then, we do not see format_g722.so or codec_g22.so in /usr/lib/asterisk/modules |
| 18:08.49 | a1fa | [TK]D-Fender : what do you think about SPA962? |
| 18:08.58 | coppice | tzanger: isn't that what telecoms does all the time? |
| 18:08.58 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:09.03 | rickross | TK - is that in 1.4.xx, or only in 1.2.xx ? |
| 18:09.41 | tzanger | coppice: just wondering how the hell to find the start of the tdm stream within the buffer |
| 18:09.44 | [TK]D-Fender | a1fa: Colour=pretty, but SPA call handling SUCKS, no presence and is pricey. IMO "fuck it" |
| 18:09.50 | tzanger | I guess I could pre-fill it with a known value and look for the change |
| 18:09.53 | [TK]D-Fender | rickross: 1.4 |
| 18:10.06 | [TK]D-Fender | rickross: 1.2 will NOT be getting any newer features. |
| 18:10.53 | a1fa | hehe |
| 18:10.56 | a1fa | polycom all the way |
| 18:11.09 | rickross | TK - should we expect it to produce modules for G.722 in /usr/lib/asterisk/modules, then? (apprently the passthrough patch does not) |
| 18:11.20 | [TK]D-Fender | a1fa: So far nobody is putting out something I would call "competition". |
| 18:11.40 | [TK]D-Fender | rickross: 1.4 trunk SVN only. |
| 18:11.43 | rickross | we can "allow g722" and the Polycom HD phones will then use it, but we cannot record voicemail, etc |
| 18:12.00 | [TK]D-Fender | rickross: and frankly, G.722.... whatever... it doesn't hit PST and its a friggen PHONE we're talking about... |
| 18:12.16 | rickross | TK - thanks, I will go ahead an pull that |
| 18:12.56 | rickross | well, when our team members are talking to one another, or using our asterisk, the higher quality is nice |
| 18:13.15 | coppice | [TK]D-Fender: was that last comment supposed to mean something? |
| 18:13.17 | rickross | but it definitely blows up the voicemail if you have G.722 enabled and try to record |
| 18:13.46 | a1fa | [TK]D-Fender : how are the low end models for polycom? |
| 18:14.02 | [TK]D-Fender | PSTN* |
| 18:14.34 | [TK]D-Fender | a1fa: All the same call handling as the higher models. IP 30X = forget. Model depended on qty & deployment. |
| 18:14.35 | coppice | wideband voice is dramatically better than normal telephony |
| 18:14.50 | lirakis | im having trouble dialing out a ZAP channel on a T1 i have. I have setup inbound DID's via pri. I am trying to dial out using Dial(ZAP/g1/${EXTEN})... my zapata.conf is here http://pastebin.com/d7cc2cde6 |
| 18:15.04 | lirakis | i get "unable to create ZAP channel reason unknown (0)" |
| 18:15.07 | lirakis | when i try to dial |
| 18:15.38 | lirakis | i have checked the upstream switch (alcatel dex) for call records on the trunk group .. and it shows nothing.. so the call isnt leaving the * box |
| 18:15.50 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:16.31 | a1fa | [TK]D-Fender : thanks.. i will do business w/ polycom next time |
| 18:16.32 | [TK]D-Fender | lirakis: Because you set group=0 a few lines lower and are OVERRIDING IT <-- |
| 18:16.43 | a1fa | no more linksys crapware firmware upgradeZ! |
| 18:16.44 | a1fa | :P |
| 18:16.51 | [TK]D-Fender | lirakis: 30 vs 39 |
| 18:17.44 | a1fa | the only "bad" part is POE |
| 18:18.12 | a1fa | i have to buy a POE brick for everyphone, if my users dont have poe switches |
| 18:18.52 | [TK]D-Fender | a1fa: You can use a PoE brick, or a standard wall-wart (sold seperately). |
| 18:19.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 18:19.24 | [TK]D-Fender | a1fa: If your PoE injectors are anywhere near the same price as the brick ($17.50) go with PoE, they are reusable later. |
| 18:19.40 | Sweeper | or you could wire the phone straight into the building's 48v power! |
| 18:19.44 | [TK]D-Fender | a1fa: But still about the same net price as Linksys, but so much better off on quality |
| 18:19.45 | Sweeper | been there, done that :D |
| 18:20.35 | a1fa | roger |
| 18:23.25 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:25.09 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 18:25.09 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 18:25.25 | *** join/#asterisk lyroy (n=lyroy@picachou.csaffluents.qc.ca) |
| 18:27.10 | lyroy | I'm trying to build a simple agi script using php... the script use the SAY PHONETIC command, the problem is when I call the agi script, I here no digit or letters at all but in my asterisk console I see something like: |
| 18:27.19 | lyroy | <PROTECTED> |
| 18:27.19 | lyroy | <PROTECTED> |
| 18:27.19 | lyroy | <PROTECTED> |
| 18:27.19 | lyroy | <PROTECTED> |
| 18:27.21 | lyroy | .. |
| 18:27.37 | lyroy | but I cant ear nothing |
| 18:28.02 | lyroy | does someone can give me a hint on howto use the SAY PHONETIC command please? |
| 18:28.38 | a1fa | hehe |
| 18:30.29 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 18:30.56 | lyroy | the person who post this http://bugs.digium.com/view.php?id=8530 have the same problem... but I dont know if it has been solve? |
| 18:31.17 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:31.28 | captiancrash | When using the Asterisk-GUI, where are configuration changes saved to? |
| 18:31.40 | a1fa | its amazing how much * is improved in 1.4. tree |
| 18:31.51 | a1fa | DTMF is much more accurate |
| 18:31.59 | a1fa | everything is smooth! |
| 18:32.02 | a1fa | ^5 developers! |
| 18:32.32 | deeperror | can anyone assist with 3-way calling? Are there issues with using zap fxs and making calls out over sip? I keep getting warnings when pressing the flash button to get a dt |
| 18:34.02 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 18:34.02 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 18:34.19 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 18:35.21 | *** join/#asterisk whhheeEEEeeee (n=nathan@adsl-68-74-132-52.dsl.klmzmi.ameritech.net) |
| 18:35.43 | Daejeo1 | lyroy: exten -> 123,1,SetLanguage(en) exten -> 123,2,SayPhonetic(Daejeo) |
| 18:36.29 | rickross | anyone here running production servers from the svn trunk of asterisk? |
| 18:36.43 | whhheeEEEeeee | that'd be ...... not smart. |
| 18:36.54 | rickross | is the project's policy to try to have the trunk be pretty stable, even though it is development |
| 18:37.03 | Strom_M | rickross: uh, no |
| 18:37.10 | Strom_M | for production, use a release branch |
| 18:37.13 | *** part/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:37.22 | rickross | whhheeEEE - there's stuff in the trunk that we need |
| 18:37.37 | rickross | we'll try to backport |
| 18:37.41 | rickross | maybe that will do |
| 18:38.39 | *** join/#asterisk didge (n=mcveighj@bas2-barrie18-1279301505.dsl.bell.ca) |
| 18:38.45 | Strom_M | what specifically do you need |
| 18:38.52 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:39.06 | rickross | G.722 codec support |
| 18:39.07 | didge | hi. I'm aware that Sphynx2 is compatible with asterisk. Is there an 8bit AM for Sphinx4 ? |
| 18:39.23 | rickross | we're copying it to our 1.4.10.1 right now - see if it builds and runs |
| 18:39.25 | Strom_M | rickross: you know that 1.4 will do passthrough, right? |
| 18:39.27 | russellb | rickross: transcoding you mean? |
| 18:39.35 | deeperror | any help on this one? http://pastebin.ca/667110 |
| 18:39.38 | *** join/#asterisk mog (i=mog@nat/digium/x-fd07897c77ccae77) |
| 18:39.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:40.00 | rickross | passthrough won't do - our voicemail is blowing up when people try to record to their mailboxes |
| 18:40.06 | Strom_M | ah |
| 18:40.14 | russellb | you should be able to record in 1.4 as well |
| 18:40.21 | russellb | well, record in g.722 format directly |
| 18:40.32 | russellb | if you want it converted to wav or something, you're right, you need codec_g722 |
| 18:40.37 | russellb | should be a trivial backport |
| 18:41.42 | rickross | fingers crossed :) |
| 18:42.14 | whhheeEEEeeee | rickross: anyways, never run cvs in production. that's basically suicidal. |
| 18:42.18 | russellb | well... there are some related configure script changes though ... |
| 18:42.22 | russellb | whhheeEEEeeee: what is cvs? |
| 18:42.37 | whhheeEEEeeee | russellb: source code repository |
| 18:42.39 | [TK]D-Fender | deeperror: I've never seen an ITSP accept a "+" in your number before. |
| 18:42.43 | coppice | its the improved evrsion of SVN |
| 18:42.53 | russellb | whhheeEEEeeee: i know that ... just trolling since we don't use cvs anymore |
| 18:42.57 | lirakis | [TK]D-Fender: i saw that.. i wasnt sure.. i didnt actually setup this T1 card.. but i saw it said it was "autogenerated" .. so i thought it was "ok" |
| 18:42.57 | whhheeEEEeeee | coppice: like hell it is. |
| 18:43.07 | whhheeEEEeeee | coppice: svn replaces cvs |
| 18:43.08 | [TK]D-Fender | rickross: compile trunk, but ONLY keep your codec_g722.so around :) |
| 18:43.14 | whhheeEEEeeee | russellb: lol |
| 18:43.19 | russellb | whhheeEEEeeee: you're missing all the fun in the sarcasm |
| 18:43.24 | russellb | cvs is totally better |
| 18:43.27 | whhheeEEEeeee | :P |
| 18:43.40 | whhheeEEEeeee | I'm not used to this channels version of 'humor'. |
| 18:43.44 | coppice | I strongly dislike CVS, but I really hate SVN |
| 18:44.30 | whhheeEEEeeee | I've come to the conclusion from one conversation with [TK]D-Fender that he's crazy. |
| 18:44.39 | russellb | whhheeEEEeeee: indeed. |
| 18:44.41 | lirakis | [TK]D-Fender: okay .. now its creating the channel and im getting a cause code back. Thanks |
| 18:44.50 | russellb | [TK]D-Fender: that won't work |
| 18:44.52 | lirakis | [TK]D-Fender: .. its my first T1 :p |
| 18:45.01 | russellb | trunk and 1.4 are not binary compatible |
| 18:45.04 | whhheeEEEeeee | lirakis: aw, so cute. |
| 18:45.11 | [TK]D-Fender | russellb: You saying yuo can't just bring over the codec from trunk to reusing in current FTP releases? |
| 18:45.20 | Qwell[] | not the .so file |
| 18:45.22 | datachomper | svn is so f'in hard to learn, but fairly easy to use once you know it |
| 18:45.23 | russellb | not as a binary, no |
| 18:45.53 | lirakis | whhheeEEEeeee: name from the firefox short ?? |
| 18:45.58 | lirakis | that is halirious btw |
| 18:46.05 | [TK]D-Fender | russellb: well how come codecs like the "freebie" G.729 don't need recompiling with new *'s? |
| 18:46.52 | russellb | all versions of 1.4 are binary compatible |
| 18:46.55 | russellb | that's why |
| 18:47.00 | russellb | it's trunk vs. 1.4 that isn't compatible |
| 18:47.21 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 18:47.51 | a1fa | [TK]D-Fender |
| 18:47.57 | a1fa | hehe |
| 18:48.20 | whhheeEEEeeee | I'm trying to find a good voip phone, sadly that doesn't seem to be happening. Instead I'm looking into getting a voip to analog device, any good iax devices you guys know of? |
| 18:48.27 | a1fa | i want to match 8-XXX-XXX-XXXX, but only dial X. |
| 18:49.04 | a1fa | _8NXXNXXXXXX |
| 18:50.23 | a1fa | i hate these patterns |
| 18:51.29 | [TK]D-Fender | russellb: that blows. |
| 18:51.33 | Strom_M | whhheeEEEeeee: good voip phone == polycom |
| 18:51.51 | *** join/#asterisk oej (n=olle@212.17.142.154) |
| 18:51.55 | datachomper | a1fa, Dial(<channel>/+${EXTEN:10}) |
| 18:52.23 | [TK]D-Fender | whhheeEEEeeee: ATA's to use analog phones can be fine, but whether I SUGGEST them depends on what kind of usage they'll see. |
| 18:52.27 | brad_mssw | whhheeEEEeeee: umm, the only iax ata I know of is the iaxy ... and it is a total POS ... best ATAs are probably the Sipura/Linksys SPA2100's |
| 18:52.50 | [TK]D-Fender | datachomper: LOL, no! |
| 18:52.51 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 18:53.58 | brad_mssw | whhheeEEEeeee: i've had good luck with linksys 942's for sip phones though |
| 18:54.35 | [TK]D-Fender | a1fa: ${EXTEN:1} <--------- |
| 18:54.57 | whhheeEEEeeee | brad_mssw: mm, I was trying to figure out the difference between the pap2 and the spa's.... I guess the spa's support iax where the pap2 only does sip. |
| 18:55.26 | brad_mssw | whhheeEEEeeee: no, nothing supports iax except digium's iaxy ... |
| 18:55.32 | *** join/#asterisk kkn088 (n=kikoun@84.6.52.109) |
| 18:55.32 | whhheeEEEeeee | Strom_M: polycom you say, hmmm. |
| 18:55.35 | whhheeEEEeeee | damnit. |
| 18:55.40 | [TK]D-Fender | whhheeEEEeeee: No, the PAP2 typically has no T.38 support, a weaker CPU, and not router function. |
| 18:55.48 | brad_mssw | whhheeEEEeeee: the pap2 is more home-user, where the spa-2100 is more business oriented, supporting things like T.38 for faxing |
| 18:55.52 | [TK]D-Fender | brad_mssw: Inaccurate :) |
| 18:56.16 | brad_mssw | [TK]D-Fender: which part? |
| 18:56.17 | [TK]D-Fender | brad_mssw: There are a number of ATA's and phones that support IAX2... |
| 18:56.32 | brad_mssw | [TK]D-Fender: really? must have changed since I last looked a year or so ago |
| 18:56.36 | [TK]D-Fender | PAP2 = bleh |
| 18:56.46 | [TK]D-Fender | brad_mssw: You never really looked then :) |
| 18:57.04 | [TK]D-Fender | brad_mssw: However : they all SUCK. **HARD** |
| 18:57.25 | whhheeEEEeeee | [TK]: in short I'm using asterisk (of course) as a backbone, and going to add voip lines for each extension. thats, in short, how it will be used. |
| 18:58.01 | coppice | there is apparently a new version of the PAP2 with T.38 |
| 18:58.19 | rickross | TK - it seems like a bunch of stuff has changed in /include/asterisk/config.h - this may not be as trivial as we had hoped :( |
| 18:58.28 | whhheeEEEeeee | excuse me: using the voip to analog converters for each line |
| 18:58.30 | brad_mssw | coppice: really? |
| 18:58.38 | Mercestes | whhheeEEEeeee, you do realize that SIp phones and VOIP connections do not operate on "lines" or any phone:lines ration, right? |
| 18:58.42 | hmmhesays | there is no pain you are receding |
| 18:58.42 | whhheeEEEeeee | gosh, I hope that wasn't confusing |
| 18:58.44 | brad_mssw | coppice: actual hardware change, or just a firmware update? |
| 18:58.50 | hmmhesays | a distant ship smoke on the horizon |
| 18:59.23 | coppice | well, people say they are trying it against my T.38 code/ dunno if its new hardware, or just a firmware change |
| 18:59.27 | whhheeEEEeeee | mercestes: no, they use chakra energy lines. ;) they're invisible. |
| 18:59.38 | russellb | rickross: how much red bull would you send me if i backported it for you? :) |
| 18:59.45 | whhheeEEEeeee | lol |
| 18:59.51 | rickross | russellb :) |
| 18:59.58 | [TK]D-Fender | rickross: as I'm told, it won't work. Trunk = incompatible with current |
| 19:00.03 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-fdec569c9a50c7cf) |
| 19:00.21 | brad_mssw | coppice: you were hot and heavy into T.38 stuff a while back ... did you end up dropping trying to get termination into asterisk? |
| 19:00.34 | Netgeeks | hey anyone know how to disable these message from the cli?: Internal RTCP NTP clock skew detected: lsr=14954456, now=101133844, dlsr=86179708 (1314:997ms), diff=320 |
| 19:00.46 | russellb | Netgeeks: it is disabled in newer versions |
| 19:00.53 | [TK]D-Fender | whhheeEEEeeee: How much tranferring of calls etc, or do you need them just to be basic seperate devices doing "their own thing" handling usually 1 call at a time? |
| 19:01.09 | coppice | brad_mssw: trying to get anything into * is too must hassle |
| 19:01.11 | brad_mssw | coppice: i know openpbx (or whatever it is now) ran with your code |
| 19:01.12 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 19:01.37 | Netgeeks | russell: ah, thanks! |
| 19:01.46 | whhheeEEEeeee | [TK]: handling one call at a time. I assumed that asterisk would hold them in a queue if the extension is busy. |
| 19:01.53 | Olobola | is it possible to convert voicemail to mp3 from within asterisk? |
| 19:02.19 | [TK]D-Fender | whhheeEEEeeee: You could go the el-cheapo rout and run a bunch of SPA-2102's for about 65$ each |
| 19:02.27 | [TK]D-Fender | whhheeEEEeeee: Handles 2 phones each |
| 19:02.31 | whhheeEEEeeee | [TK]: I want asterisk to be the monster that handles -everything- and each line just to be dumb. |
| 19:02.50 | denon | sounds like you want sccp |
| 19:02.54 | [TK]D-Fender | whhheeEEEeeee: ATA's can do fine if you're happy with the concept of Analog call handling. |
| 19:03.57 | whhheeEEEeeee | [TK]: I'd rather not have analog, but I don't like any voip phones I've seen. (at&t has this nice phone that will let you uplink your bluetooth device and cellphone to it, I mean, geeze, thats worth having an analog line. lol) |
| 19:07.09 | russellb | coppice: i'm sorry you feel that way ... we are just so backlogged. Not enough people to handle every issue and patch we get fast enough... I am certainly working my ass off to do the best I can to do my part to keep up ... |
| 19:07.56 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:09.45 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 19:10.37 | [TK]D-Fender | whhheeEEEeeee: And what have you USED? |
| 19:11.57 | Yourname` | So, here's a problem.. install of 1.4.11 is giving me a problem. codefreeze told me the other day what to do (go in manually and install asterisk addons) and I did it and it worked. And now I'm trying again but it doesn't work http://pastebin.ca/667155 |
| 19:12.10 | whhheeEEEeeee | [TK]: not much. lol |
| 19:13.56 | [TK]D-Fender | whhheeEEEeeee: You're opinions seem well founded then :p |
| 19:14.01 | [TK]D-Fender | whhheeEEEeeee: Polycom > all |
| 19:14.06 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:14.42 | deeperror | is it possible to perform 3way calling from an analog phone connected to * making the call out over SIP? |
| 19:15.11 | [TK]D-Fender | deeperror: Sure. how * is involved doesn't matter. |
| 19:15.29 | [TK]D-Fender | deeperror: Depends on the device that lets you USE an analog phone. |
| 19:15.51 | deeperror | so the analog device is what would be causing an issue if the flash button doesn't move over to dt? |
| 19:16.00 | deeperror | rhino channel bank & t1 card |
| 19:16.13 | deeperror | i can make calls inbound outbound but can't flash to dt |
| 19:16.47 | deeperror | and the errors when pressing the flash button are related to channel.c and chan_sip.c |
| 19:16.51 | whhheeEEEeeee | [TK]: polycom you say then, eh? ok, well, is that available in the USA? |
| 19:16.53 | Yourname` | Basicallly the same error as http://forums.digium.com/viewtopic.php?p=50261&sid=198db4266906de19a54aee1ea01856b0 |
| 19:17.56 | deeperror | fender: i'm just unable to really troubleshoot this issue any further without going into code |
| 19:18.00 | deeperror | any clues? |
| 19:18.14 | [TK]D-Fender | whhheeEEEeeee: www.telephonydepot.com |
| 19:18.22 | [TK]D-Fender | whhheeEEEeeee: great value. |
| 19:18.34 | Yourname` | Install problems in 1.4 |
| 19:18.43 | whhheeEEEeeee | [TK]: you rock as always |
| 19:18.43 | Yourname` | yo [TK]D-Fender |
| 19:18.50 | a1fa | ihahahah |
| 19:18.57 | Yourname` | Hey. I say the samething, whhheeEEEeeee lol |
| 19:18.57 | a1fa | I conf called two pizza joints |
| 19:18.59 | a1fa | i whish i recorded it |
| 19:19.07 | a1fa | haha |
| 19:19.14 | a1fa | "PizzaHUT" |
| 19:19.18 | a1fa | "Dominos" |
| 19:19.24 | a1fa | no, sir, this is pizza hut |
| 19:19.32 | a1fa | no, no, sir this is dominos |
| 19:19.33 | a1fa | ahhaha |
| 19:19.40 | a1fa | priceless!! |
| 19:19.47 | a1fa | maturity level: first grade |
| 19:19.51 | [TK]D-Fender | a1fa: evil bastard |
| 19:20.09 | Sci_05 | nice one |
| 19:20.14 | whhheeEEEeeee | lol |
| 19:20.31 | deeperror | my pastebin is the top of google when searching for the error |
| 19:20.38 | deeperror | thats a good sign ha |
| 19:20.46 | Yourname` | lol |
| 19:20.46 | Deeewayne | alfa: I used to do that in the early 90's with 3-way calling and my high school friends who hated each other |
| 19:21.01 | whhheeEEEeeee | deeewayne: oh gosh. I remember those days. |
| 19:21.17 | Deeewayne | I love the reactions, "you called me", "no, you called me" |
| 19:21.24 | a1fa | Deeewayne : 3-way just sounds dirty :p |
| 19:21.31 | CCFL_Man2 | polycoms look ugly |
| 19:21.40 | a1fa | i had my co-workers here |
| 19:21.43 | a1fa | we laughed our ass off |
| 19:21.46 | Sweeper | CCFL_Man2: troll! |
| 19:21.52 | a1fa | listening to these two guys go off at eachother |
| 19:21.52 | a1fa | hahaha |
| 19:23.09 | CCFL_Man2 | Sweeper: nah |
| 19:23.14 | whhheeEEEeeee | I guess a "your mom looks ugly" joke would be out of the question. |
| 19:23.19 | Yourname` | Hmm, where are the asterisk modules stores? |
| 19:23.21 | Yourname` | stored* |
| 19:23.30 | CCFL_Man2 | the ip 320 looks kind of nice |
| 19:23.37 | [TK]D-Fender | CCFL_Man2: they grow on you, fast. |
| 19:23.51 | Sweeper | whhheeEEEeeee: yea. redundant and such |
| 19:24.00 | CCFL_Man2 | [TK]D-Fender: i'm tempted to order the 320 now |
| 19:24.18 | CCFL_Man2 | $87 isn't bad |
| 19:24.20 | [TK]D-Fender | CCFL_Man2: make sure it fits your deployment, and you consider the power req. |
| 19:24.33 | [TK]D-Fender | CCFL_Man2: All of my customers with them are thrilled |
| 19:24.46 | CCFL_Man2 | [TK]D-Fender: well, it'll replace my one cisco 7912 i have |
| 19:25.08 | CCFL_Man2 | i just have one phone :P |
| 19:25.18 | whhheeEEEeeee | lol, "pap2t-na" |
| 19:25.32 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.105.176) |
| 19:25.36 | whhheeEEEeeee | "touch n affection" |
| 19:26.25 | CCFL_Man2 | [TK]D-Fender: infact i think the 302 looks prettier than the cisco 7912 |
| 19:27.06 | *** join/#asterisk danielxpt (n=danielxp@px1.xfoneusa.com) |
| 19:27.10 | whhheeEEEeeee | whats the point of having dual versus single ethernet ports, failover? |
| 19:27.40 | whhheeEEEeeee | looking at the 320 versus the 330 |
| 19:28.03 | danielxpt | can anyone help me with asterisk passing regular long distance but not passing 800 numbers? |
| 19:28.15 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 19:28.18 | ManxPower | whhheeEEEeeee: it avoids the need for 2 ethernet wall jacks (computer and phone) at every location |
| 19:28.32 | ManxPower | danielxpt: Maybe if you defined "pass". |
| 19:28.38 | whhheeEEEeeee | ManxPower: oh, so it acts like a switch then. |
| 19:28.58 | ManxPower | whhheeEEEeeee: It IS a 3-port switch (one port is internal and goes to the guts of the phone) |
| 19:29.33 | danielxpt | ManxPower: I can dial a long distance nubmer, but when it dial a 800 number, it hangs up and gives a no route to destination from the pri |
| 19:29.53 | whhheeEEEeeee | ManxPower: yeah, thats what I just said. :P |
| 19:30.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:30.09 | ManxPower | danielxpt: make sure you don't have a pridialplan= option |
| 19:30.25 | didge | whhheeEee; dual ethernet ports allow for software routers/bridges in linux |
| 19:30.40 | ManxPower | you will need to check the value of ${HANGUPCAUSE} to see the Q.931 code. |
| 19:30.48 | ManxPower | didge: I didn't know the Polycoms ran linux. |
| 19:30.52 | CCFL_Man2 | what is the ztdynamic driver used for? |
| 19:31.29 | danielxpt | I am assuming pridialplan= woudl be in zapata.conf. I don't have it in there |
| 19:31.57 | *** join/#asterisk prizark_ (i=prizark_@bb121-7-65-32.singnet.com.sg) |
| 19:32.16 | ManxPower | danielxpt: I suspect your CARRIER is not allowing you. |
| 19:32.37 | ManxPower | but knowing the value of HANGUPCAUSE might be helpful. |
| 19:33.54 | fas3r | just one question, asterisk require option in kernel ? |
| 19:34.22 | ManxPower | fas3r: no. It might need a kernel module if you have telephony hardware or require timing. |
| 19:34.44 | danielxpt | how do you find the vaule of hangupcause? |
| 19:34.49 | fas3r | require timing .. RTC ( realtimeclock ) ? |
| 19:35.08 | ManxPower | as the priority after Dial do: Noop(HANGUPCAUSE is ${HANGUPCAUSE}) |
| 19:35.25 | ManxPower | fas3r: no, for IAX2 trunking or MoH timing source. |
| 19:35.39 | *** join/#asterisk Greenbox (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:35.43 | ManxPower | Oh and Meetme. |
| 19:36.26 | fas3r | ManxPower: ok thank |
| 19:37.29 | [TK]D-Fender | ManxPower: Not needed for Moh..... |
| 19:37.32 | fas3r | what the telephone who are easy to use with asterisk ? |
| 19:37.41 | [TK]D-Fender | ManxPower: I ran MoH for a long time without needing Zaptel. |
| 19:37.50 | *** part/#asterisk prizark_ (i=prizark_@bb121-7-65-32.singnet.com.sg) |
| 19:38.19 | *** join/#asterisk mog (i=mog@nat/digium/x-4f8d1c275e13264b) |
| 19:38.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:40.16 | *** join/#asterisk Op3r (n=Op3r@125.212.63.43) |
| 19:40.49 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 19:41.25 | [TK]D-Fender | fas3r: ... |
| 19:41.26 | [TK]D-Fender | ~phones |
| 19:41.27 | jbot | [phones] http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:41.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:41.59 | *** join/#asterisk jhiver_ (n=jhiver@LReunion-151-3-191.w193-253.abo.wanadoo.fr) |
| 19:42.01 | [TK]D-Fender | ManxPower: I'm going to blog up a replacement for [av]bani's list..... |
| 19:42.07 | fas3r | ok thank :p |
| 19:42.21 | Op3r | anyone know any good inbound voip providers for UK? |
| 19:42.21 | *** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com) |
| 19:42.28 | CCFL_Man2 | seems like you can build your own isa or pci T1 card for really cheap |
| 19:42.36 | ManxPower | I hate blogs |
| 19:42.51 | whhheeEEEeeee | crap, this polycom phone doesn't support aix, only sip. |
| 19:43.17 | ManxPower | whhheeEEEeeee: NO decent phone supports IAX. Just give up on that idea right now |
| 19:43.48 | ManxPower | I think you can get one no-name chinese import that has a community developed IAX2 firmware for it. |
| 19:43.54 | fas3r | i think i wait before install it maybe it's a mistake so empty parameters ^^ |
| 19:43.58 | whhheeEEEeeee | bleh. |
| 19:45.01 | *** join/#asterisk jkimball4 (n=jerrid@ip24-252-32-248.om.om.cox.net) |
| 19:45.30 | jkimball4 | Can someone give me an idea of where the code which outputs reload information to AMI would be located at? |
| 19:46.10 | JunK-Y | jkimball4: manager.c |
| 19:46.18 | [TK]D-Fender | whhheeEEEeeee: And why does iax matter? |
| 19:46.31 | [TK]D-Fender | whhheeEEEeeee: There are no decent IAX devices. |
| 19:46.36 | jkimball4 | Okay, thanks JunK-Y. |
| 19:46.51 | ManxPower | And no, I don't consider the IAXy to be "decent". |
| 19:47.02 | [TK]D-Fender | ManxPower: Yup, those Atcom phones are GARBAGE indeed.... |
| 19:47.09 | whhheeEEEeeee | [TK]: seems like the iax protocol is better to go with. uniformity. |
| 19:47.11 | [TK]D-Fender | ManxPower: Ditto.... |
| 19:47.20 | [TK]D-Fender | ManxPower: makes me shudder... |
| 19:47.46 | CCFL_Man2 | oh damn, the pci version of the design uses BGA |
| 19:47.50 | CCFL_Man2 | i think |
| 19:47.51 | ManxPower | whhheeEEEeeee: if it was universally supported then you would be correct. |
| 19:48.14 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:48.47 | Yourname` | I saw somewhere the other day that I can set the load avg as the prompt in CLI by changing the ASTERISK_PROMPT env variable to %11, but it's still not working. What am I doing wrong? |
| 19:48.52 | whhheeEEEeeee | so, what codec is gsm? does it have a number? or just gsm. |
| 19:49.21 | ManxPower | Yourname`: What version of Asterisk did "somewhere" say this applied to? |
| 19:49.23 | [TK]D-Fender | CCFL_Man2: Well you could (poorly) build a (crap) low-end Zapata compatible card (if your time is worth the risk) |
| 19:49.53 | [TK]D-Fender | whhheeEEEeeee: You are quickly branding yourself a "cheap bastard" |
| 19:49.55 | [TK]D-Fender | ~sheap |
| 19:49.59 | [TK]D-Fender | ~cheap |
| 19:49.59 | jbot | i heard cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 19:50.09 | [TK]D-Fender | ~ygwypf |
| 19:50.10 | jbot | methinks ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 19:50.12 | Yourname` | ManxPower: 1.4 and I'm using 1.4 |
| 19:50.21 | Yourname` | ManxPower: It's in /doc/cliprompt |
| 19:51.25 | Sweeper | [TK]D-Fender: sip show peers is showing NAT as N, but it's set to yes in the config file. any ideas? |
| 19:51.47 | Netgeeks | sometimes you get more than what you paid for... unfortunately the more generally results in a trip to the doctor... |
| 19:52.16 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 19:53.40 | J4k3 | haha yep |
| 19:54.12 | CCFL_Man2 | [TK]D-Fender: the one on the zapata sit is crap and low end? |
| 19:54.48 | CCFL_Man2 | the isa one |
| 19:54.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:54.58 | [TK]D-Fender | CCFL_Man2: Because the PCI bus spec, clocking design, lack of EC, voltage handling maybe... |
| 19:55.17 | CCFL_Man2 | i mean the isa one |
| 19:55.20 | [TK]D-Fender | CCFL_Man2: ISA, LOL, so you can deal with echo problem on the SLOWEST systems possible. |
| 19:55.32 | CCFL_Man2 | heh |
| 19:55.50 | [TK]D-Fender | CCFL_Man2: You must love pain...... |
| 19:55.57 | CCFL_Man2 | so you are saying both suck? |
| 19:56.08 | a1fa | [TK]D-Fender : thanks for your help today |
| 19:56.13 | whhheeEEEeeee | [TK]: how so? |
| 19:56.22 | a1fa | bye |
| 19:56.35 | whhheeEEEeeee | [TK]: I'm new to all this, I'm only going off what I've run across so far. :P |
| 19:57.32 | fetcher | anyone messed with FOP (Flash Operator Panel) much? |
| 19:57.35 | [TK]D-Fender | a1np |
| 19:57.40 | jingles | fetcher: yes. |
| 19:58.09 | [TK]D-Fender | whhheeEEEeeee: You're aiming for ATA's, IAX2 devices, and GSM all in such a nice short stretch. |
| 19:58.27 | fetcher | I can't get it to actually update anything. All extensions & lines just flash red/green constantly |
| 19:58.32 | Sweeper | [TK]D-Fender: sip show peers is showing NAT as N, but it's set to yes in the config file. any ideas? |
| 19:58.58 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 19:59.07 | whhheeEEEeeee | [TK]: how does that make me cheap? actually, probably nixing the ata's and going with the polycom. :) |
| 19:59.32 | [TK]D-Fender | Sweeper: SHOW me something useful. |
| 19:59.35 | danielxpt | ManxPower: I just checked with the guys providing the pri. There is no blocks on it at all. |
| 20:00.45 | jingles | fetcher: you've got to have the manager account connecting, and the op_server.pl running (most often in daemon mode, using the -d switch) |
| 20:00.55 | Sweeper | [TK]D-Fender: http://pastebin.ca/667199 |
| 20:01.09 | jingles | fetcher: on the FOP site, the setup instructions are very, very straightforward. |
| 20:01.19 | jingles | just build one button, and make sure it works. |
| 20:01.29 | jingles | all too often, you build 50 buttons, and forgot to do one thing right. |
| 20:01.42 | danielxpt | my pri debug is at http://pastebin.ca/667202 |
| 20:02.13 | fetcher | jingles: yup, I did both, and running op_server.pl in debug mode shows it connecting and getting data (it does have errors querying for nonexistent SCCP extensions... hopefully that isn't causing other things to fail) |
| 20:02.14 | [TK]D-Fender | Sweeper: I'd have to see the register attempt |
| 20:02.42 | Sweeper | [TK]D-Fender: they register fine |
| 20:02.53 | Sweeper | but then go unreachable |
| 20:02.53 | fetcher | jingles: yeah, it's probably a config issue in by op_buttons.cfg |
| 20:03.06 | Sweeper | EXCEPT for the SPA2002 that is using 116 |
| 20:03.22 | CoffeeIV_ | I am using asterisk 1.4.10 on Fedora Core 7, and I have the LumenVox speech recognition thing. when my asterisk starts and tries to load res_speech_lumenvox.so, it freezes up, no messages in /var/log/asterisk/full or anything. Has anyone on here seen similar ? |
| 20:03.59 | [TK]D-Fender | Sweeper: thats a PUBLIC IP phone? |
| 20:04.25 | Sweeper | nope! |
| 20:04.28 | Sweeper | they're all behind NAY |
| 20:04.30 | Sweeper | *NAT |
| 20:04.44 | [TK]D-Fender | Sweeper: Ok well as I said, I'd have to see it register. |
| 20:04.52 | Sweeper | fine~ |
| 20:05.01 | Sweeper | you want the debug? |
| 20:05.05 | fetcher | This FOP at least seems less of a nightmare than HUD-Lite was |
| 20:05.13 | [TK]D-Fender | Sweeper: and their NAT router could be a flaming piece of shit and be killing SIP (Cisco PIX, bunch 'o' D-Links, tec) |
| 20:05.43 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 20:05.50 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 20:06.06 | Sweeper | mkay |
| 20:06.11 | whhheeEEEeeee | I used to love linksys. ... then cisco bought them. |
| 20:06.23 | ManxPower | CoffeeIV_: "asterisk -cvvv" |
| 20:06.26 | simprix | . |
| 20:06.27 | Sweeper | yea, it's def something on their network, I'm pretty sure |
| 20:06.38 | CoffeeIV_ | ManxPower: trying it . . . |
| 20:07.15 | fetcher | whhheeEEEeeee: yeah, I've yet to see any case where product quality or service actually *improved* after a merger, regardless of companies involved ;/ |
| 20:08.50 | *** part/#asterisk jkimball4 (n=jerrid@ip24-252-32-248.om.om.cox.net) |
| 20:08.55 | CoffeeIV_ | ManxPower: that gives mesome additional error out put -- it says "terminate called after throwing an instance of 'std::logic_error'" then "what(): basic_string::_S_construct NULL not valid" and "Aborted" |
| 20:09.01 | [TK]D-Fender | whhheeEEEeeee: Linksys works fine. |
| 20:09.13 | ManxPower | CoffeeIV_: now you see the error |
| 20:10.04 | danielxpt | so does anyone have any idea of what I can try to troubleshoot this failure? |
| 20:10.29 | ManxPower | danielxpt: call your carrier |
| 20:10.54 | ManxPower | danielxpt: what is the value of hangupcause |
| 20:10.57 | danielxpt | ManxPower: I work for the carrier. the guys maintaining the switch is two offices down. We just checked the PRI. |
| 20:11.27 | CoffeeIV_ | ManxPower: now I see the error . . . how do I fix it ? Does that indicate a problem in lumen.conf ? These confiigs work with other asterisk / lumenvox servers |
| 20:11.35 | danielxpt | ManxPower: Channel 0/1, span 1 got hangup, cause 3 |
| 20:11.35 | ManxPower | danielxpt: then it should be pretty easy for the tech to hook up a T-BERD and try calling the problem number. |
| 20:11.53 | danielxpt | ManxPower: I go down to our COLO to do that tomorrow morning |
| 20:12.09 | ManxPower | danielxpt: paste the ACTUAL Dial line as shown on the console (and JUST the Dial line) |
| 20:12.42 | danielxpt | ManxPower: exten => _1800NXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 20:13.05 | ManxPower | that is not the CLI output. |
| 20:13.32 | ManxPower | danielxpt: your carrier says it cannot connect to the destination. It is a CARRIER PROBLEM, regardless if what they say |
| 20:13.37 | ManxPower | Either that or your test number is not valid |
| 20:13.54 | danielxpt | ManxPower: Executing [18003104481@trunk-sip:1] Dial("SIP/astk1.asterisk.xfoneusa.net-09665160", "Zap/g1/18003104481") in new stack |
| 20:14.19 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 20:14.23 | ManxPower | good gawd, you are on XFone?????????????? |
| 20:14.38 | ManxPower | where are you located anyway? |
| 20:14.40 | danielxpt | lol, yea i've been with them for 3 years |
| 20:14.42 | danielxpt | Jackson MS |
| 20:15.08 | ManxPower | We were with I-55. I've been pretty unimpressed with them since their merget with XFone. |
| 20:15.17 | file | CoffeeIV_: you do not have the Lumenvox environment variables set |
| 20:15.32 | ManxPower | Our multilink T-1 went down. Took them a week to find anything wrong on their end. |
| 20:15.39 | danielxpt | yea it got real rough here. I didn't agree with 90% of what they did during that time. |
| 20:15.44 | file | (that's what the construct message is about) and as for freezing that is a known issue being worked on |
| 20:16.00 | danielxpt | it's gotten better in the last two months |
| 20:16.19 | ManxPower | danielxpt: since Katrina we have moved many of our PRIs away from XFone. |
| 20:16.31 | ManxPower | and I'm starting to think we should move other services off of them too. |
| 20:16.50 | danielxpt | ManxPower: I don't blame you. Small company budget with Big company wants |
| 20:17.36 | lirakis | im getting a cause code 100 about the upstream switch kicking back on a invalid "Information element" when trying to send a call out ZAP on a T1. I have properly set my callerid .. and i am uncertain how else to set a "information element" in asterisk .. any hints or ideas? |
| 20:17.45 | ManxPower | danielxpt: XFone does not have a route built for 800 numbers for your PRI. |
| 20:17.48 | Yourname` | Anddd ManxPower vanished after my question and answer round. :P |
| 20:17.53 | danielxpt | ManxPower: both i-55 and xfone was doing things halfa**. |
| 20:18.06 | *** join/#asterisk ^majik^ (n=x@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 20:18.07 | ManxPower | lirakis: show me the line where you set the CALLERID |
| 20:18.25 | ManxPower | Yourname`: you do too much asking and not enough reading. |
| 20:18.36 | *** join/#asterisk Mozzer (n=linux@87.127.154.207) |
| 20:18.41 | Mozzer | hello |
| 20:18.45 | Yourname` | ManxPower: That's what I said, lol.. I "read somewhere".. and I was asking why isn't it working :D |
| 20:18.51 | lirakis | ManxPower: exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Bob Fake <1234567890>) |
| 20:19.08 | ManxPower | lirakis: you are using 1.4? |
| 20:19.14 | lirakis | ManxPower: 1.2 |
| 20:19.21 | ManxPower | that should still work in 1.2 |
| 20:19.25 | Nugget | I don't have time to read the wiki! You should paste it into the channel so I can read the same words here on irc! Somehow that's easier. |
| 20:19.25 | danielxpt | ManxPower: I hear ya. I will test it out tomorrow with the TBERD. I personally hope it shows the exact same thing as your discussing. I can't see anything else it could be |
| 20:19.30 | CoffeeIV_ | file: I did a fresh login, ran "env" to make sure the lumenvox env's were set, and then the same hangup happened -- this time without the std::logic_error though |
| 20:19.43 | danielxpt | ManxPower: so if I may ask, who are you with? |
| 20:19.45 | ManxPower | lirakis: some carriers do not let you set your CLID, some of them make you set it to the main number of your line |
| 20:20.09 | CoffeeIV_ | but I have gotten a response from someone on the asterisk-speech-res list that indicates this is known and they are working on it, so I will just be patient |
| 20:20.09 | lirakis | ManxPower: tried without any callerid and it complaine |
| 20:20.27 | lirakis | ManxPower: the D channel shows i was sending 1etamme as a information element |
| 20:20.37 | lirakis | ManxPower: i have no idea how that got set... ? |
| 20:20.56 | ManxPower | I don't know what the 1etamme IE is. |
| 20:21.40 | syzygyBSD | is there a way to specify a directory other than the root of tftp for the 0004......-phone.cfg file for polycom phones |
| 20:22.04 | ^majik^ | hey guys, we're having the occasional problem where (most noticeably) inbound calls will get dropped after a random amount of time. when the call drops, it goes to a busy signal. here is the relevant log info: http://rafb.net/p/T2hUwo66.html |
| 20:22.05 | syzygyBSD | I want to keep my files as clean as possible, so I would love to push everything into a polycom sub dir |
| 20:22.09 | ManxPower | syzygyBSD: yes in the 0004...cfg |
| 20:22.26 | ManxPower | the MAC.cfg has to be in the root of the TFTP dir. |
| 20:22.29 | syzygyBSD | ManxPower: what if I use the default 0000000... cfg? |
| 20:22.38 | Sweeper | syzygyBSD: or use HTTP, and you can just set a subdirectory :) |
| 20:22.40 | ManxPower | We never use TFTP provisioning, we use the FTP provisioning. |
| 20:22.55 | syzygyBSD | ya, configs shoudl be close to the same |
| 20:23.30 | syzygyBSD | I am trying to setup an auto provisioning system, auto detects when there is a new phone, and sets it up in the system based on the mac |
| 20:24.19 | ManxPower | good luck with that |
| 20:24.19 | *** part/#asterisk fas3r (n=chatzill@haz95-2-82-243-75-212.fbx.proxad.net) |
| 20:24.31 | syzygyBSD | wont' be hard, almost have it done |
| 20:26.43 | deeperror | is there any way to talk with both parties using attended transfer? |
| 20:27.07 | syzygyBSD | hardest part was figuing out I had two dhcp servers running on my network, and the phones didn't like that |
| 20:27.16 | Sci_05 | I was just looking at voip-info and in 03 they say that iax2+trunk=yes is a bad thing, is this still true or did they fix that? |
| 20:27.31 | Yourname` | Nugget: No, I read the wiki too. I even read the doc. Now, if it doesn't work.. it doesn't mean I didn't read, right? |
| 20:29.02 | syzygyBSD | oh, in 0000000000.cfg, you can set the OVERRIDE_DIRECTORY to move where all the other .cfg files are located, if anyone was interested |
| 20:30.38 | *** join/#asterisk davixx (n=davixx@ASt-Lambert-151-1-63-4.w83-199.abo.wanadoo.fr) |
| 20:32.08 | ManxPower | deeperror: that is called a "three-way call" |
| 20:32.20 | *** join/#asterisk oej (n=olle@212.17.142.154) |
| 20:32.42 | deeperror | manxpower: yes i know but am having issues with the flash button when trying to make 3 way calls...it gives me a very odd error unable to find anything on it |
| 20:33.00 | deeperror | http://pastebin.ca/667110 |
| 20:34.21 | deeperror | i've been on this for about 14 hours now and still unable to make a 3 way call |
| 20:34.31 | *** join/#asterisk ToyMan (n=Stuart@user-160uamh.cable.mindspring.com) |
| 20:35.08 | ManxPower | deeperror: you do not have a correct /etc/asterisk/indications.conf |
| 20:35.28 | deeperror | how so? |
| 20:35.57 | ManxPower | deeperror: the error message you are getting happens when you do not have a correct /etc/asterisk/indications.conf |
| 20:36.16 | deeperror | are you sure on that and where did you find that? |
| 20:36.26 | ManxPower | deeperror: personal experience |
| 20:36.47 | deeperror | i see nothing in indications.conf that refers to hook flash |
| 20:36.55 | ManxPower | specifically indications.conf specifies the tones and codes used to provide tones and sounds when a call is already off hook. |
| 20:37.28 | ManxPower | You would have similar errors if you tried running the Ringing application, Congesion application or any other application that tries to use inband sounds |
| 20:37.47 | deeperror | so a flash button is tones your telling me? |
| 20:37.48 | ManxPower | well inband INDICATIONS at least |
| 20:38.21 | ManxPower | deeperror: When you press FLASH, asterisk has to play a dialtone or stutter tone. It can't do that without indications.conf and so fails. |
| 20:38.32 | ManxPower | either take my advice or not. I'm not going to argue about it. |
| 20:38.54 | deeperror | i'm not trying to argue...i do have indications.conf |
| 20:38.57 | deeperror | default |
| 20:39.39 | deeperror | http://pastebin.ca/667235 |
| 20:39.53 | ManxPower | "Unable to handle indication 9" usually means indications.conf does not exist. |
| 20:39.54 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.172.179) |
| 20:40.36 | ManxPower | deeperror: you are using a SIP phone? |
| 20:40.44 | deeperror | if you google that my pastebin comes up on top results |
| 20:40.49 | deeperror | i don't think its a very standard error |
| 20:40.56 | deeperror | analog phone |
| 20:41.00 | deeperror | channel bank |
| 20:41.04 | deeperror | t1 card in * box |
| 20:41.07 | ManxPower | connected to a Digium card |
| 20:41.14 | ManxPower | ah connected to a CHANNEL BANK |
| 20:41.22 | deeperror | rhino |
| 20:41.29 | ManxPower | and the T-1 card is..... |
| 20:41.36 | deeperror | rhino r4t1 |
| 20:41.53 | deeperror | i can make calls in and out |
| 20:42.01 | *** join/#asterisk y7n (n=na@87-194-205-33.bethere.co.uk) |
| 20:42.02 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:42.10 | deeperror | good quality sound...attended xfers work as intended |
| 20:42.16 | deeperror | just the flash errors |
| 20:42.24 | *** join/#asterisk lbow (n=lbow@41-195-77-196.access.uunet.co.za) |
| 20:42.53 | deeperror | i have threewaycalling enabled on zapata.conf and have played with flash settings in there also rxflash etc no joy as of yet |
| 20:43.03 | ManxPower | how do you do attended xfers if you are not using flash |
| 20:43.10 | ManxPower | you need transfer=yes as well |
| 20:43.14 | Op3r | anyone knows any UK voip provider that offers UK DID? |
| 20:43.23 | deeperror | using features.conf and [featuremap] |
| 20:43.38 | deeperror | didww.com |
| 20:43.44 | ManxPower | I always just used FLASH to do that |
| 20:44.00 | y7n | When a certain call hangups a function h,1,System(/home/foo.sh) is executed. foo.sh is meant to move a file into the asterisk outgoing spool. But nothing seems to happen. Any ideas? |
| 20:44.10 | ManxPower | featuremap for DTMF transfers is an ugly hack for devices too stupid to support transfer on their own |
| 20:44.15 | deeperror | y7n: permissions |
| 20:44.23 | y7n | permissions on the file? |
| 20:44.53 | y7n | i temporarily chown'd the scripts and call files to 777, that didn't seem to work |
| 20:44.58 | deeperror | y7n: script will execute with permissions * is running under |
| 20:45.26 | ManxPower | it should be pretty easy to become the asterisk user in a shell and run the command manually |
| 20:45.28 | anonymouz666 | is there a way to group the peers? for example group=10,12,23,35,26 and then GotoIf($[ "${EXTEN}" = ${group}]) ? |
| 20:45.49 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 20:45.55 | y7n | hmm |
| 20:46.07 | ManxPower | anonymouz666: your question makes no sense. |
| 20:46.13 | ManxPower | and extension is not a peer |
| 20:46.39 | southtel | Does anyone have experience with users not being able to enter extensions consistently? |
| 20:46.41 | ManxPower | if EXTEN was equal to "10,12,23,35,26" then that gotoif would work |
| 20:47.00 | ManxPower | I don't know how a user would dial a ",", however |
| 20:47.09 | anonymouz666 | forget it. |
| 20:47.11 | southtel | We have a conference machine set up, and we've got users complaining that they can't consistently enter their PINs. |
| 20:47.42 | ManxPower | southtel: relaxdtmf set to yes would do that. So would too high/low gains on your zaptel channels |
| 20:48.35 | deeperror | manxpower: any idea how to define flash in indications or to work around that error? |
| 20:48.42 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 20:48.54 | anonymouz666 | ManxPower group means extensions separated by , and these extensions are peers. |
| 20:49.01 | ManxPower | deeperror: FLASH is not the problem. the stutter dialtone that is tried to be played after the flash is the provlem |
| 20:49.20 | anonymouz666 | so instead check against every case, I want to do one time against a group |
| 20:49.25 | ManxPower | anonymouz666: good luck with that. |
| 20:49.38 | southtel | ManxPower: so I should or should not have relaxdtmf=yes ? |
| 20:49.56 | ManxPower | southtel: you should not |
| 20:50.05 | ManxPower | it. causes. problems. |
| 20:50.11 | *** join/#asterisk telmich (i=telmich@gpm/telmich) |
| 20:50.31 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 20:50.34 | telmich | could someone do me a favor and call sip:2001@home.schottelius.org? I'm wondering whether it works or not (should be a conference) |
| 20:50.36 | southtel | ManxPower: Gotcha. |
| 20:54.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:55.11 | rudholm | anyone using teliax should reconcile their asterisk logs with their teliax call history. they seem to be having billing software problems |
| 20:55.23 | deeperror | ManxPower: stutter dt is the issue how do i correct this or what do I need to look into more to determine what this should be? I don't see much documentation on that particular value |
| 20:55.37 | telmich | wow, however that was, it sounded pretty bad |
| 20:55.46 | deeperror | i've also removed this line from indications and still have same errors |
| 20:57.15 | De_Mon | I've got a call where I set SPYGROUP=2000 and then execute retryDial(), when the call connects the SPYGROUP variable gets removed! whats up with that? |
| 20:57.36 | De_Mon | this is on * 1.4.6 |
| 20:58.31 | Corydon76-vcch | What makes you say the variable is removed? |
| 20:58.47 | syzygyBSD | I wonder what some people would think if they saw my search history, |
| 20:59.00 | syzygyBSD | "man find" |
| 20:59.11 | Corydon76-vcch | syzygyBSD: pervert? |
| 20:59.13 | Sci_05 | anyone know if they fixed iax2 jitter buffer +trunk=yes so it works? |
| 21:00.28 | y7n | how do I find out what my asterisk user is, Im currently running asterisk under root |
| 21:00.29 | syzygyBSD | Corydon76-vcch: more disturbed then perverted, definitely couldn't form a real sentance |
| 21:01.38 | deeperror | ps -aux |
| 21:01.56 | syzygyBSD | wrong window? |
| 21:05.25 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-94-59.dslextreme.com) |
| 21:05.46 | y7n | asterisk is running under root and yet the script still doesn't execute, if i enter the script in the shell it works fine :-s |
| 21:06.06 | deeperror | are you logged in as root when you run the script? |
| 21:06.10 | y7n | yes |
| 21:06.20 | deeperror | chmod +x script.sh |
| 21:09.50 | y7n | nope, that didnt' work :S |
| 21:11.27 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 21:13.36 | *** join/#asterisk oej (n=olle@212.17.142.154) |
| 21:14.32 | *** join/#asterisk YoYo (n=chatzill@pigpen.office.psknet.com) |
| 21:15.07 | syzygyBSD | y7n: are you using an absolute path to the script? |
| 21:15.30 | syzygyBSD | and is the first line #!/bin/bash? |
| 21:15.47 | YoYo | is there any distro with a reasonable ports/package management system that's at least half as good as the FreeBSD ports system? (other than gentoo's portage) |
| 21:15.59 | syzygyBSD | YoYo: debian |
| 21:16.12 | syzygyBSD | aptitide |
| 21:16.33 | y7n | yes i am |
| 21:16.41 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:16.41 | YoYo | syz: so, with debian, there's a way I can build a package, say sox w/mp3 support, and have it in the package management system? |
| 21:16.45 | syzygyBSD | it is rare I need anything that isn't in the package manager |
| 21:17.05 | syzygyBSD | YoYo: you don't build on debian... you dont' need to |
| 21:17.09 | y7n | first line is #!/bin/bash |
| 21:17.29 | syzygyBSD | apt-get install sox |
| 21:17.30 | mog | lies |
| 21:17.37 | y7n | it works fine when running the script from a shell |
| 21:17.43 | YoYo | syz: do that and then run sox -h |
| 21:17.53 | YoYo | default build for sox doesn't include mp3 support |
| 21:18.07 | y7n | but it doesn't work when using System(/home/bla/foo.sh) |
| 21:18.19 | syzygyBSD | Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud mp3 (readonly) nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve |
| 21:18.37 | YoYo | readonly |
| 21:18.39 | YoYo | brb |
| 21:18.49 | y7n | i can see it being executed in the asterisk console when using verbose |
| 21:20.14 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 21:21.11 | syzygyBSD | y7n: do all the commands have absolute paths in the script? |
| 21:21.19 | y7n | yes |
| 21:21.32 | jhiver_ | what does IAX2/... stopped sounds mean? |
| 21:21.33 | syzygyBSD | hmm, I dun know then |
| 21:21.43 | y7n | thanks for your help though |
| 21:21.47 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 21:21.57 | jhiver_ | i have this |
| 21:21.58 | jhiver_ | <PROTECTED> |
| 21:22.08 | jhiver_ | doesn't sound very good |
| 21:22.16 | jhiver_ | when i test it all works though |
| 21:22.24 | jhiver_ | this with IAX2 trunking |
| 21:22.33 | jhiver_ | (using ztdummy) |
| 21:24.05 | jhiver_ | i also have this: |
| 21:24.06 | jhiver_ | [Aug 23 01:23:50] WARNING[2705]: channel.c:2612 ast_indicate_data: Unable to handle indication 3 for 'SIP/5060-081b01e0' |
| 21:24.08 | [TK]D-Fender | jhiver : ZTDUMMY isn't great as a timing source for IAX trunking |
| 21:24.11 | jhiver_ | sounds scary |
| 21:24.51 | jhiver_ | Yeah, but the box is rented in a data center far, far away, so little chance of having a hardware timing device in there =( |
| 21:25.01 | [TK]D-Fender | jhiver : 2nd warning is just where IAX2 & SIP have different call-state notification capabilities and one is sending a message that has no interpretation on the other |
| 21:25.26 | jhiver_ | ok so it's an interworking problem? |
| 21:25.44 | jhiver_ | is it something relatively benign? |
| 21:26.26 | Sci_05 | [TK]D-Fender: then what is something that should used to iax2 trunking timing source? |
| 21:26.35 | jhiver_ | i get a lot of those as well... |
| 21:26.37 | jhiver_ | [Aug 23 05:26:21] NOTICE[6009]: frame.c:216 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD fra |
| 21:27.34 | [TK]D-Fender | Sci_05, I'd say a 1-port TDM400P |
| 21:28.29 | CoffeeIV_ | I have faxdetect=incomding in my zapata.conf, but I want to take one particular DID number, and route those faxes to different fax machine. Is there a way to turn off fax detectiong in my dialplan, for a particular number ? |
| 21:29.35 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:30.35 | [TK]D-Fender | CoffeeIV_, fax detection only jumps to "fax" if it EXISTS in a context with an ivr thats listening for input. so clearly all you have to do is use an ivr that doesn't HAVE " exten => fax," in it. |
| 21:31.09 | jhiver_ | mhhhh |
| 21:31.14 | jhiver_ | lots of wierdness |
| 21:31.26 | jhiver_ | i wonder wether this IAX2 trunking thing is gonna work |
| 21:31.45 | jhiver_ | the |
| 21:31.45 | jhiver_ | <PROTECTED> |
| 21:31.51 | Sci_05 | [TK]D-Fender: do you know if they fixed the iax2 jitter buffer + trunk=yes problem? |
| 21:31.53 | jhiver_ | message scares me |
| 21:32.09 | jhiver_ | sounds like there's no more audio or something |
| 21:32.19 | jhiver_ | but pehaps it means something different alltogether |
| 21:32.45 | CoffeeIV_ | [TK]D-Fender: thanks that makes a lot of sense |
| 21:33.01 | jhiver_ | i've done trunkfreq=40, assuming it would fit 2 g729 frames per frame, is this a good value? |
| 21:33.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:33.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:38.02 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 21:39.19 | De_Mon | is RetryDial on the road to depriciation? it seems like i'm always finding bugs in it |
| 21:39.58 | *** join/#asterisk Drew__ (n=foo@zux221-160-177.adsl.green.ch) |
| 21:40.03 | Drew__ | hello |
| 21:40.15 | JunK-Y | De_Mon: not that i know. |
| 21:40.58 | Drew__ | i am trying to install *, and have some problems compiling zaptel - i allways get an "Error 2" :( |
| 21:41.08 | De_Mon | :| okay I'm gonna document and report this bug then... |
| 21:41.28 | Drew__ | can somebody point me in the right direction with this? |
| 21:41.54 | JunK-Y | Drew__: full pastebin of the error? |
| 21:42.18 | jhiver_ | ok well, off to bed, i guess i'll have to closely check QoS tomorrow =) |
| 21:42.21 | jhiver_ | cya all |
| 21:42.31 | jhiver_ | [TK]D-Fender, thanks again for all your help |
| 21:43.25 | Drew__ | JunK-Y - http://pastebin.ca/667307 |
| 21:44.08 | Drew__ | JunK-Y - i untared it, did ./configure, then make clean - and this error keeps apearing |
| 21:44.49 | JunK-Y | and with a make, not make clean? |
| 21:45.26 | syzygyBSD | Drew__: you need your kernel headers, what distro are you using? |
| 21:45.44 | syzygyBSD | debian.. |
| 21:45.54 | Drew__ | JunK-Y same error with just make |
| 21:46.04 | [TK]D-Fender | Drew__, Go check for that folder yourself |
| 21:46.06 | Drew__ | syzygyBSD - debian4 |
| 21:46.10 | syzygyBSD | Drew__: apt-get install linux-headers-2.6.18-4-686 |
| 21:46.22 | JunK-Y | u should have a warning message for that. |
| 21:46.47 | syzygyBSD | hmm, looks like you need linux-headers-2.6.18-5-686 though |
| 21:47.17 | syzygyBSD | then make clean again |
| 21:47.33 | JunK-Y | when u do a simple make, do dont get any warning message? |
| 21:47.45 | Drew__ | syz - they are already there |
| 21:48.49 | Drew__ | syzygyBSD - http://pastebin.ca/667312 <- i already had the headers |
| 21:49.36 | Drew__ | JunK-Y the error appears for make clean and make |
| 21:51.03 | JunK-Y | and you have a dir /usr/src/linux-headers-2.6.18-5-686/ ? |
| 21:51.30 | Drew__ | yes |
| 21:51.49 | JunK-Y | apparently /usr/src/linux-headers-2.6.18-5-686/Makefile isnt there. |
| 21:52.01 | Drew__ | how do i fix that? |
| 21:52.44 | JunK-Y | u sure /usr/src/linux-headers-2.6.18-5-686/Makefile is there? |
| 21:53.10 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 21:54.49 | Drew__ | JunK-Y - it exists but is empty if i open it in an editor |
| 21:55.15 | JunK-Y | thats the problem |
| 21:55.37 | Drew__ | how can i fix that? |
| 21:55.54 | Drew__ | delete the folder and apt-get it again? |
| 21:56.26 | JunK-Y | purge the package yes, |
| 21:56.34 | JunK-Y | not only deleted, purge it |
| 21:57.15 | Drew__ | purge? |
| 21:57.41 | JunK-Y | purge the debian package |
| 21:57.47 | JunK-Y | dpkg --purge |
| 22:00.20 | Drew__ | i did a purge, and got the package again - the makefile is still empty |
| 22:00.37 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 22:00.44 | whywontitwork | hi there |
| 22:02.28 | JunK-Y | Drew__: thats a problem a problem in the package, nothing we can do. |
| 22:02.37 | JunK-Y | get a proper kernel with proper headers. |
| 22:04.00 | whywontitwork | anyone? need some help with variable uniqueid, i use it to asign unique id for incoming fax files,,, it does assign a uniqueID but like this "berlioz-5279-1187815260.16FAX.tif" how do I get rid of the "berlioz"?? any help welcome been scratching my head for hours?? |
| 22:04.23 | Drew__ | JunK-Y - ok thanks |
| 22:04.58 | jingles | ok, got a strange question. |
| 22:05.00 | *** join/#asterisk Drew__ (n=foo@zux221-160-177.adsl.green.ch) |
| 22:05.13 | jingles | when you go to hit a dialing queue... |
| 22:05.31 | jingles | I've got members you dial with SIP/00EXT, and IAX2/00EXT |
| 22:05.31 | *** join/#asterisk ToTo (n=ToTo@host156-89-dynamic.56-82-r.retail.telecomitalia.it) |
| 22:05.36 | jingles | but I've also got a PRI |
| 22:06.00 | jingles | can I use ${T1TRUNK}/1XXNXXXXXXX in a queue? |
| 22:06.29 | jingles | where ${T1TRUNK} is the entry we use in the normal dialplan for dialing out through the PRI? |
| 22:08.11 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 22:08.52 | whywontitwork | anyone? need some help with variable uniqueid, i use it to asign unique id for incoming fax files,,, it does assign a uniqueID but like this "berlioz-5279-1187815260.16FAX.tif" how do I get rid of the "berlioz"?? any help welcome been scratching my head for hours?? |
| 22:09.46 | Drew__ | on a different note, does anybody know something about CLIP? |
| 22:10.09 | Drew__ | is there a scenario where you can get incoming calls on an analog line showing your own callerID? |
| 22:11.06 | whywontitwork | more info needed drew |
| 22:11.48 | whywontitwork | s,1,Answer() |
| 22:12.01 | Drew__ | whywontitwork - what kind of info? |
| 22:12.37 | jingles | ok. well, that answers my question. |
| 22:12.47 | whywontitwork | s,2,Set([${CALLERIDNUMBER} = your number)] Drew |
| 22:12.50 | jingles | No channel type registered for ${T1TRUNK} |
| 22:13.16 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:13.35 | Drew__ | whywontitwork - no thats not what i mean - no asterisk in between, i am getting incoming calls from the phone line showing my own number |
| 22:14.10 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:16.01 | whywontitwork | where does asterisk fit into the picture? |
| 22:16.32 | CCFL_Man2 | whywontitwork: why won't it work? |
| 22:17.31 | whywontitwork | why wount what work CCFL_Man2? |
| 22:17.44 | whywontitwork | sorry won't |
| 22:18.10 | Drew__ | whywontitwork - nothing to do with asterisk really, more with my phone provider, or somebody spoofing CID's |
| 22:20.18 | *** join/#asterisk Tako-san (n=Tako-san@d64-180-208-148.bchsia.telus.net) |
| 22:21.19 | whywontitwork | so you saying when someone phones you on your landline it displays your caller id ? |
| 22:21.38 | Drew__ | yes |
| 22:22.31 | Drew__ | well normally it works - all the callers show their own CID - but i am getting some calls that show my own callerID, when i answer, nobody's there.... :/ |
| 22:22.31 | Tako-san | Any recommendations on a ruggedized wireless phone that would function well wish Asterisk? I have looked at EnGenius and Senoa already. Comments about their products or other suggestions welcome (and appreciated). :) |
| 22:22.35 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 22:22.48 | Tako-san | wish=with |
| 22:23.33 | Primer | Anyone have an example for extensions.conf that detects when a registered client dials "sip:user@domain" when using asterisk as the outgoing proxy, and then just dials the sip URL? |
| 22:23.47 | whywontitwork | do you have a bri line? |
| 22:24.02 | Drew__ | whywontitwork - who? me? |
| 22:24.30 | whywontitwork | yes |
| 22:24.49 | Drew__ | no, its a POTS line |
| 22:25.16 | ToTo | what is the most advanced book in contents? Asterisk (2nd) or Asterisk Cookbook? |
| 22:25.18 | whywontitwork | looks like you got a crossed line |
| 22:25.20 | Drew__ | hooked up via a W-X100P |
| 22:25.33 | Drew__ | what is a crossed line? |
| 22:25.51 | whywontitwork | two cicuits joined |
| 22:26.37 | Drew__ | what two cicuits? how does that happen? |
| 22:26.51 | whywontitwork | does anyone know if the new asterisk book for version 1.4 is out yet? |
| 22:27.22 | whywontitwork | your telco might have linked your line with the guy next door |
| 22:29.21 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:29.37 | ToTo | whywontitwork: not at the moment, see www.play.com |
| 22:29.45 | *** part/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:30.16 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:31.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:34.31 | JT | Drew__: W-X100P eh, wow people keep making new model numbers for those things |
| 22:35.05 | Drew__ | JT - i was just too lazy to write out wildcard |
| 22:35.24 | Drew__ | JT - i assume you understood what i meant |
| 22:35.34 | JT | X100P would suffice, no-one writes wildcard |
| 22:37.04 | Drew__ | ok |
| 22:37.10 | Drew__ | i'll try to remember ;) |
| 22:37.48 | JT | that said, the chances of it actually being an X100P are slim if it was purchased anytime in the last couple of years |
| 22:44.28 | whhheeEEEeeee | ok, this is confusing me to no end. I've got this debug message coming to me and I can't figure out why: "request '<my number>'@default does not exist" |
| 22:44.42 | whhheeEEEeeee | the default extension does exist in the .conf |
| 22:46.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:48.07 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 22:49.07 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 22:49.12 | fujin | odd |
| 22:49.35 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 22:49.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:54.06 | *** part/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 22:54.07 | Qwell[] | whhheeEEEeeee: what "default extension"? |
| 22:55.23 | whhheeEEEeeee | Qwell[]: sorry, don't know what you mean |
| 22:56.21 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@64.149.29.208) |
| 22:58.10 | J4k3 | hmm, this tmo "pick any 5 numbers to have unlimited calling to" plan seems interesting |
| 22:59.34 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 22:59.41 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 23:04.32 | Trevor_b | infinity^2 |
| 23:05.00 | Trevor_b | eternity^3 |
| 23:06.55 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 23:07.38 | Trevor_b | forever, wait a little longer compile will finish |
| 23:07.42 | Trevor_b | doh, wrong room. sorry. |
| 23:08.05 | JT | "room"... cute |
| 23:08.33 | Trevor_b | channel, whatnot, all gets jumbled together in gaim for me at this point. |
| 23:08.53 | JT | eeewwww |
| 23:09.00 | JT | gaim is the worst for irc, ever |
| 23:09.06 | JT | it is not an irc client |
| 23:09.17 | JT | it's an IM client that emulates irc support VERY poorly |
| 23:09.46 | Trevor_b | yeah but i dont do anything but read and type, nothing more. Mainly its for support in IM for clients and whatnot, but simpler to keep irc in there as well while im busy during the day, dont have my third monitor anymore. |
| 23:10.23 | JT | too bad if you ever want to use any irc commands ;) |
| 23:10.35 | Trevor_b | thats why BitchX is still installed ;) |
| 23:11.05 | JT | that's also not a very good client |
| 23:11.07 | JT | irssi :) |
| 23:15.23 | fujin | irssi for the win! |
| 23:15.53 | Trevor_b | looks nice, but im sure the power is under the hood using the commands which I don't have the time to bother to read, which is why i dont even use BitchX anymore, i just type ;). |
| 23:16.12 | Trevor_b | IRC reduced to glorified IM |
| 23:17.06 | JT | i don't have time to be frustrated ;) |
| 23:17.19 | jingles | I ain't got time ta bleed. |
| 23:18.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 23:23.19 | fujin | anyone used the ip4000 polycoms? |
| 23:23.30 | fujin | thinking of getting one, I can get 25% of $1900 nzd |
| 23:23.33 | fujin | off* |
| 23:24.09 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 23:24.36 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 23:25.35 | CoolGuy21 | hi, is there any good tutorials on creating multi-tenant configs? so that i can have Company A with its own trunks and Company B with its own trunks. and they wont be able to dial each others extensions? |
| 23:26.04 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:27.07 | kiscokid | Turns out the reason I couldn't connect my Sangoma A101d to the new PRI was because the electricians cut my 25 pair house cable |
| 23:27.40 | russellb | you have the PRI going into your house? |
| 23:28.04 | kiscokid | noits our new office but the telco guy called it a house cable |
| 23:28.13 | kiscokid | no its* |
| 23:28.17 | russellb | ah. |
| 23:28.26 | russellb | jbot can fix your message foryou |
| 23:28.29 | russellb | s/foryou/for you/ |
| 23:28.34 | russellb | :) |
| 23:28.37 | kiscokid | ah |
| 23:31.15 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:32.49 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 23:33.18 | *** join/#asterisk OneWhoKnows (n=OneWhoKn@rh-la-32-99.rhythm.com) |
| 23:33.45 | OneWhoKnows | is anyone here familiar with PRIs? |
| 23:33.56 | jingles | familiar enough. |
| 23:33.59 | jingles | ask your question. |
| 23:34.07 | russellb | but you are ... the OneWhoKnows ... you should ... know! |
| 23:34.12 | russellb | sorry, couldn't resist. |
| 23:34.13 | kiscokid | somewhat, I just got one working today |
| 23:34.19 | OneWhoKnows | :( |
| 23:34.21 | OneWhoKnows | lol |
| 23:34.48 | remmo | what so hard about PRI's ? |
| 23:35.19 | kiscokid | they are fairly simple if you have the physical layer connected, as I found out |
| 23:35.30 | OneWhoKnows | actually, it's not about PRIs but more about T1 framing |
| 23:35.32 | remmo | lol would help |
| 23:35.57 | remmo | who uses T1's ? |
| 23:36.12 | OneWhoKnows | a PRI technicall is a T1 |
| 23:36.16 | OneWhoKnows | technically* |
| 23:36.26 | jingles | nothing technical about it. |
| 23:36.29 | OneWhoKnows | the better question would be who uses inband t1 |
| 23:36.33 | jingles | that is precisely what a PRI is. |
| 23:36.40 | russellb | o.O |
| 23:36.43 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 23:36.45 | russellb | one uses the other ... |
| 23:36.55 | fujin | we have a PRI circuit comign down an E1 |
| 23:36.59 | fujin | so technically, no |
| 23:37.02 | whhheeEEEeeee | so, I see the context listed in my extensions.conf, yet when inbound connections come in, its saying it doesn't exist? any suggestions? |
| 23:37.13 | asterisknerds | <PROTECTED> |
| 23:37.22 | fujin | whhheeEEEeeee: check the context your inbound calls are being sent into |
| 23:37.27 | fujin | in sip.conf/zapata.conf |
| 23:37.30 | OneWhoKnows | anyways, you know how D4 has 12 frames per superframe and ESF has 24 frames per superframe. my question is both are supposed to use the 8th bit every 6th frame for CAS (robbed bit signaling) |
| 23:37.34 | OneWhoKnows | right? |
| 23:38.02 | OneWhoKnows | on a PRI using ESF, since you're using a D-channel for out of band signaling, do the CAS bits still get reserved? |
| 23:38.09 | *** join/#asterisk l2trace9999 (n=l2trace@75.112.133.254) |
| 23:38.59 | kiscokid | fujin: inbound calls from the PRI would have their context defined in zapata.conf |
| 23:39.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:40.01 | whhheeEEEeeee | fujin: well, its being sent to the right context, it just is saying that the "default" context doesn't exist. |
| 23:40.40 | fujin | does it exists in asterisk? |
| 23:40.42 | fujin | show dialplan |
| 23:40.54 | russellb | dialplan show |
| 23:41.04 | whhheeEEEeeee | o.O |
| 23:41.33 | whhheeEEEeeee | yep, I see it. |
| 23:41.44 | whhheeEEEeeee | default created by pbx_config |
| 23:41.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:41.55 | OneWhoKnows | why would you want to use a [default] context extensions.conf? |
| 23:42.04 | whhheeEEEeeee | hell if I know know. lol |
| 23:42.06 | russellb | why wouldn't you? |
| 23:42.23 | whhheeEEEeeee | fujin: its set to 's' => that is correct, right? |
| 23:42.26 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 23:42.39 | OneWhoKnows | because... to prevent context confusion? |
| 23:42.44 | fujin | whhheeEEEeeee: depends on your setup, lol |
| 23:42.47 | OneWhoKnows | i like things neater lol |
| 23:42.50 | fujin | you need to create a context that zapata will send things into |
| 23:42.56 | fujin | and then you need to handle the calls inside that context |
| 23:42.59 | fujin | basic dialplan configuration |
| 23:43.15 | OneWhoKnows | so, russell, any input on my ESF question? :( |
| 23:43.16 | whhheeEEEeeee | fujin: for all incoming calls. :) |
| 23:45.30 | fujin | uh |
| 23:45.30 | fujin | go die |
| 23:45.36 | whhheeEEEeeee | lol |
| 23:47.17 | JT | a PRI is NOT technically a T1 |
| 23:47.32 | OneWhoKnows | so what is it? |
| 23:47.34 | jingles | *shrugs* |
| 23:47.38 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 23:47.40 | denon | a PRI |
| 23:47.42 | JT | a PRI CAN run over a T1 |
| 23:47.53 | JT | pri describes the signalling |
| 23:48.17 | JT | OneWhoKnows: as far as i know, the bits aren't reserved for a pri |
| 23:48.54 | OneWhoKnows | right, my mistake. i mix up ISDN and T carrier stuff up sometimes |
| 23:49.02 | whhheeEEEeeee | people must love typing my name. |
| 23:49.24 | whhheeEEEeeee | a T line is a collection of ISDN lines. |
| 23:49.27 | JT | whhheeEEEeeee: well |
| 23:49.35 | JT | whhheeEEEeeee: heard of a thing called tab completion? ;) |
| 23:49.43 | whhheeEEEeeee | JT: :O |
| 23:49.52 | JT | whhheeEEEeeee: no, a T line is not a collection of ISDN lines |
| 23:49.59 | OneWhoKnows | JT, you know how they say that the overhead for D4 is 56kbps per channel? is that due to the A and B robbed bits or because of the ZCS it employs? |
| 23:50.10 | whhheeEEEeeee | I'm using TinyIRC, I don't have luxuries like that. |
| 23:50.13 | OneWhoKnows | A T line is a collection of timeslots, right? |
| 23:50.17 | JT | OneWhoKnows: no idea, I don't use T1 much, more E1 |
| 23:50.23 | JT | OneWhoKnows: right |
| 23:50.33 | JT | whhheeEEEeeee: my condolences |
| 23:50.52 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:50.55 | OneWhoKnows | i think i'll read into the difference between B8ZS and HDB3 |
| 23:51.18 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 23:51.25 | OneWhoKnows | JT, have you come across D4 AMI on an E1? |
| 23:51.36 | whhheeEEEeeee | JT: yeah it is, thats why you can truncate part of your lines to voice. at least, this is how it was originally designed. |
| 23:51.50 | JT | OneWhoKnows: no |
| 23:51.56 | whhheeEEEeeee | JT: I want to say its 32 lines. |
| 23:52.15 | JT | whhheeEEEeeee: sorry, what are you talking about? |
| 23:52.19 | OneWhoKnows | whhheeEEEeeee: not true... |
| 23:52.26 | whhheeEEEeeee | JT: T1 line |
| 23:52.40 | JT | whhheeEEEeeee: be more specific, what makes it a collection of ISDN lines? |
| 23:52.42 | OneWhoKnows | whhheeEEEeeee: T1 = 24 64kbps ds0 lines aka time slots |
| 23:52.54 | whhheeEEEeeee | ah, 24 |
| 23:53.10 | whhheeEEEeeee | couldn't remember. |
| 23:53.21 | OneWhoKnows | there are two types of ISDN, BRI a PRI |
| 23:53.25 | whhheeEEEeeee | JT: thats just what it is, a bundle of ISDN lines joined together |
| 23:53.45 | OneWhoKnows | a T1 is not a bundle of ISDN lines :p |
| 23:53.46 | JT | the line speed of a T1 is not 24 * 64000 though |
| 23:53.51 | whhheeEEEeeee | yes it is. argh! |
| 23:53.55 | OneWhoKnows | right, it is with framing overhead though |
| 23:53.57 | JT | whhheeEEEeeee: wrong wrong, and uterly 100% wrong |
| 23:54.06 | JT | whhheeEEEeeee: you need to understand what ISDN is |
| 23:54.12 | JT | because clearly you don't know |
| 23:54.16 | whhheeEEEeeee | enlighten me |
| 23:54.33 | JT | it's a signalling method that involves out of band signalling using Q.931 |
| 23:54.55 | JT | a timeslot is just a timslot on a T1, whether it is isdn or not is immaterial |
| 23:55.03 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 23:55.17 | OneWhoKnows | right |
| 23:55.32 | whhheeEEEeeee | well, thats why I said the original. |
| 23:55.35 | OneWhoKnows | more importantly the out of band signaling implements a delta channel |
| 23:55.37 | whhheeEEEeeee | -_- |
| 23:55.55 | JT | the original? now you're just making stuff up |
| 23:55.59 | OneWhoKnows | lol |
| 23:56.03 | JT | i think RBS came before PRI |
| 23:56.10 | fujin | fail |
| 23:56.13 | OneWhoKnows | whhheeEEEeeee: you said that a T1 is a bundle of ISDN lines :p |
| 23:56.15 | JT | not to mention E & M and all the other crap |
| 23:56.22 | OneWhoKnows | RBS definitely came before PRI |
| 23:56.23 | whhheeEEEeeee | uh, I said originally. |
| 23:56.35 | JT | whhheeEEEeeee: you're still dead wrong |
| 23:56.38 | OneWhoKnows | what do you mean "originally" |
| 23:56.43 | JT | PRI ISDN is a MODERN form of signalling |
| 23:56.44 | whhheeEEEeeee | hell, I dunno. Its just the setup they were using at juilliard back when I worked there and it was explaine to me |
| 23:56.49 | OneWhoKnows | E & M is for analog inband T1s |
| 23:56.50 | JT | relatively speaking |
| 23:57.00 | OneWhoKnows | which is utter crap |
| 23:57.04 | jingles | oh, look who knows so much! as it turns out, your friend here is only *mostly* dead wrong. |
| 23:57.18 | whhheeEEEeeee | Well I guess I was completely, utterly wrong, and I don't understand it at all. Fine by me. |
| 23:57.21 | fujin | pwned |
| 23:57.24 | fujin | fail at everything |
| 23:57.37 | whhheeEEEeeee | ...... |
| 23:58.07 | kiscokid | How do you use Ear & Mouth signalling? Whistle into the phone? |
| 23:58.14 | JT | T and E carrier just specify framing for a timslot based TDM system |
| 23:58.30 | JT | PRI ISDN is something that can optionally be put on top of that |
| 23:58.35 | OneWhoKnows | ISDN30 in EU, right? |
| 23:58.44 | JT | that's one name |
| 23:58.50 | JT | s/EU/most of the world/ |
| 23:58.51 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:59.03 | OneWhoKnows | everywhere but japan and north america |
| 23:59.04 | OneWhoKnows | lol |
| 23:59.06 | whhheeEEEeeee | JT: thank you for explaining it and educating people |
| 23:59.10 | JT | an E1 runs at 2.048Mbit/s |
| 23:59.47 | OneWhoKnows | where a T1 runs at 1.544mbps |