00:01.57 | *** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
00:01.59 | *** part/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
00:03.52 | lmadsen | anyone ever see a linksys spa942 just hang on "Checking DNS" and not allow you to factory reset the phone, and also, it won't save any settings... (this phone has worked fine in the past... but ... ya... it's a POS :)) |
00:09.18 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
00:09.59 | entelechy | hello |
00:10.11 | entelechy | i recently was playing with the 1.6 beta & asterisknow |
00:10.25 | entelechy | i have an old 1.4.11 instllation that works fine, backed up |
00:10.42 | entelechy | but after upgrading to 1.6 beta and correcting a few semantic issues, i am getting this error: |
00:10.43 | entelechy | [Jan 21 18:04:27] WARNING[1979]: pbx_config.c:1511 pbx_load_config: ==!!== Unknown directive: gui_ring_groupname at line 653 -- IGNORING!!! |
00:11.14 | entelechy | for every line in extensions.conf that uses "gui_ring_groupname" |
00:11.38 | *** join/#asterisk kimitaka (n=swiceje@cpe-065-184-219-014.ec.res.rr.com) |
00:11.44 | entelechy | which asterisk module defines "gui_ring_groupname |
00:12.07 | JunK-Y | lmadsen: are you sure that phone has network connection? |
00:12.33 | lmadsen | JunK-Y: yes, it's plugged directly into the same router that everything else is working off of |
00:12.38 | lmadsen | it gets an IP, and I can get to the web interface |
00:12.47 | lmadsen | it just won't do anything I tell it to do |
00:13.04 | kimitaka | anyone using app_rpt or know much about it? |
00:13.14 | lmadsen | entelechy: show us the line in question |
00:13.38 | JT | kimitaka: i know a bit |
00:13.41 | JunK-Y | lmadsen: tell him to stop crack smokin' |
00:13.42 | entelechy | gui_ring_groupname = FMP |
00:13.47 | entelechy | gui_ring_groupname = BP AP |
00:13.48 | lmadsen | where is that? |
00:13.54 | entelechy | in extensions.conf |
00:13.57 | entelechy | Created by asterisknow |
00:14.04 | entelechy | part of a voice menu |
00:14.07 | lmadsen | that doesn't look like a valid dialplan line to me |
00:14.07 | kimitaka | do you have to have a repeater controller or will asterisk do that part too? |
00:14.16 | jjshoe | lmadsen 100% sure the port is good on the switch? |
00:14.24 | lmadsen | jjshoe: yep -- I've tried several ports |
00:14.43 | entelechy | lmadsen: wanna see context? this was autogenerated by the 1.4 version of asterisk now |
00:14.50 | entelechy | i wouldnt think that much would break |
00:14.59 | lmadsen | it was plugged into a Cisco phone (I have a bunch of phones daisy chained), and I moved it just to check, but it's worked prior to tonight.... |
00:15.01 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
00:15.11 | lmadsen | entelechy: nope -- I don't deal with auto-generated stuff |
00:15.12 | entelechy | i noticed the Goto(1|s|1) syntax is now completely deprecated |
00:15.19 | entelechy | yeah well |
00:15.19 | lmadsen | you can't use pipes, only commas |
00:15.28 | entelechy | yep -- I KNOW -- as i said. |
00:15.42 | lmadsen | I know -- I was stating it for the room for those who didn't know |
00:15.53 | entelechy | the error message isnt comning from asterisk GUI. it is coming from the asterisk console. /usr/sbin/asterisk -- thats why i am asking here. |
00:15.56 | lmadsen | in case anyone was wondering WHAT the correct syntax was |
00:16.49 | *** join/#asterisk Dr{Who} (n=mathewss@dev.null.nutech.com) |
00:17.02 | entelechy | i have editted extensions.conf by hand extensively in the past. i mostly know what i'm doing. thats why this error is so confusing |
00:17.16 | entelechy | this is a very simple voice menu |
00:17.36 | Dr{Who} | dho!.. anyone use ekiga? i just signed a week ago and i got a RANDOM call from someone saying they saw my # on some site.. blaa.. blaa.. |
00:17.46 | lmadsen | something other than standard dialplan logic must parse that line, because that is not standard |
00:20.52 | nhuisman_work | Question: Say I have two contexts, [phones] and [inbound-calls] My phones are all defined in [phones] and my pri is in the context [pri] I don't want to have to write out hundreds of lines like ( exten => 342,1,GoTo(phones,342,1). can I just use a pattern exten => _XXX,1,Dial(SIP/${EXTEN},) ? |
00:21.09 | nhuisman_work | and is that a best practice |
00:21.10 | nhuisman_work | ? |
00:21.26 | nhuisman_work | since you don't want your inbound context to be able to dial out |
00:21.35 | [TK]D-Fender | <PROTECTED> |
00:21.36 | nhuisman_work | my [phones] context includes my outbound patterns |
00:22.03 | [TK]D-Fender | nhuisman_work, and no thoses Gotos are completely the wrong approach |
00:22.10 | *** join/#asterisk Bleak (n=asdad@adsl-157-63.click.com.py) |
00:22.24 | nhuisman_work | here i'll show you what I have to make more sense |
00:22.31 | nhuisman_work | http://pastebin.com/m3f414fa9 |
00:22.51 | nhuisman_work | sorry i typed wrong i don't have gotos |
00:22.55 | nhuisman_work | i was just dialing the extensions |
00:23.14 | nhuisman_work | i think that's totally wrong. |
00:23.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
00:24.20 | hmmhesays | [TK]D-Fender: do you know if there is a way to make the microbrowser refresh every time the services button is pushed? |
00:25.48 | nhuisman_work | i guess one question is do includes include includes |
00:25.57 | nhuisman_work | so if I include a context that is including other contexts |
00:26.02 | nhuisman_work | do I get those by including the first context |
00:26.05 | [TK]D-Fender | hmmhesays, nope |
00:26.38 | [TK]D-Fender | nhuisman_work, exten => _XXX,1,Dial(SIP/${EXTEN},15) <- ugly becasue you can dial illegal #'s |
00:26.53 | nhuisman_work | so it's best to statically define all the numbers |
00:27.01 | nhuisman_work | and just deal with having lots of extensions in there |
00:27.49 | [TK]D-Fender | nhuisman_work, yup. |
00:28.07 | [TK]D-Fender | nhuisman_work, exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) <- "9" prefex... ew.. thats so... 80's... |
00:28.16 | nhuisman_work | that's from the asterisk book |
00:28.53 | nhuisman_work | the main reason it is forcing to dial 9 is because our current system does it and I have a feeling it would confuse the shit out of people to change now |
00:28.55 | *** join/#asterisk RoyK (n=roy@91.149.19.19) |
00:29.15 | nhuisman_work | is that what you mean? |
00:29.30 | nhuisman_work | by that I mean, do you mean that dialing 9 to get out is 80s |
00:30.09 | nhuisman_work | I think I'll change it so you can not dial 9 as well |
00:30.25 | *** join/#asterisk ManxPower (n=manxpowe@7.sub-70-218-140.myvzw.com) |
00:30.57 | hmmhesays | [TK]D-Fender: so you have to push refresh every single time? |
00:31.06 | [TK]D-Fender | hmmhesays, yup. |
00:31.39 | ManxPower | Or you can tell your polycom to check for new configs every night at X time and reboot and load them if they changed. |
00:31.50 | [TK]D-Fender | nhuisman_work, No system I depoly forces any kind of prefix. in areas where 10-digit dialing is mandatory on standard phone lines I allow 7-digit and add the default area code |
00:32.34 | Qwell | [TK]D-Fender: comes to Huntsville. Try to setup 7 digit dialing here...it's a PITA. |
00:32.37 | nhuisman_work | that makes sense, I just need to allow people to add 9 since tha's how it is now, then I'll add other patterns for people who don't dial 9 |
00:32.40 | Qwell | s/s// |
00:33.09 | Qwell | some prefixes require 7 digits, others require 10 |
00:33.13 | [TK]D-Fender | nhuisman_work, sad. tell them to change. |
00:33.16 | jjshoe | come to la and setup 7 digit :D |
00:33.16 | [TK]D-Fender | nhuisman_work, its for the best |
00:33.18 | Qwell | so you have to put all of the prefixes into extensions.conf |
00:33.39 | Qwell | jjshoe: people there expect to 10 digit dial |
00:33.39 | [TK]D-Fender | Qwell, I'm sure a lookup table would do nicely :) |
00:33.46 | Qwell | [TK]D-Fender: no such thing exists |
00:33.56 | ManxPower | or get a carrier that lets you dial all calls as 10/11 digits |
00:33.57 | Qwell | even the telco couldn't tell us |
00:34.03 | nhuisman_work | [TK]D-Fender, I was going to tell them they no longer need to dial 9 but I have a feeling If I add the rule to allow 9 it's going to make it so much easier on myself. Our other offices all still use dial 9 to get out, it would really confuse visitors |
00:34.20 | [TK]D-Fender | Qwell, neither did the pyramids... now all I need is a gang of slaves to build this list just the same! |
00:34.24 | nhuisman_work | I see where you are coming from though |
00:35.01 | jjshoe | Qwell yeah, it was in 91 when it split up |
00:35.13 | kimitaka | what does alsa console driver do? |
00:35.14 | [TK]D-Fender | nhuisman_work, no prefix also means you can call "missed calls" on your phone with impunity as their CID won't have a "9" on it. |
00:35.23 | [TK]D-Fender | nhuisman_work, its a question of functionailty too.... |
00:35.57 | nhuisman_work | wait, why would incoming calls have a 9 added to the caller id? |
00:36.17 | nhuisman_work | if asterisk strips them off before dialing out |
00:36.25 | nhuisman_work | which doesn't even affect incoming |
00:36.36 | jjshoe | although it wasn't until 99 when changes we made to force 10 digit dialing, but still, almost 10 years ago |
00:36.49 | [TK]D-Fender | nhuisman_work, incoming calls DON'T have the "9", thats the issue |
00:36.54 | jjshoe | oh more likes |
00:37.00 | jjshoe | Qwell 10 digit wasn't required until 2006 |
00:37.04 | nhuisman_work | [TK]D-Fender, oh, i understand what you mean. |
00:37.04 | jjshoe | lies* |
00:37.14 | jjshoe | Having been staved off nearly seven years, the 424 overlay was finally implemented on July 26, 2006 and new telephone numbers issued in the 310 area code may now begin with either 310 or 424. Ten-digit dialing within the 310 area code became optional on January 1, 2006 and mandatory on July 26, 2006. |
00:37.15 | Qwell | it was required before 2006 |
00:37.19 | Qwell | because there are many areacodes :p |
00:37.23 | [TK]D-Fender | nhuisman_work, if you miss a call passed on to your phone you wouldn't be able to hit the "callback" feature on the phone to call them because YOU have to have a "9" in front |
00:37.33 | Qwell | 323, 626, 213, 310 |
00:37.41 | nhuisman_work | [TK]D-Fender, yeah that dialplan wasn't complete, i'm adding the ability to also not dial 9 |
00:37.51 | nhuisman_work | sorry |
00:37.58 | Qwell | I know a lot of people who had next door neighbors with a different areacode |
00:38.14 | jjshoe | Qwell not within your own area code though |
00:38.30 | Qwell | no, but nobody ever gives a number without areacode |
00:38.41 | jjshoe | you haven't spent time in 310 :) |
00:38.54 | Qwell | I still do it here.. |
00:39.07 | nhuisman_work | so it would look more like this : http://pastebin.com/m5b05034b |
00:39.09 | jjshoe | I never give my area code out when I frequest buisnesses at home in the south bay |
00:39.15 | jjshoe | but I guess it's all personal and dependent on the area |
00:39.32 | nhuisman_work | unless i'm stupid and that won't work. |
00:39.37 | Qwell | jjshoe: you didn't grow up where that was necessary |
00:40.22 | drmessano | Theres really no such thing as a 7 digit phone number anymore |
00:40.36 | drmessano | Area codes today are what exchanges were 30 years ago |
00:42.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f6a40703eb7540c0) |
00:45.40 | drmessano | I guess I pretty well killed that convo :( |
00:45.53 | nhuisman_work | fun stuff |
00:46.19 | nhuisman_work | seppuku |
00:47.34 | [TK]D-Fender | I sliced my hand at the dojo just taking my sword out of its bag. |
00:48.28 | *** part/#asterisk RoyK (n=roy@91.149.19.19) |
00:48.49 | nhuisman_work | heh |
00:48.57 | nhuisman_work | why do you have sharp swords at your dojo? |
00:49.05 | [TK]D-Fender | it slid out of the saya as I was pulling the bad away and made an inch-long cut on my hand. Ttok about a few seconds and then just kep bleeding. fortunately nothing much to sop up |
00:49.23 | drmessano | mmmm Shinchirin |
00:49.26 | [TK]D-Fender | I just let it cake on to dry and seal up |
00:49.29 | nhuisman_work | would you just practice with wooden or shanias |
00:49.40 | drmessano | -n |
00:50.14 | [TK]D-Fender | I am a student of katori shinto so I use a katana for the iaijutsu component |
00:50.41 | nhuisman_work | ah |
00:50.47 | nhuisman_work | i guess that wouldn't really work with a wooden sword :P |
00:50.48 | [TK]D-Fender | nhuisman_work, and the rest of my session (I did this right at the START) I did with jsut my boken with the person I came to teach. |
00:51.02 | drmessano | Sensei allowed us to practice with katanas the first week of class.. good way to eliminate overcrowing and filter out the weak |
00:51.20 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
00:51.36 | drmessano | 14 ambulances later, and we had a nice small, strong class |
00:51.45 | [TK]D-Fender | drmessano, OUCH |
00:52.00 | nhuisman_work | seriously? |
00:52.08 | [TK]D-Fender | drmessano, there is only 1 person other than myself with a real sword in the class, the rest are iaito. |
00:52.32 | drmessano | no, not really.. I cant even cook in a wok without requiring medical attention |
00:52.35 | nhuisman_work | i guess it forces you to not hit yourself with a sharpened oe |
00:53.01 | drmessano | Im the second most dangerous person I know, next to my extremely clumsy wife |
00:53.13 | [TK]D-Fender | drmessano, and that other guy is a psycho and has cut himself numerous times in class. mine was a careless un-bagging mistake which shouldn't have happened. |
00:53.19 | nhuisman_work | by dangerous you mean self endangering? |
00:53.21 | drmessano | Niice |
00:53.34 | drmessano | No, we tend to harm others too in our wake |
00:53.38 | nhuisman_work | laugh |
00:53.43 | [TK]D-Fender | drmessano, eek |
00:54.13 | kimitaka | how long shoudl asterisk take to compile on a 300mhz g3? |
00:54.20 | nhuisman_work | where can I find documentation on running a stability test call type of overload test |
00:54.20 | [TK]D-Fender | I've got my custom katana still on order.... awaiting the next shipment of blades for inspection. |
00:54.30 | nhuisman_work | ie, make a whole bunch of calls to different extensions |
00:54.32 | nhuisman_work | and to the outside |
00:54.35 | nhuisman_work | to stress the system |
00:55.14 | *** join/#asterisk gmcneish (n=gary@samclay.plus.com) |
00:55.18 | jjshoe | Qwell I did actually, when they split chicago up into multiple area codes |
00:55.22 | drmessano | When I am working, I am a calm, strong thinker, who puts in 110%.. For everything else, I give about 60%, which is where the carelessness comes in |
00:56.01 | drmessano | I can wash a car and leave it dirtier than it was... do I care, not really. |
00:56.03 | gmcneish | hi can someone please look at my extensions.conf and see where im going wrong im using sip trunks and i need to delete the crap thats not related to sip out of it |
00:56.16 | nhuisman_work | gmcneish, pastebin it |
00:56.19 | gmcneish | cool |
00:57.15 | gmcneish | how do i select all in vim |
00:57.51 | nhuisman_work | you in a window manger environment? |
00:57.58 | nhuisman_work | just select it on your terminal screen |
00:58.10 | drmessano | Good god... am I the only person that uses Nano |
00:58.11 | nhuisman_work | otherwise cat the file to your terminal window and then scroll up and select it all |
00:58.12 | gmcneish | i dont know ill just put it together my self |
00:58.16 | drmessano | I feel like such a windows admin :( |
00:58.25 | nhuisman_work | yucky yucky nano |
00:58.37 | nhuisman_work | you can select all in vim but it doesn't paste it for outside applications |
00:58.46 | nhuisman_work | shift + V and then press the up and down arrows |
00:59.05 | nhuisman_work | it's pretty sad |
00:59.10 | drmessano | lol |
00:59.12 | nhuisman_work | i'll be in a normal tex editor |
00:59.24 | nhuisman_work | and i'll be going into insert context and typing :w |
00:59.29 | nhuisman_work | then go wtf, this isn't vim |
00:59.41 | gmcneish | its just im remoteing to another machine that only has comand line |
00:59.59 | nhuisman_work | gmcneish, remote in then cat the file |
01:00.04 | nhuisman_work | then copy it from your terminal |
01:00.14 | [TK]D-Fender | drmessano, I uses nano.... when I don't have MC around :) |
01:00.53 | nhuisman_work | i use vim because all unix boxes have it |
01:01.12 | nhuisman_work | sorry *nix |
01:01.46 | nhuisman_work | it's not very hard to figure out nano though if you're stuck with out vim. |
01:01.51 | NovceGuru | I'm quoting a client for a asterisk appliance :D :D |
01:01.51 | St1ckm4n | I'm hoping to get some second opinions about our current asterisk situation, short version we're on an unstable asterisk@home install v1.2.3 and I want to rebuild for a call center should I use 1.4 or stick with latest of 1.2 please pm me if you are willing to discuss in more detail |
01:01.53 | drmessano | lol |
01:02.00 | [TK]D-Fender | nhuisman_work, the reverse can't be said... |
01:02.07 | nhuisman_work | [TK]D-Fender, yeah I know |
01:02.15 | drmessano | Im gload, TK... I was getting the impression nano was some loser app |
01:02.19 | nhuisman_work | [TK]D-Fender, hence learning the harder one seemed like the way to go |
01:02.25 | drmessano | s/gload/glad |
01:02.35 | nhuisman_work | drmessano, you gloader! |
01:02.35 | [TK]D-Fender | St1ckm4n, 1.4. 1.2 is dead. |
01:02.36 | drmessano | Maybe I need its spell checker |
01:03.11 | drmessano | 1.2 is not dead.. it will be forked and live on FOREVER |
01:03.33 | [TK]D-Fender | drmessano, they mock us for it, but I like nano because I can edit in a "normal" way. VI does become incredibly fast and efficient once you've put some time into it though. |
01:03.45 | jblack | Free software never dies. It just bit-rots eternally |
01:03.46 | [TK]D-Fender | drmessano, but its still cryptic shit! |
01:04.08 | gmcneish | done http://pastebin.com/m1ef43ea8 |
01:04.26 | St1ckm4n | I'm a little scared of 1.4 since we've been on 1.2 and I hear agentcallbacklogin has been depracated |
01:04.36 | NovceGuru | vi/vim is insane once you know some of the simple shortcut |
01:04.37 | NovceGuru | s |
01:04.50 | gmcneish | i think i only need the bottom section as it says something about sip |
01:04.56 | drmessano | [TK]D-Fender, I know that merely hanging out here makes this statement complete hipocracy, but I don't think I am dorky enough for VI |
01:04.59 | [TK]D-Fender | St1ckm4n, go download 1.4 read the upgrade notes, do the same with your 1.2 notes |
01:05.01 | gmcneish | also im only going to be calling the uk |
01:05.44 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
01:05.52 | [TK]D-Fender | drmessano, I have better things to focus on that learning a cryptic text editor. I go in to get a job DONE. |
01:06.00 | drmessano | Yep |
01:06.02 | jblack | Here's a quote. "I bow down before you. I thought I had done some rather horrible things with gcc built-ins and macros, but I hereby hand over my crown to you. As my daughter would say: that patch fell out of the ugly tree, and hit every branch on the way down. Very impressive." -- Linus Torvalds |
01:06.45 | gmcneish | done http://pastebin.com/m1ef43ea8 |
01:06.55 | drmessano | I think if you've ever kissed a real woman, you should not have to prove your worth by showing some knowledge of cryptic apps like VI |
01:07.16 | drmessano | or knowing Pi to 15 places |
01:07.50 | [TK]D-Fender | gmcneish, tip : don't shove everything into a single ridiculous context making a psychotic mess. |
01:08.01 | *** join/#asterisk kyron_ (n=kyron@211-217-static-ppp.3menatwork.com) |
01:08.07 | nhuisman_work | [TK]D-Fender, heh |
01:08.10 | drmessano | "I bet you dont know all the GCC compiler error codes!" "I kissed a woman once" "Ok, fine" |
01:08.30 | drmessano | Thats an XKCD in the making |
01:08.36 | hmmhesays | bah my perl script is chomping the entire fscking line |
01:09.11 | [TK]D-Fender | hmmhesays, bits, bytes, chomping..... all this computer talk is making me hungry! |
01:09.33 | ManxPower | I know enough vi to edit and save a file. Not fast, not efficient, but I can do it. |
01:09.44 | gmcneish | me to |
01:09.47 | ManxPower | EVERYONE that works on *nix should know that much. |
01:10.12 | nhuisman_work | i would hope so |
01:10.17 | riddlebox | ManxPower, thats about as much vi as I know too |
01:10.42 | jblack | One of the best investments in time I ever made was to learn vim usage. |
01:10.43 | hmmhesays | [TK]D-Fender: I just ate |
01:10.54 | hmmhesays | chomp should only chomp \n by default in perl |
01:11.06 | riddlebox | grep is something that I need to learn more of, as well |
01:11.10 | ManxPower | joe is the editor I use. It uses WordStar keybindings. |
01:12.05 | riddlebox | I remember trying to use jove a long time ago |
01:13.44 | drmessano | I think every windows user should learn edit.com, just in case |
01:13.54 | nhuisman_work | fuck it, use ed |
01:14.06 | nhuisman_work | do you really need to learn edit.com |
01:14.13 | nhuisman_work | seriously what is there to learn |
01:14.15 | nhuisman_work | hehe |
01:14.18 | drmessano | edlin |
01:14.23 | [TK]D-Fender | nhuisman_work, thats the point, its intuitive! |
01:14.29 | drmessano | yes |
01:14.31 | nhuisman_work | it's also very limited |
01:14.39 | riddlebox | hehe I use nano a lot |
01:14.56 | drmessano | If I am gonna learn VI to be one of the cool kids, everyone should learn DOS EDIT for the same reason lol |
01:15.01 | gmcneish | so http://pastebin.com/m3ea05117 |
01:15.20 | ManxPower | drmessano: you need to learn VI because it is the only editor that is ALWAYS installed. |
01:15.22 | nhuisman_work | I'm not advocating vim and saying all other editors suck, I'm just saying you better at least know the basics of vim and other default editors as well. |
01:15.45 | gmcneish | can anyone help me first off theres a bit at the bottom about sip now im using siptrunks what part of the file will i need to delete |
01:15.51 | nhuisman_work | you don't need to be an uber im geek |
01:15.54 | nhuisman_work | im=vim |
01:15.57 | drmessano | I can open, edit, and save a file.. guess thats enough |
01:16.03 | nhuisman_work | yeah good enough then |
01:16.03 | gmcneish | :wq |
01:16.05 | jblack | gmcneish: Not one single include in that entire file? |
01:16.17 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
01:16.21 | ManxPower | drmessano: It is enough |
01:16.22 | nhuisman_work | learning emacs on the other hand |
01:16.54 | ManxPower | Ctrunk |
01:16.54 | gmcneish | whats an include |
01:16.54 | nhuisman_work | i dont' think you need to do :P |
01:16.54 | ManxPower | ~trunk |
01:16.55 | jbot | i guess trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
01:16.55 | drmessano | nano loves me more |
01:16.55 | drmessano | I have nano on windows.. |
01:16.55 | gmcneish | yeah ill put the trunk bit in tommorow |
01:16.55 | drmessano | Just because |
01:17.08 | gmcneish | i need to phone my provider i lost the details but its the bottom part thats got the ; before it |
01:17.26 | riddlebox | drmessano, the only thing I wish nano could do is search, or jump to a line |
01:17.36 | drmessano | Besides, if I run into any hardcore VI issues, jblack has my back |
01:17.43 | gmcneish | in my sip.conf file i have extensions 101 and 102 do i reference these in extensions.conf |
01:17.48 | hijacked | ^w ? |
01:17.58 | drmessano | ^w |
01:18.05 | nhuisman_work | ^X |
01:18.05 | drmessano | search works awesome |
01:18.14 | drmessano | and nano you can open a line from the CLI |
01:18.21 | nhuisman_work | so can vim |
01:18.23 | riddlebox | really? |
01:18.24 | nhuisman_work | !command |
01:18.29 | MihiNomenEst | I think it's ^W ^g to search. |
01:18.35 | drmessano | ^w to search |
01:18.38 | ManxPower | gmcneish: you do NOT have extensions 101 and 102 in your sip.conf file. You have SIP accounts/users 101 and 102 in your sip.conf. |
01:18.42 | nhuisman_work | : then type !ls |
01:18.47 | drmessano | nano +120 /etc/blah |
01:18.52 | MihiNomenEst | sorry, I meant goto line. |
01:19.04 | drmessano | nano +187 blah |
01:19.05 | ManxPower | You might have done something utterly silly and newbieish and named your SIP account names and your extensions the same, but that would be short sighted. |
01:19.11 | gmcneish | ok chers so users 101 and 102 |
01:19.20 | nhuisman_work | wait i don't think i understand what you mean by open a line from the cli |
01:19.26 | drmessano | Like |
01:19.27 | gmcneish | do they get a mension in extensions.conf? |
01:19.29 | ManxPower | exten => 101,1,Dial(SIP/101) |
01:19.32 | drmessano | I can open line 160 from the CLI |
01:19.32 | nhuisman_work | !ls is external command |
01:19.37 | drmessano | Or go to 160 |
01:19.45 | drmessano | nano +160 /etc/blah |
01:19.47 | nhuisman_work | when you say cli do you mean running nano as a command? |
01:19.56 | drmessano | yes, from the command line |
01:20.08 | drmessano | nano +160 /path |
01:20.31 | gmcneish | now do i do that for every account |
01:20.34 | nhuisman_work | displaying a given line is a common thing you can do with linux utils |
01:20.51 | drmessano | nhuisman_work |
01:20.54 | drmessano | I cant make this any clearer |
01:21.03 | drmessano | I can open nano and GO TO THE LINE |
01:21.18 | drmessano | if I told you go to line 160 in blah, I can nano +160 blah |
01:21.28 | nhuisman_work | i get it |
01:21.50 | nhuisman_work | vim has the same command |
01:21.54 | drmessano | ok |
01:21.59 | ManxPower | ^Kl is the command in jow |
01:22.04 | nhuisman_work | same syntax |
01:24.33 | drmessano | bbiaf |
01:24.41 | gmcneish | http://pastebin.com/m3ea05117 |
01:24.49 | gmcneish | what parts of this file can i delete |
01:25.49 | *** part/#asterisk Dr{Who} (n=mathewss@dev.null.nutech.com) |
01:26.25 | [TK]D-Fender | gmcneish, bad question. How are we supposed to know whats important or not in there? |
01:27.21 | gmcneish | sorry |
01:27.53 | gmcneish | im new to asterisk i have a sip fowarder account and i think there is stuff in that file that is not related to sip |
01:28.20 | gmcneish | or at the bottom of the page it says ;#### VDAD SIP UNREGISTERED TRANSFER ENTRIES #### |
01:28.20 | gmcneish | ;#### Use these entries IN PLACE OF the entries above if you are using SIP trunks |
01:28.21 | gmcneish | ;#### and are not registering your provider in sip.conf |
01:28.30 | [TK]D-Fender | gmcneish, Do you seem to understand the dialplan at all. |
01:28.58 | [TK]D-Fender | gmcneish, the dialplan controls what happens on EVERY call you make. |
01:28.59 | gmcneish | im trying to |
01:29.17 | [TK]D-Fender | gmcneish, it is not "sip" related unless you DIAL a SIP resource. |
01:29.57 | gmcneish | ok is this the only section i need http://pastebin.com/m3e2ba04c |
01:30.00 | [TK]D-Fender | gmcneish, you may use * to take calls from a SIP ITSP and have NO reference to SIP in your dialplan at all if you are only an inbound call-center for example. |
01:30.17 | [TK]D-Fender | gmcneish, We still have no idea what parts aren't needed. |
01:30.30 | [TK]D-Fender | gmcneish, Go in there and remove stuff you don't need YOURSELF./ |
01:30.35 | gmcneish | im outbound only to the uk |
01:31.34 | gmcneish | but i dont want to remove parts that will affect my dialer |
01:32.20 | gmcneish | is an 800 number america only |
01:32.32 | [TK]D-Fender | gmcneish, Ok you have NO clue whatsoever. Go ask a forum in your dialer community what parts you have to leave behind and then install * on another system and go learn it. |
01:32.58 | gmcneish | but they all tel me oh thats an asterisk question go ask them |
01:33.12 | gmcneish | is an 800 number america only |
01:33.20 | [TK]D-Fender | gmcneish, then leave this system alone and learn * on another box. |
01:33.45 | gmcneish | look my question is simple is 800 america |
01:34.20 | nhuisman_work | yes |
01:34.26 | gmcneish | thank you |
01:34.45 | nhuisman_work | that was me googling, you better double check though |
01:34.53 | [TK]D-Fender | gmcneish, nobody here know what parts your dialer really NEEDS so we can't help you. |
01:35.58 | gmcneish | its just the extensions.conf file had zap stuff in it and iax and sip so i have to delete the irelivant stuff |
01:36.41 | jblack | Woot. |
01:36.48 | jblack | http://maps.google.com/maps/ms?hl=en&gl=us&ie=UTF8&lr=lang_en&msa=0&msid=100618373221056061016.000443665b744f9405cb3 |
01:36.58 | nhuisman_work | nevermind :The toll-free prefix 800 has been widely adopted elsewhere, including as the international toll-free number. It is often preceded by a 0 rather than a 1 in many countries where "O for Operator" has no meaning in the national language. |
01:37.02 | nhuisman_work | i think it's not us only |
01:37.17 | nhuisman_work | jblack, what is that? |
01:37.20 | jblack | I think we may get a san diego node, and possibl even a colorado node. |
01:37.21 | nhuisman_work | asterisk trunks? |
01:37.25 | jblack | That's the dundi network that I'm in. |
01:37.30 | nhuisman_work | cool |
01:39.42 | nhuisman_work | so http://www.nslu2-linux.org/wiki/Optware/AsteriskStdextenMacro those extensions below the macros allow you to hit *xx and then it starts forwarding to the number you specified to |
01:39.55 | nhuisman_work | base on which one you chose |
01:40.39 | St1ckm4n | I just finished reading the upgrade notes on 1.4 but wanted to know if you guys feel that 1.4 is less or more reliable than 1.2.26? |
01:41.44 | nhuisman_work | i think the consensus is that 1.4 is ready for production |
01:41.46 | [TK]D-Fender | St1ckm4n, 1.2 is DEAD. No more bug-fixes coming, and 1.6 is is beta. |
01:41.48 | drmessano | jblack: thats hardcore |
01:41.59 | nhuisman_work | ABE is now on 1.4 and that's digiums supported version |
01:42.14 | tzanger | ABE's now 1.4? |
01:42.21 | nhuisman_work | they just realized it a few days ago |
01:42.26 | tzanger | is there any official documentation to the effect/ |
01:42.27 | nhuisman_work | i'm waiting for them to fix their installer iso |
01:42.40 | nhuisman_work | it's available in the be downloads |
01:42.41 | jblack | drmessano: Well, we're trying to be hardcore |
01:42.47 | drmessano | 1.2 is security maintainence only? |
01:42.58 | nhuisman_work | their iso isn't complete yet though |
01:43.08 | nhuisman_work | i opened a support case to find out what the deal was. |
01:43.19 | nhuisman_work | by not complete I mean there isn't one with rpath on it, the rpms are all there. |
01:43.48 | tzanger | there was some rumour that digium's internal system was 1.2 while they were suggesting for everyone to use 1.4. I don't much care, as I run 1.4 (svn trunk actually) for most systems |
01:43.57 | nhuisman_work | it was 1.2 |
01:44.07 | nhuisman_work | oh you mean their own phone system |
01:44.10 | nhuisman_work | nm i don't know anything about that |
01:48.43 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
01:49.50 | gmcneish | is the top part of this file related to sip |
01:49.55 | gmcneish | http://pastebin.com/m7a0a2f07 |
01:53.45 | gmcneish | http://pastebin.com/m7e9992bc if this is for a long distance uk number how would i change this to a local uk number |
01:54.11 | gmcneish | using sip trunks |
01:55.58 | [TK]D-Fender | gmcneish, that garbage does not mean ANYTHING. |
01:56.18 | [TK]D-Fender | gmcneish, AGI's from GUI's like that are unreadble trash |
01:56.53 | gmcneish | ok so what txt would i need to put into my dial pattern if i wanted to dial only uk numbers |
01:56.57 | [TK]D-Fender | gmcneish, you have LOT of learning to do if you want to try and admin a box thats running a dialer like that and I don't anyone here is going to give you that kind of time. |
01:56.58 | gmcneish | on sip trunks |
01:57.16 | [TK]D-Fender | gmcneish, and stop using the term "sip trunk" like is a magic term |
01:57.28 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
01:57.36 | [TK]D-Fender | gmcneish, noone can see how you are suposed to call out your provider based on that. |
01:58.32 | gmcneish | oh i thought there would just be a standard dial plan for calling local uk |
01:58.49 | [TK]D-Fender | gmcneish, if I were you I'd stop trying to mess with that system. You don't userstand Asterisk at all and we can't help you with the junk your dialer throws in. |
01:59.17 | [TK]D-Fender | gmcneish, there IS no standard with *. * is like PAINT, its whatever you want it to be and you don't even have a start. |
01:59.27 | [TK]D-Fender | gmcneish, you are in way over your head. |
01:59.39 | gmcneish | tell me about it but im getting there |
02:00.24 | [TK]D-Fender | gmcneish, You don't seem to be anywhere right now. Stop, go read the book, and install * yourself on another machine and go learn it |
02:00.33 | gmcneish | ok so how do i tell extensions.conf to call uk numbers because mine is set to american dialing rules |
02:01.49 | drmessano | lol |
02:01.57 | [TK]D-Fender | gmcneish, ok. You just don't get it. Those stupid AGI's are controlling everything and doing the actual dialing. NOBODY HERE CAN HELP YOU. |
02:02.06 | [TK]D-Fender | gmcneish, that stupid dialer OWNS YOU. |
02:02.18 | [TK]D-Fender | gmcneish, You have virtually no control. |
02:02.23 | drmessano | So you help me, no? |
02:02.49 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
02:04.35 | gmcneish | i dont see what ur problem is my question is asterisk releated imaging i have a blank extensions.conf file and i want to make my asterisk box call numbers in the uk only using siptrunks |
02:05.03 | gmcneish | like this |
02:05.03 | gmcneish | ; dial a long distance outbound number through a SIP provider |
02:05.04 | gmcneish | ; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) |
02:05.04 | gmcneish | ;; exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) |
02:05.04 | gmcneish | ; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,55,o) |
02:05.04 | gmcneish | ; exten => _91NXXNXXXXXX,3,Hangup |
02:05.19 | [TK]D-Fender | gmcneish, do NOT spam in here. |
02:05.42 | gmcneish | sorry |
02:05.43 | [TK]D-Fender | gmcneish, and if you don't like the patter on that exten, go CHANGE IT. Chapter 5 of THE BOOK |
02:05.44 | mosty | i have an odd issue, some of my sip phones will be on a call for a few minutes sometimes, then the sip phone hears music on hold, while the other end can still hear the sip phone, i can't figure out how or why this would happen, anyone have any ideas for things to look for? |
02:05.46 | [TK]D-Fender | ~book |
02:05.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
02:06.13 | [TK]D-Fender | gmcneish, and no promises that your provider will even LIKE the number you give them |
02:06.30 | mosty | [TK]D-Fender, btw the polycom v3 firmware works ok with bootrom 4.0 and even 3.something on my test phone |
02:07.03 | [TK]D-Fender | mosty, I know.... I have them both and just haven't gotten around to installing yet |
02:07.14 | [TK]D-Fender | mosty, just picked up 2.2.2 that I didn't know was released either |
02:08.06 | mosty | the other night you guys said that it required bootrom v4.1, which doesn't seem to exist, that's all |
02:08.16 | lmadsen | I just finished upgrading my IP501 |
02:08.28 | lmadsen | I think I'm runnign 2.2.0 now |
02:08.37 | [TK]D-Fender | mosty, I never said it needed it. |
02:09.04 | mosty | i forget who the other person was, the person that noticed firmware v3 |
02:09.23 | lmadsen | like... v3.x? |
02:09.25 | [TK]D-Fender | mosty, [hC] was the one who told me, |
02:09.45 | [TK]D-Fender | lmadsen, yes. SIP 3.0.0 is out with BIG changes |
02:09.51 | lmadsen | oh really.. interesting |
02:09.53 | [hC] | yeah i noticed v3 |
02:09.56 | [hC] | its not on their downloads link |
02:09.57 | lmadsen | what are some of the more interesting ones? |
02:10.03 | [hC] | i found it in some weird other upgrade matrix page |
02:10.09 | [hC] | i dont think it was supposed to be out yet |
02:10.14 | mosty | lmadsen, ldap contacts (with a specific license) |
02:10.19 | [hC] | the most interesting things are now it will do LDAP/active directory contacts |
02:10.25 | BBHoss | awesome |
02:10.28 | mosty | [hC], it's available now, if you're registered |
02:10.30 | [hC] | headset offhook functionality using the Jabra protocol |
02:10.41 | [hC] | mosty: yeah i am, it just wasnt in the download list when i found it |
02:14.39 | mosty | [hC], well it worked fine with the old bootrom on my ip550 test phone, in case you're interested |
02:14.54 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
02:15.52 | *** join/#asterisk postconf (n=marquis@gw-corp.postconf.com) |
02:16.39 | BBHoss | so you have to have a certain license to do LDAP? |
02:16.44 | mosty | yes |
02:16.57 | BBHoss | taking a page out of Ci$co's book |
02:26.49 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
02:29.42 | mpruett | Hello Everyone |
02:30.27 | drmessano | What? |
02:30.43 | Prayer | Hello mpruett |
02:30.53 | mosty | is there a way that i can see current channels that are on MOH in the cli? |
02:31.07 | mpruett | Quiet tonight |
02:32.01 | mpruett | Does anyone know if the "g" option for the Dial cmd works in a macro? |
02:32.16 | Prayer | try CLI>sip show channels |
02:32.47 | Mavvie | mpruett: why wouldn't it work? |
02:33.38 | Prayer | The "j" option doesn't work for me in 1.2 - the new s${DIALSTATUS] etc works better though. |
02:33.41 | mpruett | Don't know - I am having troubles exucuting further commands in the macro after the call has hung up |
02:33.56 | mpruett | I am on 1.2 also |
02:34.14 | Prayer | Ummm I think the macro is exited after call done ? |
02:34.38 | *** join/#asterisk theron (n=theron@65.198.151.150) |
02:35.07 | mpruett | Anyway to prevent this or send it to another macro? |
02:36.42 | mosty | Prayer, was that directed to me? i tried 'show channels' while i had a call on hold, it just said that the two sip channels were bridged, nothing about MOH |
02:37.41 | Prayer | I don't have MOH much so wasnt sure about that. |
02:39.15 | *** part/#asterisk postconf (n=marquis@gw-corp.postconf.com) |
02:39.20 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
02:39.31 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
02:39.51 | theron | Hi all, got a firewall related issue I believe, and I'm wondering if I'm down to just replacing a cheap router. I have an asterisk 1.4 server behind a netgear WGR614v6. I'm forwarding ports 5060-5082 tcp/udp, port 4569 for my provider's iax connection, and ports 10000-20000 udp. to my asterisk server on my internal network. I can receive and place calls all day long from internal to my network, however from the outside I get a connection, but no audio. |
02:40.10 | entelechy | i am STUNned |
02:40.16 | entelechy | ;-) |
02:40.20 | entelechy | hopefully someone else can help you. |
02:40.21 | *** part/#asterisk entelechy (i=user@mail.beanproducts.com) |
02:40.55 | jjshoe | theron do you see the RTP traffic at the router? |
02:41.03 | jjshoe | does the router give you the ability to see the traffic? |
02:41.13 | theron | I know that to get the rtp connection back to my client I need to use stun, I believe that I've got all the required settings in my sip.conf for that.... |
02:41.25 | theron | jjshoe: I can't see it. |
02:42.04 | drmessano | Thats NAT the problem |
02:42.07 | jjshoe | theron what happens when you take the router out of the connection and plug the asterisk box straight in? |
02:42.09 | drmessano | ~sipnat |
02:42.10 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
02:43.16 | BBHoss | theron: did you double check the rules to make sure you allowed tcp/udp instead of just tcp |
02:43.23 | theron | .. |
02:43.32 | theron | BBHoss. tcp/udp for all? |
02:43.41 | drmessano | UDP |
02:43.43 | BBHoss | well actually UDP |
02:43.52 | BBHoss | but SIP CAN run over TCP |
02:43.52 | drmessano | Check your NAT settings |
02:43.53 | [TK]D-Fender | theron, Read the guide... |
02:43.58 | drmessano | Thats likely it |
02:44.11 | [TK]D-Fender | * does not support SIP over TCP in 1.4 standard. |
02:44.26 | [TK]D-Fender | and * does not support STUN either, nor does it need it. |
02:44.58 | drmessano | [TK]D-Fender, it can't be the SIP NAT settings.. that's too easy |
02:45.15 | drmessano | Has to be some obscure firewall setting.. so start there first |
02:45.23 | drmessano | Upgrade the firmware |
02:45.27 | drmessano | Wait a week |
02:45.29 | drmessano | Complain |
02:45.33 | theron | drmessano: running latest. |
02:45.36 | drmessano | Then check the sip.conf |
02:45.38 | theron | not interested in complaning. |
02:45.49 | theron | I "believe" it's correct. |
02:46.01 | drmessano | Its probably not |
02:46.12 | [TK]D-Fender | theron, Hitler thought he was "correct" too :) |
02:46.17 | theron | bet you a sandwich ;) |
02:46.26 | [TK]D-Fender | theron, PASTEBIN is your friend. |
02:46.27 | [TK]D-Fender | ~pb |
02:46.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
02:46.48 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
02:46.50 | drmessano | Most people with external audio issues that insist their settings are correct are dead fscking wrong |
02:47.24 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
02:47.26 | lmadsen | wow... my polycom takes 2-3x as long to boot on the new firmware :) |
02:47.45 | jjshoe | of course this is why I asked him to plug directly into his source around his router |
02:47.48 | jjshoe | rather then bitch at him |
02:47.54 | jjshoe | just ask him to take the router out of the picture |
02:47.57 | jjshoe | it's far quicker. |
02:48.01 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
02:48.03 | theron | http://pastebin.ca/868040 |
02:48.12 | jjshoe | lmadsen polycom wins the race for slowest booting phone I've ever seen |
02:48.32 | mpruett | sorry guys I got disconnect - Prayer (anyone?) do you know a way for me to execute commands after hangup of a call in a macro? |
02:48.39 | lmadsen | I'm shocked at how much longer it takes on the 2.x from the 1.x |
02:49.24 | drmessano | Far quicker until he puts the router back |
02:49.30 | drmessano | Youre not solving the problem |
02:50.15 | jjshoe | drmessano you're proving to him where the issue is |
02:50.45 | *** join/#asterisk krapper (i=krapper@wsip-64-58-154-130.oc.oc.cox.net) |
02:50.51 | jjshoe | drmessano until you can make him believe what you and I know, he'll never look at it, so this method is insanely quicker |
02:51.07 | [TK]D-Fender | jjshoe, and Polycom wins the race for "needs rebooting the fewest number of times" |
02:51.10 | Prayer | mpruett, I think there is a way I read yesterday but cant remember details. Maybe executions after the macro() call ? Or a variable set to make that occur. |
02:51.34 | drmessano | jjshoe.. So? |
02:51.44 | theron | even if I wanted to take the router out of the picture atm it's at a remote location. |
02:51.46 | drmessano | Its his box, not mine... if he wants it to FAIL, thats up to him |
02:51.55 | drmessano | paste your damn config pls |
02:51.58 | theron | did. |
02:52.03 | [TK]D-Fender | theron, Where is 6000? |
02:52.14 | theron | 6000 is my softphone... |
02:52.25 | theron | pasting extensions.conf .... |
02:52.28 | drmessano | uh |
02:52.30 | [TK]D-Fender | theron, Ah. for [6000] add "nat=yes", "qualify=yes" |
02:52.37 | Prayer | mpruett, I was trying to use SET(TIMELIMIT(absolute)) to limit call times. I wanted a warning though and found something better. The macro ending was hurting the absolute sometimes. |
02:54.21 | jjshoe | drmessano nice attitude. |
02:54.41 | jjshoe | [TK]D-Fender I don't have to reboot an aastra more then once and I can make every change I want *shrug* |
02:54.41 | mpruett | That gave me an idea to try Prayer - thanks for the help |
02:55.06 | [TK]D-Fender | jjshoe, I have an Aastra 57i CT that likes to spontaneoulsy lock up... |
02:55.28 | drmessano | jjshoe, I dont need a lecture |
02:55.39 | [TK]D-Fender | jjshoe, Aastra is fairly decent though... but the 5i Series manufacturing build BLOWS |
02:56.03 | Prayer | mpruett, tell about idea and if it worked or not when you are done with experiment. |
02:56.48 | theron | [TK]D-Fender, adding now... question though, if they're in the general, dosen't it apply to all below? |
02:56.57 | jjshoe | [TK]D-Fender yes, we can reproduce lockups in house, we're working with aastra engineering to get it cleared up |
02:57.12 | jjshoe | [TK]D-Fender with the 5 series anyways, the 9 and 4 series we don't have issues with |
02:57.25 | jjshoe | drmessano You consider two words a lecture? |
02:57.35 | drmessano | the beginning of one |
02:57.38 | [TK]D-Fender | theron, just add it, and you should probably tell your phone not to do any kind of nat-keep-alive. Also you should not be forwarding ports to your phone, and you SHOULD be forwrarding ports to *. |
02:57.40 | *** join/#asterisk krapper (i=krapper@wsip-64-58-154-130.oc.oc.cox.net) |
02:57.44 | drmessano | So I halted it before it began :) |
02:58.02 | theron | ok, I'm not, and I am. |
02:58.09 | jjshoe | drmessano I didn't plan on talking to you any more, *shrug* you have a shitty attitude, why would I want to continnue? |
02:58.10 | theron | checking. |
02:58.21 | drmessano | Fair enough :) |
02:58.31 | [TK]D-Fender | jjshoe, 480i was ok, but the 5i was a step backwards. |
02:58.49 | theron | [TK]D-Fender, sip reload enough for that? |
02:59.33 | [TK]D-Fender | jjshoe, NO weight to the unit or handset, tinny speakerphone (wieght would help), shitty rubber buttons, pixel based screen they are still treating like char matrix, PITA call handling, and well.. the lokcing up bit... |
02:59.38 | [TK]D-Fender | theron, yup |
02:59.49 | theron | [TK]D-Fender, same behavior. |
02:59.56 | [TK]D-Fender | jjshoe, drmessano : calm down already. |
03:00.04 | drmessano | Im calm dude |
03:00.19 | theron | [TK]D-Fender, i can see asterisk doing what it does..... vm-login....etc. |
03:00.32 | [TK]D-Fender | theron, jsut no audio, right? |
03:00.42 | theron | [TK]D-Fender, correct. |
03:00.44 | [TK]D-Fender | theron, What routers are you using? |
03:01.00 | [TK]D-Fender | theron, Cisco PIX, and many D-Links are key-offenders. |
03:01.45 | theron | [TK]D-Fender, cisco something between me and internet, then WGR614v6 |
03:01.53 | *** join/#asterisk _ShrikE-cell (n=_ShrikE-@32.162.249.118) |
03:02.01 | theron | [TK]D-Fender, I had a coworker check from his end though. |
03:02.05 | [TK]D-Fender | theron, that may be part of the issue. |
03:02.13 | jjshoe | [TK]D-Fender I don't like the 5 series either, look/feel is awful, but most everyone I know likes it more then the 9 look |
03:02.15 | [TK]D-Fender | theron, PM. |
03:02.17 | theron | [TK]D-Fender, (no firewall that side) |
03:04.49 | drmessano | double nat |
03:13.00 | jjshoe | [TK]D-Fender I'm checking to see if they've gone public with a specific phone yet... sec |
03:13.11 | drmessano | jjshoe, who do you work for? |
03:13.15 | drmessano | Im just curious |
03:13.25 | jjshoe | drmessano fonality. |
03:13.35 | drmessano | Ohh |
03:13.38 | [TK]D-Fender | jjshoe, new model? |
03:13.39 | drmessano | What do you do there? |
03:13.50 | jjshoe | [TK]D-Fender yeah, not in the 5 series |
03:13.51 | [TK]D-Fender | jjshoe, the 5i series had a lot of promise, but came up quite short. |
03:14.01 | Prayer | mpruett, experiment done yet ? |
03:14.09 | [TK]D-Fender | drmessano, .... don't go there! |
03:14.27 | nhuisman_work | [TK]D-Fender, I've created two small extension.conf examples and I was wondering if you would comment on which method is the proper way to setup inbound and outbound calls in concurrency with internal phones. The pri is on context = inbound. I read that you don't want outbound capabilities on your inbound because that would allow callers to call in and then dial out. But I also wanted to find out if it was possible to not have to write |
03:14.27 | nhuisman_work | each extension twice |
03:14.36 | nhuisman_work | http://pastebin.com/m4dda816f |
03:15.07 | JT | eww fonality |
03:15.09 | nhuisman_work | part of the question is: if I include the internal context, does it in turn include the outbound context |
03:15.22 | drmessano | jjshoe: What do you do at Fonality? |
03:15.30 | drmessano | Sounds cool |
03:15.43 | nhuisman_work | everyone else also could comment |
03:16.03 | [TK]D-Fender | nhuisman_work, You don't seem to allow 10-digit dialing... |
03:16.25 | [TK]D-Fender | nhuisman_work, also I don't see ${OUTBOUNDTRUNK} defined anywhere |
03:17.09 | nhuisman_work | oops forgot to paste that in, in the example that issue isn't what i'm worried about. |
03:18.00 | nhuisman_work | i'm wondering what the best way to go about it is, do you have double entries for all your phones, or just include the context. |
03:18.08 | [TK]D-Fender | nhuisman_work, and you should leave [internal] to contain ONLY extens related to ringing phones connected to your system |
03:18.10 | nhuisman_work | and does including the context leave me open to bad things since it includes the outbounds |
03:18.36 | [TK]D-Fender | nhuisman_work, and then make a NEW context that includes [internal] and [outbound] and point your phone's sip.conf entries to THAT. |
03:19.00 | nhuisman_work | i see |
03:20.30 | nhuisman_work | when you say phones sip.conf you mean where I specify what context each phone is in right? |
03:21.35 | jjshoe | [TK]D-Fender http://www.abptech.com/products/Aastra/sip_dect.html |
03:21.39 | nhuisman_work | adjusting the example now to see if i'm undestanding it properly. |
03:22.06 | [TK]D-Fender | jjshoe, those available now? |
03:22.19 | [TK]D-Fender | nhuisman_work, Correct |
03:22.32 | drmessano | Trixbox Pro Engineer.. hmm |
03:24.11 | jjshoe | [TK]D-Fender dunno, but they rock. |
03:24.37 | JT | trixbox pro is lame, they don't even "certify" polycom |
03:24.41 | [TK]D-Fender | jjshoe, tease :p |
03:25.21 | nhuisman_work | [TK]D-Fender, ok so sip.conf phones are in the [phones] context, pri in zapata.conf is in [inbound]. http://pastebin.com/m3c7a9e3c |
03:25.43 | drmessano | Ask jjshoe why, JT |
03:25.52 | *** join/#asterisk Dayver (n=user@ip65-44-153-126.z153-44-65.customer.algx.net) |
03:26.04 | nhuisman_work | trixbox also won't sell you support unless you use their hardware |
03:26.09 | nhuisman_work | or a list of two motherboards |
03:26.15 | nhuisman_work | .... |
03:26.18 | nhuisman_work | fuck that |
03:26.28 | jjshoe | nhuisman_work won't they? I just thought it was more *shrug* |
03:26.31 | jjshoe | I don't do shit with trixbox :) |
03:26.43 | drmessano | You dont? |
03:26.54 | Dayver | I am looking for a good SIP provider with unlimited inbound lines. Pls help. |
03:27.08 | jjshoe | [TK]D-Fender range is INCREDIBLE |
03:27.14 | drmessano | http://www.trixbox.org/forums/trixbox-pro/trixbox-pro-general-information/new-reporting-engine-released |
03:27.21 | drmessano | Not the same guy? |
03:27.22 | [TK]D-Fender | jjshoe, Yeah I have a 5iCT... I know about the range... |
03:27.38 | [TK]D-Fender | jjshoe, how many phones can an AP like that handle INDEPENDANT of each other? |
03:27.54 | nhuisman_work | drmessano, snap. |
03:28.48 | jjshoe | [TK]D-Fender not sure, I don't test it |
03:29.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
03:29.09 | [TK]D-Fender | jjshoe, I'll have to investigate, thanks for the tip though, I'm in the market for a new one. |
03:29.09 | nhuisman_work | someone was trying to sell me a different wifi band of phone |
03:29.22 | jjshoe | [TK]D-Fender there's no "base" phone, just the white thingy you see, it's awesome |
03:29.30 | jjshoe | nhuisman_work they won't sell you support but at a higher cost? |
03:29.33 | nhuisman_work | supposedly worked better for in buildings. |
03:29.49 | nhuisman_work | jjshoe, i dunno they said they only supported their list of supported hardware |
03:29.55 | jjshoe | nhuisman_work how long ago? |
03:30.00 | nhuisman_work | a month? |
03:30.03 | [TK]D-Fender | jjshoe, will investigate. Thanks. |
03:30.28 | jjshoe | nhuisman_work pro or cce? |
03:30.31 | nhuisman_work | perhaps that just means no hardware support |
03:31.10 | nhuisman_work | trying to find the links i was reading before |
03:31.16 | jjshoe | nhuisman_work pro or ce? |
03:31.28 | nhuisman_work | i can't remember |
03:31.56 | drmessano | Did I mention my favorite color is lime green? |
03:32.01 | nhuisman_work | i'll be honest I was looking at so many different choices I may have gotten them confused |
03:32.05 | nhuisman_work | i'll go find out |
03:32.09 | jjshoe | nhuisman_work ok just asked, if it's not supported hardware you have to buy hourly |
03:32.43 | jjshoe | like we've all said a million times if it's not for you, stick with ce or whatever you already use *Shrug* |
03:33.00 | nhuisman_work | yeah that's fine |
03:33.37 | *** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.mn.comcast.net) |
03:33.57 | jjshoe | [TK]D-Fender the only complaint about the wireless handsets, even on the 480ict, is that it doesn't take much to break the headset jack |
03:34.14 | drmessano | Trix are for kids |
03:34.19 | nhuisman_work | [TK]D-Fender, any comment on that example I posted last? "[TK]D-Fender, ok so sip.conf phones are in the [phones] context, pri in zapata.conf is in [inbound]. http://pastebin.com/m3c7a9e3c" |
03:34.30 | jjshoe | alright, I'm out, later home slices |
03:34.32 | nhuisman_work | speaking of wireless handsets |
03:34.40 | nhuisman_work | i need to find some |
03:34.47 | nhuisman_work | not 802.11 stuff |
03:35.03 | nhuisman_work | but just a wireless handset that is sip |
03:35.10 | nhuisman_work | i guess the other option is to buy an ata |
03:35.27 | jjshoe | nhuisman_work I just dropped a link... |
03:35.41 | jjshoe | nhuisman_work down two floors and half way through the parking garage it sounds great :) |
03:35.41 | [TK]D-Fender | nhuisman_work, ATA is a good idea. |
03:35.43 | nhuisman_work | oh, let me scroll up |
03:36.01 | jjshoe | [19:21] <jjshoe> [TK]D-Fender http://www.abptech.com/products/Aastra/sip_dect.html |
03:36.03 | [TK]D-Fender | there is this too : http://www.telephonydepot.com/product_p/105-059-m3basic.htm |
03:36.14 | nhuisman_work | ah yes |
03:36.16 | nhuisman_work | those use dect |
03:36.24 | jjshoe | ata is good if you have idjits who are going to drop them in a deep fat fryer |
03:36.33 | nhuisman_work | so it has dect in the handset and in the base? |
03:36.36 | nhuisman_work | or do I need aps everywhere |
03:36.40 | nhuisman_work | bleh aps everywhere |
03:36.56 | jjshoe | but yeah, out & |
03:37.44 | nhuisman_work | what do you mean drop them in a deep fat frier... |
03:37.46 | nhuisman_work | fryer |
03:38.25 | drmessano | Hmmm fried Trixbox |
03:39.59 | nhuisman_work | deep fried. |
03:40.06 | JT | considering that fonality seems to never contribute back to asterisk |
03:40.22 | JT | the question on when to buy fonality products seems like it has a clear answer |
03:40.23 | JT | never |
03:43.25 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
03:44.29 | nhuisman_work | context includes are one way right? |
03:44.37 | *** join/#asterisk PepOSX (n=angeldav@190.78.221.19) |
03:45.20 | [TK]D-Fender | nhuisman_work, Correct. |
03:45.52 | nhuisman_work | k that makes more sense then, I kept getting afraid that by including outbound and inbound in the phones context it was letting inbound access outbound |
03:46.20 | nhuisman_work | that makes it much simpler :) |
03:47.01 | Olobola | what is the going rate for asterisk setup/dialplan? |
03:47.46 | [TK]D-Fender | Olobola, Everything depends. What do you need? |
03:47.50 | nhuisman_work | 2 headaches for the first 10 hours |
03:48.01 | nhuisman_work | after that it's only 1 headache every 72 hours |
03:48.02 | [TK]D-Fender | nhuisman_work, nice! |
03:48.11 | nhuisman_work | ;) |
03:49.17 | Olobola | Someone wants to know to know the 'going rate'.. I suppose it could be any number. |
03:49.26 | nhuisman_work | that's for a basic 10 extension plan. |
03:49.50 | nhuisman_work | digium charges 400 for 2 hours |
03:49.57 | nhuisman_work | that's for anything you could think to have them do though |
03:50.28 | [TK]D-Fender | nhuisman_work, 400 headaches?!?! OMG my skull would fracture! |
03:50.46 | nhuisman_work | dollars. but there is a conversion from dollars to headaches, let me just look it up |
03:51.45 | nhuisman_work | 1 U.S. dollar = 0.00683293475 headaches |
03:52.17 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
03:54.59 | nhuisman_work | are includes transitive ? |
03:55.21 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
03:56.45 | [TK]D-Fender | nhuisman_work, ? |
03:57.08 | nhuisman_work | a includes b, b includes c, does a include c |
03:57.13 | [TK]D-Fender | nhuisman_work, yes |
03:57.15 | nhuisman_work | kk |
04:01.54 | nhuisman_work | is there a way to play an all circuits busy message if the call can't be completed due to insufficient available channels? |
04:02.24 | nhuisman_work | well obviously there is, should I just google for documentation or does someone already know where it can be found. |
04:03.04 | [TK]D-Fender | nhuisman_work, look at ${DIALSTATUS} after your call and play a message back if its called for |
04:03.16 | nhuisman_work | ah yes that makes sense |
04:03.25 | nhuisman_work | i have a macro I can change to do that, thx. |
04:03.51 | *** join/#asterisk _ShrikE-cell (n=_ShrikE-@32.162.249.118) |
04:05.06 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
04:07.53 | nhuisman_work | hmm i need an extra enter for 011 + country code + city code + number |
04:08.00 | nhuisman_work | enter=entry |
04:12.52 | nhuisman_work | how do you do that? the country codes are not a set number of digits |
04:13.56 | [TK]D-Fender | nhuisman_work, you just do 011. and don't restrict the rest. |
04:14.01 | nhuisman_work | oh |
04:14.12 | nhuisman_work | so "011." |
04:14.20 | [TK]D-Fender | nhuisman_work, yup |
04:14.26 | nhuisman_work | ah thanks. |
04:18.09 | *** join/#asterisk ncampion (n=ncampion@24-159-207-172.dhcp.roch.mn.charter.com) |
04:24.15 | ncampion | I need some help with asterisk-gui's configure script, specifically the --with-zaptel option. Anyone around? |
04:25.21 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
04:25.58 | [TK]D-Fender | ncampion, GUI's are not supported in this channel. |
04:26.08 | ncampion | ok |
04:26.14 | nhuisman_work | asterisknow? |
04:26.15 | [TK]D-Fender | ncampion, #asterisk-gui is their channel |
04:26.16 | nhuisman_work | #asterisknow |
04:26.20 | ncampion | thanks guys |
04:26.20 | nhuisman_work | ah nm |
04:26.27 | nhuisman_work | didn't know about asterisk-gui |
04:26.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
04:27.23 | nhuisman_work | it would be nice to see an addon for provisioning phones for asterisk |
04:27.30 | nhuisman_work | instead of having to write all the scripts |
04:27.57 | [TK]D-Fender | nhuisman_work, GUI's do things like that. |
04:28.10 | nhuisman_work | i don't think asterisk-gui does |
04:28.21 | [TK]D-Fender | nhuisman_work, And the moment it configures your phones fo you it starts thinking its running the show, and not you. |
04:28.47 | [TK]D-Fender | nhuisman_work, yes, they has scripts for configuring Polycom's at least that I've heard of. not sure of others. |
04:28.51 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
04:29.16 | [T]ank | anyone know how to disable the do not disturb button on a polycom 301? |
04:29.32 | nhuisman_work | i know asterisknow has no provisioning, i thought that it used the asterisk-gui |
04:29.34 | nhuisman_work | double checking |
04:30.32 | [TK]D-Fender | [T]ank, you can remap the key in your provisioning. Go check your admin guide. |
04:32.14 | [TK]D-Fender | [T]ank, in SIP 2.1.0+ you seem to be able to set it for server-side DND to which I presume you can send it to a dead-end as well |
04:36.25 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
04:40.52 | nhuisman_work | what are the cheapest atas for the price these days? |
04:41.02 | nhuisman_work | sorry i meant cheapest atas that aren't crap |
04:41.08 | [TK]D-Fender | nhuisman_work, Linksys. |
04:41.14 | *** join/#asterisk hmodes (n=hmodes@ipv6.matrix.gs) |
04:41.30 | nhuisman_work | they don't have 1 port ones eh |
04:41.30 | [TK]D-Fender | nhuisman_work, SPA-2102 is the norm. PAP2's are a bit too low. |
04:41.35 | [TK]D-Fender | nhuisman_work, they do. |
04:41.52 | nhuisman_work | i only see 2 port ones |
04:41.53 | Prayer | Sipura2000 I got several from eBay |
04:42.11 | [TK]D-Fender | http://www.voipsupply.com/product_info.php?products_id=320 |
04:42.35 | nhuisman_work | funny that linksys doesn't have that on their site |
04:42.42 | nhuisman_work | ah, discontinued |
04:42.56 | [TK]D-Fender | nhuisman_work, there's a LOT they don't have on their site, including a lot of current stuff... |
04:43.12 | nhuisman_work | what the hell is that, the replacement is a 2 port |
04:43.35 | nhuisman_work | oh well i'll just use those and leave them at the patch panel level |
04:43.41 | nhuisman_work | then I can buy less |
04:44.16 | *** join/#asterisk dcmwai (n=dcmwai@75.217.95.219.brf02-home.tm.net.my) |
04:44.28 | nhuisman_work | each interface is a separate phone line eh |
04:44.36 | nhuisman_work | i guess that's why there are 2 rj45 interfaces |
04:44.44 | nhuisman_work | each is a different ip acting as a sip device |
04:44.47 | jblack | oh boy. Duck |
04:44.55 | *** join/#asterisk dkatz334 (n=guest@h-72-245-152-13.nycmny83.covad.net) |
04:45.17 | [TK]D-Fender | nhuisman_work, nope. |
04:45.26 | dcmwai | hello all. |
04:45.32 | [TK]D-Fender | nhuisman_work, the SPA-2102 can be a ROUTER. |
04:45.44 | dcmwai | I would like to have your advise on linux-ha + Asterisk |
04:45.47 | [TK]D-Fender | nhuisman_work, and it does SIP out the single RJ-45 |
04:45.50 | jblack | Nugget: Nah. You're buying a stripped down version of an spa8k. |
04:45.53 | nhuisman_work | ooh |
04:46.04 | [TK]D-Fender | jblack, the SPA-8000 is *new* :) |
04:46.22 | jblack | [TK]D-Fender: Ok, then the 8k is a stripped up. :) |
04:46.29 | nhuisman_work | so if I don't need a router there is nothing less then this eh |
04:46.30 | [TK]D-Fender | jblack, So its more like a duct-taped bunch of SPA-2102's by comparison :) |
04:46.42 | dcmwai | My coulleauge haev been configuring 2 Asterisk server + Linux HA but we have some problem. |
04:47.01 | jblack | point being, rj11-1 is ethernet in, and rj11-2 is a masqued network -- forced to 192.168.0.0 |
04:47.08 | dcmwai | the things work fine when primary fail... backup will take over. |
04:47.34 | jblack | gah gah gah. rj45! rj45, I mean |
04:47.55 | dcmwai | but when primay come back online... something wong, the backup seem are unable to send things back... anyone have setup things like that before? |
04:48.20 | nhuisman_work | when you say unable to send |
04:48.22 | nhuisman_work | what do you mean |
04:48.26 | nhuisman_work | heartbeat doesn't fork over the ip? |
04:48.38 | nhuisman_work | you probably need to ask that question in another room if so, more of a linux ha question |
04:49.11 | nhuisman_work | LOL |
04:49.13 | dcmwai | nhuisman_work, it does fork over the ip. However, it was to the backup Mac Address |
04:49.35 | nhuisman_work | on voipsupply you click on sipura spa-1000 it says discontinued |
04:49.37 | dcmwai | nhuisman_work, so I also confused and wonder if that is the right way .. |
04:49.42 | nhuisman_work | which sends you do the spa 10001 |
04:49.46 | [TK]D-Fender | nhuisman_work, then SPA-2102 it is. |
04:49.54 | nhuisman_work | which is then discontinued and sends you do the spa 2xxxx |
04:50.17 | nhuisman_work | i guess i'll just not use the router |
04:50.42 | [TK]D-Fender | nhuisman_work, exactly. Its good for re-use later. |
04:51.21 | [TK]D-Fender | nhuisman_work, And supports T.38 which may come in handy later (esp having a 2nd port for it) |
04:51.46 | nhuisman_work | what about the pap2t-na |
04:52.36 | jblack | Yeah, I suppose you can always get more on them... 8/250, or 6/240 |
04:52.39 | [TK]D-Fender | nhuisman_work, no T.38, no router, wimpier CPU (not sure about multiple G.729, etc) |
04:52.54 | nhuisman_work | honestly i'll never reuse them |
04:52.57 | nhuisman_work | is why i ask |
04:53.06 | [TK]D-Fender | nhuisman_work, PAP2 works... your call... |
04:53.15 | nhuisman_work | k |
04:54.51 | nhuisman_work | fax + asterisk = not worth trying? |
04:57.56 | nhuisman_work | well thanks for the help today. I'm out for the night. |
04:57.59 | nhuisman_work | *gone* |
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05:08.59 | dkatz334 | Hello all. |
05:09.19 | dkatz334 | I'd like to know recommendations for RX faxing with asterisk. |
05:09.30 | dkatz334 | Not intereted in TX, just RX. |
05:09.33 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
05:10.02 | dkatz334 | Currently using spandsp with a digium TE212P and ecm, it doesn't work with one station, the fax never comes through. |
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06:06.06 | UnixDog | this bites |
06:06.31 | UnixDog | some of the patches wont work right on 1.6 |
06:06.40 | UnixDog | this is going to be painfull |
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06:24.50 | UnixDog | wow o far 4 patches can not be used |
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06:36.48 | CCFL_Man2 | cli-msn seems to suck |
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06:44.22 | fozzmoo | Good evening/morning. |
06:45.01 | fozzmoo | quick question: I'm trying to set some variables in AMI with the Set: parameter so that a dialplan that gets run by Originate has some data. |
06:45.30 | fozzmoo | I'm using 1.2.x. Is "Set: variable=value" correct styntax? |
06:45.40 | fozzmoo | Or do I need to use Action: SetVar? |
06:46.35 | Corydon76-dig | fozzmoo: who told you it was ever Set in AMI? |
06:47.41 | Corydon76-dig | fozzmoo: Try running Action: ListCommands |
06:47.58 | fozzmoo | Corydon76-dig: jsmith. :) |
06:48.13 | fozzmoo | I need to pass some variables to the dialplan from AMI. |
06:48.13 | Corydon76-dig | fozzmoo: never has been and still isn't |
06:48.31 | Corydon76-dig | That's very risky |
06:49.11 | fozzmoo | here's what jsmith told me a couple days ago: |
06:49.13 | fozzmoo | (03:07:32 PM) jsmith: Action: Originate |
06:49.13 | fozzmoo | (03:07:54 PM) jsmith: Channel: Local/18012545677@longdistance/n |
06:49.13 | fozzmoo | (03:07:59 PM) jsmith: Context: somecontext |
06:49.14 | fozzmoo | (03:08:07 PM) jsmith: Exten: 1234 |
06:49.14 | fozzmoo | (03:08:11 PM) jsmith: Priority: 1 |
06:49.14 | fozzmoo | (03:08:29 PM) jsmith: Set: seller_id=51293851 |
06:49.28 | fozzmoo | And that "seller_id" line is the one I'm trying to work with. |
06:49.35 | Corydon76-dig | Are you sure he was talking about AMI and spool files? |
06:49.52 | fozzmoo | yeah- he explained how to telnet into port 5038 and all that before this. |
06:49.52 | Nugget | telnet is eeeeeeevil! |
06:49.54 | Corydon76-dig | err, and NOT spool files? |
06:50.03 | fozzmoo | telnet |
06:50.18 | Corydon76-dig | I'll have to beat him down next time I see him |
06:50.33 | fozzmoo | So, I'm up a creek or what? |
06:50.44 | Corydon76-dig | SetVar is the name of the action |
06:50.50 | fozzmoo | Okay. |
06:51.08 | fozzmoo | So if I do a bunch of those before the Originate action, my dialplan will be aware of the variables? |
06:51.35 | Corydon76-dig | There is no way to set arbitrary variables from an Action: originate |
06:51.54 | fozzmoo | Hmm. I'll have to talk to jsmith about that inaccuracy. :) |
06:52.04 | fozzmoo | Feeding me full of crap. |
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06:52.36 | fozzmoo | Corydon76-dig: Thank you very much for your assistance. |
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07:03.40 | fozzmoo | Alright. Another AMI question: How can I use the SetVar: action to set variables for a local psuedo channel? In the realm of possible? |
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07:10.17 | Corydon76-dig | fozzmoo: get on the Asterisk CLI and type: show manager command setvar |
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07:17.10 | citats | Corydon76-vcch: i thought you could use Variable: to set variables with originate, but maybe that has changed since i did it last |
07:18.06 | fozzmoo | citats: That's what I just read about |
07:18.13 | fozzmoo | I tried it and it worked for the first Variable: |
07:18.21 | fozzmoo | Now I've got to figure out what's wrong with the rest of my variables. :) |
07:18.49 | citats | fozzmoo: iirc you could only pass one Variable: arg and there was some kind of seperator |
07:19.04 | fozzmoo | citats: that makes sense too. |
07:19.08 | citats | like Variable: var1=this&var2=that |
07:19.17 | citats | or maybe Variable: var1=this|var2=that |
07:19.29 | fozzmoo | A HA! |
07:19.31 | fozzmoo | Variable: var1=23|var2=24|var3=25 |
07:19.33 | tsabi | hi |
07:19.35 | fozzmoo | (voip-info.org) |
07:19.45 | fozzmoo | citats: THANKS! |
07:19.46 | tsabi | i have a small problem with a grandstream BT101: |
07:19.59 | tsabi | asterisk writes this on console when i wish to make a call with it: |
07:20.01 | tsabi | chan_sip.c:3670 sip_write: Asked to transmit frame type 4, while native formats is 0x1 (g723)(1) read/write = 0x8 (alaw)(8)/0x4 (ulaw)(4) |
07:20.17 | tsabi | what this means? |
07:20.32 | J4k3 | tasbi: I'd guess your codec selections aren't the same on both sides |
07:20.36 | J4k3 | does the call actually complete? |
07:20.52 | tsabi | on asterisk i allow=all |
07:21.04 | tsabi | the call is done, but the phone is silcence |
07:21.17 | J4k3 | I'd try forcing a setting in asterik |
07:21.20 | Corydon76-dig | tsabi: one side negotiated g.723, but you don't have a translation path from G.723 to anything else |
07:21.51 | tsabi | Corydon76-dig: what thsi means? :) |
07:21.57 | Corydon76-dig | The only way to get G.723 working (legally) is to buy the hardware transcoder board |
07:22.06 | J4k3 | your codec settings aren't the same on both sides. |
07:22.14 | Corydon76-dig | otherwise, you CANNOT use G.723 |
07:22.27 | tsabi | ok, no G723, what other codec can i use? |
07:22.41 | J4k3 | g711 ulaw/alaw and g729 (with license, of course) |
07:22.42 | Corydon76-dig | Try ulaw |
07:22.58 | tsabi | ulaw, that didnt worked last time |
07:23.06 | tsabi | i tryed every coedec on the phone :( |
07:23.17 | tsabi | but i will try again, maybe the error message will be different |
07:23.28 | J4k3 | make sure you've updated the firmware on the phone to the latest |
07:23.36 | tsabi | yeah i updated |
07:23.56 | tsabi | i can make SIP calls with this settings, but no ISDN calls |
07:24.00 | tsabi | via ZAP channel |
07:24.32 | J4k3 | try alaw |
07:25.18 | tsabi | ok, i tryed alaw |
07:25.35 | tsabi | now i made a call, and it says no translation path from g723 to alaw |
07:25.51 | tsabi | so my ISDN is in g723 codec? |
07:26.09 | tsabi | but my linksys phone works well :S |
07:26.10 | J4k3 | afaik its alaw except in north america and japan (ulaw) |
07:26.12 | tsabi | i dont understand this |
07:26.25 | J4k3 | I suspect you've got the codec settings on the phone all confused |
07:26.34 | tsabi | and i use 711 alaw on the linksys too |
07:26.45 | J4k3 | and/or, you need to force codecs in asterisk (disallow=all, allow=[codec]) |
07:26.53 | J4k3 | thats how my bt101s work |
07:27.05 | tsabi | hmm, i see |
07:27.09 | tsabi | ok |
07:31.25 | tsabi | hmm, it semms it works |
07:31.56 | tsabi | bug in asterisk? :S |
07:32.04 | J4k3 | bug in grandstream |
07:32.17 | J4k3 | I'd trust asterisk a lot more than I'd trust a $30 pos phone |
07:32.27 | J4k3 | don't get me wrong, I use gs101's every day, they do work |
07:32.28 | jblack | Grandstreams sure do have a poor reputation here |
07:32.32 | J4k3 | but they're.... grandstreams |
07:32.46 | tsabi | hmm, ok |
07:33.07 | jblack | I was actually chased down by people after they found out I had bought a gs, and they got me to return it |
07:33.33 | fozzmoo | heh. |
07:33.41 | hmodes | asterisk still needs some serious work re: codec handling, grandstreams aside, imo |
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07:34.05 | jblack | what piece of free software is done... :) |
07:34.33 | tsabi | yeah i tested it in every way now, its works now |
07:34.39 | tsabi | thny for big help :) |
07:35.00 | hmodes | yeah, truedat |
07:35.05 | hmodes | i'm not complaining, just saying :) |
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07:35.48 | tsabi | in my opinion, if i should force codec usage in asterisk, and asterisk says about codec confusion, it seems that asterisk confused, and asterisk have a bug |
07:36.13 | hmodes | eh, it's not a 'bug' per say, it's working as designed |
07:36.25 | hmodes | just, the design could use some work |
07:36.47 | hmodes | once you get used to it, it's not such a big deal except in a few very specific situations |
07:37.23 | hmodes | also seems to be a big hangup for people who don't already understand how the codec negotiation 'works' |
07:37.27 | bintut | when to use realm on sip.conf ? is there a standard value/format for the realm? |
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08:15.53 | awk | ~the book |
08:15.56 | awk | !the book |
08:16.07 | awk | somebody give me a link to that asterisk book |
08:16.18 | J4k3 | ~thebook |
08:16.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
08:16.22 | awk | thanks |
08:16.41 | mort_gib | http://www.asteriskdocs.org/ |
08:17.05 | awk | thanks |
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09:06.01 | yang | In grandstream phones, I have the option enabled when I dial the number it tells me in voice which number is calling me, where can i disable that, I looked all over the options |
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09:21.31 | ^shark_ | hi guys one question -- if i connect my asterisk box to another box that has got a PSTN connection, will my extensions be able to dial through to this PSTN network....? is this possible..? |
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09:35.15 | dominic1 | Hi, anybody knows what happened to openhardphone.org? |
09:37.50 | hi365 | it went soft |
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09:45.57 | ^shark_ | if i connect my asterisk box to another box that has got a PSTN connection, will my extensions be able to dial through to this PSTN network....? is this possible..? |
09:50.56 | ronr | ^shark_: yes, that's possible |
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09:57.23 | ^shark_ | ronr: any guides to make this connection to the PSTN network from my box |
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10:05.22 | Zeeek | so who's on 1.6 ? |
10:11.13 | DarKnesS_WolF | evening geeks |
10:11.21 | Zeeek | morning! |
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10:15.12 | DarKnesS_WolF | Zeeek: what with 1.6 ? |
10:19.31 | Zeeek | anyone using it? |
10:19.39 | Zeeek | I'd like to hear about it if so |
10:23.38 | DarKnesS_WolF | Zeeek: not yet but yes i'll upgrade my asterisks to it |
10:26.03 | Zeeek | from 1.4? |
10:26.16 | ronr | ^shark_: just connect the two boxes using sip or iax (see the book) and forward the call from the second box to pstn |
10:27.35 | DarKnesS_WolF | Zeeek: yep |
10:27.48 | DarKnesS_WolF | IAX much better ^shark_ |
10:27.53 | DarKnesS_WolF | ~book |
10:27.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
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10:46.23 | DarKnesS_WolF | if have a sip phone and i'm dilaing using zap card what i the varibale that holds the sip phone number ? |
10:46.26 | DarKnesS_WolF | to use it in file ? |
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10:47.27 | DarKnesS_WolF | ${CALLERID(num)} |
10:47.29 | DarKnesS_WolF | sweet ! |
10:47.31 | Zeeek | CLI> show application dial |
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10:51.46 | ^shark_ | ronr: thanks man -- i had gone out for lunch |
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11:03.55 | ^shark_ | !dual servers |
11:04.02 | ^shark_ | sorry guys |
11:08.13 | DarKnesS_WolF | what is wrong wiht that ? exten => s,n,MixMonitor(${FILENAME}.wav|system(/usr/bin/lame -S -V7 -B24 --tt ${FILENAME} --add-id3v2 /var/spool/asterisk/monitor/${FILENAME}.wav /var/spool/asterisk/monitor/${FILENAME}.mp3)) |
11:08.32 | DarKnesS_WolF | after the call done it don't create the mp3 even it is executes the sytem function |
11:11.00 | DarKnesS_WolF | and even i did replace | with || |
11:11.08 | nebojsajsimic | hi all |
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11:12.12 | nebojsajsimic | little help when i call 2N gateway i don't have callerID can it be fixed |
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11:12.33 | nebojsajsimic | in asterisk i get only name of pear ..... |
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11:17.16 | DarKnesS_WolF | set(callerid(num)=1234567890) |
11:19.40 | nebojsajsimic | where??? |
11:19.43 | nebojsajsimic | to set |
11:19.46 | DarKnesS_WolF | in ur dialplan |
11:19.56 | DarKnesS_WolF | what is the problem i mean what if u don't have CallerID |
11:19.57 | DarKnesS_WolF | so what :D? |
11:20.01 | DarKnesS_WolF | what is wrong? |
11:20.10 | nebojsajsimic | i have ID |
11:20.31 | nebojsajsimic | call come from 2N gateway |
11:20.35 | nebojsajsimic | from GSM |
11:20.56 | DarKnesS_WolF | where is asterisk here? |
11:21.00 | nebojsajsimic | and i try to make something to put my Agents to queue |
11:21.28 | nebojsajsimic | but y try to use CallerID for identification |
11:23.31 | nebojsajsimic | any idea??? |
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11:27.39 | DarKnesS_WolF | nebojsajsimic: copy in pastebin ur errors and ur exntesions.conf part for this |
11:27.47 | ArchSSM | If the channel.c returns "didn't get frame from channel: SIP/.... ". Where do I start to debug? |
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11:31.00 | DarKnesS_WolF | ArchSSM: i think codecs |
11:31.15 | DarKnesS_WolF | make sure ur using compatible codecs |
11:31.53 | ArchSSM | Hmm.. I see. |
11:32.31 | ArchSSM | The thing is: It happens during a conversation. I can have an active conversation for everything from 10 seconds to 30 minutes, and then it just hangs up. |
11:32.37 | ArchSSM | With the error mentioned. |
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11:33.20 | BipBip | hello all |
11:33.31 | ArchSSM | Hello :) |
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11:35.09 | BipBip | ii'm having trouble with one config wich i think it should be simple but i'm going nuts. anyone with some energy to waste in me? |
11:35.36 | ArchSSM | Maybe ;). Just ask and we'll help if we know. |
11:37.08 | BipBip | nice :) I'm trying to configure asterisk to register a sip account, the issue is that i need to use a proxy and it's not working as i thought it should |
11:37.39 | BipBip | register => user:pass@proxy.voip.sapo.pt:5070 |
11:38.00 | BipBip | then [proxy.voip.sapo.pt] fromdomain=sapo.pt |
11:38.15 | BipBip | the user should be user@sapo.pt |
11:38.17 | mosty | BipBip, use pastebin.com |
11:38.27 | BipBip | mosty: k |
11:38.39 | mosty | just x out the password and username |
11:39.56 | BipBip | http://pastebin.com/mfc36ba0 |
11:40.40 | FlatFoot | morning all |
11:40.47 | BipBip | hello FlatFoot |
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11:41.12 | ArchSSM | Good morning |
11:41.25 | FlatFoot | got a bit of trouble with res_odbc.so and res_config_odbc.so they are not present on my box |
11:41.28 | mosty | BipBip, what happens when you try to register? |
11:41.37 | FlatFoot | using FreeTDS do i need them installed ? |
11:41.57 | mosty | FlatFoot, i think you need unixodbc dev libraries for odbc, not freetds |
11:42.11 | BipBip | with this config, he trie to use port 5060 |
11:42.20 | BipBip | tries |
11:42.31 | FlatFoot | i had this working on another box but this install does not seem as happy |
11:44.30 | BipBip | this is the config that i need: http://pastebin.com/m4fdc766f |
11:44.47 | FlatFoot | the old box was running 1.4.11 this one is on 1.4.17 can't find too much info on this . Running FreeBSD can anyone suggest an answer |
11:45.38 | BipBip | ... using 1.4.17 compiled |
11:45.46 | FlatFoot | from ports |
11:45.56 | mosty | FlatFoot, so do you have the package i mentioned? |
11:46.10 | ArchSSM | BipBip: register => UUU:PPP@proxy.voip.sapo.pt:5070/sapo <--- should be correct |
11:46.27 | ArchSSM | BipBip: You had the username mentioned twice. And what does the cli say? |
11:46.45 | FlatFoot | mosty: i have not installed it myself cos all the info about FreeTDS said it has some of the same files as unixODBC so i did not need it |
11:46.56 | BipBip | ArchSSM: http://pastebin.com/m3e96da37 |
11:47.17 | mosty | BipBip, tried setting outboundproxyport? and does it register ok? |
11:47.23 | BipBip | humm, he is trying sapo.pt:5070 wich is wrong |
11:49.21 | BipBip | ok, i changed host to proxy.voip.sapo.pt and now he tries to register, but with domain proxy.voip.sapo.pt and not with sapo.pt |
11:49.27 | FlatFoot | mosty: i am installing it now |
11:50.27 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
11:51.16 | AlienPenguin | hi, i noticed that in the 200 that asterisk responds to my client i do not have the INFO in the Allow header. Where can i change this behaviour? |
11:54.57 | BipBip | is there a way to force the realm? from what i'm seeing in the logs, the realm used is voip.sapo.pt and not sapo.pt. When i force the user to have the domain, the realm is changed to sapo.pt but i receive bad password |
11:56.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:56.10 | mosty | BipBip, there is an example at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
11:56.35 | BipBip | mosty: thanks, i'll have a look |
12:01.02 | *** join/#asterisk neuwald (n=neuwald@189.50.81.11) |
12:01.26 | neuwald | Hi folks. Does bindport=5060, 1020 will work in sip.conf ? |
12:01.44 | neuwald | to sip listen in two or more udp ports... |
12:02.23 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
12:03.00 | mosty | neuwald, i don't think so, do you get an error message when you try? |
12:03.17 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
12:03.27 | neuwald | I didn't try, first asking here, cause it's a production server |
12:03.32 | ArchSSM | neuwald: Why not use iptables for this? |
12:03.57 | mosty | neuwald, if you have a production server, you should also have a test server |
12:03.58 | neuwald | I'm using pf on firewall, I think it's a good idea too |
12:04.06 | neuwald | mosty ok, I'll test, just a minute |
12:05.25 | neuwald | bindport=5060,1020 |
12:05.34 | neuwald | didn't worked. only 5060 is listen |
12:11.44 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
12:12.24 | AlienPenguin | i have seen some posts in asteriskguru that other ppl share my problem: asterisk responds 403 to any SIP INFO message an UA tries to send to the remote party. Is there a way to change this behaviour? |
12:16.03 | *** join/#asterisk javar (n=javar@69.79.134.24) |
12:17.12 | troubled | mornin's |
12:23.14 | nebojsajsimic | i try to fix something in sip header i have callerID |
12:23.22 | nebojsajsimic | hov can i catch it |
12:23.25 | nebojsajsimic | ???? |
12:23.30 | nebojsajsimic | *how |
12:23.37 | *** join/#asterisk whymarkwhy (n=koos@nas-mid.nashuamp.co.za) |
12:23.55 | whymarkwhy | hi there |
12:23.59 | ArchSSM | Hello |
12:24.09 | nebojsajsimic | Hi |
12:24.36 | whymarkwhy | is there anyway you can edit you extension.conf file from the dial plan using one of aterisk's applications |
12:25.48 | nebojsajsimic | i think YES you hav on Asterisk GUI file editor :) |
12:25.55 | nebojsajsimic | you hawe |
12:25.57 | nebojsajsimic | *** |
12:27.43 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
12:27.45 | nebojsajsimic | How to Get parameter from sip header???? |
12:27.48 | nebojsajsimic | eny help |
12:27.53 | nebojsajsimic | any help ...... |
12:28.01 | nebojsajsimic | poooooor my eng |
12:28.28 | ArchSSM | I'm afraid I don't know. |
12:29.01 | ArchSSM | And just a tips - The more question mark you have, the less of a chance you have of getting help ;) |
12:29.53 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
12:30.29 | ^shark_ | hello |
12:30.52 | ^shark_ | what is the logic behind connecting dual servers, i have tried to check this out but have failed |
12:31.21 | ^shark_ | ok i know u have to connect dial peers... but i dont know the full details and howto |
12:31.28 | ^shark_ | i am using SIP |
12:31.47 | ^shark_ | Oooops i meant dial plans |
12:32.16 | *** join/#asterisk zaur_pronet (n=zaur_pro@85.132.55.134) |
12:33.57 | Zeeek | so installed 1.66beta |
12:34.12 | Zeeek | s/1.66/1.6 |
12:34.19 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
12:34.51 | tzafrir | The computer has not gone up in smoke yet? |
12:37.02 | Zeeek | no and zaptel is working |
12:37.19 | Zeeek | but I haven't seen anything register (network changes so I need to chg extensions and sip) |
12:37.30 | Zeeek | tzafrir you tried it yet? |
12:37.42 | tzafrir | Not really |
12:38.11 | Zeeek | Not really? Is that a no? |
12:38.29 | Zeeek | so what can I do with 1.6 that I couldn't with 1.2? |
12:38.42 | Zeeek | probably nothing with my antiquated equipment :) |
12:38.59 | ArchSSM | I'm actually a bit disappointed. It's not much new when you look at the change log |
12:39.25 | tzafrir | Zeeek, you mean: compared to 1.4, right? |
12:39.34 | Zeeek | NO 1.2! |
12:39.48 | Zeeek | ArchSSM I'm thinking a lot of the stuff is total re-write, though ? |
12:40.04 | Zeeek | IOW, maybe not a lot of new features, but lots of better code |
12:40.20 | ArchSSM | Well. both sip handler and the channel handler has increased performance and readability |
12:41.28 | *** join/#asterisk the_5th_wheel (n=edd@dsl-242-89-144.telkomadsl.co.za) |
12:41.57 | Zeeek | Sounds right out of the Digium flyer :) |
12:42.15 | ArchSSM | hehehe |
12:42.18 | ArchSSM | exactly :) |
12:42.26 | Zeeek | Funny how "legacy" used to me inheritance (good) and now it means crappy code (bad) |
12:42.39 | ArchSSM | true,true :) |
12:42.42 | Zeeek | as in "we've removed the legacy code" |
12:42.57 | Zeeek | like "legacy application" = MS-DOS |
12:43.10 | the_5th_wheel | hi. are there any CLI checking avaliable for linux? My phone doesnt support CLI, but i the my predecessor got CLI working on his windows pc, but i dont use that. |
12:43.25 | the_5th_wheel | (and my predecessor doesnt return my calls) |
12:43.31 | Zeeek | heh |
12:43.37 | Zeeek | they rarely do |
12:43.44 | the_5th_wheel | (caller line identification) |
12:43.54 | Zeeek | callerid |
12:44.05 | the_5th_wheel | yeah |
12:44.07 | Zeeek | as against Command Line Interpreter |
12:44.34 | Zeeek | the_5th_wheel I don't understand the question, though |
12:44.44 | whymarkwhy | if you have your sip client(xlite) connected to more than one asterisk server how do you dial out on say a zap one one server and sip on the other? |
12:45.31 | the_5th_wheel | on the windows pc next to me, they have a bubble popping up every when his phone rings, saying who is phoning |
12:45.38 | the_5th_wheel | is there such an application for linux |
12:46.13 | Zeeek | I hear Jabber can do that for you. On WIndows I use ... |
12:46.28 | ArchSSM | yep |
12:46.29 | Zeeek | "YAC" |
12:46.31 | ArchSSM | freeswitch :) |
12:46.39 | Zeeek | Yet Another Callerid |
12:46.52 | Zeeek | it's actually done for TiVo |
12:47.16 | the_5th_wheel | that is what my neighbour has running on his pc |
12:47.20 | the_5th_wheel | but i dont use windows |
12:47.32 | Zeeek | nice, eh? Linux, dunno. Try FOP |
12:47.34 | DarKnesS_WolF | whymarkwhy: i think u do ad #1 or something like that not sure i don't use SIP since ages. |
12:47.48 | DarKnesS_WolF | the_5th_wheel: which application is that ? |
12:47.54 | the_5th_wheel | Yac |
12:48.16 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
12:48.38 | Zeeek | ... |
12:48.43 | Zeeek | mercilessly kicked |
12:49.05 | ^shark_ | hi |
12:49.14 | ^shark_ | in this scenario >> exten => _8XXX,2,Dial(SIP/myserver:passwordB@SIPserverB/${EXTEN:1},30,r) |
12:49.14 | Zeeek | yes hit #1 #2 etc for multiple servers |
12:49.46 | tzafrir | Zeeek, well one new feature is the ability to write dialplan logic in Lua, in case you're interested |
12:49.54 | ^shark_ | myserver: is an ip address and the password: is the userpassword!? |
12:50.26 | tzafrir | chan_zap has many small improvements |
12:51.08 | Zeeek | Lua? Never hoid of it |
12:51.31 | Zeeek | so far I just realized I did a make config and blew away the old dialplan :) |
12:52.14 | Zeeek | but it is running, with one FXS so I just need to find something to call |
12:52.26 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
12:52.32 | tzafrir | e.g: try: zap set <tab><tab> with a recent Zaptel |
12:53.53 | tzafrir | You can always call an echo test |
12:54.06 | tzafrir | Originate Zap/1 application Echo |
12:54.23 | tzafrir | did I mention originate is new as of 1.4? highly useful |
12:54.36 | Zeeek | how do I reload extensions now? |
12:54.59 | tzafrir | dialplan reload |
12:55.02 | Zeeek | zap set - cool! |
12:55.07 | Zeeek | ok, thx |
12:55.40 | tzafrir | (yeah I know: much worse that 'extensions reload' for tab completion) |
12:56.09 | Zeeek | IT WORKS!!!! asterisk 1.6beta1 WORKS!!!! |
12:56.17 | Zeeek | absolutely amazing |
12:56.18 | ArchSSM | ehm... shouldn't it? |
12:56.26 | ArchSSM | it's a beta... not pre-alpha0 |
12:56.27 | ArchSSM | :) |
12:56.32 | Zeeek | they don't always :) |
12:56.45 | ArchSSM | true |
12:56.57 | Zeeek | note : http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/README?revision=99123&view=markup |
12:57.05 | Zeeek | refers to 1.2 in the text |
12:57.33 | Zeeek | "To discover the major new features of Asterisk 1.2, please visit ..." |
12:57.50 | ArchSSM | hehehe |
12:57.51 | Zeeek | that's the thing about the INternet |
12:57.54 | ArchSSM | nice |
12:58.01 | Zeeek | it's like an exploding star |
12:58.20 | Zeeek | You can NEVER find and correct all those little oversights |
12:58.50 | Zeeek | anyway, so far I'm impressed. |
12:59.03 | Zeeek | tzafrir what else can I try for fun, eh? |
12:59.31 | Zeeek | tzafrir ext<tab>rel<tab> was a lot better |
12:59.51 | Zeeek | maybe terminal function keys are the answer to that |
12:59.53 | tzafrir | e<tab><tab>, actually, IIRC |
13:00.22 | Zeeek | you're right, just tried it on 1.2 |
13:00.39 | Zeeek | and all these years I've been tyoing 12 more keys than nec. |
13:01.16 | Zeeek | so, no more "core"? good, I hated that |
13:02.00 | tzafrir | why no more core? it's there |
13:02.53 | Zeeek | sip show peers works |
13:03.04 | Zeeek | wasn't it core sip.... in 1.4? |
13:03.10 | tzafrir | core show globals |
13:03.18 | tzafrir | no, it was sip |
13:03.49 | hmmhesays | tzafrir: your guys must not like emailing me back |
13:05.04 | Zeeek | ok, get ready, I'm going to try to call from 1.2 to 1.6 |
13:05.07 | tzafrir | Zeeek, core set chanvar is also nice |
13:08.15 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
13:10.40 | DarKnesS_WolF | tzafrir: what is chanvar? |
13:10.46 | DarKnesS_WolF | tzafrir: any updtes about the suse init ? |
13:10.54 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.235.168) |
13:10.59 | tzafrir | Set the value of a normal channel variable |
13:11.06 | tzafrir | (as opposed to a global variable) |
13:13.40 | DarKnesS_WolF | ah i c |
13:13.41 | DarKnesS_WolF | nice nice |
13:15.42 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
13:17.18 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
13:28.53 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
13:31.24 | nebojsajsimic | me again |
13:31.44 | nebojsajsimic | how to substring something |
13:32.00 | nebojsajsimic | i fing way to logon into queue |
13:32.20 | nebojsajsimic | when i call ${CALLERIDNUM} |
13:32.37 | nebojsajsimic | i get sip/123456-897946546 |
13:32.45 | nebojsajsimic | i need only 123456 |
13:32.54 | nebojsajsimic | how can i do it ???? |
13:33.07 | ArchSSM | regexp it :) |
13:33.27 | nebojsajsimic | please help little |
13:33.29 | nebojsajsimic | :) |
13:33.57 | penguinFunk | regex is easy |
13:34.06 | penguinFunk | get your dog/cat to walk on the keyboard |
13:34.07 | Kigh | 1,Set(FOO=${CUT(CALLERIDNUM,/,2)} |
13:34.13 | ArchSSM | hehehe |
13:34.14 | Kigh | 2,Set(FOO=${CUT(FOO,-,1)} |
13:34.26 | Kigh | FOO contains 123456 |
13:34.28 | *** join/#asterisk af_ (n=getsmart@88-149-230-244.dynamic.ngi.it) |
13:34.49 | Kigh | this is no job for a regexp imho .. simple cutting |
13:34.49 | nebojsajsimic | thx Kigh |
13:34.51 | Kigh | np |
13:34.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
13:34.56 | ArchSSM | Kigh: Is that a 'better' way than exporting it and doing a real regexp of it? |
13:34.58 | ArchSSM | ah... ok :) |
13:35.17 | Kigh | ArchSSM: it is basic asterisk extension language, yes its 'better' |
13:36.37 | ArchSSM | Ok. |
13:40.38 | [TK]D-Fender | And that isn't a callerid... thats a CHANNEL NAME |
13:40.47 | ^shark_ | i need some basic help -- if my asterisk box is not on the domain but only has a static local IP on the LAN and my mail server is also on the LAN but in the domain, how do i tweak postfix to send my voicemails to my inbox? |
13:40.54 | ^shark_ | do i just install postfix |
13:41.19 | [TK]D-Fender | ^shark_: this is a question for #postfix , not here |
13:41.39 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
13:42.09 | ^shark_ | [TK]D-Fender: ok -- i was just wondering how you guys set this up |
13:42.31 | [TK]D-Fender | ^shark_: most of us just let sendmail do its thing |
13:43.09 | nebojsajsimic | is there ${chanelname} |
13:43.10 | nebojsajsimic | ??? |
13:43.10 | ^shark_ | [TK]D-Fender: so its juts a matter of installing send mail is that right>? |
13:43.50 | nebojsajsimic | and is there way to get Callerid from sip header |
13:43.55 | nebojsajsimic | ??? |
13:44.14 | [TK]D-Fender | ^shark_: funny how you think that everything magically "jsut works" and will also "just work the way *I* want it to". Do all of your corporate functions happen without any configuration as well? |
13:44.48 | [TK]D-Fender | nebojsajsimic: "show functions" <- go look at the list and see what stands out. |
13:44.51 | nebojsajsimic | i receive call from GSM gateway and when i need CALLERID from real caller who call gsm gateway |
13:45.04 | nebojsajsimic | thx |
13:45.23 | ^shark_ | [TK]D-Fender: let me try and do something, will be back laters |
13:45.27 | [TK]D-Fender | nebojsajsimic: "sip/123456-897946546" is a channel name. If that somehow ended up in the callerid its because someone overrode it manually. |
13:46.15 | ManxPower | The answers you seek are *within*, grasshopper. |
13:46.39 | [TK]D-Fender | ManxPower: and coated with grease so the hard you grasp, the more slips through your fingers! |
13:46.40 | Zeeek | seek *this* |
13:47.06 | ManxPower | nebojsajsimic: ${CALLERID} contains the callerid. |
13:47.12 | Zeeek | so I tried 1.6 |
13:47.19 | ManxPower | If it is not in ${CALLERID} then is is not Caller*ID. |
13:47.28 | ManxPower | Zeeek: What did you think of it? |
13:48.05 | Zeeek | so far, no make problems, but I need to make a dialplan as I deleted the old (unused 1.4) one |
13:48.22 | ManxPower | Zeeek: I have high hopes for 1.6. |
13:48.23 | Zeeek | I made one SIP call thru USA, worked fine. |
13:48.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
13:48.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
13:48.58 | Zeeek | what I really need to do is delete the entire extensions.conf and start from scratch with the years of collected wisdom |
13:49.18 | Zeeek | but playtime is limited |
13:49.44 | Zeeek | and now I have no fixed ip on that connex so it makes things a little harder |
13:50.39 | ManxPower | Once 1.6 hits RC status I might just take the pain and start testing it in semi-production enviroments. |
13:50.56 | *** join/#asterisk hijacked (n=argh@66.255.220.17) |
13:51.18 | [TK]D-Fender | ManxPower: I'll probably wait about 2-3 weeks after 1.6.0 and see if the body-count slows. |
13:51.21 | ManxPower | And trust me, testing PBX upgrades in my environment means pain, usually caused by headaches caused by whiney users. |
13:51.41 | Zeeek | heh, well I don't have *that* problem |
13:51.51 | ManxPower | [TK]D-Fender: I'm making the effort to help with 1.6 if I can. |
13:51.51 | Zeeek | my wife is used to it |
13:52.01 | *** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
13:52.52 | nDuff | Is there a way to tell if I'm providing ringback on an outgoing call or if it's being provided by the remote end? I have "pri debug" logs, but don't know what to look for. |
13:53.20 | ManxPower | nDuff: if the call is not answered, you are not providing ringback |
13:54.08 | *** join/#asterisk ming_zym (n=ming_zym@124.14.237.225) |
13:54.09 | nebojsajsimic | is there some good book with implemented functions and examples ..... |
13:54.22 | nDuff | s/outgoing call/inbound call being forwarded to an outside line/ |
13:54.23 | ManxPower | now you might not be sending the Q.931 messages to make the remote telco provide ringback -- but that was not your answer. |
13:54.54 | nDuff | hmm. |
13:55.27 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
13:55.34 | *** join/#asterisk mkl1525 (n=qwertz@mail.cp-net.de) |
13:55.46 | ManxPower | nDuff: is the forwarded call answered before it is forwarded? |
13:56.31 | nDuff | yes. |
13:56.33 | mkl1525 | Hi, (* 1.2 and 1.4) haven't found it in voip wiki but is it possible to use extconfig.conf for agents.conf? |
13:56.53 | [TK]D-Fender | nebojsajsimic: "show function [FUNCTIONNAME]" and ... |
13:56.55 | [TK]D-Fender | ~book |
13:56.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
13:56.58 | [TK]D-Fender | ^^^^^^^^^^ |
13:57.07 | [TK]D-Fender | nebojsajsimic: then there is the |
13:57.09 | [TK]D-Fender | ~wikis |
13:57.10 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
13:57.27 | ManxPower | then Asterisk would have to provide INBAND (controlled by /etc/asterisk/indications.conf) ringback and you would not see that in pro debug messages |
13:58.17 | nDuff | ahh; thank you. |
13:58.50 | ManxPower | "Don't know how to handle indication 15" or some sort of message like that is common if you don't have an indications.conf and you need it. |
13:59.38 | nDuff | (the remote end says they don't see my outgoing calls at all, though I'm receiving CALL PROCEEDING, ALERTING and DISCONNECT on them; they were wondering who was providing ringback on forwarded test calls -- but I can clarify that by changing indications.conf to make mine sound different). |
14:00.16 | ManxPower | "see"? |
14:01.03 | nDuff | "Not sure what to tell you. I'm not seeing the inbound call attempt at all, but the system doesn't have a low level protocol debug option." |
14:01.16 | ManxPower | nDuff: Aha! |
14:02.02 | ManxPower | nDuff: We had a similar issue with outgoing leg of an automatically forwarded call failing. Turned out we added a 9-1- to the CLID when the call came in and our telco got VERY upset when it saw calls being dialed with that callerid. |
14:02.23 | ManxPower | fixed the callerid to be CORRECT (nothing but numbers, 10 digits only) and it started working just fine. |
14:02.29 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
14:03.04 | nDuff | ManxPower: hmm; I'll try that. |
14:03.37 | ManxPower | and for good measure, overide the callerid on the outgoing leg to be a number on your PRI. |
14:03.48 | *** join/#asterisk destructure (n=de@66.193.229.254) |
14:04.07 | ManxPower | I guess I should get to work soon. |
14:04.14 | ManxPower | I have to deal with end users today. *sob* |
14:04.31 | ManxPower | (all non-VoIP stuff) |
14:04.32 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
14:05.24 | ManxPower | We are VLANing one of the offices, did most of the work last night, but a few people locked their offices and I could not get in to renumber their printers, etc. |
14:05.29 | Zeeek | ah. |
14:05.35 | Zeeek | of course. |
14:07.17 | ManxPower | There are other issues too like the electricition never putting the power outlets required for the security cameras (wondered why they were not working...) |
14:08.32 | ManxPower | I just dropped my cig lighter in my coffee. 8-( |
14:08.54 | d3wayne | that's not a good start to the day :-\ |
14:09.08 | ManxPower | d3wayne: no, it is not. |
14:10.14 | Uni | how goes brudda man? |
14:10.17 | ManxPower | BTW, everyone, I am now accepting a limited number of clients. VoIP, WAN, etc. Preference is for clients in the SE USA, but not a requirement. I do consulting. |
14:10.24 | nDuff | It goes, it goes. |
14:11.00 | ManxPower | But I do require that a potential client is not a cheap bastard. |
14:11.09 | hmmhesays | ManxPower: don't we all? |
14:11.15 | ManxPower | hmmhesays: apparently not. |
14:11.38 | hmmhesays | I specialize in being macgyver |
14:12.06 | ManxPower | "I've got 7 SPA-2100s and my boss wants to convert the PBX to VoIP. Oh, we have no budget!" |
14:12.16 | shido6 | lol |
14:12.50 | ManxPower | Call me selfish, but if you want me to get out of bed and help you, you had better compensate me for the pain of being awake. |
14:13.27 | Zeeek | thanks for that image |
14:14.18 | ManxPower | I *LIKE* a challenging project. I do not like futile projects. 8-) |
14:14.54 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
14:15.24 | ManxPower | Well, challenging technical projects at least. I'm not fond of trying not to call someone a moron when they are connected to both the corporate network and the coffee shop's WiFi network at the same time and are having problems. |
14:15.43 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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14:16.38 | ManxPower | I compromised with saying something like "Don't tell me this, if HR finds out, you could be fired." That shut them up pretty fast. |
14:17.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
14:17.41 | nebojsajsimic | ok i get AgentCallbackLogin("SIP/81733-08215090", "81733#1234#") |
14:18.01 | nebojsajsimic | what i do wrong |
14:18.40 | [TK]D-Fender | nebojsajsimic: go read the instructions again. |
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14:19.42 | d3wayne | ManxPower: where in the SE ? |
14:28.06 | *** join/#asterisk ncampion (n=ncampion@nat/ibm/x-19a4bc0c59812909) |
14:28.36 | *** join/#asterisk tiav (n=tiav@212.73.244.125) |
14:29.26 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
14:29.44 | flujan | hi guys... :) |
14:30.13 | *** join/#asterisk BaseJam (n=BaseJam@gw2.imperia.net) |
14:30.18 | flujan | simple question... Is the busy-level feature available in version 1.4.15 and above? |
14:30.37 | BaseJam | have anybody a german howto for me ?! |
14:30.54 | ManxPower | d3wayne: I'm based near Birmingham AL, but I have several clients in the New Orleans, LA area. |
14:32.02 | ManxPower | BaseJam: Not really, but I think you have to start out by not having a sense of humor and speaking like you have a cold all the time. |
14:36.01 | *** join/#asterisk sob0l (n=sobol@devel4.net) |
14:36.37 | De_Mon | BaseJam a what? |
14:36.37 | [TK]D-Fender | flujan: huh? |
14:36.52 | flujan | hi [TK]D-Fender . How are you doing? |
14:37.02 | [TK]D-Fender | flujan: Muh.... need vacation. |
14:37.16 | flujan | I googled about a cal-limit issue that I am having... |
14:37.51 | BaseJam | De_Mon, i am searching a german tutorial to configure an asterisk server, because i am a beginner in asterisk ;-) so i can learn faster and more abour asterisk |
14:37.52 | flujan | [TK]D-Fender: Wow, I recently passed through a surgery... Man I love to work.. It is too bad staying on a hospital.. :( |
14:38.09 | sob0l | I have a problem with CDR'a billsec > duration and duration=0 |
14:38.22 | *** join/#asterisk nejme_eddinne (n=nejm@41.225.251.170) |
14:38.37 | nejme_eddinne | hi |
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14:38.42 | flujan | [TK]D-Fender: http://svn.digium.com/view/asterisk/trunk/CHANGES?revision=62672&view=markup |
14:38.44 | [TK]D-Fender | sob0l: Perhaps you are billing for "ringing time" on a call that was never answered... |
14:39.03 | sob0l | [TK]D-Fender: the call was answered |
14:39.16 | [TK]D-Fender | sob0l: with a duration of 0? odd. |
14:39.19 | flujan | [TK]D-Fender: There is the new call-limit/busy-level stuff that I need... |
14:39.32 | sob0l | [TK]D-Fender: and CLID=src=dst |
14:39.38 | De_Mon | BaseJam oh a asterisk setup tutorial in german.. Nope none that I know of |
14:39.39 | sob0l | that is stranger |
14:39.43 | flujan | guys who answers a queue need to have a call-limit of 1 to optimize the queue delivery.... |
14:39.47 | nejme_eddinne | I have some questions can any one help me please ? |
14:40.04 | flujan | but when the guys need to answer a call and placed another... it does not work ... |
14:40.10 | De_Mon | flujan do you have a mantis ticket number for that feature? |
14:40.27 | [TK]D-Fender | flujan: if you are using a direct channel interface you should be able to put "call-limit=1" in their sip.conf entries for example. |
14:40.41 | [TK]D-Fender | nejme_eddinne: ... |
14:40.42 | flujan | [TK]D-Fender: I am already doing it... |
14:40.44 | [TK]D-Fender | ~ask |
14:40.45 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
14:40.53 | [TK]D-Fender | ^^^^ |
14:41.09 | BaseJam | De_Mon, damn, do you have a good beginner setup tutorial in english ? |
14:41.19 | [TK]D-Fender | BaseJam: here : |
14:41.21 | De_Mon | ~book |
14:41.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
14:41.25 | [TK]D-Fender | ~jerjerguide |
14:41.25 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
14:41.25 | *** join/#asterisk nejme_eddinne (n=nejm@41.225.251.170) |
14:41.28 | flujan | [TK]D-Fender: but the guys who have the call-limit=1 needs to answer a call, and place another call in most of the cases... this is issue I am having... |
14:41.44 | BaseJam | thanks at all ;-) |
14:41.54 | [TK]D-Fender | flujan: I think there was an "incominglimit" somewhere as well. |
14:42.12 | flujan | I need to allow the guys to receive just one call and place another... so call-limit=2 that is it... |
14:42.13 | nejme_eddinne | check ur trunk configuration... |
14:42.16 | nejme_eddinne | call-limit |
14:43.20 | flujan | [TK]D-Fender: incominglimit and call-limit are equivalent. outgoinglimit is not supported anymore. |
14:43.22 | [TK]D-Fender | nejme_eddinne: So, just ask your question, don't ask to ask. |
14:43.38 | BaseJam | De_Mon, and that is easy to use ? i hate to be a beginner in something |
14:43.40 | BaseJam | ;-) |
14:43.46 | [TK]D-Fender | flujan: Ah, well you could always run through Local and check if their on a call first |
14:44.04 | flujan | [TK]D-Fender: do you mean using the dialplan? |
14:44.05 | [TK]D-Fender | BaseJam: Go read the book. Its the best thing out there. |
14:44.25 | [TK]D-Fender | flujan: yup |
14:44.30 | BaseJam | [TK]D-Fender, sure, i'm downloading atm |
14:45.15 | [TK]D-Fender | BaseJam: the quick guide I linked you is a good small sample for some learning concepts onces you've gone through chapter 5 of the book. |
14:46.06 | BaseJam | thank you |
14:48.15 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
14:48.22 | whymarkwhy | how do you check witch linux version are you running suse or fedora? |
14:48.54 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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14:48.54 | whymarkwhy | i know its not a linux question |
14:49.14 | tzafrir | cat /etc/redhat-version |
14:49.21 | whymarkwhy | thx |
14:49.39 | De_Mon | flujan I'm not sure if http://bugs.digium.com/view.php?id=11180 is the right ticket for your question or not, if it is no it's not in 1.4 |
14:49.55 | tzafrir | sorry: /etc/redhat-release . There should also be a similar for for SuSE |
14:51.39 | whymarkwhy | can't belive it the most stable system of them all is a fedora 4 box, never gave my a days problems |
14:52.05 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
14:52.15 | whymarkwhy | can one install yast on fedora? |
14:52.17 | Dr-Linux | anybody tried vicidial? |
14:52.31 | nejme_eddinne | thank you, so I want to know if I can execute AGI script and making a DIAL() at the same time... |
14:52.42 | tzafrir | whymarkwhy, why do you think it would still be stable after you install yast on it? |
14:52.50 | tzafrir | yast is very intrusive |
14:52.59 | tzafrir | So it surely changes things |
14:53.05 | whymarkwhy | so better not then |
14:53.09 | whymarkwhy | thx |
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14:53.47 | nejme_eddinne | Dr-Linux ca i Help you ? |
14:53.58 | nejme_eddinne | can* |
14:54.06 | Dr-Linux | sure |
14:54.12 | Dr-Linux | nejme_eddinne: can i /msg you? |
14:54.26 | nejme_eddinne | yes of corse |
14:54.27 | *** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de) |
14:54.30 | nejme_eddinne | course* |
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14:55.12 | De_Mon | Woooo... backport of audio hooks to 1.4? |
14:55.15 | lilalinux | is it possible to push a text instead of a phone number to a sip phone (e.g. SL75 WLAN) instead of the callerid? |
14:55.37 | De_Mon | lilalinux I believe so |
14:55.44 | De_Mon | lilalinux try it and let us know :) |
14:55.53 | [TK]D-Fender | lilalinux: Sure |
14:56.53 | nejme_eddinne | to explain more, for exemple if I want to use the DIALSTATUS variable I have to wait until the call is hunged up to user it... |
14:57.12 | nejme_eddinne | I want to do it in real time... |
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15:02.26 | nejme_eddinne | no one have an idea ? :( |
15:05.18 | lilalinux | thx |
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15:08.26 | [TK]D-Fender | nejme_eddinne: :"M" is the only important option. |
15:08.56 | nejme_eddinne | Macro ? |
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15:09.12 | nejme_eddinne | I can Dial and execute a Macro at the same time ? |
15:09.31 | zuchmir | can anyone help me with this message: Registration for '61300xxxx@sip2.bbpglobal.com' timed out, trying again (Attempt #177) |
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15:10.06 | skopii | I am trying to use txfax and get TIFF/F format is not compatible. I used the ghostscript command from voip-info (http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax) any advice? |
15:10.09 | Dr-Linux | zuchmir: see if the host sip2.bbpglobal.com is up |
15:10.29 | skopii | nejme_eddinne: your macro can dial or goto if you want |
15:10.30 | zuchmir | it is, i think it's a nat issue |
15:10.45 | Dr-Linux | zuchmir: try ip address instead |
15:11.00 | nejme_eddinne | thanks |
15:11.49 | [TK]D-Fender | zuchmir: read up : |
15:11.51 | [TK]D-Fender | ~sipnat |
15:11.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
15:11.58 | [TK]D-Fender | ^^^^^^^^^^^^ |
15:12.05 | eric_hill | skopii: I think the txfax uses a CCITT/G3 fax, not G4 for some reason. You'll need to transcode before sending. |
15:12.37 | eric_hill | skopii: (And that's 1+year old information from my brain so it may have changed recently) |
15:13.58 | skopii | eric_hill: I will google for that...thanks |
15:14.37 | *** join/#asterisk Teeli (n=tili@cm48.gamma244.maxonline.com.sg) |
15:15.08 | *** join/#asterisk ThoMe (n=tm@keks.be) |
15:15.10 | ThoMe | hi. |
15:15.13 | ThoMe | kann hier auch wer deutsch? |
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15:15.51 | lilalinux | ThoMe: koennen ja, duerfen nein ;-) |
15:16.16 | ThoMe | lilalinux: hm. :-/ kurz #asterisk.de ? |
15:18.43 | zuchmir | does this make sense: http://pastebin.com/d27688dea |
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15:20.03 | zuchmir | my * is behind NAT, but sip2.bbpglobal.com is presumably not, do i specify NAT=yes? in the [bbpglobal] section? |
15:21.23 | skopii | hey eric_hill the command on the tx/rx page says to use -sDEVICE=tiffg3 for ghostscript.....did I miss something? |
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15:22.44 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
15:25.21 | km- | how do I make the bot talk to me |
15:25.23 | km- | !tfot |
15:25.32 | km- | wanna find the free copy of the tfot :) |
15:25.34 | km- | ~tfot |
15:25.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
15:25.36 | eric_hill | skopii: I don't think you missed anything. The inbound format needed to be G3 (tiffg3), but you should be able to use imagemagick in a script to convert an inbound G4 to G3. |
15:25.37 | km- | there we go |
15:26.01 | hmodes | hey hey, it's a pete |
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15:30.38 | skopii | eric_hill: I am just trying to send a fax, I am converting a pdf to a tiff using gs, and then I made a callfile to call my fax machine and set the data to sendfax.agi (which just starts the txfax app with options $faxfile|caller |
15:31.16 | *** join/#asterisk atop (n=a@oaktyres.force9.co.uk) |
15:31.56 | skopii | I am wondering...does anyone actually use txfax? |
15:33.06 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
15:33.11 | lmadsen | no one uses fax :) |
15:33.24 | eric_hill | We're using LightningFAX since it supports a hardware board, but we do 300+ faxes per day... |
15:33.53 | eric_hill | Anyone know if I can "clone" a voicemail after it's left into a second mailbox? |
15:34.18 | atop | The telephone number shown for inbound calls is missing the first zero, so a call from 0151 2223333 shows as being from 151222333, this is using the latest version of asterisk 1.4 and anciliary files, calls are being presented over ISDN line connected to Sangoma card with latest version on wanrouter. I've searched the log files, config files and web and I'm finding nothing. |
15:34.24 | skopii | eric_hill: you can just symlink two mailboxes together |
15:34.26 | atop | Anyone recognise the problem? |
15:34.51 | eric_hill | I need about 12 separate mailboxes that copy all of their voicemail into a central "logging" mailbox... |
15:35.37 | eric_hill | atop: Turn on debugging on your PRI and verify that the provider is indeed passing all of the digits to you. |
15:35.50 | zuchmir | tk-dfender: does my scenario fit into: "Asterisk as a SIP client behind nat, connecting to outside SIP Proxies "? |
15:36.13 | skopii | hmmm not sure about that one....sorry eric_hill I would say "make a script to rsync every 5 min or something" but that is surely not a good solution lol |
15:36.30 | atop | I did that, and I've checked with the provider who /claims/ they are sending it, but the asterisk debug does not show it. Given that, should I assume the provider is wrong and go back to them? |
15:38.13 | eric_hill | atop: I would say that the provider may be wrong, but probably doesn't have the technical expertise to add that extra digit. |
15:38.50 | eric_hill | atop: Can you just work around it? exten => 151XXXXXXX,1,Goto(realcontext,0${EXTEN},1) |
15:38.58 | atop | thought so. I was getting frustrated looking for something at this side. Is there a way to make asterisk.... |
15:39.02 | atop | ah, yeah something like that! |
15:39.36 | atop | not sure how that would work tho, as the inbound number could be anything |
15:42.11 | skopii | couldn't you just use a dialing rule like _X.? |
15:43.06 | x86 | _ShrikE: you around? |
15:43.22 | *** join/#asterisk zenobic (n=zenobic@174-204-116-85.dsl.manitu.net) |
15:43.31 | x86 | anyone ever mess with Adit 600 channel banks? |
15:43.54 | x86 | I seem to have a major alarm light on the TDM controller |
15:43.54 | atop | I'll get on the line to the carrier again and see what I can find out; thanks all. |
15:44.04 | zenobic | hello:i have a voicemail problem. |
15:45.30 | [TK]D-Fender | zuchmir>my * is behind NAT, but sip2.bbpglobal.com is presumably not, do i specify NAT=yes? in the [bbpglobal] section? <- for your peer/user, nat=no. |
15:47.15 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
15:47.39 | zenobic | Can anyone explain me a the difference between Local Channel and SIP (Voicemail) . |
15:48.29 | FlatFoot | anyone got a copy of asterisk-1.2.12.1-codec-negotiation-20060926.diff.gz ? |
15:48.50 | zenobic | http://pastebin.com/d4f237837 |
15:50.10 | x86 | ah ok, cleared the TDM controller alarm... only using one T1 interface on the Adit 600, but both were enabled... |
15:50.32 | x86 | so now I've got all green lights, but for some reason I'm still not getting dial tone from asterisk |
15:51.36 | [TK]D-Fender | zenobic: You clearly have a Dial(Local//..... line in your dialplan, go LOOK FOR IT. |
15:52.13 | [TK]D-Fender | zenobic: And "Local" and "SIP" in this case have absolutely nothing to do with each other. |
15:52.47 | [TK]D-Fender | zenobic: You just compared the originating channel of 1 call to a channel DIALED because of your dialplan. These are apples & oranges. |
15:53.15 | jpsharp | Apples & sausage |
15:53.29 | [TK]D-Fender | jpsharp: Sure, why not... |
15:54.12 | x86 | [TK]D-Fender: ever mess with an adit 600? |
15:54.14 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
15:54.19 | [TK]D-Fender | x86: nope |
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15:54.38 | [TK]D-Fender | x86: Stop wasteing your time on channel-banks you schmuck! :p |
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15:57.50 | skopii | anyone using txfax? |
15:59.21 | supjigator | we tried but hylafax seems to just work. |
15:59.51 | skopii | supjigator: can hylafax send faxes out? |
15:59.53 | zuchmir | tkd-fender: still getting Registration ... timeout |
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15:59.58 | skopii | I just need to send faxes |
16:00.03 | [TK]D-Fender | zuchmir: pastebin your sip.conf |
16:00.24 | skopii | supjigator: nm me checks docs thanks =]] |
16:01.24 | supjigator | yea hylafax does that well. |
16:02.15 | tzanger | Adit600 rocks |
16:02.39 | tzanger | I need to experiment with iaxmodem and hylafax |
16:02.49 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
16:02.52 | zenobic | [TK]D-Fender: thx. i checked the dialplan: exten => 3847413,10,Dial(Local/413@default,240,tw) ...and later this: exten => 413,1,Voicemail(su413@default) |
16:03.15 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
16:03.53 | supjigator | tzanger: hylafax can do t38 sip |
16:04.34 | [TK]D-Fender | zenobic: well.. you put it there. Not much more to say. |
16:04.57 | zuchmir | http://pastebin.com/d42e1f79e |
16:05.58 | [TK]D-Fender | zuchmir: externip = my.fqdn.com <- NO. this is for IP, not HOST. use externhost / externrefresh for that. |
16:07.06 | zuchmir | tkd-fender: as far as i can see the only diff between the two is that exernhost gets updated more frequently (it does a DNS lookup more often) |
16:07.22 | *** part/#asterisk BaseJam (n=BaseJam@gw2.imperia.net) |
16:07.45 | tzanger | supjigator: sure, but I still need to feed it t38 from somewhere |
16:11.08 | zuchmir | http://pastebin.com/d2fc724d1 |
16:12.17 | andrewn | maps.google.com working for anyone? all i see is a man in a uniform |
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16:13.04 | andrewn | strange, it's ok now |
16:13.04 | zuchmir | tk-defender: http://pastebin.com/d2fc724d1 |
16:13.46 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
16:14.48 | zuchmir | would bbpglobal be initiating a connection (and thus not being able to get in)? |
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16:15.35 | nny_1 | anyone who deals a lot with polycoms seen issues where they don't notify the user of new voicemails? (over a course of a couple of days..) |
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16:16.45 | penguinFunk | mine work perfect |
16:16.50 | penguinFunk | snom300's |
16:17.09 | nny_1 | yeah i have snoms out there too, they work well |
16:17.15 | _ShrikE | nny_1: I may have seen something like that way back. What firmware are you running? |
16:17.17 | penguinFunk | do you have mailbox parameters set in sip.conf ? that match the mailboxes in voicemail.conf? |
16:17.31 | nny_1 | _ShrikE: let me check stand by |
16:18.27 | [TK]D-Fender | nny_1: Nope. |
16:20.32 | *** join/#asterisk galeras (n=Martin@201.245.228.127) |
16:21.22 | nny_1 | _ShrikE: I am assuming by polycom "Bootrom" is the firmware on the phone, in that case, it is 3.2.2.0019 |
16:21.29 | galeras | Dear Sirs, please which is de default set for memberdelay parameter ? |
16:21.40 | nny_1 | _ShrikE: let me see if I can peep the provision FTP server and see what version resides on there |
16:21.41 | _ShrikE | nny_1: No the bootrom is not the firmware |
16:21.47 | nny_1 | _ShrikE: heh ok stand by |
16:22.17 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
16:23.16 | zenobic | [TK]D-Fender: do i have to define: Local/413 => 413 in voicemail.conf (there's: 413 => 413), too? |
16:23.26 | nny_1 | _ShrikE: is it under SIP app version? |
16:23.34 | [TK]D-Fender | zenobic: that has NOTHING to do with voicemail! |
16:23.35 | nny_1 | _ShrikE: in that case 2.1.0.2708 |
16:23.48 | [TK]D-Fender | zenobic: You don't even know why you're doing that in your dialpllan do you? |
16:23.48 | _ShrikE | nny_1: That's it |
16:24.19 | _ShrikE | nny_1: Thats way newer than what I was running back when I saw that. |
16:24.46 | nny_1 | _ShrikE: yeah I have a polycom 501 here same model firmware that has no issues |
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16:25.27 | *** mode/#asterisk [+o russellb] by ChanServ |
16:25.57 | zuchmir | tk-defender: any ideas? |
16:26.05 | nny_1 | anyone know of an easy way to drop a voicemail in someones box without calling? :) |
16:29.19 | nny_1 | hmm maybe this error in messages might* offer a clue |
16:29.35 | nny_1 | er nm |
16:29.43 | nny_1 | thats a app_queue.c error :) |
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16:34.13 | brodiem | has anyone done a chan_local style agentlogin()? |
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16:35.58 | *** part/#asterisk galeras (n=Martin@201.245.228.127) |
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16:36.10 | brodiem | or better yet, pstn agents |
16:38.14 | [TK]D-Fender | zuchmir: check your forwarding |
16:40.18 | nny_1 | hmm.. I guess the only way to test voicemail is to leave one.. problem is right now their dialplan doesn't go to vm during the day.. it rings two sip clients, and if no one answers, autoanswers, puts the caller on hold, and continues to ring the clients... |
16:40.46 | nny_1 | can you just cp a msg0001.WAV to the voicemail folder to test? |
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16:51.01 | fedya | what does the config line that sets up the cdr fields look like? |
16:51.16 | fedya | i'm trying to figure out where in my config the dst field is set |
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16:54.42 | dishem | it doesnt use system time? |
16:55.56 | rrittenhouse | Is it possible to use a vonage voip "box" as an ATA with asterisk? |
16:57.44 | [TK]D-Fender | rrittenhouse: iF YOU CAN UNLOCK IT AND KEEP IT UNLOCKED, SURE |
16:57.55 | nny_1 | heh well |
16:58.05 | nny_1 | just left a voicemail here and the polycom ist silent |
16:58.12 | nny_1 | so whatever I did there I managed to dupe here :\ |
16:58.15 | hesco | If a caller ID phone number I'm seeing on calls coming from the * server is not set in the call file and its not in extensions.conf, where might it be getting into the configuration? |
16:59.10 | jpsharp | Do you have a a "mailbox" set in the phone's sip.conf entry? |
16:59.11 | rrittenhouse | [TK]D-Fender, I would assume you could just "redirect" the request to get the config file to your own config file.. if its that easy |
17:01.00 | dishem | rrittenhouse: yeah but the configs are encrypted |
17:01.12 | nny_1 | so is this right? 1.) asterisk notifies of new voicemails when the sip client re-registers, and you can add a checkmwi to sip.conf to change the message waiting indicator timing? |
17:01.34 | rrittenhouse | dishem, ah.. darn. Know where I can read up on the subject? |
17:01.43 | rrittenhouse | Not sure if this is the right place to discuss it |
17:02.25 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
17:02.27 | dishem | rrittenhouse: Not sure of any place specific unfortunately.. I'd google for how to unlock your specific device |
17:03.12 | dishem | also voip-info.org wiki sometimes has info on devices |
17:03.56 | dishem | I think with a lot of vonage devices you had to have never connected the device to the internet |
17:04.15 | dishem | I have a pap2 that I hope is easy to unlock some day. |
17:05.41 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
17:06.03 | nny_1 | well crap |
17:06.13 | nny_1 | it seems that default checkmwi is already 10 |
17:06.38 | nny_1 | although it seems that the host=dynamic means the client registers itself. not sure if this could be an issue with the voicemail NOTIFY |
17:06.57 | nny_1 | wonder if there is a way to send the NOTIFY from console to test |
17:11.39 | davevg-btwtech | nny_1, see sip_notify.conf, probably what you are looking for |
17:12.40 | skopii | nny_1: it may be an issue with your firewall...I haven't messed around with polycom phones for a while. there's an option in the polycom config (sip.cfg or phone.cfg I forget) to make the phone reregister every 120S or something |
17:13.04 | skopii | http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk |
17:13.35 | nny_1 | davevg-btwtech: thanks will look, i have the same configs here and our polycom works nicely |
17:13.36 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
17:13.44 | nny_1 | skopii: they are both on the local LAN |
17:13.50 | nny_1 | skopii: but thanks for the suggestion |
17:14.05 | nny_1 | I had qualify off on the system, wondering if that would do anything |
17:14.55 | skopii | in the sip user? that just displays them differently in sip show peers |
17:15.03 | *** join/#asterisk fozzmoo (n=fozz@66.7.122.158) |
17:15.05 | nny_1 | skopii: yeah |
17:15.07 | skopii | it will show the latency |
17:15.16 | nny_1 | skopii: yeah thats what I figured |
17:15.21 | fozzmoo | Hey. Can someone show me an example of using func_curl in a dialplan? |
17:15.21 | skopii | at least that's what I thought it does heh |
17:15.22 | nny_1 | need to find a way to test it |
17:16.05 | nny_1 | can I just cp a new voicemail message to their INBOX? |
17:16.05 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
17:16.05 | skopii | not sure, my guess is yes though |
17:16.55 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
17:18.20 | nny_1 | yeah it works |
17:18.21 | nny_1 | cool |
17:18.30 | nny_1 | easy way to test it at least |
17:18.54 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
17:20.25 | _ShrikE | When did polycom release SIP 3.0? |
17:20.47 | zenobic | nny_1: asterisk 1.2? Theres also a txt. |
17:21.34 | nny_1 | zenobic: nah 1.4 |
17:21.43 | nny_1 | zenobic: er but yeah I copied both |
17:21.48 | nny_1 | is the txt file for 1.2? |
17:22.31 | zenobic | not only 1.2 (i just asked). tried also with both :) |
17:22.41 | nny_1 | ahh thanks |
17:22.46 | skopii | isn't the txt file just the callerid? |
17:23.02 | nny_1 | yeah it works here.. sad thing is the issue I am chasing is only* on that system.. methinks may be pebkac somehow |
17:23.13 | nny_1 | skopii: and time called, duration, etc |
17:23.30 | *** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk) |
17:23.52 | skopii | well if I was still at my old job I could dig up the * and polycom configs for ya =\ |
17:24.10 | nny_1 | hehe nah i think this problem isn' |
17:24.27 | nny_1 | t related to the configs, I think it may be a user error or similar ilk |
17:24.29 | *** part/#asterisk fozzmoo (n=fozz@66.7.122.158) |
17:24.44 | nny_1 | but now that I can force a voicemail... |
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17:40.13 | bsdwarrior | I can call out with a .call file fine, but If I send the same info to the manager (using originate), I get this error.Starting SIP/104 at mycontext,,1 failed so falling back to exten 's' |
17:41.52 | De_Mon | bsdwarrior mycontext,,1 isn't valid so that would make sense. What *Exactly* are you sending with originate? |
17:42.28 | bsdwarrior | de_mon, mycontext - its really outbound, i will pastebin |
17:43.16 | outtolunc | bsdwarrior: use Exten: not Extension: |
17:43.48 | bsdwarrior | hmm |
17:43.58 | bsdwarrior | http://pastebin.com/d238ed8eb |
17:44.25 | bsdwarrior | thats awesome |
17:44.26 | bsdwarrior | thanks man |
17:44.30 | outtolunc | np |
17:45.01 | bsdwarrior | outtolunc - Its correct in the docs too. I can't read |
17:45.23 | outtolunc | it is one of those 'differences' that has gotten a LOT of people over the years |
17:46.43 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
17:47.02 | [T]ank | is there a dialtone sound file? |
17:48.12 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
17:48.13 | [T]ank | i am doing a section of an auto attendant where an outbound call may be placed... i want it to do something like: exten => blah,1,Background(dialtone) |
17:48.28 | [T]ank | but i do not see any sound files that are named what I would expect for dial tome |
17:48.29 | [T]ank | tone |
17:48.56 | [T]ank | exten => 9833899,1,Answer() |
17:48.56 | [T]ank | exten => 9833899,n,Background(tt-monkeys) |
17:48.56 | [T]ank | exten => 9833899,n,Set(TIMEOUT(digit)=5) |
17:48.56 | [T]ank | exten => 9833899,n,Set(TIMEOUT(response)=10) |
17:49.12 | bsdwarrior | is there any way to show on the phone who they are calling? I.E. Im sending calls to the phones but the callerid gets cutoff |
17:50.27 | outtolunc | bsdwarrior: the easiest way is to set a var, then set that var as teh callerid on its way back in |
17:50.39 | outtolunc | you can do that with 'Variable: ...' |
17:51.03 | outtolunc | remember to use __VARS |
17:51.08 | bsdwarrior | im setting the caller Id as "Callback phonenum", but the phone number is cut off ? |
17:51.43 | outtolunc | your callerid should be complete as in 'Some Name <xxxxxxxxxx>' |
17:52.13 | bsdwarrior | thanks. fixed |
17:52.59 | x86 | [TK]D-Fender: hey i forgot the value in the polycom config file to have the phone auto-check for updates (firmware updates), do you know it off the top of your head? |
17:54.58 | jjshoe | [T]ank why not play dtmf digits? |
17:55.42 | kyron | Q: does * follow the even/odd version release scheme such as the Linux kernel? |
17:56.04 | bsdwarrior | once I send a call with the manager and originate, is there any way to get the status of the call besides using cdr ? |
17:56.22 | [T]ank | jjshoe: found DISA, seems to do what I am after. |
17:56.30 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
17:56.48 | skopii | bsdwarrior: maybe AGI... |
17:57.58 | outtolunc | bbiam nature |
17:58.05 | jjshoe | bsdwarrior what exactly are you doing? |
17:59.06 | x86 | kyron: 1.0 == stable, 1.2 == stable, 1.4 == stable, 1.6 == development... |
17:59.12 | x86 | so i dont think so |
17:59.49 | kyron | hmm...weird.. so 1.5 == twilight zone... (wooOOOooooOOooo) |
18:00.03 | bsdwarrior | jjshoe, im sending outbound calls , I want to know if they dont answer, etc so I can flag it in the db and try again later. |
18:00.06 | kyron | x86, also, aren't 1.0==1.2==deprecated? |
18:01.08 | x86 | kyron: now they are, but at one time they were stable releases |
18:01.26 | *** join/#asterisk whymarkwhy (n=koos@196.211.34.2) |
18:01.37 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.129) |
18:02.06 | kyron | of course... |
18:02.30 | skopii | I thought 1.2 was supported? |
18:02.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
18:02.36 | outtolunc | bsdwarrior: most use async: true so it does not hangout waiting for originate to end, and add exten => failed,1,Hangup to the context.. this is create either OriginateSuccess/Failure OR OriginateResponse events |
18:03.04 | kyron | From what I am reading on the * web site, I am better to start off *Now than trixbox (does this imply that *now is more compatible in the way it generates/manages the config files?) |
18:04.03 | whymarkwhy | hi there could anyone please help me i dont grasp the consept if i dial a sip channel(x-lite(Horaaaaaaaaaaaaaaaaa) finaly got that out of the way, my brain tells me if a phone is busy it must be ingaged, how do you send a ingaged signal if sip phone is busy currently i get more than one call ringing on my sip phone ps: did i tel you i got my first sip call |
18:04.05 | kyron | (yeah, I know this ain't #asterisk-gui nor #asterisknow but I want the pov from #asterisk people ;) ) |
18:04.19 | *** join/#asterisk kamanashisroy (n=root@202.56.7.142) |
18:04.31 | *** join/#asterisk grayhame (n=grayhame@adsl-070-148-122-203.sip.bna.bellsouth.net) |
18:04.37 | skopii | kyron: try them both? |
18:04.50 | kamanashisroy | hi, anyone initiated a call from manager and tracked that ? |
18:05.43 | kyron | skopii, uhm...well...am running trix at the moment but was considering asteriskNOW for better support/compatibility/flexibility+manual configurability... |
18:06.03 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
18:06.11 | whymarkwhy | it says you can specify to jump to exten +101 if busy how does one implement that and could some please set some light on this |
18:06.16 | skopii | freepbx has all that...there should be a foo_custom.conf file that is empty |
18:06.29 | ArchSSM | Is either trixbox/asteriskNow really something for a 'professional' installation ? ... |
18:06.33 | whymarkwhy | is this a freepbx channel? |
18:06.48 | skopii | isn't asterisk the freepbx? |
18:06.54 | [TK]D-Fender | whymarkwhy: BOOK... and priority jumping like that is deprecated... |
18:06.57 | [TK]D-Fender | ~book |
18:06.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
18:06.58 | fiXXXerMet | I have asterisk working just fine, but as soon as I install asterisk-addons, asterisk stops working. Stops as in when I do an asterisk -r, it doesn't bring me to the command line. It shows the copyright info and then puts me on a blank/empty line. |
18:07.03 | skopii | i mean it's free and it's a pbx right? |
18:07.12 | whymarkwhy | they told me the best way to learn asterik is from the command line |
18:07.28 | [TK]D-Fender | kyron: All GUI's own your ass, and IMO are to be avoided. |
18:07.44 | whymarkwhy | i installed asterisk 1.4 does this book cover it? [TK]D-Fender? |
18:07.58 | fiXXXerMet | whymarkwhy: yes, give the book a read. It's good. |
18:08.00 | [TK]D-Fender | whymarkwhy: because as I said, priority jumping is DEPRECATED |
18:08.00 | skopii | [TK]D-Fender: I think gui's are like training wheels |
18:08.04 | *** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
18:08.08 | [TK]D-Fender | ~zeeek |
18:08.08 | jbot | well, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
18:08.10 | [TK]D-Fender | ^^^ |
18:08.17 | kyron | [TK]D-Fender, hehe, so not even worth looking into then...nuff said ;) |
18:08.31 | kyron | LOOL |
18:08.31 | whymarkwhy | good one |
18:08.34 | kyron | lmao |
18:08.36 | jjshoe | bsdwarrior I'd sent it all to a context and do the work there *Shrug* |
18:08.37 | *** part/#asterisk whymarkwhy (n=koos@196.211.34.2) |
18:08.52 | [TK]D-Fender | skopii: Training wheels? No... at least YOU are riding the bike. FreePBX is like having a private chauffeur. You SIT in the car while someone else does everything and you learn NOTHING. |
18:09.06 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
18:09.37 | skopii | [TK]D-Fender: well in all honesty I learned * by setting up freepbx |
18:09.45 | skopii | it was all very overwhelming at first for me |
18:10.00 | [T]ank | is it possible to call into a telephone number and capture digits entered in to the phone and use them as a variable? |
18:10.05 | [TK]D-Fender | skopii: and then when your driver runs into a wall you come out screaming and trying to fix the car but you know nothing, and nobody wants to touch the car your driver though was "well desinged and easy to maintain" |
18:10.22 | [TK]D-Fender | skopii: its doesn't teach anything about dialplan... theirs is a psycho mess. |
18:10.30 | [T]ank | what I am trying to do is call a number enter my desired caller id and then use it when dialing out to a new telephone number via disa |
18:10.31 | jjshoe | [T]ank do you mean Read ? |
18:10.41 | skopii | they have some cool macros.. |
18:10.42 | [T]ank | maybe, let me go to the wiki and see that one. |
18:10.44 | jjshoe | [T]ank Read. |
18:10.48 | [TK]D-Fender | [T]ank: "show application read" ,_ |
18:10.53 | [T]ank | awesome |
18:10.54 | [T]ank | thanks |
18:10.56 | bsdwarrior | outtolunc - I have that setup, so then I can just lookup what happened to the call in the cdr db ? |
18:11.08 | [TK]D-Fender | [T]ank: Don't use the WIKI for this stuff until you have gone through the * CLI |
18:11.16 | [T]ank | ok |
18:12.18 | outtolunc | just parse that Event: and use the Reason: x code |
18:13.01 | bsdwarrior | forgive me but parse it from where? |
18:13.33 | outtolunc | umm.. from the manager interface.. where else would an "Event: .." be |
18:13.58 | bsdwarrior | ok |
18:15.23 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
18:20.37 | fiXXXerMet | Is asterisk-addons-1.4.5 broken? As soon as I install it, even with just the cdr_addon_mysql and res_config_mysql modules, asterisk stops working. |
18:21.22 | fiXXXerMet | Or should I just use odbc with mysql instead of mysql directly? |
18:21.55 | *** join/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
18:25.10 | De_Mon | fiXXXerMet look at asterisks logs or start it in console mode and find out why it "stops working" an fix it. |
18:25.48 | fiXXXerMet | De_Mon: The problem is that I can't get to the console and that the logs are empty, even with debug turned on. |
18:26.00 | *** join/#asterisk prabu^ (i=prabu@prabu.hitbsecconf.org) |
18:26.25 | prabu^ | Hi guys, i just wanted to know does SIP/SIMPLE messaging work better in 1.6.0-Beta1 |
18:27.06 | De_Mon | fiXXXerMet -f doesn't work? |
18:27.32 | fiXXXerMet | De_Mon: Doesn't just 'asterisk' start it then 'asterisk -r' to get to the console? |
18:27.38 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
18:29.49 | De_Mon | -f tells asterisk not to fork, so there is no need to re-attach |
18:30.01 | De_Mon | (-r) |
18:31.51 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
18:32.04 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
18:35.04 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
18:39.23 | fedya | http://pastebin.com/d75e29e4f |
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18:39.32 | fedya | does this mean that the context isn't found? |
18:41.50 | [TK]D-Fender | fedya: no exten that matches the number is in a context with that name |
18:42.44 | fedya | http://pastebin.com/d275ba4ea <-- this is in sip_custom.conf |
18:43.25 | ZPertee | I have Linksys Spa8000. I have setup sip extensions in asterisk w/ voicemail. Also I have setup SPA8000 to give phones a stuttered dial tone. The SPA detects messages as I can tell from the web interface info page. However no MWI or VMWI. Any ideas? |
18:43.31 | [TK]D-Fender | fedya: so? |
18:43.50 | UnixDog | ok I have a question about 1.4 and 1.6 with respects to users.conf |
18:44.22 | UnixDog | I like the idea of users.conf but why did they not make a trunks.conf for trunk setting to match |
18:44.37 | UnixDog | that way you seperate the 2 and have better control |
18:47.18 | fedya | as far as i understand i want to go into this context to extract the DID, then go to from-trunk like it was doing originally |
18:47.34 | [TK]D-Fender | UnixDog: And we used to have "type=peer/user/friend" for sip.conf, but thats being consolidated as well. 2 files would eb a step back for them. |
18:47.42 | *** join/#asterisk RoyK (n=roy@91.149.21.205) |
18:47.51 | [TK]D-Fender | fedya: well that context you showed is clearly not being used for anything. |
18:48.09 | [TK]D-Fender | fedya: So you can go ahead and create 500 more just like it and it won't get you any farther. |
18:48.19 | UnixDog | why would users.conf and trunks.conf be a step back ? |
18:48.39 | UnixDog | its just putting trunks in 1 file to better control them |
18:48.42 | UnixDog | but ok |
18:48.55 | [TK]D-Fender | UnixDog: because Asterisk is CONSOLIDATING account-types, not splitting up. |
18:48.56 | fedya | i guess that's why it rings busy, i dont know why it doesn't do anything though... |
18:49.16 | [TK]D-Fender | fedya: it doesn't because the context that is being refernced doesn't have a match. |
18:50.23 | pagec | does asterisk support NI1 or NI2 better for PRI lines? |
18:50.48 | [TK]D-Fender | pagec: Either jsut as well |
18:51.17 | [TK]D-Fender | pagec: I think NI2 is offering 2BCT soon for * though, so if you can, thats probably a better choice. |
18:52.32 | jpsharp | 2b call transfer? |
18:52.43 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
18:53.11 | [TK]D-Fender | UnixDog: And a thing to understand : SIP is SIP. * has no clue about HOW you want to use this "account". What makes one a "phone" versus a "trunk" (another term that should NEVER be used) |
18:53.27 | [TK]D-Fender | jpsharp: 2 B Channel Transfer |
18:53.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:53.44 | pagec | [TK]D-Fender: tnaks |
18:53.49 | pagec | *thanks |
18:57.43 | UnixDog | well then tk it is trunks vs user exten vs devices |
18:58.27 | [TK]D-Fender | UnixDog: users.conf = flaming pile of SHIT. |
18:58.45 | UnixDog | ? |
18:59.09 | UnixDog | it allows you to set a single exten as both iax and sip is a plus |
18:59.46 | UnixDog | it allows you to create a single exten and set it up insted of having to waste time and do it twice in 2 files |
19:00.00 | UnixDog | and the have to write dial plan to ring both |
19:01.07 | UnixDog | and the idea of doing the same for trunks that would be put in sip and iax.conf into a trunks.conf would allow the same if you have a provider that allows both type of trunks for fail over. |
19:01.39 | UnixDog | but I was just asking |
19:01.47 | UnixDog | not starting a flamewar |
19:02.45 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
19:03.46 | pagec | are there any good pages out there on asterisk and fax setups? |
19:03.57 | hmmhesays | I'm having a hell of a time compiling gnutls on this blackfin |
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19:19.07 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
19:20.37 | *** part/#asterisk CF (n=tal@bzq-79-177-128-189.red.bezeqint.net) |
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19:28.37 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
19:29.14 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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19:36.49 | d-tech | anyone here have access to a live ccm5 or ccm6 system ... I need 'LdapDialingRules.xml' config file ... TIA |
19:39.59 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
19:41.41 | [TK]D-Fender | d-tech: thats like asking for a Whopper at McDonalds..... |
19:42.30 | SwK | can asterisk do do CSD on NI? |
19:42.49 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
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19:45.16 | hmmhesays | uclibc |
19:45.28 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
19:45.46 | The_X | anyone ever used an adtran TA 90x with asterisk for voicemail? |
19:46.50 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
19:49.04 | *** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
19:49.20 | [TK]D-Fender | The_X: Asterisk already does VM, why would we use something external? |
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19:50.55 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
19:52.18 | The_X | customer doesn't have an asterisk box |
19:52.22 | The_X | only an adtran with analog ports |
19:52.38 | The_X | I want the adtran to fwd voicemail to a centralized asterisk server |
19:53.09 | jjshoe | The_X I'm sure someone has, are you having a specific issue you need help with? |
19:55.04 | [TK]D-Fender | The_X: And how would this Adtran be "forwarding" on voicemail? |
19:55.41 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
19:56.04 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
19:57.04 | The_X | fwd on busy? |
19:57.18 | The_X | or after 5 rings, I'm sure it's doable |
19:57.30 | [TK]D-Fender | The_X: HOW is it doing it? What is * expected to do? |
19:57.46 | [TK]D-Fender | The_X: Where do those ports go now? |
19:58.40 | The_X | ever used an adtran TA? |
19:59.57 | *** join/#asterisk dho_ragus (n=dho_ragu@cup1.sugarcrm.net) |
20:00.21 | dho_ragus | does anybody know a good polycom vendor? dell is being stupid with polycom right now, and my other polycom vendor is totally giving me the run-around. |
20:00.54 | *** part/#asterisk kamanashisroy (n=root@202.56.7.142) |
20:01.04 | jblack | [TK]D-Fender: Still not today. |
20:01.05 | Qwell | dho_ragus: I always see [TK]D-Fender recommend telephonydepot |
20:02.58 | *** part/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
20:04.45 | dho_ragus | thanks Qwell |
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20:06.56 | *** mode/#asterisk [+o anthm] by ChanServ |
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20:14.12 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
20:14.12 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) |
20:19.20 | *** join/#asterisk nick-temp (n=spid3r@229.87.modemcable.oricom.ca) |
20:21.50 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
20:21.50 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) |
20:22.16 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
20:22.48 | *** mode/#asterisk [-b %#phix!*@*] by russellb |
20:22.52 | *** mode/#asterisk [-b %phix!*@*] by russellb |
20:23.04 | phix | :D |
20:23.19 | phix | That's better |
20:25.33 | phix | ok, I want to hook up an ATA or TDM to an existing PBX system (as extra lines for it). Since it will be providing extra lines I will need FXO modules right? |
20:26.09 | jpsharp | FXO modules for the PBX, yes. |
20:26.48 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
20:27.00 | phix | nice, now my next question (I have a few cached up :)), ATA or TDM? I have 3 lines left over on my existing PBX system that I can use. |
20:27.15 | jpsharp | Depends on how much you want to spend. |
20:27.50 | phix | well the only ATAs I can find are 2 or 4 fxs :( not fxo. |
20:28.13 | _ShrikE | phix: audiocodes MP114-FXO |
20:28.16 | phix | a TDM with 3 FXO modules would set me back around $300 or so I think |
20:28.28 | _ShrikE | pretty decent ata |
20:28.39 | jpsharp | No, you'll need FXS ports on the ATAs, FXO ports on the PBX. |
20:28.54 | phix | _ShrikE: ok, does it produce an echo? I am having echo / feedback issues on my linksys supara |
20:29.06 | jpsharp | Assuming that you want the ATAs to show up as lines not phones. |
20:29.47 | _ShrikE | echo is not really a product of the ATA. I have sipuras that work just fine. |
20:30.03 | jpsharp | All 8 of the lines on the Lucent PBX I have here at my office are driven off of 4 Cisco ATA-186s. |
20:30.19 | phix | jpsharp: 4 or so PSTN lines plug into the PBX atm, they are PSTN lines so they are fxo? |
20:30.37 | jpsharp | Yes. They're FXO ports on the PBX. |
20:30.43 | phix | oh ok |
20:30.54 | phix | lol I keep getting them mixed up |
20:30.55 | jpsharp | So, a couple of plain, dumb ATAs will plug into the other 3 lines on your PBX. |
20:31.17 | phix | jpsharp: I am not happy with my supara :( but I can get them on special |
20:31.21 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-190-155.dsl.irvnca.pacbell.net) |
20:31.25 | flujan | hi guys... I have a wav file working as a music on hold file... |
20:31.26 | phix | $60 each |
20:31.40 | phix | flujan: I have that too on a system, but it chunks :( |
20:31.42 | flujan | I have two queues, One receives calls from a sip trunk and the musiconhold is working... |
20:32.10 | bsdwarrior | I want to prevent 411,911 ,etc to be called back. does anyone have a good list of blocked outbound numbers? |
20:32.12 | flujan | the other queue receive calls from a pri/e1 digium board. |
20:32.20 | flujan | guys that call on the sip trunk hear the music. |
20:32.36 | flujan | calls incoming from the queue does not work. The caller hear nothing... |
20:32.38 | jpsharp | phix: Look at other vendors for ATAs? |
20:32.41 | flujan | any idea? |
20:32.59 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
20:33.06 | jpsharp | er, rather, looking? |
20:33.41 | flujan | phix: why it chunks? |
20:34.00 | phix | jpsharp: maybe, but I don't know which ones are good |
20:34.04 | phix | flujan: nfi :( |
20:34.19 | [TK]D-Fender | bsdwarrior: its your dialplan, what are you LETTING them dial those numbers? |
20:34.53 | bsdwarrior | tkd-fender, Im just looking for a blacklist of numbers. |
20:34.55 | [TK]D-Fender | bsdwarrior: You don't seem to be following the big picture. * accepts those calls because you made extensions for them. Simply don't DO it for those. |
20:34.57 | jpsharp | I'm happy with the ATA-186s I have. I've also used Grandstream HT-486s. |
20:35.46 | [TK]D-Fender | phix: I've never had any issue with Sipura/Linksys |
20:35.53 | phix | I have heard bad things about grandstream |
20:35.57 | [TK]D-Fender | phix: then tend to be the most predictable and flexible. |
20:36.19 | phix | [TK]D-Fender: ok. well maybe it is my asterisl system producing the echo / feedback |
20:36.28 | bsdwarrior | tkd-fender forget it your just talking over me |
20:36.32 | phix | [TK]D-Fender: how can I tell? |
20:36.45 | flujan | [TK]D-Fender: any idea about the issue I am experiencing? |
20:37.06 | [TK]D-Fender | flujan: pastebin it. |
20:37.08 | flujan | [TK]D-Fender: some time ago you said to me use the wav file to avoid the mp3 problems of transcoding... |
20:37.12 | phix | [TK]D-Fender: I have an analog phone connected to sipura, I can hear my voice intermittently |
20:37.14 | flujan | [TK]D-Fender: ok |
20:37.26 | [TK]D-Fender | phix: Sounds like your gains are shot |
20:37.39 | phix | where are gains set? |
20:37.52 | phix | sipura or asterisk? |
20:38.06 | [TK]D-Fender | phix: Sipura |
20:38.24 | [TK]D-Fender | phix: also on those older models be careful which firmware you use. |
20:38.32 | [TK]D-Fender | phix: the stock one was pretty decent. |
20:38.47 | [TK]D-Fender | phix: I set up a friend of mine on my old SPA-2000 and its working 100% fine |
20:39.58 | flujan | [TK]D-Fender: http://pastebin.com/m1ebb4959 |
20:40.01 | ddunavant | Is there any reason why * wouldn't read the priority labels I've assigned? |
20:40.23 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
20:40.24 | [TK]D-Fender | ddunavant: if you showed us your dialplan and the CLI output of your failed attempt perhaps we could. |
20:40.26 | [TK]D-Fender | ~pb |
20:40.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
20:40.28 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
20:40.41 | phix | [TK]D-Fender: Product Name:SPA-3102, Software Version:3.3.6(GW) |
20:41.26 | [TK]D-Fender | phix: go to www.voxilla.com and check out their forums. There are plenty of threads about which FW's work best and how to tweak them |
20:41.37 | [TK]D-Fender | flujan: pastebin something COMPLETE |
20:42.33 | flujan | [TK]D-Fender: but that are my zapata.conf and musiconhold.conf files... which file do you need me to pastebin? |
20:42.35 | phix | [TK]D-Fender: ok thank you! |
20:42.57 | [TK]D-Fender | flujan: full call showing the failure, queue config, etc. |
20:43.18 | ddunavant | [TK]D-Fender: http://pastebin.com/d1ccdbd0a |
20:43.21 | flujan | [TK]D-Fender: the queue is configured in the realtime table |
20:44.07 | [TK]D-Fender | ddunavant: Jan 22 15:36:45 NOTICE[17677]: pbx.c:1753 pbx_extension_helper: No such label ' CallID' in extension 's' in context 'macro-IntMenu' <-- see this? |
20:44.40 | [TK]D-Fender | ddunavant: No such label ' CallID' <- this means stop putting whiespace after the "?" in your GotoIf's |
20:44.59 | ddunavant | Ahhhh |
20:45.00 | [TK]D-Fender | ddunavant: exten => s,n,GotoIf($[${Option} = 2]? CallerID) <- Whitespace = BAD |
20:45.17 | ddunavant | gotcha |
20:45.17 | [TK]D-Fender | NEXT!@!@ (c) BKW |
20:47.04 | flujan | [TK]D-Fender: here is what is happening:http://pastebin.com/m58d23327 |
20:47.14 | flujan | [TK]D-Fender: the cli output from the file |
20:48.01 | flujan | [TK]D-Fender: as you can see, asterisk shows that it is playing the moh but I hear nothing... :( |
20:48.32 | flujan | [TK]D-Fender: with the mp3 files, I hear it well. :( |
20:48.56 | _ShrikE | Some of the features in the new polycom 3.0 sip firmware look neat. |
20:49.35 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
20:51.10 | nick-temp | hi, is there a way to *block* incoming calls to an agent when the extension he is registered to dial out on pstn ? |
20:51.37 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
20:51.42 | Katty | hi! |
20:51.55 | Katty | hewwoes? |
20:52.00 | [TK]D-Fender | <PROTECTED> |
20:52.04 | Katty | [TK]D-Fender: i have NEWS! |
20:52.06 | Katty | also, mews. |
20:52.10 | Katty | mew mew. |
20:52.12 | Katty | etc. |
20:53.40 | lmadsen | wem |
20:53.47 | [TK]D-Fender | Katty: :O |
20:54.06 | flujan | [TK]D-Fender: do you think it could be a problem with the .wav file? But if the file was the problem the callers from the sip trunk should hear nothing too right? |
20:54.46 | Katty | i officially have a timeline for leaving the united states. |
20:54.53 | [TK]D-Fender | flujan: no idea, that doesn't tell me much... |
20:54.55 | Katty | as in work visa leaving. |
20:55.00 | Katty | hopefully. |
20:55.14 | [TK]D-Fender | Katty: Looking to move permanently and this is the first step? |
20:55.25 | Katty | [TK]D-Fender: *nod* |
20:55.35 | [TK]D-Fender | Katty: Neato... Whereabouts? |
20:55.35 | flujan | [TK]D-Fender: which information may help you help me??? I just loose some hair trying to discover what happened. :( |
20:55.45 | Katty | [TK]D-Fender: narrowed it down to europe. |
20:55.54 | Katty | [TK]D-Fender: due to speaking english and such. |
20:55.56 | [TK]D-Fender | flujan: can't concentrate on this now, only much later tonight |
20:56.11 | flujan | [TK]D-Fender: ok |
20:56.15 | [TK]D-Fender | Katty: wow... big. Solo? |
20:56.18 | flujan | [TK]D-Fender: thanks anyway for the help |
20:56.19 | flujan | :) |
20:56.22 | Katty | i'm not sure how work visas work, tho i think a talk with Junky will help |
20:56.37 | Katty | [TK]D-Fender: mostly, solo. |
20:56.58 | Katty | [TK]D-Fender: no Big Move can be done entirely alone |
20:57.36 | flujan | hi jblack |
20:57.43 | jblack | hi |
20:57.47 | jblack | what's going on? |
20:59.21 | jblack | flujan: So, can you recap for me what's up? |
20:59.32 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
20:59.42 | flujan | jblack: sure... |
20:59.59 | fiXXXerMet | The ODBC -> MySQL works so well that I'm having trouble seeing why I need asterisk-addons at all. |
21:00.09 | flujan | I have two queues with musiconhold class=music1 |
21:00.32 | flujan | one queue receive calls from a sip trunk... This queue is working and all users here the musiconhold while holding... |
21:00.32 | Katty | anyone here have experience with work visas? |
21:00.56 | flujan | the other queue receive calls from a pri/e1 line... these guys hears nothing while waiting... :( |
21:01.13 | jblack | Ok. |
21:01.18 | flujan | jblack: asterisk shows that started music on hold on the channel, but the end point just do not hear it... :( |
21:01.49 | jblack | Ok. First, let's make sure you're playing the same moh channel to both incoming contexts. |
21:02.21 | jblack | Can you pastebin your sip.conf (don't forget to XXX passwords) and your zapata.conf ? |
21:02.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
21:02.52 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
21:03.05 | jblack | or, if you feel comfortable, make sure mohsuggest is appropriately set. |
21:03.51 | [TK]D-Fender | flujan: go prove that class even works normally |
21:04.19 | [koss] | don't say XXX passwords you're getting me excited |
21:05.00 | fiXXXerMet | hah |
21:05.09 | jpsharp | coldshower.conf |
21:05.14 | jpsharp | or app_coldshower |
21:05.16 | jblack | Since he's claming that moh is working on sip, but not on zapata, I'm thinking he's setting different contexts; either suggesting an old, boken music class in zapata, or overriding a broken default class in sip.conf (which, iirc, can override the default moh) |
21:05.33 | [TK]D-Fender | jblack: he has 3 classes, none of which I trust |
21:05.49 | flujan | jblack: http://pastebin.com/m4e66feb5 |
21:06.11 | jblack | flujan: Then while I'm reading this pastebin, paste your musiconhold.conf as well. |
21:06.31 | jpsharp | and your extensions.conf too. |
21:06.51 | flujan | jblack: http://pastebin.com/m6c7d4370 |
21:07.05 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:07.35 | jblack | Yuck. Do you really need 6 different moh channels? |
21:08.11 | jblack | Hmmm. |
21:08.35 | jblack | according to your sip.conf and your zapata.conf, you are sending them to two different places. |
21:08.45 | jblack | musicclass=default |
21:08.51 | jblack | musiconhold=music1 |
21:09.20 | jblack | You're sending sip to directory=/home/totalip/sounds/music, and zapata to directory=/home/totalip/sounds/music1 |
21:10.34 | flujan | jblack: fixed it.. the file to the default and music1 is the same... the others are department specific mohs |
21:10.37 | jblack | So, if the mp3s in /music/ are close enough to acceptable, they'll play... but the music1 ones may be loaded with id3tags and all sorts of stuff that the builtin format_mp3 inappropriately chokes on. |
21:10.53 | jblack | flujan: What do you mean "fixed it" ? |
21:10.54 | flujan | both files are .wav |
21:11.06 | flujan | jblack: put both to the same directory. |
21:11.25 | jblack | Not according to the config files you pasted. You're definitely pointing to two different places for the channels |
21:11.30 | lesouvage | is it technically possible that two inbound lines spontanious get bridged to each other. A costumer claims that that happened today but the loggings and cdr look quite normal. It is an Asterisk 1.4 box and snom 320 phones with scopserv as gui. Any suggestion? |
21:12.24 | jblack | lesouvage: It's technically possible for two inbound lines to get bridged (there's tools to do it on purpose). I've never heard of it happening magically on it's own. |
21:12.39 | file | there was actually a discussion on the -users list about that... |
21:13.04 | flujan | jblack: yeap... I saw it and changed the sip.conf to the /home/totalip/sounds/music1 location... musicclass=music1 |
21:13.04 | jblack | lesouvage: file to your rescue. ;) |
21:13.10 | file | "Calls Being Randomly Bridged" specifically with Snom phones |
21:13.39 | hmmhesays | I hate cross compile never fun, EVER |
21:13.42 | file | had to do with transfers and the way Snom handles them on the way, people accidentally transferring the wrong legs |
21:13.49 | jblack | flujan: Um,... you changed the one that works to the one that doesn't? |
21:13.58 | lesouvage | file & jblack: thanks |
21:14.13 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
21:14.20 | jblack | lesouvage: Don't thank me. I unintentionally misled you |
21:15.05 | jblack | file: Ok... well then, reload your sip.conf, call in, and see if things are now broken on sip too. |
21:15.24 | jblack | If so, it's time for you to examine the permissions for your audio files more carefully, the format of those wavs, etc. |
21:15.43 | Katty | hmmhesays: hai |
21:16.08 | jblack | file: No, I was talking to you. |
21:16.10 | flujan | jblack: reloaded and it works for the sip don't work for the zap |
21:16.15 | flujan | I will check the files now |
21:16.16 | jblack | file: Sorry. |
21:16.21 | jblack | I am talking to flujan. |
21:16.30 | flujan | jblack: but they are exactly the same. |
21:16.42 | file | if I'm asking for help in here something is wrong with me |
21:17.01 | jblack | Nah. It's me. I'm only on my 3rd cup of coffee |
21:17.34 | Katty | file: ask for help. i dare you. |
21:17.36 | hmmhesays | Hello Katty |
21:17.42 | file | Katty: nooooooo |
21:17.47 | flujan | jblack: http://pastebin.com/m2f1a8604 |
21:17.49 | jblack | flujan: So you're telling me that you ahve changed things so that both sip and zapata are both pointing (this time, really?!?) at teh same moh channels, and sip still works, and zapata doesn't? |
21:18.05 | flujan | jblack: yeap. |
21:18.07 | jblack | flujan: You did do a sip reload, and called back in? |
21:18.16 | hmmhesays | I can't get libgpg-error compile on this blackfin for the life of me |
21:18.16 | flujan | I reload asterisk |
21:18.22 | flujan | stop now and safe_asterisk again |
21:19.31 | jblack | Ok. Let me see your sip.conf and zapata.conf again |
21:19.50 | jblack | and if you can run "file fila_espera.wav" for me, that would be nice |
21:20.01 | *** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.ca.comcast.net) |
21:20.05 | jblack | (the file fila...) can be pasted here, since it's 1 line |
21:22.09 | jblack | be back in a few minutes |
21:23.20 | Katty | file: :> |
21:24.05 | [TK]D-Fender | ok, heading home, later all. |
21:24.07 | defsdoor | are there any freely available english accent recordings available ? |
21:25.02 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
21:25.13 | jameswf | there are some ausie prompts never heard uk |
21:25.50 | jameswf | some southern hillbilly redneck prompts would be cool |
21:25.59 | Katty | jameswf: :< |
21:26.01 | jameswf | hey yall press 1 |
21:26.05 | jpsharp | lol |
21:26.06 | Katty | :<<< |
21:26.08 | Katty | )_= |
21:26.19 | Katty | jameswf: you're redonkerus! |
21:26.25 | jpsharp | Here, hold my beer and press 1 |
21:26.40 | defsdoor | hmm found some |
21:27.02 | russellb | Asterisk: brought to you by the deep south. |
21:27.07 | russellb | seriously, it is. |
21:27.15 | defsdoor | http://www.enicomms.com/cutglassivr/ |
21:27.43 | flujan | jblack: |
21:27.44 | flujan | http://pastebin.com/meb77495 |
21:27.57 | jameswf | I set up a test dial plan and thought all-your-base might be cool, but it looses something when allison says it in broadcastereze |
21:28.14 | jblack | looking |
21:28.46 | jblack | That's as far as I can take you, flujan |
21:29.26 | flujan | jblack: thanks anyway... I just have no idea why it is happening... :( |
21:29.28 | flujan | danmed. |
21:30.43 | *** join/#asterisk lelandg (i=leland@24-116-151-196.cpe.cableone.net) |
21:31.42 | lelandg | Hi, I get "configure: error: *** termcap support not found" when I run './configure'; running debian |
21:32.15 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
21:32.44 | fedya | exten => _.,1,Set(CDR(accountcode)=4949494) <-- is there any reason why this won't work, i tried userfield too, and i have it enabled in cdr_mysql |
21:33.00 | *** join/#asterisk RoyK (n=roy@91.149.3.9) |
21:35.01 | *** join/#asterisk janinge (i=j@ninge.net) |
21:37.10 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
21:37.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
21:38.36 | jblack | fedya: I'd try using external (say, madplay) on music1, and see if mp3s play ok. |
21:38.44 | jblack | fedya: Never mind. someone else |
21:39.03 | jblack | flujan: I'd try using external (say, madplay) on music1, and see if mp3s play ok. The internal moh player has plenty of problems. |
21:39.07 | *** join/#asterisk RoyK (n=roy@91.149.3.9) |
21:39.33 | jblack | flujan: I imagine that its also possible that in extensions.conf, you're not really sending people into moh like you think |
21:39.41 | jblack | or into the wrong one |
21:40.21 | jblack | fedya: _. is really, really buggy. It tells you that on the console. Try _X. |
21:40.52 | fedya | what does that part mean, where can i read something about it? |
21:41.12 | fedya | i put s now, i saw that in some other guides |
21:41.38 | jblack | In the console. |
21:41.56 | jblack | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns explains some though. |
21:44.25 | flujan | jblack: sure with madplay and mp3 it worked for both... :D |
21:44.33 | flujan | does madplay supports .wav files? |
21:44.38 | jblack | flujan: No idea. |
21:44.45 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
21:44.49 | jblack | madplay does support a variety of formats, though. |
21:45.53 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
21:46.40 | jblack | At least on output it does. I don't know what it supports on input. |
21:47.17 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
21:47.28 | fedya | .X,1 does the 1 indicate the instruciton number? |
21:47.33 | fedya | i'm still here |
21:47.59 | jblack | donadie: Please take me off your list of bot recipients. |
21:48.53 | jblack | fedya: the 1 represents the priority for that extension. Please reread chapter 5 of the * book |
21:53.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-71-49.lns10.syd6.internode.on.net) |
21:55.19 | fedya | should i use s, or _X? |
21:55.29 | jblack | You should use the book. |
21:57.26 | jameswf | ~book |
21:57.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
21:57.34 | jblack | I'm not being an asshole. There's a lot there you have to read and understand. walking it through will take longer, be less comprehensive, and take longer than for you to read the chaper yourself |
21:57.55 | jblack | much longer, considering how i said longer twice. =) |
21:57.57 | jameswf | yes you are jblack but its cool we still love you |
21:58.14 | jblack | You callin me an asshole? |
21:58.25 | jameswf | :) |
21:58.32 | jblack | Let me get this right. You. are calling. Me. An asshole? |
21:58.39 | JT | jameswf: btw, it's "aussie, not ausie |
21:58.48 | jblack | I'm gonna come down there and kick your parakeet's ass! |
21:58.52 | jameswf | technicaly I was agreeing with you in an inverse manner |
21:59.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
21:59.42 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
22:00.15 | jblack | [TK]D-Fender: I think flujan is basically covered. By getting him in the same moh class and convincing him to use an external player, he's much happier |
22:00.49 | flujan | jblack: I will test it... :D |
22:01.12 | jblack | flujan: Testing something.... That's a good idea! |
22:01.43 | flujan | jblack: I really dunno what this damned this is happening... I will stop asterisk and try to use the .wav with madplay.. |
22:01.48 | defsdoor | anyone use debian ? I trying to listen to some .gsm files and can't find a player that will play them |
22:01.52 | flujan | I dont have the mp3 file right now... |
22:02.01 | jblack | defsdoor: I found one. |
22:02.03 | jblack | let me find it again |
22:02.42 | jblack | here it is. the play command from sox plays them. |
22:02.44 | defsdoor | I've got libgsm installed so figured sox might |
22:03.01 | defsdoor | jblack: play doesn't recognise them here :o |
22:03.09 | defsdoor | play soxio: Failed reading `0.gsm': unknown file type `gsm' |
22:03.18 | defsdoor | aah libsox-fmt-gsm |
22:04.12 | jblack | And that makes 3. I think I have fulfilled my civic duty for the day and can spend the rest of my day with obnoxious-but hilarious-trolling |
22:04.26 | defsdoor | jblack: nah that don't count |
22:04.55 | defsdoor | jblack: you were no help to me at all - I'd already tried play - I used my aptitude search skills to find the lib needed |
22:04.58 | jblack | You're at death's door. Your mind is muddled. So I'm right, and you're wrong. |
22:05.04 | defsdoor | sorry - troll later :) |
22:05.50 | fedya | woo i got it to set the account code, using _XXXXXXXXXX |
22:06.02 | jblack | fedya: Figured. |
22:06.06 | defsdoor | fedya: :| |
22:06.09 | jblack | fedya: Congrats |
22:06.24 | fedya | is that the solution? |
22:06.34 | fedya | hopefully it works the same way for userfield |
22:06.38 | jblack | fedya: Avoiding _. ? Yup. |
22:06.45 | fedya | the guide i was looking at did it with _. |
22:06.53 | fedya | i don't understand the 's' |
22:07.03 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
22:07.04 | jblack | fedya: that's why I told you to read chapter 5 of the book. |
22:07.25 | jblack | fedya: Buy the book if you can. If you can't, use the pdf (type ~book) until you can afford it. |
22:07.47 | defsdoor | fedya: _X. no good ? |
22:08.21 | defsdoor | I bought the book - never read it :) |
22:08.23 | fedya | interesting assumption |
22:08.37 | fedya | i've got the book right here, i didnt realize you were talking about the same book i have |
22:08.40 | defsdoor | read the pdf instead as I left the book at home |
22:08.59 | jblack | fedya: No assumption at all. There's only one book, as far as I know. |
22:09.07 | defsdoor | _the_ book |
22:09.11 | fedya | afford it :0 :) |
22:09.32 | *** join/#asterisk SteveTotaro (n=root@pool-70-16-26-249.balt.east.verizon.net) |
22:09.50 | jblack | Ohhh, you can afford that. That's great. Can you afford to send me a check for $2,500? Daddy wants a new computer. |
22:09.53 | [TK]D-Fender | _. = evil |
22:10.15 | defsdoor | _X. is ok though isn't it ? |
22:10.17 | fedya | interesting assumption because i can't afford it |
22:10.24 | [TK]D-Fender | defsdoor, much better. |
22:10.25 | ZX81 | hey, anyone know how to get ${CDR(duration)} in a deadagi run in h exten? |
22:10.26 | jblack | defsdoor: Much safer. |
22:10.29 | fedya | but i do have the book, company property |
22:10.51 | *** join/#asterisk echosyp (n=echosyp@75.111.175.135) |
22:10.57 | fedya | jblack: a macbook air/ |
22:11.00 | fedya | ? |
22:11.07 | defsdoor | jblack: btw - still can't play .gsm file :) play writes to some odd default audio device :) |
22:11.13 | echosyp | can someone clarify the difference between a sip server and a pbx |
22:11.18 | fedya | i'll try the _X. see if it works |
22:11.34 | echosyp | im assuming the sip server can't be terminated to the pstn |
22:11.38 | defsdoor | echosyp: (possibly) a sip server just routes sip to sip |
22:11.50 | echosyp | that was my thought |
22:11.51 | jblack | fedya: I didn't assume you could afford it or not. I gave you a simple code path while (toobroke) ReadPdf(); BuyBook(); |
22:11.57 | SteveTotaro | a pbx might not even do VoIP |
22:11.58 | echosyp | i want a pbx |
22:12.11 | echosyp | i see |
22:12.12 | defsdoor | echosyp: I can sell you one |
22:12.13 | jblack | echosyp: Asterisk is your char. |
22:12.14 | fedya | HAH!! |
22:12.26 | echosyp | i don't pay for things i can do myself |
22:12.27 | echosyp | sorry |
22:12.31 | fedya | thx for helping me with that :0, weird glitch with the _. |
22:12.39 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
22:12.46 | SteveTotaro | you must be young echosyp |
22:13.05 | jblack | echosyp: Asterisk is free software. that said, do you bake your own bread? Sew your own clothing? Make your own movies? |
22:13.06 | echosyp | no |
22:13.12 | defsdoor | echosyp: paying me to do it is much more fun (and helps my mortgage) |
22:13.30 | echosyp | jblack missed the point |
22:13.37 | echosyp | hah |
22:13.38 | SteveTotaro | i can do plenty of things myself but my time is better spent doing things that make money |
22:13.47 | defsdoor | though I am still a novice - building my 3rd * install now |
22:13.49 | echosyp | its fun for me learning something new |
22:13.50 | [TK]D-Fender | jblack, Just that once, but Paris' lawyers are trying to block its release... |
22:14.03 | echosyp | im installing asterisk on my router |
22:14.07 | echosyp | dd-wrt |
22:14.08 | jblack | Nope. Not at all. I just happen to know that in a world like ours, that people can often do things for you cheaper than one can do it for oneself. |
22:14.13 | defsdoor | SteveTotaro: I can do some things myself but I much prefer my wife to do them ;) |
22:14.25 | echosyp | using gizmo to connect, or so the plan is |
22:14.30 | SteveTotaro | lol def |
22:14.36 | jblack | I can make my own cheeseburgers. But if I give mcdonald's a dollar, they'll give me one that's nicer than what I can make. |
22:14.37 | SteveTotaro | that is a good point too |
22:14.38 | echosyp | eventually i'll terminate it |
22:14.49 | fedya | ok i can set the userfield now, but the call never makes it through now.. 1 problem fixed but now the call doesnt come to the phone |
22:14.58 | jblack | Yeah. And don't forget children. They're virtually slaves until they're old enough to get a job. |
22:15.02 | [TK]D-Fender | jblack, You clearly suck if yours doesn't kill McD's :0 |
22:15.11 | echosyp | yeah |
22:15.24 | SteveTotaro | wendy's makes a killer triple quarter pounder |
22:15.31 | echosyp | listen to this guy |
22:15.42 | defsdoor | fedya: trixbox isn't it - where have you added your bit ? |
22:15.42 | echosyp | saying children are slaves |
22:15.51 | echosyp | i bet he runs a sweat shop in his attic |
22:15.54 | SteveTotaro | ~trixbox |
22:15.55 | jbot | [trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
22:16.00 | jblack | McDonald's can make a better burger for a buck than I can. |
22:16.29 | defsdoor | jblack: dunno - you'd have to make a pretty bad burger |
22:16.44 | SteveTotaro | for $1 is his point |
22:16.51 | echosyp | in any case |
22:17.03 | echosyp | has anyone ever setup asterisk on a dd-wrt router? |
22:17.09 | SteveTotaro | i have |
22:17.20 | [TK]D-Fender | echosyp, yes, a whole bunch of people have |
22:17.23 | jblack | Have you seen how much food costs these days? |
22:17.24 | SteveTotaro | you should use an old pc or something |
22:17.32 | fedya | yes tb |
22:17.40 | defsdoor | I used a laptop with a busted screen |
22:17.42 | echosyp | whats sad is my old pc's are weaker than my router |
22:17.48 | echosyp | i have a wrt350n |
22:18.06 | echosyp | with usb |
22:18.12 | SteveTotaro | yeah but you cannot stick a modem in it and plug a phone line in |
22:18.29 | SteveTotaro | if you really want to explore asterisk, i would go with a pc |
22:18.36 | phix | hi |
22:18.41 | jblack | The 1/3 of a pound of beef costs 66 cents alone. Add the cheese, and that's probably another 30 cents. |
22:19.07 | echosyp | why can't i plug a modem in it |
22:19.09 | echosyp | its got usb |
22:19.10 | [TK]D-Fender | SteveTotaro, you just said "really want to explore asterisk" and " stick a modem in it".... LOL. |
22:19.13 | jblack | That leaves 4 cents for bun and cooking. I'll either have to skip the bun, or eat a raw hamburger. |
22:19.13 | echosyp | there are usb modems |
22:19.33 | [TK]D-Fender | echosyp, those will not work with *. |
22:19.35 | jblack | echosyp: That's not the funny part. |
22:19.41 | SteveTotaro | he can use a "modem" as an FXO |
22:19.46 | jblack | asterisk is software. there's no plug to stick a modem into. |
22:19.52 | [TK]D-Fender | echosyp, You are working with a toaster, don't expect more than toast.... BAGELS if you're lucky |
22:20.08 | defsdoor | jblack: got a 5 star name badge ? |
22:20.11 | Corydon76-dig | You are not a member of the A-Team. You cannot take a pile of junk and make it into a mansion in 30 minutes. |
22:20.20 | echosyp | iv already done taht |
22:20.26 | [TK]D-Fender | SteveTotaro, There is only a specific chitset that can be used, the other 99.999% of them are 100% worthless. |
22:20.31 | echosyp | asterisk is installed on it |
22:20.35 | [TK]D-Fender | SteveTotaro, and even then the X100P SUCKED. |
22:20.39 | echosyp | i just need to configure it and test it |
22:20.39 | jblack | defsdoor: Heh. my technical skillset has ensured plenty of jobs. My lack of a personality skillset ensures that I don't keep them. =) |
22:20.47 | echosyp | i can get an fxo for it |
22:21.07 | [TK]D-Fender | echosyp, well if * is installed go get the book, and start reading. |
22:21.10 | defsdoor | jblack: if you don't mind me asking what are you doing now then ? |
22:21.13 | SteveTotaro | you could buy a linksys or something i suppose |
22:21.21 | SteveTotaro | not much room for VM on a router though |
22:21.24 | jblack | defsdoor: I breathe for a living. |
22:21.37 | defsdoor | you got a good deal there then |
22:21.41 | SteveTotaro | and eat $.99 burgers |
22:21.53 | defsdoor | occasionally a happy meal ? |
22:22.06 | SteveTotaro | sticks to the dollar menu |
22:22.08 | defsdoor | just - for a treat ? |
22:22.12 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
22:22.14 | jblack | (/me takes a deep breath in). See that? I just made ten cents |
22:22.39 | lesouvage | I found the info about the romdomly bridged calls on the user list (http://readlist.com/lists/lists.digium.com/asterisk-users/14/70908.html) and I'm astonished about the bug. How to explain to the costumer? |
22:22.58 | bsdwarrior | im calling a perl script with agi(myscript.pl) . the script uses Asterisk::AGI; when I call $AGI->set_variable($variable, $value) in the script, I should be able to read the variable in asterisk? |
22:23.21 | SteveTotaro | lesouvage, just tell them it is their imagination |
22:23.30 | SteveTotaro | until the bugfix comes out soon |
22:23.37 | SteveTotaro | user error |
22:25.25 | lesouvage | SteveTotaro: To be hones, I did that a couple of month ago because I couldn't believe that it was caused by a software bug. It happens now 4 times with 60 phones in 5 month. |
22:25.45 | SteveTotaro | same here |
22:26.01 | tzanger | lesouvage: I saw that on the list too, never experienced it though |
22:26.33 | SteveTotaro | the best thing to do is to show them the bug report or at least explain it to them and that a fix is on the way |
22:26.34 | fiXXXerMet | Is there a way to have the Directory() application read back the names to you, maybe using Festival? |
22:26.50 | lesouvage | SteveTotaro: How did you deal with it, the imagination option? |
22:27.07 | SteveTotaro | i seriously thought it was their imagination |
22:27.08 | fedya | http://pastebin.com/d75ae2124, before changing to _X. it would go through this context, not set the userfield and return to the previous context and ring, now it sets the userfield but doesn't ring |
22:27.22 | [TK]D-Fender | fiXXXerMet, your users should be recording their names for the directory.... |
22:27.37 | SteveTotaro | it did not happen enough to recreate |
22:29.21 | lesouvage | SteveTotaro: did you have contact with Snom? |
22:31.47 | defsdoor | fedya: I'm talking to you in #trixbox :O |
22:31.58 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
22:32.12 | SteveTotaro | no, i don't do work for that company anymore |
22:33.53 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
22:36.24 | [TK]D-Fender | fedya, it wouldn't ring or do anything proactrical... because there is nothing more to do! You ran out of priorities and aren't sending it off or doing anything. |
22:40.35 | fiXXXerMet | Testing out Directory(). When I type the 3 digits of the person's last name, asterisk hangs up. console and extensions @ http://pastebin.com/m1f2f9001 |
22:42.17 | fedya | if i take out the CDR() line, it goes on to the next context from the context that called from-pstn-custom, and rings, but when the CDR() line is in there it rings busy |
22:42.46 | [TK]D-Fender | fiXXXerMet, voicemail.conf plesae... |
22:43.27 | [TK]D-Fender | fedya, doesn't matter. that exten is a dead-end |
22:43.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
22:43.53 | *** mode/#asterisk [+o anthm] by ChanServ |
22:43.58 | [TK]D-Fender | fedya, and the end of that last priority your call will terminate |
22:44.01 | ZX81 | anyone here used the maxtnt or the lucent apx8000? |
22:44.20 | fedya | a. howcome it works without the line? b. where do i tell it to go? |
22:44.31 | [TK]D-Fender | fedya, Show me. |
22:44.33 | fedya | http://pastebin.com/d43187118 |
22:44.44 | fedya | first 3 lines show what it does without the CDR() line |
22:44.52 | fedya | even with the deadend |
22:44.55 | defsdoor | [TK]D-Fender: it's included from [from-pstn] |
22:45.04 | fedya | so i guess it needs to go to ext-did |
22:45.30 | [TK]D-Fender | Executing Goto("SIP/64.2.142.30-088e9d08", "ext-did|s|1") in new stack <- where is YOUR goto? You showed me a DEAD-END |
22:45.34 | fedya | from-pstn: http://pastebin.com/d1dbaf65 |
22:45.45 | ZX81 | I need a box that can handle ~1xds3 of fxs |
22:45.55 | [TK]D-Fender | fedya, And you do NOT merge contexts with confliscting extens in them! |
22:46.15 | fedya | what does that mean? |
22:46.21 | fedya | what is merge contexts? |
22:46.28 | [TK]D-Fender | fedya, INCLUDE. |
22:46.28 | fedya | and what conflicts? |
22:47.52 | defsdoor | your _X. hits everything - so the following stuff included in ext-did is never reached |
22:48.49 | lelandg | Hi, I get "configure: error: *** termcap support not found" when I run './configure'; running debian; I know its pretty basic but if someone has a suggestion to get past this... very appreciated |
22:48.58 | fedya | not sure what you mean |
22:49.02 | fedya | why is it never reached? |
22:50.05 | fedya | ah |
22:50.10 | fedya | s,1 then s,n |
22:51.29 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
22:51.29 | defsdoor | this netgear poe switch is louder than my pcs |
22:51.56 | SteveTotaro | fedya if you use _X. you need to include that INSIDE a context with with things you want to match on first |
22:52.02 | ZX81 | lol just got a job to do 4400 analogue extensions |
22:52.10 | ZX81 | with 8 port gateways would be a little insane |
22:52.11 | ZX81 | :) |
22:52.13 | SteveTotaro | otherwise it will always hit _X. first |
22:52.40 | ZX81 | :) ~500 power sockets |
22:52.41 | ZX81 | mmm |
22:53.01 | SteveTotaro | ZX81, look at the tenor AX by quintum |
22:53.10 | ZX81 | sweet, thanks man |
22:53.12 | SteveTotaro | highly recommended |
22:53.15 | ZX81 | cool |
22:53.33 | fedya | hm, i changed it to s,n and s,1; it rings now, but doesn't record the CDR() again |
22:53.40 | fedya | it's either one thing or the other |
22:54.05 | ZX81 | that 48? |
22:54.29 | ZX81 | still 91 power sockets :) |
22:54.33 | SteveTotaro | i only used 24s not sure if they have 48 yet, they didn't when i bought |
22:54.47 | SteveTotaro | yeah, you have a big install there! |
22:54.52 | ZX81 | :) |
22:55.04 | ZX81 | university |
22:55.07 | ZX81 | funnnnnn |
22:55.08 | ZX81 | :) |
22:55.24 | SteveTotaro | there must be a higher density product out there |
22:55.28 | SteveTotaro | just not sure what it is |
22:55.38 | ZX81 | yeah I was looking at maxtnt and apx8000 |
22:55.49 | SteveTotaro | how many ports? |
22:55.51 | ZX81 | but I used the tnt before for a ds3 for a predictive dialing customer |
22:55.55 | ZX81 | ds3 |
22:56.01 | ZX81 | but high cps killed it |
22:56.02 | fedya | http://pastebin.com/d34046678 |
22:56.20 | SteveTotaro | cps = calls per second? |
22:56.21 | fedya | using this setup: http://pastebin.com/d488825c3 |
22:56.23 | ZX81 | yeah |
22:56.43 | SteveTotaro | i wouldn't expect that to be a problem with a university |
22:56.44 | ZX81 | the uni would be pretty normal CPS, so I'm kinda looking at it again |
22:56.48 | ZX81 | exactly |
22:56.52 | SteveTotaro | how much $$$ |
22:57.03 | ZX81 | don't remember - customer bought it direct |
22:57.12 | ZX81 | have put out a quote request |
22:57.17 | ZX81 | so will have to wait and see |
22:57.29 | ZX81 | had another install of 600 to satisfy as well |
22:57.33 | SteveTotaro | probably not many of those on ebay ;) |
22:57.36 | ZX81 | so could be used for that too |
22:57.38 | ZX81 | :) yeah |
22:57.55 | fedya | so when i use the s it doesn't even execute |
22:57.58 | SteveTotaro | how are you getting these large jobs? |
22:58.03 | SteveTotaro | if you don't mind me asking |
22:58.44 | SteveTotaro | i did a ds3 call center for inbound |
22:58.46 | ZX81 | :) we got investment from a guy who was working with the incumbent telco for 41 years |
22:58.56 | ZX81 | was the account manager for all the large accounts |
22:59.01 | SteveTotaro | oh, it's who you know, should have figured |
22:59.08 | ZX81 | he left the telco and now works full time |
22:59.10 | ZX81 | yeah 100% |
22:59.16 | ManxPower | (consulting) |
22:59.20 | ZX81 | :) |
22:59.23 | ZX81 | cool |
22:59.51 | ZX81 | http://cgi.ebay.com/Lucent-Max-TNT-for-VOIP-with-spares_W0QQitemZ110215871831QQihZ001QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
22:59.52 | ZX81 | :) |
22:59.56 | ZX81 | 8 pri |
23:00.42 | lesouvage | I'm reading Snom320 issue posts and there seem to be a problem with "engin calls". What is ment with "engin calls"? |
23:01.06 | fiXXXerMet | [TK]D-Fender: http://pastebin.com/m445a890b I have the actual accounts stored in a mysql database |
23:01.27 | ManxPower | ZX81: I'm located in the Birmingham, AL area (close to Atlanta, GA), but I'm also in New Orleans frequently for customers. I do mostly WAN, LAN (VLAN), and VoIP (Asterisk, Polycom,etc) consulting. |
23:01.32 | SteveTotaro | my ds3 implementation was seven servers with quad port sangoma cards running asterisk |
23:01.35 | mosty | lesouvage, engin is a voip provider |
23:01.41 | SteveTotaro | an NFAS |
23:01.59 | ZX81 | sweet |
23:02.01 | SteveTotaro | so seven trunk groups as the telco called them |
23:02.11 | SteveTotaro | it worked flawlessly |
23:02.11 | ManxPower | ZX81: So if you have need for my skills... |
23:02.13 | lesouvage | mosty: thanks |
23:02.30 | jpsharp | Or if you need someone *in* Atlanta.... |
23:02.58 | ManxPower | jpsharp: I suspect I've been doing this sort of stuff a while longer than most. |
23:03.27 | SteveTotaro | longer than me? |
23:03.42 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
23:03.47 | ManxPower | I've been doing the non-VoIP stuff for a little over 10 years. |
23:04.00 | lesouvage | ManxPower: You were around when I asked my first x100p related question on this forum 4 years ago <;-) |
23:04.02 | ManxPower | the voip stuff since before Asteris, 0.65. |
23:04.10 | SteveTotaro | six years traditional voice |
23:04.16 | jpsharp | I've had my hands on * since 2001ish. |
23:04.37 | ManxPower | SteveTotaro: I've only doing voice for 4 - 5 years. Not sure. |
23:04.48 | jpsharp | regular telco stuff since 1995ish. |
23:05.08 | SteveTotaro | ccna since 96 |
23:05.24 | ManxPower | And here I thought almost everyone here was a total noob 8-) |
23:05.34 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net) |
23:05.56 | SteveTotaro | well i see alot of names but only a few speaking up |
23:06.07 | jpsharp | We just like to lurk and glare at the newbies. |
23:06.18 | hmodes | mmhmm |
23:06.24 | SteveTotaro | we should form a conglomerate |
23:06.30 | SteveTotaro | and blanket the US |
23:06.56 | jpsharp | The Asterisk Strike Force |
23:07.07 | SteveTotaro | separate companies under one umbrella company |
23:07.20 | SteveTotaro | or at least one name |
23:07.38 | SteveTotaro | for marketing and passing leads around |
23:08.19 | jpsharp | I'm a little out of practice with bleeding edge *. I just use it to handle IP phones for our VSAT customers. |
23:08.25 | SteveTotaro | i just setup a bunch of regional self storage companies up like that so they can compete with PODS and the like |
23:08.58 | SteveTotaro | I don't like bleeding edge * because it is not stable enough for my customers |
23:09.00 | ManxPower | I speak far too much |
23:09.19 | SteveTotaro | that is good because i am a man of few words |
23:09.24 | SteveTotaro | ;) |
23:09.53 | fedya | thanks for all your help, i have to pause for today |
23:10.26 | *** join/#asterisk goodmove (n=yves@209.88.71.201) |
23:11.06 | goodmove | hello asterisk community |
23:14.18 | cappiz | someone knows of a fax-client that uses SIP/IAX? |
23:14.34 | mosty | cappiz, fax over voice over ip is stupid and unreliable |
23:14.58 | mvanbaak | it's not stupid |
23:15.04 | mvanbaak | but it is unreliable |
23:15.10 | drmessano | lol |
23:15.10 | cappiz | mosty, sometimes i have to |
23:15.24 | drmessano | Thats damn well put |
23:16.14 | mosty | cappiz, why is that? |
23:16.25 | SteveTotaro | it can be made reliable but you have to eliminate as much IP as possible |
23:16.29 | goodmove | I am relatively new to asterisk. I have a production system using 1.4.13. Whenever an IP phone develops a problem while a call is in progress, Asterisk does not delete the bridged connection. This connection will remain bridged until a "soft hangup" is issued. The symptom of this problem is that only one half of a subsequent telephone conversation from the problem extension is heard. How to fix this? |
23:16.37 | cappiz | cause i dont have a real faxmachine |
23:16.41 | mvanbaak | SteveTotaro: indeed |
23:16.50 | mvanbaak | we do fax over ip in lan setups |
23:16.54 | mvanbaak | no problem there |
23:17.06 | mvanbaak | but you dont want it over the public internet to unkown connections |
23:17.51 | lesouvage | mvanbaak: I read your name on the mailing list with the snom320 issue and randomly bridged incoming calls. How often does that happen (every hour, every day, once a month)Edw ooo |
23:18.09 | mvanbaak | more like every minute |
23:18.10 | SteveTotaro | i suggest a crossover cable or a T1 channel bank for fax solutions |
23:18.21 | goodmove | How to fix this problem so that Asterisk will realise that one of the channels is dead and therefore teardown this bridge..? |
23:18.28 | mosty | cappiz: there are email to fax providers that are pretty cheap |
23:18.41 | SteveTotaro | yes, i personally use www.trustfax.com |
23:18.58 | mvanbaak | our ITSP's offer fax2mail |
23:19.00 | SteveTotaro | but when you are talking about thousands of pages a day, it ain't cheap |
23:19.16 | mvanbaak | lesouvage: every minute. We replaced the snom phones with aastra's and all is fine now |
23:19.22 | mosty | goodmove, probably worth trying the latest 1.4 version to see if it's been fixed already |
23:19.45 | mvanbaak | goodmove: SIP ? |
23:20.13 | mosty | mvanbaak, fax over ip is not stupid, fax over voice over ip is |
23:21.10 | cappiz | mosty, okey... still, i want to check it out :) |
23:21.32 | mvanbaak | try, and be disappointed by it :) |
23:21.50 | goodmove | mosty, I noticed that this problem was apparent since version 1.4.9. I also noticed this in version 1.2 as well. I also searched the bugs.digium.com but came up empty. |
23:22.13 | mvanbaak | goodmove: is this with SIP channels ? |
23:22.58 | goodmove | mvanbaak, yes with SIP channels bridged to Zap Channels using a Sangoma A400 card |
23:23.31 | mvanbaak | goodmove: try 1.6 beta. It has SIP session timers |
23:23.37 | mvanbaak | that will help with this stuff |
23:23.54 | mvanbaak | we noticed it on 1.0, 1.2 and 1.4 and the new session-timers stuff fixes it |
23:24.14 | SteveTotaro | but what else doesn't work in 1.6? |
23:24.19 | SteveTotaro | scary |
23:24.34 | mvanbaak | SteveTotaro: I run -trunk in production |
23:24.48 | mvanbaak | if you follow the commits mailinglist it can be rockstable |
23:24.55 | goodmove | mvanbaak, is 1.6beta stable enough to use in my production environment without disruption? |
23:25.20 | mvanbaak | goodmove: depends on the setup. try it in your test setup |
23:25.23 | SteveTotaro | i wouldn't bet on it unless you get a call an hour |
23:25.26 | mvanbaak | you do have a test setup right |
23:25.46 | mvanbaak | SteveTotaro: we run like 1500 calls an hour |
23:25.48 | SteveTotaro | rockstable under what load? |
23:25.49 | mvanbaak | without trouble |
23:26.01 | SteveTotaro | doing what? |
23:26.04 | SteveTotaro | just a pbx? |
23:26.12 | mvanbaak | iax to sip and vice versa |
23:26.17 | goodmove | mvanbaak, i will try upgrading a mirror system of my production system tonight |
23:26.28 | SteveTotaro | so you just transcode and put packets on the wire |
23:26.32 | mvanbaak | with media handling, queues, mixmonitor, voicemail |
23:27.08 | mvanbaak | adaptive odbc for cdr handling |
23:27.34 | mvanbaak | we use the manager interface for originating calls and querying device status |
23:27.35 | SteveTotaro | are these real calls or some kind of test calls? |
23:27.43 | mvanbaak | SteveTotaro: production |
23:27.46 | mvanbaak | real calls |
23:28.00 | SteveTotaro | somehow i think you are exaggerating |
23:28.35 | mosty | mvanbaak, how many iax/sip clients approximately? |
23:29.00 | mosty | defsdoor, FXS? you always need a power connector for that |
23:29.02 | defsdoor | unfortunately the internal power connector doesnt reach |
23:29.07 | defsdoor | mosty: fx0 |
23:29.27 | mvanbaak | mosty: 6 IAX ITSP's and like 300 sip clients |
23:29.32 | mosty | i've never had anyone use that many FXO ports in a PBX :) |
23:29.32 | mvanbaak | SteveTotaro: why ? |
23:29.39 | SteveTotaro | the power supplies ring voltage to the phone |
23:30.05 | defsdoor | mosty: got 6 analog lines |
23:30.05 | mosty | mvanbaak, i have similar numbers of calls, but many more sip/iax clients, and we have big threading issues with 1.4.17 :( |
23:30.37 | mosty | defsdoor, that many channels usually means BRI here |
23:30.39 | SteveTotaro | because 1.4 is a disaster and i cannot imagine 1.6 being better than 1.4 at this point |
23:30.45 | SteveTotaro | it is BETA |
23:30.49 | mvanbaak | so ? |
23:31.00 | defsdoor | mosty: yeah - but it's not - 6 separate lines |
23:31.14 | defsdoor | mosty: I'd would have liked to move to PRI but they didn't want to spend |
23:31.23 | mvanbaak | I started running trunk at home 1.5 years ago with the skinny phones, and moved our production platform to it 4 months later |
23:31.28 | SteveTotaro | you asked why and I told you, that's why |
23:31.34 | mvanbaak | 1.4 is not a disaster |
23:31.43 | mosty | defsdoor: can you get BRI in the uk? might be cheap enough |
23:31.46 | mvanbaak | maybe 1.4.0 and 5 following releases |
23:31.47 | SteveTotaro | is ABE using it? |
23:32.02 | mvanbaak | but latest 1.4 versions are fine |
23:32.11 | defsdoor | mosty: yes - but would need 3 - too expensive |
23:32.30 | SteveTotaro | i subscribe to the bug list and my folder fills every day with major bugs |
23:33.00 | SteveTotaro | again, i just don't buy it |
23:33.08 | mvanbaak | fine |
23:33.12 | mvanbaak | call me a liar |
23:33.34 | SteveTotaro | you said first that you were just bridging calls |
23:33.39 | mosty | i think there's plenty of bugs in 1.4, but not everybody encounters many of them |
23:33.40 | SteveTotaro | i believe that |
23:34.13 | SteveTotaro | that is my point, if you are just bridging calls then it's probably plenty stable |
23:34.26 | mvanbaak | bridging calls is easy, and you dont need asterisk for that. ser does that way better |
23:34.43 | SteveTotaro | ser does not transcode |
23:34.55 | mvanbaak | who needs transcoding ? |
23:35.06 | mvanbaak | I dont |
23:35.19 | mvanbaak | all traffic here is alaw |
23:35.21 | SteveTotaro | does ser bridge iax to sip? |
23:35.44 | mvanbaak | nope, not that I know |
23:36.27 | SteveTotaro | pstn connectivity? |
23:36.44 | mvanbaak | we dont have that |
23:36.44 | Shaun2222 | I'm using dial()+GOSUB and when i set GOSUB_RESULT=CONTINUE for some reason the call is bridged. |
23:36.48 | mvanbaak | pure voip setup |
23:37.01 | drmessano | Give your users a Cell/SIP handset and you can force them to GSM.. they wont know |
23:37.02 | mvanbaak | we use ITSP to do the PSTN stuff |
23:37.07 | drmessano | Bam... worldwide GSM |
23:37.30 | mvanbaak | We do SIP, IAX2 and Skinny |
23:37.32 | mvanbaak | that's it |
23:37.52 | mvanbaak | IAX2 to ITSP and local pbx-en |
23:37.59 | mvanbaak | SIP and Skinny for phones |
23:38.16 | mosty | can anyone recommend a good way to to iax load testing? |
23:38.32 | mvanbaak | mosty: 2 asterisk boxen with loopbacks |
23:38.40 | SteveTotaro | doesn't iax2 have a limit of 254 calls? i thought i read that somewhere |
23:38.52 | mvanbaak | SteveTotaro: erm, no |
23:38.58 | mosty | mvanbaak, i've thought of that, i was hoping for something more lightweight on the client side |
23:39.12 | ManxPower | SteveTotaro: I don't know, we don't use SIP, but I'm pretty sure if it ever was (maybe for an iax2 trunk?) it's been fixed. |
23:39.21 | mosty | SteveTotaro, we have thread iax issues at around that point |
23:39.45 | ManxPower | later versions of 1.4 has MASSIVE IAX2 fixes, including threading issues, IIRC |
23:39.46 | SteveTotaro | yeah, i thought there was an issue there |
23:40.05 | mvanbaak | we have had issues with 1.2 |
23:40.19 | mvanbaak | so might be that that was the issue there |
23:40.27 | mosty | ManxPower, yeah but this issue is with 1.4.17 |
23:40.38 | SteveTotaro | yes, that is what i thought |
23:40.41 | mvanbaak | we switched from 1.2 to trunk around the time 1.4.7 was released |
23:40.53 | SteveTotaro | i believe you are living in a dream world mvanbaak |
23:41.16 | mvanbaak | mihau did find and fix a lot of issues in chan_iax2.c |
23:41.21 | SteveTotaro | maybe 1,500 one minute or less phone calls an hour |
23:41.45 | goodmove | I am contemplating setting up an IPPBX with two SATA drives using RAID 1(mirroring). I have been searching for an adapter that is compatible with RedHat Enterprise Linux 5.0. Can anyone recommend a suitable product that provides a good performance/price. |
23:42.02 | mosty | goodmove, what kind of adapter? |
23:42.04 | mvanbaak | SteveTotaro: or we do it the right way |
23:42.24 | SteveTotaro | ok guy |
23:42.31 | goodmove | mosty, sata raid adapter |
23:42.51 | mosty | goodmove, the best value in terms of performace/price is linux software raid, mdadm |
23:42.59 | SteveTotaro | want a mid range server for a good price with a 3yr warranty? |
23:43.05 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
23:43.25 | mvanbaak | mosty: get a sun X2200 |
23:43.40 | mvanbaak | they are nice and supported with redhat |
23:44.16 | mvanbaak | SteveTotaro: I really cant see why you think we are not able to do stuff like this |
23:44.25 | SteveTotaro | http://www.dell.com/content/products/features.aspx/pedge_r200?c=us&cs=04&l=en&s=bsd |
23:44.34 | SteveTotaro | i would go for the $799 deal |
23:44.42 | goodmove | mosty my experience with software RAID is that the boot partition is not mirrored. It therefore means that should there be a failure on the boot disk, then the system will not boot, even though you may be able to save the data. |
23:44.48 | Shaun2222 | is this a bug... I'm using dial()+GOSUB and when i set GOSUB_RESULT=CONTINUE for some reason the call is bridged. |
23:45.02 | ManxPower | goodmove: Have you considered asking on the correct channel? |
23:45.17 | ManxPower | Shaun2222: using M() option to Dial, I assume. |
23:45.19 | mosty | goodmove, i do software raid1 for boot here, works fine |
23:45.26 | mvanbaak | goodmove: the boot partition only needs the kernel. It's easy to copy a couple of files everytime you install a new kernel |
23:45.41 | Shaun2222 | ManxPower: no that would be a macro... i'm using U() |
23:45.48 | mvanbaak | and with newer grub/linux kernel you can even boot from raid1 |
23:45.54 | mosty | goodmove, any modern linux kernel with initramfs can mount a raided root/boot/anything |
23:45.55 | goodmove | mosty are you using RedHat? |
23:46.06 | mosty | goodmove, no, debian |
23:46.11 | mvanbaak | I dont like redhat |
23:46.20 | ManxPower | Shaun2222: You know the issues with M(), I would not be surprised if they happened with U() as well. |
23:46.41 | Shaun2222 | what issues with M() do these issues exist in trunk/ |
23:47.11 | goodmove | mvanbaak, Redhat was the first distro I leaned linux therefore I have a sentimental attachment. I am also fairly familiar. |
23:47.50 | mvanbaak | guess I was lucky with being introduced to Linux with Debian |
23:48.00 | ManxPower | Shaun2222: I don't know about trunk. There are various issues with M() macros jumping out if it gets DTMF, etc. I assume they are design decision rather than a bug. |
23:48.11 | mvanbaak | as far as I've tested it's the least worst of them all |
23:48.14 | codefreeze | ~pb |
23:48.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
23:48.21 | mvanbaak | still, it is linux so it's flawed by design |
23:48.49 | SteveTotaro | freeBSD rulez |
23:49.03 | SteveTotaro | solaris as well |
23:49.04 | mvanbaak | that's a lot better then linux indeed |
23:49.15 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
23:49.20 | mvanbaak | but I prefer OpenBSD |
23:49.42 | Shaun2222 | ManxPower: well i suppose that could make sense since the callee is hitting 3 to send the CONTINUE |
23:50.24 | Shaun2222 | ManxPower: but actually... i see the dialplan run for the Set()... so it didnt jump out. |
23:50.42 | Shaun2222 | not unless it decided to run one more exten before it bails |
23:51.27 | ManxPower | Shaun2222: what happens if the caller presses a DTMF that you do NOT trap for? |
23:51.28 | *** join/#asterisk MaliutaWrk (n=nikolai@kiev.lusan.id.au) |
23:52.11 | ManxPower | I always liked CPM |
23:52.30 | Shaun2222 | ManxPower: not sure, right now the caller and callee is me just on seperate phones... the caller isnt hitting any DTMF's while the gosub is running. |
23:52.45 | Shaun2222 | http://pastebin.ca/869203 |
23:53.19 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
23:54.16 | Shaun2222 | also i dont understand this either.... == Spawn extension (app_dial_gosub_virtual_context, s, 1) exited KEEPALIVE on 'SIP/306-ac0331f0' |
23:55.37 | mvanbaak | SteveTotaro: did you report your issues under load to the bugtracker ? |