| 00:02.13 | jamko | http://pastebin.com/MkCcF2Nf --- how much worst is that? |
| 00:03.26 | [TK]D-Fender | jamYou have repeat tags |
| 00:03.30 | [TK]D-Fender | jamko: |
| 00:03.56 | [TK]D-Fender | jamko: And it will never be "ANSWER" so your checking is a waste of time really |
| 00:04.46 | jamko | Should it not be ANSWER after I park a call? |
| 00:05.18 | [TK]D-Fender | jamko: Actually your call should die off period |
| 00:05.26 | jamko | It doesn't |
| 00:05.37 | jamko | It keeps going through the dialplan after parking. |
| 00:06.29 | jamko | And it does this on 3 boxes.. 1.4 and 1.6 .. |
| 00:07.06 | [TK]D-Fender | jamko: Bug report time. Also stop using features.conf for this. That should solve it. |
| 00:07.54 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 00:08.18 | jamko | What should I use instead? |
| 00:08.25 | jamko | I need a parking lot. |
| 00:09.16 | nny | just so I am clear, is there a document that states which versions of asterisk, if any, support HINT:nnn@exten in queues.conf ? |
| 00:11.17 | [TK]D-Fender | jamko: us a normal attended transfer |
| 00:11.57 | [TK]D-Fender | nny: where do you see any refrence to it? |
| 00:12.47 | nny | [TK]D-Fender: http://www.freepbx.org/news/2010-02-26/heavy-queue-usage-in-freepbx, gimme a sec, i'll try and explain what/why/where |
| 00:13.25 | jamko | [TK]D-Fender: I am using an attended transfer, with the Park app from the dialplan. Don't I still need features.conf to define the parking lot, and context? |
| 00:16.24 | [TK]D-Fender | jamko: Once you complete an attended transfer the call should simply die off. |
| 00:16.48 | [TK]D-Fender | jamko: And I don't don't eman some bullshit * DTMF transfer. Use REAL phones with REAL transfer capabilities |
| 00:17.33 | Gugge | nny: it seems to have been added around a year ago - https://issues.asterisk.org/view.php?id=15168 |
| 00:17.40 | jamko | [TK]D-Fender: You're the f-ing man bro... I had changed my method of xfer, and stopped using the default parking lot, and never bothered to test if the issue stopped. Well you saved my night.. Thanks! |
| 00:17.43 | Gugge | nny: but i have no idea what releases its in :) |
| 00:18.09 | jamko | Seems to only be an issue with the default lot. |
| 00:18.12 | nny | [TK]D-Fender: That article states that it makes patches to asterisk to allow HINT:nnn@context a viable option in queues. I have a system that after an update to freepbx is complaining about the syntax as explained. If i disable the option, it works, but queues are now unbalanced and there is some residual weirdness in FOP2. Both problems seem directly related to the differences. |
| 00:18.45 | nny | [TK]D-Fender: so about to patch, my one issue is how it went from working to not, when asterisk was never recompiled |
| 00:19.06 | nny | Gugge: yeah that's the patch I have found, grepped through the source on this box and it has not been applied afaik |
| 00:19.53 | nny | [TK]D-Fender: when I say "it complains" I mean asterisk core, it states the members are invalid due to the syntax difference |
| 00:20.01 | nny | scratches head |
| 00:20.16 | nny | (fleas) |
| 00:20.37 | [TK]D-Fender | nny: dependswhat KIND of members they are as well |
| 00:21.58 | nny | [TK]D-Fender: simple deskphones, no agent or non Dial/SIP type members. I imagine the local channel is to control the dialplan once the channel is called |
| 00:22.03 | nny | no hotdesking |
| 00:22.19 | *** join/#asterisk guilhermebr (~Guilherme@189.63.52.140) |
| 00:22.37 | [TK]D-Fender | nny: Local channels alone can get "invalid" simply because of module load order. |
| 00:22.52 | nny | I am assuming a large part of this is an incorrect intial setup btw. |
| 00:22.54 | [TK]D-Fender | nny: if app_queue loads before chan_local all the members are "invalid" |
| 00:23.04 | [TK]D-Fender | nny: You need to preload +> chan local |
| 00:23.08 | nny | [TK]D-Fender: true, I have tried the changes to modules.conf as suggested in other places |
| 00:23.57 | nny | [TK]D-Fender: the error pops up only when hint:nnn is used. if I manually change it or tell freepbx "USEQUEUESTATE=false" it works, but with unbalanced queues and some FOP2 oddness, the latter being the least of my concerns atm |
| 00:24.08 | nny | i'll pastebin modules.conf stand by |
| 00:25.14 | jamko | [TK]D-Fender: For the sake of discussion... I had been told that the DIAL command was not meant to roll over to multiple providers the way I have always done, and that I had to use DIALSTATUS to go to the next provider in line. Is that complete BS ? |
| 00:25.19 | Gugge | nny: as far as i can tell the HINT: support is not in 1.6.2.11 ... so i would guess freepbx requires custom patches to use HINT: |
| 00:25.58 | nny | Gugge: yes I am seeing that as well. So odd how it changed form working to not, when the updates I did were just modules and 4 digit version updates (i.e. X.X.X.1 to X.X.X.2) |
| 00:26.18 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 00:26.22 | Gugge | nny: maybe from one version with patches, to one without? :) |
| 00:26.36 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 00:26.42 | nny | the explanantions I have are: 1.) amportal got changed to USEQUEUESTATE=true, 2.) asterisk somehow lost this ability or ????/Profit |
| 00:26.56 | nny | Gugge: the updates should not have affected asterisk as far as I understand it |
| 00:27.02 | [TK]D-Fender | jamko: * will execut the next priority as long as the call was NOT answered. for ANY reason |
| 00:27.04 | nny | Gugge: the two are mutually exclusive |
| 00:27.25 | nny | as far as updates/installation. Obviously packages come to mind, but.. |
| 00:27.25 | [TK]D-Fender | nny: please do not refer to ANYTHING coming from FreePBX. Ever. |
| 00:27.32 | [TK]D-Fender | nny: * config backup only. |
| 00:28.08 | [TK]D-Fender | nny: also considering when the patch came out is hsould not have made it into anything other than 1.6.2 AFAICT |
| 00:28.19 | nny | [TK]D-Fender: yeah i hear yah, this has been tough considering what two aspects are at play. |
| 00:28.31 | [TK]D-Fender | nny: because that would be a new feature and would not be included in. |
| 00:28.37 | nny | [TK]D-Fender: http://pastebin.com/6GVEXJUW |
| 00:28.41 | nny | (btw) |
| 00:28.43 | nny | yeah |
| 00:28.51 | nny | well i'll patch and try it the way they intend. |
| 00:29.01 | nny | just so god damn odd that it ever came about this way |
| 00:29.37 | jamko | [TK]D-Fender: Right so if it's answered, then it won't execute the next priority, as I always assumed... F**kers had me chasing my tail for nothing. thanks again.. |
| 00:30.50 | Gugge | jamko: not answered means busy too .... |
| 00:31.23 | Gugge | so if you do Dial(privider1), and then Dial(provider2) |
| 00:31.51 | Gugge | you would annoy the recipient if he denies the call :) |
| 00:36.28 | *** join/#asterisk SirThomas_Home (~tomc@209-169-199-174.mn.warpdriveonline.com) |
| 00:38.21 | FiReSTaRT | declares his idiocy for the whole world to know |
| 00:38.47 | FiReSTaRT | i have spent days trying to compile zaptel drivers on like 4 different linux distros until i found out that it got replaced by dahdi |
| 00:38.52 | FiReSTaRT | dope-slaps himself |
| 00:43.14 | titter | [TK]D-Fender: Do you have any info on using DNS SRV records with IAX? I get registration to work correctly, however after the primary system goes down, IAX just reports as unreachable, and doesn't try the next priority of the DNS record. |
| 00:45.20 | [TK]D-Fender | titter: no |
| 00:45.50 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 00:47.11 | titter | [TK]D-Fender: Do you have any idea why it won't work? No errors, nothing in debug, just sits idle once it reports |
| 00:50.44 | ManxPower | titter, maybe handle it in the dialplan. |
| 00:51.02 | ManxPower | I've always handled those sorts of issues with failover in my dialplan. |
| 00:51.24 | titter | ManxPower: I have as well, just trying to do something a little different |
| 00:52.31 | titter | ManxPower: I have DNS SRV setup for my SIP phones, and have a fail over server at another datacenter that my itsp also will failover incoming calls to. So I figured why not see if IAX can do the same. |
| 00:54.06 | titter | IAX can definitely use srvlookups, but for whatever reason, probably my own fault, doesn't try the next priority once the first has been reporting as down |
| 00:58.19 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 00:58.39 | a1fa | grr... some stupid war dialer called me last night/morning at 5AM |
| 00:58.51 | a1fa | i need a human check |
| 00:59.06 | ChannelZ | 'press X if you're a human' and WaitExten for a couple of seconds |
| 00:59.13 | a1fa | Captcha for the * |
| 00:59.14 | ChannelZ | that's all I do |
| 00:59.23 | a1fa | may have to do that |
| 00:59.37 | ChannelZ | Write an AGI that makes up a 3 digit random number if you're really having a problem |
| 00:59.45 | a1fa | its not that bad |
| 00:59.48 | [TK]D-Fender | ChannelZ: Unncessary |
| 00:59.52 | a1fa | it was a first call in months |
| 00:59.57 | ChannelZ | but I've not had my phone ring in 3 days now with all these stupid political calls with a single static digit |
| 01:00.19 | ChannelZ | in fact here comes one now across the console |
| 01:00.30 | a1fa | i really need to re-do my dial plan |
| 01:00.31 | ChannelZ | hahah irony |
| 01:00.34 | a1fa | it sucks and its all over the place |
| 01:00.53 | [TK]D-Fender | a1fa: Certainly not how *I* left it ;) |
| 01:01.03 | a1fa | ;) |
| 01:01.20 | a1fa | [TK]D-Fender: i kept adding custom numbers, and crap, and its now all over the place |
| 01:01.22 | a1fa | this is what kilss me |
| 01:01.32 | a1fa | exten => t,n,Dial |
| 01:01.41 | a1fa | i probably need to hangup on timeout |
| 01:01.48 | a1fa | does quick audit of incoming calls |
| 01:02.10 | a1fa | granted; i only use this for my family |
| 01:02.49 | a1fa | anyway.. if you guys need a cheap wireless SIP phone; may I suggest one of the Android phones :) |
| 01:03.22 | Gugge | you found an android sip client that does not suck? :) |
| 01:03.40 | a1fa | sipdroid is ok -- i guess |
| 01:03.45 | a1fa | works ok on G1 |
| 01:03.46 | a1fa | :) |
| 01:04.00 | a1fa | i retired the G1 6 months ago, and now it is a wireless SIP phone |
| 01:04.01 | Gugge | last i checked it did not support srv lookup :) |
| 01:04.19 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:04.36 | a1fa | srv lookup ? |
| 01:05.01 | Gugge | http://www.voip-info.org/wiki/view/DNS+SRV |
| 01:05.13 | a1fa | oh |
| 01:05.21 | a1fa | no big deal for me |
| 01:05.29 | a1fa | since i use it as a phone ;) |
| 01:06.15 | a1fa | Wright R & D 12198668421 |
| 01:06.18 | Gugge | it would allow it to failover to another asterisk box if the primary is down |
| 01:06.24 | a1fa | that's the call i got last night at 5AM |
| 01:06.54 | a1fa | fuckers called my 1-800 number |
| 01:07.51 | a1fa | damn broadvoice they need to fix their crap.. you cant tell the difference on what number a call is coming in on |
| 01:07.54 | a1fa | its making me mad |
| 01:08.56 | a1fa | they only show the primary number as inbound |
| 01:09.55 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 01:12.13 | a1fa | [TK]D-Fender: who is your prefered sip trunk these days? |
| 01:14.02 | a1fa | the only thing that stops me from dumping broadvoice is the damn caller id |
| 01:14.05 | a1fa | err.. DID |
| 01:19.55 | *** join/#asterisk wafflehead_horro (~admin@67.216.145.164.pool.hargray.net) |
| 01:21.15 | a1fa | http://bewareofbroadvoice.com/ |
| 01:21.17 | a1fa | LOL |
| 01:21.18 | [TK]D-Fender | a1fa: doubt that. Show me your reg & full INVITE |
| 01:21.50 | a1fa | k |
| 01:22.02 | a1fa | we went through this once |
| 01:22.02 | a1fa | remember? |
| 01:22.16 | a1fa | Broadvoice is messed up. they only send invite to the primary number |
| 01:23.00 | a1fa | i am going to dump them; but need a new provider that can port numbers over |
| 01:23.01 | Gugge | did you check the To: header too? |
| 01:26.47 | a1fa | yes |
| 01:27.00 | a1fa | it goes to an extension as specified via register-> |
| 01:27.35 | [TK]D-Fender | a1fa: PASTEBIN <----- |
| 01:27.57 | a1fa | sip.conf and the debug peer info? |
| 01:28.51 | [TK]D-Fender | a1fa: Yes. Mask only passwords |
| 01:32.23 | a1fa | sent |
| 01:32.40 | a1fa | the same sip debug is generated when calling the alternate number |
| 01:32.43 | a1fa | or the primary number |
| 01:38.28 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 01:49.03 | FiReSTaRT | hey guys.. quick question.. the arch wiki asterisk page mentions asterisk-addons and asterisk-sounds packages.. have the addons been discontinued? |
| 01:49.20 | FiReSTaRT | for 1.8 |
| 01:49.23 | [TK]D-Fender | yes |
| 01:49.32 | [TK]D-Fender | FiReSTaRT: they should now be included |
| 01:49.55 | FiReSTaRT | [TK]D-Fender: good.. and what about the sounds? also included or should i dl from digium? |
| 01:50.00 | [TK]D-Fender | FiReSTaRT: Used to be separate for legal reasons. |
| 01:50.21 | [TK]D-Fender | FiReSTaRT: Typically your install should DL them itself in menuselect |
| 01:51.27 | FiReSTaRT | [TK]D-Fender: i actually installed 1.8.0-1 through AUR (on an arch system) so i didn't do menuselect... how would i check if they're there? just browse to the folder and see what's in there? |
| 01:52.13 | [TK]D-Fender | FiReSTaRT: Funny, I don't see a 1.8.0.1 relase mentioned... |
| 01:52.38 | [TK]D-Fender | FiReSTaRT: And we don't support their (anyone elses) packages |
| 01:53.28 | FiReSTaRT | [TK]D-Fender: could be an internal aur thing.. 1.8 and then it depends on who created the install script |
| 01:54.26 | FiReSTaRT | [TK]D-Fender: i see a folder asterisk_sounds_cache, but the files in there are archived |
| 01:56.09 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:56.38 | FiReSTaRT | [TK]D-Fender: so if i want to add sounds i just download the tgzs to that directory? |
| 01:57.39 | *** join/#asterisk Besticles (~larry@209-58-227-178.static-ip.telepacific.net) |
| 01:59.29 | Besticles | I am using FastAGI. I am now trying to program the detection of Operator Intercepts (Disconnected Numbers). However CLI is showing me that it: PROGRESS with cause code 1 received, and does not any extension. Is there a way I can maybe... turn this on as a feature, hopefully? |
| 01:59.50 | Besticles | does not go to any extension* |
| 02:03.04 | *** part/#asterisk wafflehead_horro (~admin@67.216.145.164.pool.hargray.net) |
| 02:21.49 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
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| 02:33.27 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 02:36.03 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 02:38.41 | *** join/#asterisk fireman_biff (~biff@65.48.132.153) |
| 02:42.16 | *** join/#asterisk coppice (~chatzilla@m121-202-10-105.smartone-vodafone.com) |
| 02:49.04 | titter | a1fa: I get invites on both my BV numbers. |
| 02:50.28 | a1fa | how are they setup |
| 02:50.39 | a1fa | i have 1 number, and then alternative number |
| 02:50.44 | a1fa | the alternative number is 1-866 |
| 02:51.34 | *** part/#asterisk fireman_biff (~biff@65.48.132.153) |
| 02:53.39 | p3nguin | firestart: I recommend pulling in the pkgbuild in a local build directory, then extracting the source, then going into the source dir and running ./configure ; make menuselect to check what is enabled/disabled. Then you can go back to the build dir and run makepkg if you're satisfied with what you say in the menu. |
| 02:54.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:55.31 | FiReSTaRT | p3nguin: sensible idea.. i figured i already installed it, i just wanna set it up for learning, so i'll leave it as is, but i will save this for future installs :) |
| 02:56.06 | FiReSTaRT | tomorrow, when i have a bit of spare time, i'll finally start screwing around with * itself |
| 02:56.46 | titter | a1fa: That's what I had, but I cut back to a single number. Do you have an example of your sip.conf? I made these changes about two months ago, but maybe I can recognize the difference |
| 02:57.31 | [TK]D-Fender | Arch... something usually followed by "enemy". Sounds like a guaranteed fight.... abort, ABORT... |
| 02:57.38 | p3nguin | firestart: I often use makepkg -e and makepkg -o for scenarios exactly as you've encountered. |
| 02:58.41 | FiReSTaRT | p3nguin: something else to look in on the documentation.. today was my first day with arch, so i haven't picked up all of the ins and outs.. it'll happen.. i'm lovin' it so far |
| 02:58.43 | a1fa | titter: http://pastebin.ca/1972351 |
| 02:59.02 | p3nguin | Extract the sources but don't build anything. Then I can tinker around manually. Then follow up with making the package from the already extracted source. If you don't use the right option with makepkg, it'll delete the sources you've just manually worked on first. |
| 02:59.40 | titter | a1fa: I had mine split into two register => and two contexts |
| 02:59.50 | p3nguin | Once you get the feel for pkgbuilds and how AUR stuff works, you'll enjoy it even more. |
| 03:01.04 | FiReSTaRT | i'm loving it even from my limited exposure to it, but you are right and i have a whole lot more learning to do |
| 03:01.30 | a1fa | hm :0 |
| 03:01.36 | a1fa | using different numbers? |
| 03:01.41 | a1fa | are you sure you did not have Line2? |
| 03:01.48 | a1fa | i only have 1 Line with 2 numbers |
| 03:02.38 | titter | hmm |
| 03:02.45 | titter | Now I honestly forget |
| 03:03.00 | FiReSTaRT | off for a cancer stick :) |
| 03:03.17 | FiReSTaRT | thanks for all the tips p3nguin |
| 03:03.18 | titter | I added a second line actually, not two numbers |
| 03:04.03 | a1fa | there ya go |
| 03:04.08 | a1fa | i have 1 line, two numbers |
| 03:04.12 | a1fa | attached to 1 line |
| 03:05.32 | *** join/#asterisk echinos (~echinos@211.136.196.67.static.heavycomputing.ca) |
| 03:09.52 | titter | a1fa: Gotcha, what do the settings look like on their page? Do they just give you one auth_id and auth_pw? |
| 03:11.36 | a1fa | yup |
| 03:13.30 | *** part/#asterisk FFForever (FFForever@unaffiliated/ffforever) |
| 03:15.14 | titter | Well they suck lol |
| 03:15.37 | titter | Ironically I have never had an outage with them or issues in almost two years |
| 03:15.43 | titter | So for the price, eh, its not bad |
| 03:18.16 | vinhdizzo | For Background(), can we play mp3 files? |
| 03:20.29 | vinhdizzo | suppose I call my GV number that's set up on asterisk, and i select an extension that dials out via Dial(). Does the receiver (that of Dial()) see the original caller's phone number or the GV number on caller ID? |
| 03:22.52 | [TK]D-Fender | vinhdizzo: Depends on the provider you dial out of |
| 03:23.23 | vinhdizzo | i see |
| 03:23.37 | vinhdizzo | no mp3 support for Background()? |
| 03:25.13 | [TK]D-Fender | vinhdizzo: Yes, install Asterisk-addons for versions below 1.8 |
| 03:25.25 | vinhdizzo | thanks |
| 03:29.40 | vinhdizzo | anyone in here have google voice set up to asterisk via jabber.conf and gtalk.conf? I have in extensions.conf set up to pick up and say "hello world". It works when I call via computer in gmail, but not when I call the GV #. Can anyone help? |
| 03:32.33 | [TK]D-Fender | vinhdizzo: Go look at the call |
| 03:32.59 | vinhdizzo | [TK]D-Fender: what do you mean by that? |
| 03:34.15 | [TK]D-Fender | vinhdizzo: means go to * CLI and LOOK at the call you get sent |
| 03:34.56 | vinhdizzo | ok |
| 03:35.44 | vinhdizzo | i see the incoming call |
| 03:36.05 | vinhdizzo | but asterisk doesn't pick up |
| 03:39.32 | p3nguin | Did you tell it to? |
| 03:40.03 | vinhdizzo | yea |
| 03:40.24 | vinhdizzo | if I call from a different gmail account, to call computer, then asterisk picks up |
| 03:40.33 | vinhdizzo | but if i call from a phone to the GV#, it doesn't |
| 03:40.44 | p3nguin | I'm not seeing the call. |
| 03:41.31 | vinhdizzo | p3nguin: do you want me to paste it or something? |
| 03:42.01 | vinhdizzo | p3nguin: oh i see you called |
| 03:42.56 | [TK]D-Fender | ~pb |
| 03:42.56 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 03:44.44 | vinhdizzo | p3nguin: u hear "hello world" when u calling me from computer? |
| 03:50.55 | vinhdizzo | u guys still here? |
| 03:52.26 | p3nguin | I'm here, but I'm not seeing your debug info yet. |
| 03:52.28 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 03:53.05 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:54.17 | vinhdizzo | oh sorry one sec |
| 03:55.09 | vinhdizzo | http://pastebin.com/gcvPDqfm |
| 03:55.37 | vinhdizzo | thats calling from the GV # from a phone |
| 03:55.39 | vinhdizzo | not computer |
| 03:55.45 | vinhdizzo | the one im having issues with |
| 03:57.07 | p3nguin | Starting Gtalk/+17142711815-6b90 at default,vinhqn@gmail.com,1 failed so falling back to exten 's' |
| 03:59.15 | p3nguin | That, as well as the rest of the info after that, tells me that Asterisk took the call. |
| 03:59.20 | p3nguin | What was the problem? |
| 04:00.17 | vinhdizzo | on the caller side |
| 04:00.34 | vinhdizzo | i dont hear the pickup (hello world and hangup), but i get to the google voice voicemail |
| 04:00.37 | vinhdizzo | that's the problem |
| 04:01.08 | p3nguin | Do you agree that Asterisk is processing the call? |
| 04:02.40 | vinhdizzo | p3nguin: yea, sure. |
| 04:03.08 | vinhdizzo | not so sure why it shows "q failed" and falling back on exten "s" |
| 04:03.17 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 04:03.29 | vinhdizzo | let me show u my config |
| 04:03.57 | vinhdizzo | http://pastebin.com/1VpiRGrt |
| 04:05.25 | vinhdizzo | p3nguin: looking at the debug log again, after that line u mentioned, it shows jabber making an out call to the caller? |
| 04:10.37 | vinhdizzo | p3nguin: the "Starting Gtalk line", does that say that the caller dialed extension "1"? |
| 04:21.16 | vinhdizzo | p3nguin: hmm, very weird, after looking at the debug log some more, it appears that hello world is played and the hangup is called. however, on the caller side i dont hear anything and i go to the google voice voicemail....very weird |
| 04:23.34 | *** join/#asterisk jamko_ (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 04:26.31 | jamko_ | Is it absolutely certain that if DIRECTMEDIA and DIRECTRTPSETUP are specified in the GENERAL section of sip.conf , that it will apply to all UA unless otherwise specified on an individual basis? |
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| 04:40.14 | kaldemar | jamko_: yes, unless there is a bug. |
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| 04:53.03 | sshock | when using CALLERID(num) in a GotoIf statment, do I have to worry about if the number may or may not have a "+" on the front? |
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| 04:58.36 | sshock | sorry, my Internet is being flaky tonight; did you get my question? |
| 05:00.14 | sshock | about callerid having a + on the front |
| 05:00.59 | p3nguin | Yes, you have to be concerned about it. |
| 05:01.01 | ChannelZ | I'm not versed in international calls but I generally thought the CID number should always be digits |
| 05:01.15 | sshock | ok |
| 05:01.39 | p3nguin | Some providers think some of us want our inbound caller ID to contain the plus symbol. I strip it if it exists. |
| 05:01.44 | sshock | well, I just added a Verbose(incoming number is ${CALLERID(num)}) and it shows with a + on the front |
| 05:02.09 | sshock | next question: setting my callerid number for outgoing calls is working great, |
| 05:02.16 | p3nguin | ExecIf($["${CALLERID(num):0:2}" = "+1"]|Set|CALLERID(num)=${CALLERID(num):2}) |
| 05:02.30 | sshock | but I can't seem to be able to set my callerid name; is that pretty common for that to not work? |
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| 05:02.51 | sshock | p3nguin: thanks; I think I'll probably need that |
| 05:03.20 | p3nguin | Almost no telco is going to allow you to set your outgoing name. CID name lookup is done at the receiving end by a dip in a database. |
| 05:03.36 | ChannelZ | The name isn't actually transmitted anywhere |
| 05:03.43 | sshock | now my power is about to go out... |
| 05:03.51 | ChannelZ | sweet |
| 05:03.52 | p3nguin | Hold on to your socks! |
| 05:04.03 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 05:04.05 | sshock | ok, so what does it take to get my name in that database? |
| 05:04.07 | ChannelZ | arm up and wait for the zombies |
| 05:04.24 | sshock | (power keeps browining out like every 2 seconds) |
| 05:04.51 | ChannelZ | Your telco might be able to put the text of your choice into whatever database they use for a nominal fee |
| 05:05.24 | p3nguin | If you have a regular telco, they'll likely insert your name into their LIDB. If you are using an ITSP, some of them have the ability and some don't. |
| 05:05.58 | sshock | is it my telco or the recipient's telco that looks it up? |
| 05:06.19 | p3nguin | Also, even the ITSPs that have access to LIDB... there is no guarantee that the person you are calling will be using that DB for CNAM lookups. |
| 05:06.20 | sshock | I'm using strictly an ITSP |
| 05:06.37 | p3nguin | Which one? |
| 05:06.40 | sshock | Hmm, how many of these DBs are out there? |
| 05:06.44 | ChannelZ | it's the remote telco that looks it up ultimately and sends to its customer whose phone is ringing |
| 05:06.44 | sshock | flowroute |
| 05:07.14 | sshock | ok, so it all depends on the remote telco; that sucks :( |
| 05:08.31 | sshock | So for it to really work I'd have to call up Qwest, AT&T, Verizon, and who knows who else to get my name into all their databases... |
| 05:08.31 | p3nguin | I don't recall if Flowroute has access to databases or not. You'll have to ask them if they can submit your CNAM into LIDB. |
| 05:08.35 | sshock | sounds like a lost cause |
| 05:08.47 | sshock | ok; worth checking I guess |
| 05:09.08 | p3nguin | If they can, they'll probably charge you like $10 to submit it where they have access. |
| 05:09.45 | sshock | ok |
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| 05:11.43 | ChannelZ | Hmm. Vitelity wants $10 and $1.49/mo to be in the DB |
| 05:12.13 | ChannelZ | oh no wait that's directory. Just $10 one-time for the CNAM entry. |
| 05:13.43 | sshock | ok, hopefully flowroute has something similar |
| 05:13.50 | sshock | I'd pay $10 to get my name working |
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| 05:21.12 | ChannelZ | I don't care if people know who I am |
| 05:22.23 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 05:23.08 | p3nguin | I would prefer to have a valid name appear on CID along with my number. "OUT OF AREA" or "TOLL FREE CALL" kinda sucks. |
| 05:26.47 | ChannelZ | Mine is just blank or Unavailable. Cell phones don't know anyway and half the people I know don't even have real phones, only cells |
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| 06:00.00 | pbxdude | Hello, is the latest asterisk-addons (1.6.2.2) compatible with asterisk-1.8.0? |
| 06:05.13 | WIMPy | No. It's included. |
| 06:06.40 | ChannelZ | people don't read UPGRADE anymore I guess |
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| 06:13.27 | pbxdude | ok, thanks |
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| 06:34.18 | SeTTleR | hi |
| 06:39.31 | ChannelZ | hai |
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| 09:37.12 | Besticles | Bit of a problem here. I got my agi script working. It just dawned on me that, because I originate using AMI, asterisk is only mapping to the appropiate extension if the call gets answered. If it's busy, or disconnected, or even no answer, it never executes the code in the dialplan. Can someone steer me to the right direction? |
| 09:39.29 | kaldemar | originate the call using Local/exten@context instead of the "real" tech and put you're dialplan core there. |
| 09:41.25 | Besticles | Won't I lose control of what channel I want the call to be placed on, using that method? |
| 09:43.36 | *** join/#asterisk coldsteal (~Antonni@unaffiliated/coldsteal) |
| 09:44.39 | coldsteal | Hello, does anyone know how to solve "Music class default requested but no musiconhold loaded." |
| 09:45.38 | kaldemar | Besticles: no. |
| 09:47.36 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 09:48.04 | Besticles | Oh I see. Originating using local, won't actually start the call, it will map it to the extension, I think physically make the call, then AGI it, and bam, I'm happy. |
| 09:48.05 | Besticles | Thanks. |
| 09:48.15 | Besticles | think = then |
| 09:49.31 | E-bola | WARNING[12553]: chan_sip.c:6011 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) |
| 09:49.38 | E-bola | ^^^Why is that a problem? |
| 09:51.31 | petern_ | ulaw isn't alaw |
| 09:53.00 | E-bola | so? it apears its working fine, and im asuming asterisk is simply converting, so why do i need to be warned? |
| 09:53.16 | E-bola | or is something else going on? |
| 10:04.16 | Besticles | Kaldemar, you're the man. Thanks, it works. |
| 10:09.07 | kaldemar | Besticles: no problem. |
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| 10:51.51 | Besticles | kaldemar: the only issue i found is that ${DIALSTATUS} seems to be corrupted now. It returns CHANUNAVAIL. Is that normal, if so is there any work arounds? |
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| 11:33.04 | [sr] | howdy |
| 11:38.42 | FlashDeluxe | Hi @all! Does anybody know a good GUI for asterisk, exept freepbx and asterisk-gui? |
| 11:39.20 | *** join/#asterisk bjornts (~Adium@2001:700:200:10:d69a:20ff:fe00:c54e) |
| 11:39.21 | FlashDeluxe | which can handle * 1.8? |
| 11:51.34 | *** part/#asterisk drepan (~drepan@apcdns2.autopage.co.za) |
| 11:57.00 | krion | i should i manage emergency numbers correctly ? |
| 11:57.26 | krion | i face a situation today, with a customer who got short number like 110 - 112 - 118 |
| 11:57.41 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 11:57.52 | krion | the 112 was unreachable, so it then call 112, who's the european emergency number... |
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| 12:10.07 | moos3 | good morning, how can i make a queue thats using mysql for real time queues, accept press 1 to leave a message ? |
| 12:14.32 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 12:16.13 | doolittlework | hi there can someone please inlighten me, i took over the maintenance of a server that has strange g729 license, i see they were downloaded from http://asterisk.hosting.lv/ how does this compare to the paid for codex? |
| 12:25.14 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 12:26.26 | hrhrhr | chan_sccp seems to have a lot of good feedback. anyone else using it? |
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| 12:28.19 | doolittlework | hrhrhr: still trying i see, buy a snom |
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| 12:32.05 | hrhrhr | thank you for that helpful feedback |
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| 13:07.38 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 13:07.40 | evangelion | hello |
| 13:08.35 | evangelion | does anyone know why this ( http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch ) patch has been discontinued? |
| 13:09.31 | evangelion | asterisk bug traking system ( https://issues.asterisk.org/view.php?id=4825#82279 ) reports "The development team has given the area of codec negotiation a lot of thought" |
| 13:09.43 | evangelion | since ~2K8 |
| 13:10.01 | evangelion | but i can't find anything related to transcoding avoidance in changelog |
| 13:12.35 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:16.02 | [TK]D-Fender | evangelion: Check Mantis for other related bugs/features in progress and reopen if needed |
| 13:16.19 | Katty | hi |
| 13:17.38 | [TK]D-Fender | Katty: Mew. |
| 13:17.43 | evangelion | [TK]D-Fender: thank you |
| 13:25.27 | evangelion | are there any chances to force asterisk to negotiate the _SAME_ code on both sides of the call when possible using official releases? |
| 13:26.12 | Katty | [TK]D-Fender: mew |
| 13:26.54 | evangelion | i mean: now the two sides are negotiated individually ignoring the bridging logic |
| 13:27.57 | evangelion | this could potentially lead to useless transcoding |
| 13:31.09 | E-bola | hmmm my 1.8 rc2 just froze again, really need to upgrade to the final. Sure hope its more stable |
| 13:46.43 | marksaitis | can anybody recommend me a sip softphone? :) |
| 13:47.10 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 13:53.03 | [TK]D-Fender | marksaitis: Ekiga |
| 13:53.05 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-jqgeofystmaxtvvg) |
| 13:53.06 | [TK]D-Fender | ~ekiga |
| 13:53.06 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 13:54.40 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:57.51 | marksaitis | Ekiga? tls and srtp? |
| 13:58.04 | marksaitis | I think I tried that crap yesterday |
| 13:58.36 | evangelion | according to this( https://issues.asterisk.org/view.php?id=10500 ) chan_sip try to avoid transcoding moving first calling channel's codec in the CODEC_PREFS list |
| 13:59.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 14:03.02 | marksaitis | I believe that CounterPath products are the best, been the best and will be for a long time ;] |
| 14:09.22 | *** join/#asterisk shapr (~shapr@nat/digium/x-wtqwnmqettlioppy) |
| 14:14.05 | [TK]D-Fender | marksaitis: If you feel like paying for it, sure |
| 14:19.44 | marksaitis | [TK]D-Fender, well, there are no other TLS SRTP SIP apart from this one |
| 14:19.49 | marksaitis | feels like a monopoly |
| 14:20.42 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:20.42 | marksaitis | the strange thing is that I have bria on iphone 4 and bria on my PC. I have installed the same self signed cert on both for tls |
| 14:20.50 | marksaitis | iphone client works fine with it |
| 14:20.58 | marksaitis | pc client complains about its name or smth |
| 14:21.00 | marksaitis | ... |
| 14:21.11 | marksaitis | I dont get it |
| 14:21.48 | p3nguin | If it complains about Smith, change it to Jones. |
| 14:22.02 | shapr | marksaitis: What does it complain about? |
| 14:22.37 | marksaitis | dunno,m litttle bastard, say smth about cert name mismatch |
| 14:24.33 | *** join/#asterisk polemon (mcp@polemon.org) |
| 14:24.58 | polemon | hey can someone give me a quick snippet? |
| 14:25.25 | polemon | I want two SIP phones to be rinning when a number is dialed |
| 14:25.53 | polemon | once one of them picks up, the other phone should stop ringing |
| 14:25.56 | polemon | how do I do that? |
| 14:27.48 | [TK]D-Fender | polemon: Dial both. "core show application dial" |
| 14:32.54 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:33.50 | polemon | [TK]D-Fender: what should I do with what you've put in '"'? |
| 14:34.21 | p3nguin | Wow, first day using Asterisk? |
| 14:34.23 | polemon | I simply Dial(Phone1&Phone2) and have a Hangup on next? |
| 14:34.26 | polemon | yeah |
| 14:34.35 | polemon | not first but second... |
| 14:34.41 | p3nguin | Run his suggested command in the CLI. |
| 14:36.43 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:36.52 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:36.59 | ManxPower | polemon, looks like you should be reading, the asterisk book |
| 14:37.26 | *** join/#asterisk marksaitis (~Mk@207.243.46.84.ip.erdves.lt) |
| 14:37.42 | polemon | All other channels that were requested will then |
| 14:37.42 | polemon | be hung up. |
| 14:37.49 | polemon | that explains it |
| 14:37.52 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 14:37.55 | polemon | ManxPower: I have the book right here |
| 14:37.59 | polemon | and I'm working with it |
| 14:38.24 | polemon | the book by O'Reilly press, that is. Don't know if that's the one you mean |
| 14:38.30 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 14:38.52 | polemon | "Asterisk - The future of telephony" |
| 14:38.59 | ManxPower | That is the one |
| 14:40.10 | ManxPower | polemon, there is information in there about the CLI commands? |
| 14:40.17 | ManxPower | <polemon> [TK]D-Fender: what should I do with what you've put in '"'? |
| 14:40.26 | fullstop | I have an asterisk system which handles ~96 concurrent calls, acting as an IVR before possibly sending them off to a second party agent. Due to heavy usage, we may increase the number of inbound calls, but the other end will probably still have the same number of agents. |
| 14:41.04 | fullstop | Is there a way for me to put the callers into a queue and limit it with GROUP and GROUP_COUNT? How can I remove people from the queue and transfer them after a call clears? |
| 14:42.22 | polemon | there's a variable read, which goes like this: ${EXTEN:6} in a code sample, waht does the ":6" mean? |
| 14:42.24 | marksaitis | I have two IP Phones connected using TLS and SRTP. When I try to call from one to another, system says that this person is unavailable! HELP |
| 14:42.26 | marksaitis | :) |
| 14:42.40 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 14:43.04 | *** join/#asterisk flavioribeiro (~flavio@li20-198.members.linode.com) |
| 14:43.27 | polemon | or can I check the CLI for that syntax |
| 14:43.28 | fullstop | polemon: first 6 characters of the extension. |
| 14:43.53 | polemon | aha, so it doesn't make sense for that example, since the extension is just 4 digits long... |
| 14:44.45 | fullstop | polemon: sorry -- last 6 |
| 14:44.50 | fullstop | read here http://www.voip-info.org/wiki/view/Asterisk+variables |
| 14:44.53 | fullstop | under substrings |
| 14:45.17 | fullstop | and, again, I was wrong about the last 6.. :) |
| 14:45.36 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:45.41 | fullstop | everything after the 6th character |
| 14:48.00 | fullstop | needs to wake up |
| 14:48.16 | marksaitis | can anybody spot smth wrong in this cli dump. I been trying to call from one phone to another, but it says its unavailable even though asterisk shows that both of them are online! |
| 14:48.17 | marksaitis | http://pastebin.com/MwHuAhcM |
| 14:50.07 | polemon | ok, so when the EXTEN is just 4 digits long and it reads ${EXTEN:6} then nothing is returned? |
| 14:52.13 | *** join/#asterisk eternis (~proba@cpe-67-244-127-222.nyc.res.rr.com) |
| 14:52.16 | eternis | hi |
| 14:52.20 | eternis | anyone here? |
| 14:52.27 | thehar | sleeping |
| 14:52.31 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 14:53.15 | polemon | or does ${EXTEN} contain the full number? |
| 14:53.19 | eternis | hi |
| 14:53.25 | polemon | not just the extension in question |
| 14:53.33 | p3nguin | ${EXTEN} is the extension called. |
| 14:54.06 | p3nguin | letters, numbers... whatever the extension is. |
| 14:57.19 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:59.14 | *** join/#asterisk Denial- (Denial@drgi.co.uk) |
| 15:00.09 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:03.30 | eternis | let me get this straight, there's hardware to hook up the old cable phone, the one that has 2 copper wires, then there's the new type that uses straight internet. RIGHT? |
| 15:03.47 | eternis | which one should I get started with? |
| 15:03.49 | p3nguin | Sounds pretty accurate to me. |
| 15:03.59 | p3nguin | I like IP phones. |
| 15:04.48 | eternis | right, sounds more streamlined and elegant solution. the wave of the future. |
| 15:05.13 | elred_ | Hi. I updated my kernel and want to keep ISDN support, i was running an old version of asterisk (1.4.24), i looked in latest 1.4 (.36) and i don't see any chan_lcr support. Can someone tell me please in which version chan_lcr got in ? (I want to use new kernel ISDN interface rather than the old chan_misdn). Thanks you ! |
| 15:05.28 | p3nguin | I think IP phones will give you more useful features than analog phones will. |
| 15:06.44 | eternis | analog = RJ14 |
| 15:07.02 | eternis | un-plugs his phone wire and looks |
| 15:08.09 | fullstop | I've been thinking about this -- I really like the ip phones.. but, at home, I like the convenience of the cordless phone. |
| 15:08.23 | eternis | confirmed the phone got just TWO!! wires, unbelievable! |
| 15:08.32 | p3nguin | There are cordless IP phones. |
| 15:08.51 | fullstop | My kids can accidentally drop the phone in the tub and it's not a huge financial burden. |
| 15:08.54 | p3nguin | I actually use one at home. |
| 15:09.01 | fullstop | Now, they haven't done this yet... |
| 15:09.11 | fullstop | but I'm okay with them using the analog phone. |
| 15:09.19 | p3nguin | You can also use a cheap cordless phone paired with an ATA. |
| 15:09.33 | fullstop | I don't even want to think about how much a cordless IP phone costs. :) |
| 15:09.44 | p3nguin | I spent $20 on mine. |
| 15:09.59 | fullstop | p3nguin: model / talk time? |
| 15:10.01 | eternis | ip phones will become common place hopefully |
| 15:10.31 | eternis | so let's say, if I want to try both methods, what extra hardware do I need? |
| 15:10.49 | fullstop | Analog phones will die, but mostly because of the abundance of cell phones |
| 15:10.54 | eternis | I am not setting up a business or a residential area |
| 15:11.19 | fullstop | eternis: for an ip phone, there are a lot to choose from. We use polycom phones here. |
| 15:11.20 | elred_ | ok just found it's apart from release and available on http://www.linux-call-router.de/ (LCR --> chan_lcr). !! |
| 15:11.38 | Qwell | eternis: either an ATA, or a PCI card with FXS modules (like the Digium TDM410) |
| 15:11.56 | p3nguin | It says ATS on the device, but the user agent is Elite 6011S. I don't know about talk time, since I don't sit on it for a long time -- I use it when I need to be portable around the house. |
| 15:12.27 | eternis | ATA as the hard drive cable PATA and SATA? |
| 15:12.31 | Qwell | no |
| 15:12.36 | Qwell | ~ATA |
| 15:12.36 | infobot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 15:12.56 | Qwell | also I move to rename PATA to PITA |
| 15:13.02 | fullstop | p3nguin: thanks. With my old cordless phone (which finally died after 15 years of use). That one was awesome. I could leave it off of the charger for over a week and it would be fine. |
| 15:13.08 | polemon | will the comments made with NoOp show up in the logs? |
| 15:13.19 | Qwell | eternis: That of course assumes that the "2-wire" phone you got is actually analog. |
| 15:13.57 | p3nguin | fullstop: This cheap-o ATS phone isn't that good. |
| 15:14.02 | polemon | or what should I or should I use another application instead? |
| 15:14.26 | p3nguin | After a few days off the charging base, it'll go dead. |
| 15:14.29 | eternis | Qwell: of course, it's verizon RJ11 wire |
| 15:14.41 | Qwell | verizon? O.o |
| 15:14.46 | Qwell | thought you said it was a phone |
| 15:15.13 | fullstop | Qwell: I think he is saying that he has verizon phone service |
| 15:15.15 | p3nguin | I of course have not replaced the battery with a brand new one. The battery in it could be the cause of the short standby times. |
| 15:15.17 | eternis | alright so the Digium TDM410 is for the asterisk+analog (rj11) set up right? |
| 15:15.36 | Qwell | eternis: yes |
| 15:15.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:15.43 | eternis | Qwell: verizon company, that's my account. |
| 15:15.55 | p3nguin | Compare the price of a PAP2T-NA to the TDM410 and decide which one fits your needs right now. |
| 15:15.57 | fullstop | p3nguin: I have a set of DECT phones that I replaced the old lucent cordless phone... they last maybe 5 days without a charge. |
| 15:16.23 | fullstop | p3nguin: audio sounds better, but I miss my old lucent. |
| 15:18.23 | eternis | damn $10 and I'd get the latest Radeon HD 6850 |
| 15:18.33 | polemon | hello? |
| 15:18.37 | polemon | or what should I or should I use another application instead? |
| 15:18.46 | eternis | can the analog part be done in GPU off loaded? |
| 15:19.22 | ChannelZ | polemon: Log() |
| 15:19.28 | Qwell | GPU offloaded? |
| 15:20.22 | eternis | oh the PAP2T-NA is cheaper. |
| 15:24.00 | eternis | in the meantime, the IP way I don't need any extra hardware? |
| 15:24.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 15:24.37 | eternis | right of the bat do I need the DAHDI and MISDN part in asterisk? |
| 15:24.59 | fullstop | dahdi only if you are using the TDM card. |
| 15:24.59 | eternis | just trying to sort out what are the analog and digital elements of asterisk installation |
| 15:25.00 | Qwell | Only DAHDI, and only if you use PCI hardware (for the most part) |
| 15:25.21 | polemon | ChannelZ: thanks |
| 15:25.30 | p3nguin | If you get a PAP2, that converts your analog phone to an IP phone. |
| 15:25.32 | eternis | do I need a TDM card? |
| 15:25.40 | fullstop | not if you get the ATA |
| 15:25.41 | Qwell | that is what the TDM410 is |
| 15:25.48 | Qwell | (hence the TDM in the model # :p) |
| 15:25.53 | eternis | ah right.. sorry |
| 15:26.22 | eternis | wow there are tons upon tons of voip sites. |
| 15:26.53 | eternis | great so the digital way I don't need MISDN nor DAHDI |
| 15:27.16 | Qwell | Do not say digital when referring to VoIP. Ever. |
| 15:27.32 | eternis | so what do I say? |
| 15:27.34 | Qwell | "digital phones" are something VERY different |
| 15:27.39 | Qwell | IP phone, or VoIP phone |
| 15:27.45 | eternis | ah yes |
| 15:27.58 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 15:28.18 | eternis | so DAHDI and MISDN aren't needed for VoIP, right? |
| 15:28.29 | bougyman | when did an ip or voip phone stop being digital? |
| 15:28.31 | p3nguin | They are if you're running analog phones. |
| 15:28.32 | [TK]D-Fender | eternis: Correct |
| 15:28.39 | eternis | I ask because I am using gentoo and has the dahdi and misdn USE flags. |
| 15:28.42 | eternis | thanks |
| 15:28.43 | bougyman | what analog voip phone are you using? |
| 15:28.56 | Qwell | bougyman: It didn't, but confusing the terms is very very bad. |
| 15:29.07 | [TK]D-Fender | eternis: DAHDI is also needed for support of 2-3 other things actually. One is IAX2 Trunk Mode which makes that answer sort of a "probably not" |
| 15:29.36 | eternis | none, yet. so I will try go the VoIP way first, until I acquire the TDM or the PAP2T-NA. |
| 15:29.44 | [TK]D-Fender | bougyman: My analog phone has a digital clock on it. It is very confused |
| 15:29.53 | [TK]D-Fender | sends his phone to therapy |
| 15:30.12 | bougyman | funny, my digital phone sports an analog clock (snom 370) |
| 15:30.39 | eternis | [TK]D-Fender: 2-3 things linked to VoIP or the analog part, or do you mean that work for both? |
| 15:30.39 | p3nguin | You'll spend $30 on a PAP2, and you can use it with your existing bell phone. It'll cost you about the same to pick up a used low-end IP phone. |
| 15:30.46 | tzafrir | bougyman, it's not analog. It's printed using digital pixels |
| 15:31.10 | bougyman | fauxnalog |
| 15:31.13 | [TK]D-Fender | eternis: YES, you should install DAHDI regardless. |
| 15:31.37 | p3nguin | In 1.8, do we still need dahdi for anything? |
| 15:31.38 | eternis | is Voiplink.com a reputable place to purchase? |
| 15:31.44 | [TK]D-Fender | bougyman: Your phone needs SERIOUS therapy... its delusions are having delusions :p |
| 15:32.09 | p3nguin | eternis: Check ebay if you need to save some money. |
| 15:32.16 | [TK]D-Fender | ~savemoney |
| 15:32.17 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 15:32.17 | eternis | I am running kernel 2.6.36 and asterisk --> asterisk-1.6.2.13-r2 |
| 15:32.48 | [TK]D-Fender | eternis: They look big enough. Before you jump, what are you looking at? |
| 15:34.11 | eternis | [TK]D-Fender: first time setting up asterisk at home. |
| 15:34.28 | [TK]D-Fender | eternis: What do you have now, and what do yuo want to do? |
| 15:34.29 | eternis | I just got sorted out the analog part and the VoIP side. |
| 15:34.43 | [TK]D-Fender | eternis: Im not so sure... |
| 15:34.50 | eternis | during the installation I was confused about some modules and hardware requirements. |
| 15:34.51 | The_Boy_Wonder | asterisk is the bomb! |
| 15:34.57 | The_Boy_Wonder | fo sho! |
| 15:35.14 | bougyman | bomb like *BOOM*? |
| 15:35.18 | eternis | I have a quad-core with internet connection... do you want the specs of my computer? |
| 15:35.43 | eternis | and an analog phone with rj11 wire with a verizon number. |
| 15:36.23 | *** join/#asterisk ccomp5950 (~ccomp5950@24.204.47.5) |
| 15:36.28 | The_Boy_Wonder | my computer has a turbo button |
| 15:36.42 | p3nguin | Are you wanting to connect your Asterisk system to your Verizon land line wall jack? |
| 15:36.45 | The_Boy_Wonder | why would I ever not want to be in turbo mode? |
| 15:37.21 | p3nguin | when stability is a concern. |
| 15:38.03 | p3nguin | eternis: Are you wanting to connect your Asterisk system to your Verizon land line wall jack? |
| 15:38.14 | The_Boy_Wonder | yeah, i guess limiting my cpu to 4mhz instead of 8 would help stability |
| 15:38.19 | eternis | err there are several TDM410 models, ending letters are P, PLF, BF, EF, ELF, E, B... |
| 15:38.39 | dmast | Is there any way to use the unsolicited_mailbox option multiple times for the same peer in order to keep track of message counts for multiple phones? |
| 15:39.47 | [TK]D-Fender | eternis: Ok, so you have a normal land line, right? |
| 15:39.51 | eternis | p3nguin: well, first the pure VoIP way since I don't have any hardware card yet. If there's a way to start setting up asterisk for pure VoIP mode, that doesn't require extra hardware, that'll be great. |
| 15:40.05 | eternis | [TK]D-Fender: right. |
| 15:40.11 | [TK]D-Fender | eternis: Do you want * to USE it? |
| 15:40.37 | p3nguin | eternis: If you intend to keep your Verizon number, you have two choices: connect Asterisk to your land line, or port your number out of Verizon to an ITSP. |
| 15:40.46 | eternis | [TK]D-Fender: when I get the extra hardware piece, sure why not. |
| 15:41.03 | [TK]D-Fender | eternis: Ok, what do yuo wnt to do using VoIP tech? |
| 15:41.05 | p3nguin | eternis: So then you have to decide if you're keeping the land line or if you want to run strictly over the internet. |
| 15:42.14 | eternis | first strictly over internet. |
| 15:42.15 | p3nguin | eternis: If you're going to use the land line, you can get a different type of device (similar to the PAP2) to connect the land line to Asterisk via IP. |
| 15:42.35 | eternis | [TK]D-Fender: to make calls, receive calls, and to learn about asterisk. |
| 15:43.14 | [TK]D-Fender | eternis: Ok, you don't need any special hardware for that. However you might want a little something to start with and your goals/needs seem small. |
| 15:43.28 | [TK]D-Fender | eternis: That in mind I'd recommend a Linksys SPA-3102. |
| 15:43.40 | eternis | for a home user, what are reasonable goals/uses? |
| 15:43.47 | [TK]D-Fender | eternis: This will let you use your land line with *, and ALSO use a regular phone with it as well. |
| 15:44.25 | [TK]D-Fender | [11:43]<eternis>for a home user, what are reasonable goals/uses? <- precisely what you stated. Just to learn *, maybe use an ITSP for some calls, maybe use your home line for basic low-volume calling,e tc |
| 15:44.54 | [TK]D-Fender | eternis: The SPA-3102 is like the PAP2 you were previously recommended, but instead of supporting 2 phones, it does 1 phone, 1 line. |
| 15:45.04 | [TK]D-Fender | eternis: $60 well spent |
| 15:45.19 | eternis | great. |
| 15:45.21 | p3nguin | It's a nice piece of equipment for home use. |
| 15:45.46 | eternis | what's the pure VoIP way? the one porting the number to an ITSP? |
| 15:46.01 | eternis | does that mean I kill my verizon account for good? |
| 15:46.27 | p3nguin | yes |
| 15:46.35 | eternis | oh I see |
| 15:46.35 | [TK]D-Fender | eternis: You'd port your number if you care to KEEP it. Clearly an option, but not always necessary |
| 15:46.52 | [TK]D-Fender | eternis: And you could kill off Verizon either way. Depends on your goals. |
| 15:46.59 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:47.04 | [TK]D-Fender | eternis: Sometimes having a hard-line is a good thing. Depends on the details |
| 15:47.16 | [TK]D-Fender | eternis: Alarm systems is a good one, so is faxing |
| 15:47.23 | eternis | aha |
| 15:47.27 | [TK]D-Fender | eternis: Depends on your piece of mind, etc |
| 15:47.29 | eternis | I thought fax was dead |
| 15:47.47 | [TK]D-Fender | eternis: I only use my cell really, and use VoIP at home for LD calling. |
| 15:48.02 | [TK]D-Fender | [11:47]<eternis>I thought fax was dead <- No, just a decades-long death-rattle |
| 15:48.27 | eternis | is not possible to use VoIP with a cell? |
| 15:48.40 | eternis | wait yes, sype and all. |
| 15:48.44 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:48.55 | eternis | but that's not homebrew |
| 15:49.19 | [TK]D-Fender | eternis: My cell runs Android, and VoIP is just internet traffic... I could do it if I cared. |
| 15:49.21 | eternis | for the SPA-3102 do I need the DAHDI and MISDN modules? |
| 15:49.29 | [TK]D-Fender | eternis: Of course I use my * for consulting as well |
| 15:49.49 | [TK]D-Fender | eternis: No, it does not require DAHDI, but as I said earlier I recommend you installing it ANYWAY |
| 15:49.58 | eternis | yes yes. |
| 15:50.06 | [TK]D-Fender | eternis: For other reasons than just supporting hardware to use your lines/phones |
| 15:50.06 | eternis | [TK]D-Fender: same with MISDN? |
| 15:50.33 | [TK]D-Fender | eternis: that is absolutelyworthless to you. mISDN is for ISDN BRI which frankly does nt exist in your world anyway |
| 15:50.53 | [TK]D-Fender | eternis: USA != BRI |
| 15:51.00 | eternis | lol, ok |
| 15:51.02 | [TK]D-Fender | eternis: Whoever suggested that wasn't thinking at all |
| 15:51.09 | eternis | BRI stands for... |
| 15:51.13 | titter | During a meetme conference, is there a way to disable moh? |
| 15:51.16 | [TK]D-Fender | ~bri |
| 15:51.16 | infobot | hmm... bri is [~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
| 15:51.35 | [TK]D-Fender | titter: Change the calls before entering |
| 15:51.37 | Naikrovek | I had a BRI when I lived in Anchorage |
| 15:51.41 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:51.41 | eternis | bri is also a delicious cheese |
| 15:51.48 | [TK]D-Fender | eternis: bri+e |
| 15:51.54 | Naikrovek | I used it for internet access because it was the fastest thing available |
| 15:51.54 | eternis | :D |
| 15:51.56 | [TK]D-Fender | And indeed |
| 15:52.22 | *** part/#asterisk flavioribeiro (~flavio@li20-198.members.linode.com) |
| 15:52.27 | [TK]D-Fender | Naikrovek: Anchorage -> what happens when you beach a boat in an ice-cap ;) |
| 15:52.58 | Naikrovek | nothing. nothing ever happens on/to that boat again |
| 15:53.14 | eternis | also is this a reputable guide to follow? --> http://www.asteriskguide.com/mediawiki/index.php/Main_Page |
| 15:53.18 | Naikrovek | stripped and abandoned |
| 15:53.18 | eternis | I downloaded the PDF |
| 15:53.19 | [TK]D-Fender | Naikrovek: That wasn't a question :p |
| 15:53.25 | Naikrovek | oh |
| 15:53.26 | p3nguin | ~book |
| 15:53.26 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 15:53.29 | p3nguin | eternis: ^^ |
| 15:53.33 | Naikrovek | everything goes over my head |
| 15:53.34 | [TK]D-Fender | p3nguin: You win... THIS TIME |
| 15:53.59 | [TK]D-Fender | Naikrovek: May I suggest you spend more time on the rack ;) |
| 15:55.15 | polemon | i need some more help |
| 15:55.19 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.173.194) |
| 15:55.38 | eternis | --> For Asterisk 1.4 |
| 15:55.43 | eternis | I have 1.6 |
| 15:55.47 | polemon | Mar 13 13:13:16 asterisk[1098]: VERBOSE[1245]: -- Executing [20949600@ISDN-PROVIDER-6267687004c90f6df4fa85-incoming:2] Dial("DAHDI/1-1", "Local/20949600@/n") in new stack |
| 15:55.48 | [TK]D-Fender | eternis: With the SPA-3102 you'll be able to use a regular analog phone with *, as well as your line. It will bridge the 2 in case of a power failure for failover. Also if you decide to ditch your line altogether you can still use the Phone port so it isn't a loss as an investment. |
| 15:55.51 | polemon | Mar 13 13:13:16 asterisk[1098]: NOTICE[1245]: chan_local.c:550 in local_call: No such extension/context 20949600@ while calling Local channel |
| 15:56.02 | polemon | why do I get this error? |
| 15:56.20 | eternis | [TK]D-Fender: ok |
| 15:56.24 | [TK]D-Fender | eternis: Later you can of course deploy it remotely if you want to access someone elses line via your server etc. Or give to a friend/reletive so they can call you for free |
| 15:56.45 | [TK]D-Fender | polemon: Because there is no context |
| 15:57.08 | [TK]D-Fender | polemon: and why are you even using DIAL to call that instead of a Goto()? |
| 15:58.21 | polemon | wait, another example |
| 15:58.34 | polemon | I have a SIP fone defined for caller ID 02 |
| 15:58.46 | polemon | and I can call all internal phones each other fine |
| 15:58.57 | polemon | and also call outside to an ISDN line |
| 15:59.06 | polemon | but incomming calls don't get through |
| 15:59.29 | polemon | the number 20949602 is supposed to go to CallerID 02 |
| 15:59.34 | p3nguin | You can't seriously expect it to work when you've set it up wrong. |
| 16:00.05 | polemon | well, what did I do wrong, then? |
| 16:00.37 | WIMPy | What? A call going to a CallerID? What does that mean? |
| 16:01.16 | polemon | to a local sip phone |
| 16:01.31 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 16:01.49 | *** join/#asterisk Glasswalker (~Glasswalk@CPE005056ad47df-CM001225e00d58.cpe.net.cable.rogers.com) |
| 16:03.11 | [TK]D-Fender | polemon: Your DIAL is bad. You are dialing a local channel without specifying the CONTEXT. Yuo may as well have dumped them off a cliff |
| 16:03.25 | [TK]D-Fender | lunch time, BBIAB |
| 16:05.23 | *** join/#asterisk Gianlu (~gianluca@89.251.177.19) |
| 16:05.29 | Gianlu | hello everybody |
| 16:06.48 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:07.13 | polemon | how is my dial bad, it's an incomming call from an external source |
| 16:08.11 | p3nguin | "You are dialing a local channel without specifying the CONTEXT." |
| 16:08.53 | Gianlu | guys, I am quite new to Asterisk. and I have some newbie questions... is there any good soul who can help me? |
| 16:09.18 | fullstop | ask and see what you get |
| 16:09.20 | *** join/#asterisk MrWork (~Mr@host216.ezlinx.net) |
| 16:09.24 | fullstop | don't ask to ask. :) |
| 16:09.27 | Qwell | ~help |
| 16:09.31 | Gianlu | @fullstop: ok thanks. |
| 16:09.35 | Qwell | glares at infobot |
| 16:09.39 | fullstop | hahaha |
| 16:11.20 | Naikrovek | Gianlu: yeah, just ask the questions and we'll answer |
| 16:12.38 | Gianlu | I understand the different between SIP account, extension and channels. I am trying to use the Asterisk Manager API in a project and I need to match SIP peer and extensions... how do I do that? I mean, it is just a convention that the SIP number is the same than the extension... but that could be completely different, right? |
| 16:13.12 | Gianlu | sorry, I am quite confused.. I guess I still have a lot to learn :) |
| 16:13.33 | p3nguin | Some people prefer to make extension 1000 dial SIP/1000, but you can do anything you want. |
| 16:14.06 | Gianlu | p3nguin: that's it. so how does one know that SIP/1000's extension is - say - 12345? |
| 16:14.24 | Gianlu | do I really have to parse the extensions.conf file? isnt there a easiest way? |
| 16:14.47 | Gianlu | and also, can more extensions be assigned to the same SIP peer? I guess so.. right? |
| 16:15.00 | Gianlu | *easier not easiest :) |
| 16:15.18 | vinhdizzo | p3nguin: any additional comments to add to where we got to last ngiht with google voice? |
| 16:15.18 | p3nguin | Sure, extension 12345 and 54321 can both Dial SIP/1000. |
| 16:15.43 | Naikrovek | you'll just have to enforce some convention when you assign extension numbers |
| 16:16.48 | Gianlu | so, show channels gives me the list of the active channels (streams between peers and asterisk server), and that's not what I need. show hints gives me the.. hints :) but they are not mandatory as far as I understand. sip show peers gives me the list of SIP peers defined in the system, but they don't necessary represent the extensions... |
| 16:17.07 | eternis | anyone running gentoo around here? what's the status of asterisk 1.8 in gentoo? |
| 16:17.27 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 16:17.30 | zplinux | hi all |
| 16:18.23 | eternis | hi |
| 16:18.24 | Gianlu | hi |
| 16:18.26 | zplinux | is anyone aware of an issue using dahdi 2.4.0+2.3.0 on kernel 2.6.32 |
| 16:18.28 | zplinux | ? |
| 16:19.09 | eternis | also I am intriguied by someone mentioning that DAHDI wasn't needed in asterisk 1.8. |
| 16:19.09 | zplinux | I used gcc 4.1.3 |
| 16:19.09 | polemon | ok, I think, I'm on the right path right now, how do I specify the context inside a Dial()? |
| 16:20.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:21.41 | p3nguin | gianlu: Devise your own convention for extensions and device names. Device names are often the MAC address of the phone. Extensions could be the first three/four letters (converted to numbers) of the person's name. For example, to call Mark, you might call extension 6275 which runs Dial(SIP/0011aabb1234,30). |
| 16:24.53 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:24.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:25.00 | Gianlu | p3nguin: ok. I try to turn the question around. I am sure all of you know FOP & FOP2. Please have a look here: http://www.fop2.com/img/gallery/fullsize/fop2screenshot1.png The question is how do they manage to get the extension list even if they are not called/in use? The previous FOP version was using AMI to obtain the various information... I guess FOP2 is doing the same.. |
| 16:28.55 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:29.47 | [TK]D-Fender | Gianlu: Of course |
| 16:30.56 | Gianlu | Fender: ok. but what AMI command is using? SipShowPeers doesnt give those data. |
| 16:31.26 | [TK]D-Fender | Gianlu: What data? |
| 16:31.48 | p3nguin | I doubt it uses magic -- the peer/extension was configured from inside FOP, so it wrote the information to a database. |
| 16:32.27 | p3nguin | That's my guess; I don't use FOP to know for sure. |
| 16:32.35 | Gianlu | look at the link I sent, it is an image, a screenshot taken from FOP2. They show the "extensions" list, even the "unused" ones for example. |
| 16:32.48 | E-bola | I'm not sure what the question really, is. I've used both fop and fop2 |
| 16:32.52 | p3nguin | If you have a system configured already and you toss FOP on it, does it know that info? |
| 16:33.00 | Gianlu | ok, gotcha. so p3nguin, you say that the matches have been previously configured both. |
| 16:33.35 | [TK]D-Fender | Gianlu: that looks like UNREGISTERED. I presume this is what is meant by "unused" |
| 16:33.37 | Gianlu | e-bola: do the require some initial configuration? |
| 16:33.49 | E-bola | yes? |
| 16:33.54 | E-bola | have u read the docs? |
| 16:34.28 | p3nguin | I have to assume it operates similar to FreePBX, in that the associations are created automatically when you add an "extension." |
| 16:34.29 | Gianlu | I went throught a lot of docs lately, as I said I am quite of a newbie and I am trying to catch up with all these info. |
| 16:34.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 16:34.59 | marksaitis | any clues why I cant call from 7001 to 7002? http://pastebin.com/mRjWUp3f |
| 16:35.17 | p3nguin | If you go in and create a new "extension," it will create the SIP device and the extension. |
| 16:35.53 | [TK]D-Fender | marksaitis: No, because we don't see the actual SIP DEBUG to see what was actually called (if anything) and what the reposnse might have been |
| 16:36.31 | p3nguin | scroll down |
| 16:36.32 | p3nguin | # |
| 16:36.32 | p3nguin | From: "Redc" <sip:7001@192.168.3.199>;tag=A7hbOi2TQJ9wUra-a6xGc60RTrL9Q4HX |
| 16:36.32 | p3nguin | # |
| 16:36.33 | p3nguin | To: sip:7002@192.168.3.199; |
| 16:36.58 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 16:38.27 | marksaitis | [TK]D-Fender, in that lastest pastebin I have enabled sip debug |
| 16:38.28 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 16:38.28 | marksaitis | ;] |
| 16:39.02 | [TK]D-Fender | marksaitis: http://pastebin.com/mRjWUp3f <-- this has debug for 7001, not 7002 |
| 16:39.08 | Gianlu | guys, thank you all for the help. |
| 16:39.13 | [TK]D-Fender | marksaitis: The guy yuo are CALLING is the problem |
| 16:39.31 | Gianlu | bye bye |
| 16:40.41 | marksaitis | [TK]D-Fender, well................. ur saying guy 7002 is the problem? If I call from 7002 to 7001 - the same shit |
| 16:40.44 | marksaitis | exactly the same shit |
| 16:40.55 | *** join/#asterisk ThoMe (tm@tm.muc.de) |
| 16:40.57 | ThoMe | helloh |
| 16:41.04 | marksaitis | ;] |
| 16:41.07 | [TK]D-Fender | marksaitis: I don't see debug for the one you are CALLING. |
| 16:41.37 | [TK]D-Fender | marksaitis: And if there is none to be had you'd better be looking at the PEER STATUS |
| 16:41.38 | ThoMe | I use asterisk 1.4 and would try the queue-function. i have 5 members in the queue but the members have a forward to a external number (isdn) |
| 16:41.59 | ThoMe | is it posible a queue with sip-offline user but i have this sip-id(number) forward to an external number? |
| 16:42.16 | marksaitis | [TK]D-Fender, they are both connected, BOTH |
| 16:42.39 | [TK]D-Fender | marksaitis: PASTEBIN <- |
| 16:42.40 | marksaitis | Name/username Host Dyn Forcerport ACL Port Status |
| 16:42.41 | marksaitis | 7001/7001 192.168.3.98 D N A 52887 OK (213 ms) |
| 16:42.41 | marksaitis | 7002/7002 84.46.243.207 D A 17753 OK (422 ms) |
| 16:42.41 | marksaitis | 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 16:43.08 | marksaitis | I can call from 7001 to 7002 and I cant call from 7002 to 7001 |
| 16:44.04 | *** join/#asterisk Tim_Toady (~moi@178.128.10.210.dsl.dyn.forthnet.gr) |
| 16:45.00 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 16:45.06 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 16:45.06 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 16:45.22 | drmessano | marksaitis, arent you using FreePBX and are trying to get support for your complex FreePBX dialplan in here? |
| 16:50.25 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 16:52.20 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 16:56.14 | *** join/#asterisk prank (~prank@97-81-251-38.dhcp.leds.al.charter.com) |
| 17:00.47 | *** join/#asterisk citywok (~Andrew@70.35.113.66) |
| 17:01.44 | citywok | anybody have any idea on this one? |
| 17:01.45 | citywok | [Oct 25 09:58:36] WARNING[22556]: res_calendar_ews.c:530 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://mail.mycompany.com/ews/Exchange.asmx': Could not read status line: connection timed out |
| 17:03.58 | eternis | it's ok to kill asterisk with killall -9? |
| 17:04.21 | eternis | I got into its cli and checked it out. |
| 17:05.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:08.10 | titter | [TK]D-Fender: Is it Set(CHANNEL(musicclass)=whatever)? |
| 17:08.51 | [TK]D-Fender | titter: Depends on your version. GO LOOK. |
| 17:09.31 | *** join/#asterisk citywok (~Andrew@173-114-145-37.pools.spcsdns.net) |
| 17:09.41 | titter | [TK]D-Fender: nvm, I am using a Dial to an IAX I need to set it there ... brain fart. Thanks for the suggestion, working now |
| 17:20.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:25.12 | *** join/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 17:26.46 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:27.58 | *** join/#asterisk dacm_work (~dan@host109-156-125-175.range109-156.btcentralplus.com) |
| 17:28.06 | dacm_work | Hi guys. |
| 17:29.11 | dacm_work | If I'm running asterisk as `asterisk -vvvgc' is there a way to start running it as a server instead and still access that output in realtime? Perhaps using the asterisk-manager thingy? |
| 17:29.56 | fullstop | dacm_work: asterisk -rvvvgc |
| 17:30.21 | ChannelZ | you generally run it as a server (no 'c') and then asterisk -r to connect to the running process |
| 17:30.46 | dacm_work | ah lovely |
| 17:30.55 | dacm_work | That's nice and easy. |
| 17:31.04 | dacm_work | fullstop, ChannelZ: Thank you. |
| 17:31.23 | ChannelZ | yup |
| 17:31.28 | *** join/#asterisk nwidger (~nwidger@steerpike.iol.unh.edu) |
| 17:32.43 | dacm_work | Hmm actually that isn't working for me. |
| 17:32.46 | dacm_work | Says: |
| 17:32.47 | dacm_work | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 17:33.00 | dacm_work | That file does seem to exist. |
| 17:33.16 | dacm_work | Don't suppose any one here has any ideas? |
| 17:34.00 | dacm_work | SELinux |
| 17:34.03 | dacm_work | Ignore me. |
| 17:34.06 | dacm_work | Bloody thing. |
| 17:35.32 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 17:35.34 | dacm_work | Oh god I hate SELinux. This doesn't even look easy to fix. (Unless I turn it off!) |
| 17:38.58 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:39.53 | jdoe | dacm_work: shrug. Find a policy or turn it off. Or install asterisk somewhere selinux doesn't care about (/opt works) |
| 17:39.54 | p3nguin | Creating rules for SElinux isn't really hard. It does take time and patience to create each rule, though. |
| 17:41.08 | jdoe | never been a big fan, personally. |
| 17:41.12 | dacm_work | Well maybe I'll teach myself how to create a policy then. I don't see why the damn thing can't just work though. |
| 17:41.28 | jdoe | good starting point :P |
| 17:43.35 | *** join/#asterisk SirThomas (~tomc@mail.kendeco.com) |
| 17:43.50 | p3nguin | audit2allow will help you. |
| 17:45.37 | dacm_work | reboots to see if it magically fixes selinux. (It has in the past...) |
| 17:46.20 | p3nguin | Must be a Windows admin. |
| 17:46.32 | titter | http://pastebin.com/7aD5LLm4 |
| 17:46.58 | titter | Is musicclass not working because it is setting the SIP channel and not the IAX channel? |
| 17:48.22 | *** join/#asterisk hesco (~hesco@c-76-97-185-49.hsd1.ga.comcast.net) |
| 17:48.24 | ChannelZ | well yes the musicclass would be on the SIP channel |
| 17:49.03 | ChannelZ | but whose putting who on hold? |
| 17:49.06 | titter | I am |
| 17:49.11 | titter | I am the calling parting |
| 17:49.48 | titter | What happens is people will be on a conference, and answer an incoming call which turns on MOH and pisses off the rest of the conference lol |
| 17:50.07 | titter | So my idea was to create a dummy moh class, and set that on the calls |
| 17:50.12 | hesco | curious to know why this line: exten => 7115,n,Set(CDR(userfield)=${RECIPIENT}-${userfield}-${DIALSTATUS}-${CAUSECODE}) would not result in data in my cdr table. Can anyone here please suggest somewhere I ought to poke about to fix this, please? |
| 17:52.02 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 17:52.18 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 17:53.11 | titter | ChannelZ: http://pastebin.com/5DfctBkk this doesn't work either, looks like it is setting the class for that IAX channel now correct? |
| 17:53.58 | p3nguin | Started music on hold, class 'default', on IAX2/medfordro-17539 |
| 17:54.13 | titter | correct |
| 17:54.14 | ChannelZ | it might be the MOH class isn't loading |
| 17:54.20 | ChannelZ | and defaulting back to default |
| 17:54.44 | p3nguin | moh show classes |
| 17:54.45 | titter | It says it didn't find files since its set to a null dir ... that might be it |
| 17:54.51 | ChannelZ | you will probably have to have a file of silence |
| 17:54.59 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 17:55.06 | titter | Bingo |
| 17:55.11 | titter | Thanks guys |
| 17:56.13 | TobSnyder | hello |
| 17:56.21 | TobSnyder | how to enable monitor |
| 17:56.25 | TobSnyder | having an active call |
| 17:56.36 | p3nguin | I'd be surprised if you can't disable moh on a per-call basis. |
| 17:56.39 | ChannelZ | ChanSpy |
| 17:58.10 | *** join/#asterisk drbrown (~drbrown@rrcs-24-172-144-11.central.biz.rr.com) |
| 17:59.53 | fullstop | is there any way to validate a dialplan syntax without actually loading it and making a test call? |
| 18:00.10 | p3nguin | dialplan show |
| 18:00.22 | p3nguin | You do have to load it, though. |
| 18:00.24 | fullstop | p3nguin: without loading it into asterisk |
| 18:00.29 | fullstop | p3nguin: okay.. too risky |
| 18:00.36 | p3nguin | Learn syntax, read it. |
| 18:00.55 | fullstop | p3nguin: it would still be nice to be able to validate |
| 18:01.13 | fullstop | p3nguin: Despite being skinny, I do have fat fingers from time to time. |
| 18:01.41 | p3nguin | You can always hire a consultant that knows dialplan syntax. |
| 18:01.53 | fullstop | p3nguin: You are missing the point |
| 18:02.33 | ChannelZ | The entire dialplan shouldn't fail if one extension is wrong |
| 18:02.46 | titter | fullstop: Setup a test box |
| 18:03.02 | p3nguin | Only if it blows up pbx_config would it be a problem. |
| 18:03.12 | fullstop | titter: that's the next step. It would be nice to be able to validate, though. |
| 18:03.30 | fullstop | p3nguin: you are still missing the point |
| 18:05.20 | *** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 18:05.42 | Get_The_Fish | is there any way to get the channel that the Queue application connected a call to? |
| 18:06.46 | *** join/#asterisk GameGamer43|Mac (~GameGamer@66.87.1.91) |
| 18:08.15 | titter | p3nguin: ChannelZ: http://pastebin.com/NdNuu93P -- this works instead of creating a blank audio file |
| 18:08.34 | titter | custom doesn't require a dir, just an application |
| 18:09.14 | fullstop | Setting GROUP()=${EXTEN}... what is the scope of this? |
| 18:09.25 | fullstop | Only within a context? Global? |
| 18:11.30 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 18:13.33 | Get_The_Fish | Does anyone have a method of getting the channel that a queue connected a call to? |
| 18:14.20 | fullstop | Get_The_Fish: You mean which queue member picked up the call? |
| 18:15.28 | Get_The_Fish | sorry, yes, that is correct |
| 18:17.01 | marksaitis | titter, can u call from one sip phone to another? |
| 18:17.06 | marksaitis | do they ring and can u hear them |
| 18:17.16 | titter | Ya |
| 18:17.47 | marksaitis | good for u |
| 18:17.48 | marksaitis | ;] |
| 18:18.00 | eternis | for VoIP am I looking for SIP or IAX? |
| 18:18.01 | fullstop | Get_The_Fish: I am curious.. what will be evaluating this information? |
| 18:18.27 | titter | marksaitis: I would hope so, I have about 600 SIP phones I administer |
| 18:18.39 | fullstop | Get_The_Fish: that is, once an agent picks up, the channel is still on the "Queue" line in the context. |
| 18:18.42 | WIMPy | eternis: For example. |
| 18:18.48 | marksaitis | titter, I bet u tested havent u |
| 18:19.18 | titter | marksaitis: Of course |
| 18:19.25 | fullstop | eternis: I use IAX to connect between asterisk servers and SIP to go to the phones themselves. |
| 18:19.28 | Get_The_Fish | fullstop: will be using this to determine the "username" (which is the voicemail mailbox in our case) that the call was sent to for the CDR's. |
| 18:19.52 | marksaitis | titter, on mines it doesnt work ;] |
| 18:20.00 | marksaitis | I cant ring another phone |
| 18:20.15 | fullstop | marksaitis: same network or different networks and firewalls are involved? |
| 18:20.32 | titter | marksaitis: Then your dialplan is messed up, or your phones have a NAT/registration issue |
| 18:20.40 | eternis | aha |
| 18:20.53 | fullstop | Get_The_Fish: That information may be exposed through AMI.. |
| 18:20.54 | marksaitis | fullstop, one test sip is inside nat and another one is outside nat ;] |
| 18:21.03 | p3nguin | eternis: epiphany? |
| 18:21.05 | marksaitis | bot can call to 100 test phone and hear pbx voice |
| 18:21.18 | titter | marksaitis: sip show peers do all phones show as registered? |
| 18:21.19 | marksaitis | but cant ring each other as systems says user is offline |
| 18:21.25 | Get_The_Fish | fullstop: hmmm, yes it would wouldn't. That doesnt do me much good. |
| 18:21.29 | fullstop | marksaitis: which one has problems, or is it both? |
| 18:21.47 | marksaitis | titter, yes they do really show as online and registered :) |
| 18:21.55 | fullstop | Get_The_Fish: would it be possible to have the queue time out and send to a generic voice mail box? |
| 18:22.09 | marksaitis | fullstop, both ;]]]] |
| 18:22.14 | Get_The_Fish | fullstop no it wouldnt |
| 18:22.31 | titter | marksaitis: Add qualify=yes to your sip.conf for those users, do a sip reload, and then another sip show peers ... tell me if any show as unreachable |
| 18:22.44 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-36.telkomadsl.co.za) |
| 18:23.12 | fullstop | Get_The_Fish: I suppose that I don't exactly see the big picture.. You wish to know which agent took the call, even if it goes to VM? |
| 18:23.19 | fullstop | and each agent has VM? |
| 18:23.32 | marksaitis | titter ok ;] |
| 18:23.36 | p3nguin | If an agent takes the call, it didn't go to VM. |
| 18:23.51 | fullstop | p3nguin: it depends how the queue members are set up |
| 18:23.54 | eternis | p3nguin: is there a command line SIP client? |
| 18:24.56 | Get_The_Fish | fullstop, no sorry, I just wish to know which agent took the call. Disregard the mailbox bits, it's immaterial to you |
| 18:25.02 | marksaitis | titter, shit, it changed |
| 18:25.05 | p3nguin | Actually it doesn't. If app_queue gives the call to a member who is a logged in agent, the call is connected to the member not to voicemail. |
| 18:25.22 | titter | marksaitis: then you have a NAT issue |
| 18:25.38 | titter | marksaitis: you are using TLS right? Try opening tcp 5061 |
| 18:26.04 | marksaitis | yeah tcp 5061 is open :) |
| 18:26.21 | marksaitis | as I said they both connect fine and can do test call |
| 18:26.23 | titter | marksaitis: what kind of firewall? is the centos firewall disabled? |
| 18:26.24 | marksaitis | mhmhm |
| 18:26.26 | fullstop | p3nguin: It depends on how your agents are set up. If they are set up where they are always on, listening to music on hold, then yes. |
| 18:26.33 | marksaitis | I will try to put both of them outside nat |
| 18:26.36 | marksaitis | and try again |
| 18:26.41 | titter | marksaitis: it changed to unreachable correct? |
| 18:26.51 | fullstop | p3nguin: If they are dynamically added, they can do whatever you want. |
| 18:26.55 | marksaitis | 8001 (Unspecified) D N A 0 UNKNOWN |
| 18:26.55 | marksaitis | 8002/8002 84.46.243.207 D N A 18327 OK (316 ms) |
| 18:26.56 | fullstop | including voicemail |
| 18:27.00 | marksaitis | thats what it changed to |
| 18:27.11 | titter | Then 8001 isn't registered |
| 18:27.17 | *** join/#asterisk dacm_work (~dan@host109-156-125-175.range109-156.btcentralplus.com) |
| 18:27.33 | titter | That client needs to be correct |
| 18:27.34 | titter | ed |
| 18:27.46 | eternis | epiphany is a webbrowser |
| 18:27.48 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 18:28.26 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 18:28.27 | marksaitis | titter, trust me it is connected ;] im making calls from it etc... |
| 18:28.54 | titter | marksaitis: Asterisk doesn't know that, it doesn't see it |
| 18:29.06 | fullstop | marksaitis: watch your console and see if you see messages such as "8001 is LAGGED" or "8001 is now UNREACHABLE" |
| 18:29.24 | fullstop | Your connection may have too much latency |
| 18:29.37 | titter | Do another sip show peers and see if that message has changed |
| 18:29.52 | marksaitis | same |
| 18:32.41 | fullstop | Get_The_Fish: My queues are set up as local channels... and agents are logged in from other asterisk systems. The extensions are numbered 25XX, so I have the local channel dial Q25XX so that the remote system knows that it comes from a queue.. and the CDR records show who answered the call from that. |
| 18:34.19 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 18:35.03 | Get_The_Fish | very interesting. So what do the CDR's look like? Are there multiple entries per call with that setup, or just one? |
| 18:35.53 | Get_The_Fish | fullstop: I would love to see your dialplan code if you wouldn't mind, get some ideas, because that might work for me... |
| 18:36.08 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 18:39.13 | fullstop | Get_The_Fish: In my setup, I unfortunately, get a CDR record for agents who do not pick up, because the call is answered by the remote asterisk server. |
| 18:39.18 | eternis | err.. bindaddr exists in 1.6? I just see tcpbindaddr=0.0.0.0 |
| 18:39.40 | Qwell | eternis: it exists. it's documented in 1.6 as udpbindaddr, I believe |
| 18:39.46 | Qwell | though bindaddr works just fine |
| 18:40.21 | Get_The_Fish | ah, I see.... I wonder if this will still work that way. I am actually thinking of using some shared variables. I think that I may submit that as a feature request- I am a little surprised no one has asked about this before or requested t |
| 18:40.40 | fullstop | Get_The_Fish: http://pastebin.com/vgQBaT88 |
| 18:41.09 | fullstop | Get_The_Fish: The queue logging in and out was adapted straight from one of the docs in the doc directory in the asterisk source tree. |
| 18:41.16 | eternis | ok |
| 18:41.48 | p3nguin | How could that dialplan possibly work with those spaces in it? |
| 18:42.26 | fullstop | p3nguin: is that directed at me? |
| 18:42.39 | p3nguin | Not necessarily. |
| 18:42.49 | p3nguin | It's regarding your dialplan, though. |
| 18:42.54 | fullstop | p3nguin: The spaces are not really there. |
| 18:43.03 | Get_The_Fish | p3nguin, are you talking about fullstop's dialplan? If you think that's bad, you should see mine |
| 18:43.10 | marksaitis | Another strange thing is that if I am in a call with 8002 phone, and try to call it from 8001 phone, PBX says "user is busy". But if 8002 is not busy, pbx says "user is not available" :D and its vice versa |
| 18:43.11 | fullstop | p3nguin: it is how the copy + paste works with my terminal emulator |
| 18:43.15 | p3nguin | Interesting. Virtual spaces. |
| 18:43.18 | eternis | so bindport and bindaddr are unrelated to tcpbindaddr? also I see that defaults to port 5060 so no need to specify port, I guess |
| 18:43.28 | fullstop | p3nguin: yeah, it's annoying actually. |
| 18:43.37 | p3nguin | I'd imagine. |
| 18:49.10 | marksaitis | how to simulate a call and ring a siphone in CLI? |
| 18:49.12 | marksaitis | anybody? |
| 18:49.13 | marksaitis | :) |
| 18:49.48 | WIMPy | Channel originate |
| 18:50.39 | marksaitis | WIMPy, lets say I want to ring 8002 |
| 18:50.43 | marksaitis | what do I type in? |
| 18:50.59 | fullstop | marksaitis: AMI, or a .call file with the spool app |
| 18:51.02 | WIMPy | type just that and read what it says |
| 18:51.18 | p3nguin | It's a lot less work to use the originate command on the CLI. |
| 18:51.30 | marksaitis | ? |
| 18:52.09 | eternis | in bindaddr which IP should I use? the wireless or the ethernet? |
| 18:52.21 | eternis | the wireless is the one connected to internet |
| 18:52.30 | eternis | ethernet is just doing some NFS |
| 18:52.37 | WIMPy | eternis: The one you want to use. |
| 18:52.37 | p3nguin | eternis: Is there any reason you wouldn't use both addresses (all addresses = 0.0.0.0)? |
| 18:52.50 | WIMPy | Or that |
| 18:52.59 | brad_mssw | in the CLI, when I do a 'sip show channels' I have 3 channels to my sip provider that say Last Message: "Rx: BYE" ... as far as I can tell, they've been like that for days |
| 18:53.08 | marksaitis | channel originate/originate help text doesnt ring any bells to me whatsoever |
| 18:53.08 | brad_mssw | how can I kill them? |
| 18:53.31 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:54.00 | eternis | ok |
| 18:54.23 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 18:56.55 | *** join/#asterisk amarz (~chatzilla@41.232.211.19) |
| 18:57.21 | brad_mssw | 'channel request hangup' doesn't appear to actually destroy the channel |
| 18:59.20 | amarz | Hello, I am trying to use Voicetronix openline4 with asterisk, I have setup the card and the drivers correctly and can run all the test , however when I load the driver chan_vpb.so asterisk tells me that "No Voicetronix cards detected" has anyone faced anything like that |
| 19:00.15 | *** join/#asterisk coppice (~chatzilla@122.177.46.85) |
| 19:00.50 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 19:03.28 | eternis | WTFF!! the book has a m$$ program as an example!! |
| 19:03.30 | eternis | :( |
| 19:04.02 | WIMPy | o.O |
| 19:04.07 | eternis | I am at 'Configuring the SIP devices' |
| 19:04.15 | eternis | ok I need a soft-phone |
| 19:04.30 | fullstop | zoiper |
| 19:05.20 | eternis | mm.. is not on my repos |
| 19:05.25 | eternis | any other one? |
| 19:05.58 | WIMPy | ekiga |
| 19:11.57 | *** join/#asterisk fofware (~Fabian@host174.190-31-223.telecom.net.ar) |
| 19:12.21 | *** join/#asterisk Abel408 (429863dc@gateway/web/freenode/ip.66.152.99.220) |
| 19:14.09 | Abel408 | Hey everyone. I'm having a ringback problem on sip calls. I have progressinband=yes, but when I view the sip debug logs during an active call I never get "183 Session Progress". Any reason why asterisk doesn't seem to be taking my changes? I issued a sip reload command. Would I have to do anything else? |
| 19:14.22 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 19:15.03 | *** join/#asterisk mahables (~Jason@208.86.215.10) |
| 19:15.33 | mahables | hi guys, does anyone know of a SIP Trunk provider that supports g.722? |
| 19:16.14 | Qwell | Probably none... It makes no sense to. |
| 19:16.22 | Qwell | Think about it. Once it gets to your provider, what happens? |
| 19:16.52 | Qwell | (hint: It gets sent over a T1 to the PSTN, which is 8kHz) |
| 19:17.42 | p3nguin | heh |
| 19:17.48 | marksaitis | can any expert spot why I cant call other phones? http://pastebin.com/Fnkkwhxw |
| 19:17.57 | mahables | haha true... but I was kinda hoping that there was a colony of wdeband enthusiasts out there somewhere |
| 19:18.28 | p3nguin | I think that colony uses an internal network for calling each other. |
| 19:18.54 | fullstop | a colony of wide enthusiasts does not sound fun |
| 19:18.55 | mahables | LOL |
| 19:19.34 | p3nguin | Once you hit the PSTN, all the fun ends. |
| 19:19.43 | paulc | SetLanguage() is deprecated, in favour of Set(CHANNEL(language)=blah), right? |
| 19:20.49 | WIMPy | Qwell: PRI can do G722 as well. |
| 19:21.10 | WIMPy | Unfortunaletly, dahdi can't. |
| 19:21.28 | p3nguin | What happens to it once your call hits the telco's switches at the other end of your PRI? |
| 19:21.29 | WIMPy | It's everything but common, however. |
| 19:21.49 | p3nguin | You're back down to narrowband again. |
| 19:22.06 | WIMPy | Depends on the calld partys equipment. |
| 19:24.26 | jkroon | hi guys, all the DUNDi examples I can locate uses RSA to authenticate, is there any other options available? |
| 19:25.05 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 19:27.00 | mahables | I understand how the PSTN will be 8k, but sip to sip could be wideband if the callee's sip provider suports g.722 right? |
| 19:28.25 | p3nguin | Once your call gets to your ITSP, they stick the call onto the PSTN. End of the line for G.722. |
| 19:29.46 | mahables | :-( |
| 19:30.23 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 19:31.38 | muiro | In a Realtime Queue, can I point to a local interface and use an IAX2 interface for status, like in queues.conf? (ala Local/XXX@YYY,0,Name,IAX2/xxx) |
| 19:33.13 | WIMPy | mahables: It would make sense. Like a lot of other things that aren't done. Ask some providers why they don't offe the service, you'd like to see. |
| 19:35.03 | coppice | sadly, even where BRI is common, support for G.722 isn't |
| 19:35.25 | WIMPy | True |
| 19:35.48 | WIMPy | Asterisk _could_ make a difference there, I'd guess. |
| 19:36.36 | coldsteal | can sln be used for music on hold? |
| 19:37.07 | WIMPy | Unfortunaletly phones supporting G.722 are more than rare. |
| 19:37.36 | WIMPy | Not IP phones, obviousely. |
| 19:39.29 | WIMPy | has just been told, that some DECT phones do support G.722. |
| 19:39.51 | puzzled | iirc some Siemens Giga... support it |
| 19:40.59 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 19:42.02 | coppice | I'm not sure if those SIemens DECT phones with G.722 actually support it over the PSTN |
| 19:42.06 | Get_The_Fish | fullstop: ${MEMBERINTERFACE} and ${MEMBERNAME} become available when you add the option setinterfacevar=yes to the queue in question in queues.conf |
| 19:42.42 | WIMPy | coppice: yes |
| 19:45.53 | p3nguin | Hmm, Lowe's doesn't have DSL in-line filters? Amazing! |
| 19:46.27 | fullstop | Get_The_Fish: Good to know. I still need the _Q25XX so that I can tell it to ring differently for calls coming to the support queue. |
| 19:46.27 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:46.57 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-phhjbnzcdwupdmsr) |
| 19:47.09 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 19:47.37 | marksaitis | titter |
| 19:47.41 | marksaitis | u there? |
| 19:47.44 | marksaitis | I need your help |
| 19:47.45 | fullstop | p3nguin: my grandma was able to somehow get DSL while I was still stuck on dial up (in 2003). She was having very poor performance, so I called up tech support for her. |
| 19:47.45 | marksaitis | a lot |
| 19:47.47 | marksaitis | ;] |
| 19:47.50 | marksaitis | titter |
| 19:47.53 | titter | ? |
| 19:48.06 | marksaitis | u remember that script yesterday |
| 19:48.10 | marksaitis | to generate certs |
| 19:48.10 | titter | Yes |
| 19:48.18 | marksaitis | specifically for asterisk18 |
| 19:48.26 | fullstop | She lived in rural PA, and I had to tell the support person "Hang on, I have to check if there is a filter on the line out in the barn" |
| 19:48.28 | titter | http://svn.digium.com/svn/asterisk/branches/1.8/contrib/scripts/ast_tls_cert |
| 19:48.41 | marksaitis | when u launched it, did you provide a password for it? |
| 19:48.49 | titter | yes |
| 19:49.25 | marksaitis | where else do u use that password? |
| 19:49.44 | titter | No where that I remember |
| 19:49.56 | p3nguin | waits for the conclusion |
| 19:50.11 | fullstop | As it turned out, there was a filter missing on a line she didn't know about in the basement. |
| 19:50.47 | fullstop | However, I felt quite strange having to tell someone that I had to check a line in the barn. Especially since I live in a more urban area and still couldn't get DSL. |
| 19:50.49 | marksaitis | titter ok |
| 19:50.56 | p3nguin | Missing filters shouldn't cause bad DSL/Internet connection performance. |
| 19:51.09 | p3nguin | Missing filters just make phones sound really bad. |
| 19:51.11 | fullstop | It was connected to a direct TV receiver.. |
| 19:51.26 | fullstop | and, after removing it, the connection started working perfectly. |
| 19:51.45 | marksaitis | titter, if I use that script, it wouldnt work for me strangely enough |
| 19:52.23 | titter | I imported the ca.crt to my phone, and it worked |
| 19:52.43 | titter | Make sure you cp the certificate.pem I think it was to your hostname .pem |
| 19:52.52 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 19:53.07 | p3nguin | I was just trying to figure out how I was going to get some DSL filters by 8:30 tomorrow morning. First thought was to stop at Lowe's, but their web site doesn't indicate that they have any. The Home Depot also doesn't show any. |
| 19:53.41 | fullstop | Their online presence is not the same as the store. They can (and usually do) stock different items. |
| 19:53.50 | fullstop | Give them a call and ask. |
| 19:53.53 | [TK]D-Fender | p3nguin: RadioScrap |
| 19:54.06 | [TK]D-Fender | RadioShack : You'ev got questions .. we've got batteries |
| 19:54.08 | fullstop | They will ask you for your phone number |
| 19:54.26 | fullstop | The last time I was in there, they tried to sell me a cell phone plan, even though I was not interested in the least. |
| 19:54.44 | muiro | In a Realtime Queue, I know I can use Local/ for the interface, but can I use another interface for devic status, like in queues.conf? (ala Local/XXX@YYY,0,Name,IAX2/xxx) |
| 19:54.59 | p3nguin | I probably won't even need the filters, but I need to take them along just in case. You know how people expect you to carry every possible component along with you. |
| 19:55.46 | titter | p3nguin: Half the time its not even the people, but the fact you go why in the world would I need this? Then leave it behind, and it turns out the moon and mars aligned and now you need it. |
| 19:57.15 | Abel408 | Hey everyone. I'm having a ringback problem on sip calls. I have progressinband=yes, but when I view the sip debug logs during an active call I never get "183 Session Progress". Any reason why asterisk doesn't seem to be taking my changes? I issued a sip reload command. Would I have to do anything else? |
| 19:57.16 | p3nguin | "What do you mean you don't have a spare Catalyst 4507 chassis in your car?!" |
| 19:57.38 | titter | lol |
| 19:58.15 | *** part/#asterisk mahables (~Jason@208.86.215.10) |
| 19:58.23 | fullstop | http://www.radioshack.com/product/index.jsp?productId=2103593 |
| 19:59.10 | fullstop | http://www.staples.com/Staples-DSL-Filter/product_837569?cm_mmc=GoogleBase-_-Shopping-_-Technology%3ECables_%26_Hubs-_-837569-18767&cid=CSE:GoogleBase:Technology:Cables_%26_Hubs:837569:18767 |
| 19:59.20 | fullstop | $10 at staples.. |
| 19:59.21 | p3nguin | I might not have a huge core switch chassis in my car, but I do have a 9mm for people who expect me to have a huge core switch chassis in my car. :D |
| 19:59.37 | drmessano | OMG |
| 19:59.38 | fullstop | a 9mm screw driver |
| 19:59.47 | p3nguin | Staples is next door to Lowe's so I'll have to stop off there. |
| 19:59.48 | drmessano | Why nto buy a DSL filter off of ebay for 99 cents? |
| 19:59.52 | drmessano | $20? |
| 20:00.01 | fullstop | drmessano: he needs it by 8:30 AM tomorrow |
| 20:00.08 | p3nguin | I need a handful of them by the start of business tomorrow. |
| 20:00.12 | drmessano | Ouch |
| 20:00.46 | p3nguin | If people would send purchase orders during the same WEEK that they ask for work to be done, this wouldn't be a problem. |
| 20:01.02 | *** join/#asterisk stephenfranks (~stephenfr@79-79-184-129.dynamic.dsl.as9105.com) |
| 20:01.09 | p3nguin | Now they'll pay a premium if the parts are required. |
| 20:01.13 | drmessano | I also install one of the outdoor DSL filters and split the line.. I hate those damn inline filters |
| 20:01.23 | drmessano | s/also/always/ |
| 20:01.44 | p3nguin | I do like the "whole house" filters when it is possible to use them. |
| 20:02.00 | stephenfranks | Hi does anyone know if a guy called Bananaskin hangs around here? |
| 20:02.37 | drmessano | ~seen bananaskin |
| 20:02.38 | infobot | bananaskin <n=mike@user-514d4e32.l1.c1.dsl.pol.co.uk> was last seen on IRC in channel #asterisk, 673d 3h 49m 14s ago, saying: 'interesting fact, the Clarke Belt is named after him'. |
| 20:02.38 | drmessano | No clue what the trigger is |
| 20:02.43 | drmessano | oh |
| 20:02.47 | drmessano | slow bot |
| 20:02.49 | fullstop | been a while |
| 20:03.08 | drmessano | 673 days.. I would guess "no" |
| 20:03.17 | stephenfranks | Thanks - that was a long time ago!! |
| 20:03.27 | stephenfranks | Appreciate your help guys |
| 20:04.53 | p3nguin | Good grief. |
| 20:05.22 | p3nguin | Why would the demarc need to be extended if there is already an existing DSL installation? |
| 20:05.29 | muiro | Can anyone here assist me with Realtime Queue syntax for the Interface? I can't seem to get it to recognize a status interface |
| 20:06.27 | muiro | I've gone through all the documentation I can find but I'm not seeing any reference to this |
| 20:07.44 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 20:08.38 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 20:10.37 | eternis | what's the REGISTAR field in ekiga when setting up a SIP account? |
| 20:11.05 | eternis | I mean what option in sip.conf corresponds to? |
| 20:11.25 | p3nguin | The registrar is the address/name of Asterisk. |
| 20:11.46 | p3nguin | IP address or host name |
| 20:12.53 | eternis | thanks |
| 20:14.39 | eternis | 'Could not register (Method not allowed)' |
| 20:14.59 | eternis | when I put the ip in the registar field |
| 20:15.07 | p3nguin | I wonder what method it could possibly use that isn't allowed. |
| 20:16.14 | marksaitis | im done today with this peace of crap |
| 20:17.02 | eternis | p3nguin: when testing everything on a single machine, the IP's should be the same everywhere? in sip.conf and iax.conf? |
| 20:17.25 | p3nguin | You've got both Asterisk and ekiga on the same computer? |
| 20:17.27 | eternis | maybe because I didn't start asterisk I get that error |
| 20:17.34 | p3nguin | haha |
| 20:17.35 | eternis | p3nguin: yes |
| 20:17.43 | eternis | wrong? |
| 20:17.53 | p3nguin | Asterisk certainly needs to be running to use it. |
| 20:17.59 | eternis | ok |
| 20:19.23 | eternis | right I did it wrong |
| 20:20.28 | drmessano | that's right |
| 20:20.29 | drmessano | I mean, wrong |
| 20:22.02 | *** join/#asterisk Niklas- (~niklas@217.116.253.195) |
| 20:23.02 | Niklas- | Hey. Is there a variable / function that returns the number of current active channels? Similar to the count from 'core show channels'. |
| 20:25.29 | p3nguin | I guess you could always do something like System(asterisk -rx "core show channels"|grep "active channels") |
| 20:28.06 | Abel408 | How can I get asterisk to send sip "183 Session Progress" during an incomming call? |
| 20:28.21 | p3nguin | Progress() maybe? |
| 20:28.30 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 20:28.37 | Niklas- | p3nguin: so far it seems the only option, thanks :) |
| 20:29.18 | Abel408 | Would I need to use progress() even if I have progressinband = yes? |
| 20:30.00 | Abel408 | this is for a call coming into asterisk |
| 20:30.57 | Abel408 | So I don't think Progress() would work... |
| 20:31.05 | *** part/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 20:33.09 | muiro | alright, looks like I figured out my problem. A (seemingly) undocumented field added to the member table takes care of it (state_interface) |
| 20:35.05 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 20:38.48 | eternis | I don't think everything on the same computer is doable |
| 20:38.58 | eternis | I will have to install some virtual thingy |
| 20:39.43 | eternis | vm |
| 20:42.46 | eternis | I get this message 'Could not register sip:6000@192.168.0.12' |
| 20:43.27 | jkroon | is there any way to get asterisk to reload keys (ie, rescan for new keys on disk) without doing a full restart? |
| 20:44.30 | eternis | dialplan reload? or sip reload? |
| 20:46.34 | jkroon | eternis, even "reload" doesn't get it. |
| 20:47.06 | eternis | keys reload? |
| 20:47.26 | jkroon | nope |
| 20:47.40 | jkroon | and can't unload res_crypto just to reload it either ... |
| 20:49.38 | Niklas- | jkroon: 'reload' is enough for me to reload the keys, have you checked if the key files have proper permissions? |
| 20:49.59 | shapr | jkroon: What sort of keys? license keys? |
| 20:53.03 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 20:54.45 | p3nguin | eternis: Using Asterisk and a softphone on the same computer *is* okay. Many people do it for testing; I've even done it. |
| 20:57.08 | jkroon | Niklas-, ast version? |
| 20:57.21 | jkroon | shapr, no, rsa keys. |
| 20:57.48 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:58.05 | *** join/#asterisk bcrisp (~bcrisp@wsip-184-191-141-38.ph.ph.cox.net) |
| 20:58.06 | p3nguin | eternis: I must say that I never used ekiga on the same computer as asterisk, though. Although twinkle works fine on the same computer. |
| 20:58.30 | p3nguin | eternis: Just make sure you change the sip port in twinkle to avoid the port conflict and it works fine. |
| 20:58.37 | bcrisp | what could it mean if our IVR menu responds to dtmf sent from a cell phone but not when using other phones? |
| 20:58.50 | jkroon | define "other phones" |
| 20:58.59 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 20:59.12 | bcrisp | desk phones, other voip phones |
| 20:59.38 | bcrisp | outside of our network |
| 20:59.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:00.19 | bcrisp | no debug is showing, just timeouts on the waitexten() |
| 21:00.31 | bcrisp | this problem comes and goes and am not sure what is going on |
| 21:01.00 | jkroon | bcrisp, make sure that their DTMF method matches that which asterisk is configured to. the rfc method seems to work best for most phones. |
| 21:01.13 | bcrisp | we're using rfc2833 |
| 21:01.18 | bcrisp | in sip.conf |
| 21:01.19 | *** join/#asterisk Guest66070 (~saint42@c-76-98-53-209.hsd1.pa.comcast.net) |
| 21:02.25 | bcrisp | perhaps i should try auto |
| 21:03.19 | jkroon | just get your phones to also use rfc2833. |
| 21:03.36 | bcrisp | well i don't control people calling our support line |
| 21:04.08 | [TK]D-Fender | bcrisp: And HOW are they calling you? |
| 21:04.31 | bcrisp | we use an inbound sip provider |
| 21:05.20 | Glasswalker | how functional is a raw asterisk system out of the box? like as a PBX, can I create extensions, have IVRs, make recordings, voicemail, voicemail to email, confrencing, hold music, and so on? or is it pretty raw, and I need a distro like trixbox to do that stuff? |
| 21:05.20 | [TK]D-Fender | bcrisp: then it isn't the "people calling your support line", it is your PROVIDER |
| 21:05.26 | Glasswalker | (or write the functionality myself) |
| 21:05.28 | bcrisp | ok |
| 21:05.35 | *** join/#asterisk [cannibalera] (~cannibale@200.193.14.217) |
| 21:05.35 | p3nguin | ~toolkit |
| 21:05.36 | infobot | Remember, Asterisk isn't really a PBX: Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 21:05.48 | *** part/#asterisk [cannibalera] (~cannibale@200.193.14.217) |
| 21:06.09 | [TK]D-Fender | Glasswalker: There is no "out of the box. trixbox and other similar distros bundle an OS with packaged * and management GUI's etc. |
| 21:06.15 | p3nguin | You can run Asterisk on almost Any Linux distro. |
| 21:06.40 | Glasswalker | well we're running trixbox now |
| 21:06.45 | [TK]D-Fender | Glasswalker: As to how long it will take you to set things up, that depends on your experience when you get started, and what you need to set up |
| 21:06.56 | Glasswalker | but are running into many problems due to our environment's needs, and bugs in the versions trixbox is using |
| 21:07.12 | Glasswalker | I've tried compiling latest build of asterisk and upgrading trixbox but it broke hard |
| 21:07.12 | [TK]D-Fender | Glasswalker: Could be a few minutes, could be a few hours, could be a few days/ |
| 21:07.33 | p3nguin | ~trixbox |
| 21:07.33 | infobot | trixbox is probably SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 21:07.35 | [TK]D-Fender | Glasswalker: Did you already try upgrading them via their docs and support tools? |
| 21:07.37 | Glasswalker | we've been working on solving these problems for weeks |
| 21:07.56 | Glasswalker | yes, the latest version they have a repo package for is still old |
| 21:08.04 | Glasswalker | I need a version with a particular unistim fix |
| 21:08.20 | Glasswalker | which is in 1.8 but wasn't merged into any of the 1.6 builds that I can confirm |
| 21:08.35 | [TK]D-Fender | Glasswalker: then you're in TFB territory, and things may break depending on differences |
| 21:08.44 | Glasswalker | fair enough |
| 21:09.29 | Glasswalker | That's why I was checking what the base asterisk install does "out of box" but if it's purely a toolkit/framework then I'm probably SOL... |
| 21:09.38 | [TK]D-Fender | Glasswalker: Descibe the amount of things you've configured on our existing system. How many phones, menus, providers, various routes, etc. |
| 21:09.44 | [TK]D-Fender | your* |
| 21:09.49 | jkroon | Glasswalker, i've been doing this for more than two years now - i'm still learning. |
| 21:10.43 | Glasswalker | I'm not concerned about learning. Asterisk is not my job. I'm the IT Manager of this company, and they need to upgrade their phone system... I presented several options, and they chose asterisk for cost/benefit. |
| 21:11.05 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:11.16 | Glasswalker | probelm is my time has been taken up trying to get everything just right, in order to finish this deployment, for like the last 2 months, which has ended up being extremely costly |
| 21:11.25 | Glasswalker | so I'm trying to decide if it's time to pack it in and choose another solution |
| 21:11.27 | jkroon | Glasswalker, i can recommend running asterisk without the configuration GUIs. got rid of a LOT of my problems. |
| 21:11.28 | Glasswalker | or keep fighting with this one |
| 21:12.41 | jkroon | setting up a basic dialplan is NOT difficult. and if you use templates properly for SIP extensions and stuff it actually becomes quite easy to get things set up properly quite quickly. |
| 21:12.53 | Glasswalker | jkroon: ok, so how much work is it to configure say 100 extensions, 4 branch offices with 20 extensions each. Voicemail for everyone, a few confrence hubs, voicemail to email, and a few IVRs. Plus call routing rules for intra-office routing and outbound analog and sip trunks. |
| 21:13.22 | jkroon | creating the recordings for the IVRs is the most work. |
| 21:13.29 | Glasswalker | that's easy |
| 21:13.40 | [TK]D-Fender | Glasswalker: When you say "other solution", what are those so far? |
| 21:13.40 | Glasswalker | it's called hire a voice talent and hand them a script (already written) |
| 21:13.42 | jkroon | the rest is less than a days work if you know what you're doing. |
| 21:13.47 | jkroon | probably less if you can script a bit. |
| 21:14.11 | jkroon | it still takes time :) |
| 21:14.15 | Glasswalker | [TK]D-Fender: Avaya, Cisco |
| 21:14.33 | Glasswalker | Commercial Asterisk derivitive with paid support contract |
| 21:14.42 | Glasswalker | and implimentation services included |
| 21:14.58 | jkroon | switchvox ? |
| 21:15.06 | Glasswalker | We looked at a few. that was one of them |
| 21:15.27 | Glasswalker | The other consideration is replacing all our phones |
| 21:15.34 | Glasswalker | we have like 150 nortel IP phones |
| 21:15.45 | Glasswalker | (hence the dependance on unistim) |
| 21:15.45 | jkroon | from what i've seen switchvox is quite good, albeit serious overkill for most people. |
| 21:16.08 | Glasswalker | and unistim being an option for asterisk was one of the major factors in it's favor |
| 21:16.22 | p3nguin | its |
| 21:16.24 | Glasswalker | replacing all our IP phones with equivalent featured phones now would cost us nearly $350 per phone |
| 21:16.51 | Glasswalker | which is $50 Grand on it's own |
| 21:16.55 | [TK]D-Fender | galssHow big is your current setup? |
| 21:16.56 | p3nguin | its |
| 21:16.59 | jkroon | i'm not familiar with unistim - but surely nortel should have support for sip too ? |
| 21:17.02 | Glasswalker | 100 extensions at head office |
| 21:17.10 | Glasswalker | and about 20 extensions in 4 branch offices |
| 21:17.15 | Glasswalker | (20 per) |
| 21:17.25 | Glasswalker | jkroon: nope |
| 21:17.43 | Glasswalker | nortel went under because of their ungodly tendancy to make things proprietary |
| 21:17.46 | [TK]D-Fender | Glasswalker: Ok, since you have a GUI already I'd recommend Switching to AsteriskNOW and porting your EXISTING GUI configs over where compatible. |
| 21:18.46 | jkroon | dude, are you sure? last nortel phone i saw looked more like a rebranded cisco phone than anything else. in fact, if somebody didn't point out it was nortel to this day i would have sworn it's cisco. |
| 21:18.47 | Glasswalker | ok. Does AsteriskNOW work with 1.8? or is it a little easier to integrate hand compiled code? |
| 21:19.23 | Glasswalker | jkroon: nope, nortel built almost all of their own stuff. they were one of the largest telecom manufacturers in the world, but they just went under last year. |
| 21:19.34 | Glasswalker | however they sold out their telecom products to avaya |
| 21:19.40 | Glasswalker | the very latest nortel stuff was SIP |
| 21:19.45 | Glasswalker | but what we had was a generation earlier |
| 21:19.47 | Glasswalker | all unistim only |
| 21:20.22 | jkroon | damn. well, good luck. seeing that you won't get support on the phones getting going on replacing them might be the way to go. |
| 21:20.36 | jkroon | but it is a lot of $$$ |
| 21:20.45 | [TK]D-Fender | [17:18]<Glasswalker>ok. Does AsteriskNOW work with 1.8? or is it a little easier to integrate hand compiled code? <- YES, |
| 21:20.53 | [TK]D-Fender | Glasswalker: and its documented on their channel |
| 21:20.54 | Glasswalker | awesome |
| 21:20.58 | Glasswalker | that's what I wanted to hear |
| 21:21.16 | Glasswalker | ok, so I have one or two more things to stab at with this. if not then I'll fire up a POC AsteriskNow VM and try that. |
| 21:21.24 | [TK]D-Fender | Glasswalker: So far the latest FreePBX (native) seems to work with 1.8 |
| 21:21.31 | Glasswalker | So I've read in their forums |
| 21:21.35 | Glasswalker | but I can't make it go |
| 21:21.38 | jkroon | Glasswalker, or you possibly just backport the specific patch onto your existing setup (backup first please), seeing that it sounds like that's your only problem. |
| 21:22.10 | Glasswalker | if I compile in 1.8 it looses audio on the unistim phones, and call routing for newly created sip extensions is broken. Plus the web gui refuses to trigger a reload |
| 21:22.10 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 21:22.26 | Glasswalker | I don't know if that's the limit of the problems, but with that many, I suspected it simply wasn't integrating |
| 21:22.36 | Glasswalker | and didn't want to go on a bug hunt trying to plug leaks one at a time |
| 21:23.29 | Glasswalker | it either works or it doesn't. If it's well documented to work, and I have a todo list of items to close off, that's easy. Finite time. But if it's "Well now find X random unknown incompatibilities and problems" that's not cool. open ended. |
| 21:24.03 | Glasswalker | anyway, I have a lead on a guy in town that supposedly is using these nortel phones without problems on an asterisk based setup, I'll talk to him see if it's trixbox and if so what he's doing. If not, maybe he can help |
| 21:24.15 | eternis | now, finally! I don't have the error anymore, I had to add the port |
| 21:24.19 | eternis | in ekiga |
| 21:24.23 | Glasswalker | next I'll try asterisknow |
| 21:29.54 | eternis | what's my next step? |
| 21:30.09 | p3nguin | Place a call. |
| 21:39.24 | eternis | ahh, gotta change ports on the client |
| 21:42.02 | *** join/#asterisk jsgoecke (~Adium@12.130.118.7) |
| 21:42.10 | jsgoecke | Hello |
| 21:45.27 | Dovid | hi |
| 21:47.17 | thehar | H/win 11 |
| 21:47.19 | thehar | ail |
| 21:49.21 | Dovid | oolah |
| 21:57.00 | eternis | can reload from within asterisk? |
| 21:57.06 | eternis | if I changed a port? |
| 21:57.17 | p3nguin | Don't change the bindport in asterisk. |
| 21:57.43 | eternis | p3nguin: this one?? --> 5060 |
| 21:58.09 | p3nguin | The problem is that the sip client isn't really just a client, just like the server isn't really just a server; both user agents have roles as client/server. The phone listens on a port just like asterisk does, so you have to shift the phone's port when it operates on the same interface. |
| 21:58.15 | eternis | ekiga uses the same one, accerding to the book server client need to have differing ports |
| 21:58.28 | [TK]D-Fender | eternis: You are running a softphone on your * box? |
| 21:58.37 | eternis | [TK]D-Fender: yes |
| 21:58.46 | [TK]D-Fender | eternis: then yes, set Ekiga to use 5061 |
| 21:59.02 | eternis | ekiga doesn't have a meaningful way to change its port. |
| 21:59.15 | p3nguin | I would have already dumped ekiga and started using twinkle by now. At least I know the port setting is readily available in it. |
| 22:00.18 | p3nguin | I've never seen any setting in ekiga to change the port. |
| 22:00.24 | [TK]D-Fender | eternis: http://wiki.ekiga.org/index.php/Internet_ports_used_by_Ekiga |
| 22:00.47 | [TK]D-Fender | eternis: Took me considerably less than a minute to find |
| 22:02.14 | eternis | I was actually here --> http://wiki.ekiga.org/index.php/Enable_port_forwarding_manually |
| 22:02.17 | p3nguin | Ah, gconf editor. |
| 22:02.47 | eternis | I took this 'SIP 5000 to 5100 UDP SIP signalling, listen port: 5060 ' as ekiga has it hardcoded to 5060 |
| 22:02.49 | p3nguin | starts singing twinkle twinkle little star |
| 22:02.50 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:03.23 | p3nguin | Hard-coded, no... a bother to change because you need gconf-editor, probably. |
| 22:03.32 | eternis | unfortunately twinkle isn't in my repos |
| 22:03.41 | p3nguin | Which distro are you using? |
| 22:03.51 | eternis | Gentoo |
| 22:04.28 | p3nguin | Looks like you're doing it wrong. |
| 22:04.42 | p3nguin | net-voip/twinkle in portage |
| 22:05.37 | p3nguin | "emerge -v twinkle" does what? |
| 22:06.00 | eternis | mm.. do you have an overlay? |
| 22:06.01 | *** join/#asterisk jsgoecke (~Adium@12.130.118.7) |
| 22:06.14 | eternis | neither eix nor emerge -s find it |
| 22:06.53 | eternis | I've checked manually in /usr/portage/net-voip/ as well |
| 22:07.17 | [TK]D-Fender | eternis: just fix up ekiga now |
| 22:07.39 | p3nguin | heh, gconf-editor is too much work to launch! |
| 22:08.00 | eternis | I am now. |
| 22:17.16 | vinhdizzo | anyone have time to help me solve this call GV #, asterisks process call, but caller doesn't observe anything and goes straight to GV voicemail, issue? |
| 22:25.38 | *** join/#asterisk [cannibalera] (~cannibale@200-193-14-217.fnsce703.dsl.brasiltelecom.net.br) |
| 22:27.26 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:30.43 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:33.29 | bcrisp | didww is aweful |
| 22:41.05 | p3nguin | ~didww |
| 22:41.06 | infobot | didww is, like, awful |
| 22:41.10 | p3nguin | noted. |
| 22:51.13 | BesticlesWork | I'm having abit of a problem. I originate a call via AMI to Local, then I FastAGI out, to send a Dail on a DAHDI channel. The number I am dialing is not in service. My problem is that ${DIALSTATUS} = CANCEL. I am completly out of ideas to get the result that I need. |
| 22:51.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 22:51.52 | eternis | I am not sure if this means something but when doing the 'Echo test' in ekiga which is this number sip:500@ekiga.net I get this error --> 'Could not connect to remote host' |
| 22:53.26 | eternis | what am I doing wrong |
| 22:53.28 | [TK]D-Fender | eternis: Why are you touching ekiga.net at all? |
| 22:53.41 | p3nguin | Did you try your own local echo test first? |
| 22:53.51 | [TK]D-Fender | eternis: FIRST mistake |
| 22:53.53 | eternis | how do I perform a local echo test? |
| 22:54.05 | p3nguin | Use Echo() in your asterisk dialplan. |
| 22:54.16 | [TK]D-Fender | eternis: Set Ekiga up with YOUR server and dial YOUR echo test that YOU set up in YOUR dialplan |
| 22:55.45 | eternis | wait... dialplan is a command in *CLI> |
| 22:56.03 | [TK]D-Fender | eternis: dialplan = EXTENSIONS.CONF |
| 22:56.45 | p3nguin | You've already read The Book, right? |
| 22:57.25 | eternis | oh man, sorry about this. |
| 22:57.55 | *** join/#asterisk jsgoecke (~Adium@12.130.118.7) |
| 22:58.22 | eternis | I am trying to sort out the parts that don't need extra hardware and I got confuzzled. |
| 22:58.45 | [TK]D-Fender | eternis: "extra hardware"? WTF? |
| 22:58.57 | p3nguin | The only extra hardware will be needed to hook up POTS phones and wiring to Asterisk. |
| 23:03.00 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 23:11.34 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 23:12.55 | *** join/#asterisk x86 (~x86@kitrich.net) |
| 23:13.08 | x86 | so how well does 1.8's Google Voice integration work? |
| 23:13.39 | jsgoecke | What is the GV integration exactly? |
| 23:13.45 | jsgoecke | I thought it was for the GTalk integration? |
| 23:14.09 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 23:27.54 | BesticlesWork | Found alittle more info on my problem. I'm AMI originating local into my dialplan, using context=outbound. In context outbound, I am then dialing out on my dahdi. Then next priority hangs it up. I am seeing behavior of 2 threads. If I answer the call, the current thread completes the priorities, And a new thread starts at 1, making another call. What is causing that? |
| 23:29.17 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:31.09 | bcrisp | ugh |
| 23:33.02 | bcrisp | [TK]D-Fender, got in touch with the sip provider who insisted that DTMF was being relayed properly out of band but some callers' dtmf is not coming through |
| 23:33.16 | bcrisp | tried switching dtmfmode to auto, info, inband, rfc2833 and nothing seems to work |
| 23:35.14 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 23:36.01 | lvlolvlo | @bcrisp were you able to use tcpdump and look at the event durations to see if they were above ~700 |
| 23:36.22 | bcrisp | .. no |
| 23:37.14 | lvlolvlo | prolly should do that |
| 23:37.19 | bcrisp | lvlolvlo: could you help me with the command? |
| 23:37.25 | lvlolvlo | it's fairly simple |
| 23:37.38 | bcrisp | any siwtches on that command i should use? |
| 23:37.41 | lvlolvlo | from SSH first check if you have tcpdump installed |
| 23:37.46 | bcrisp | i do |
| 23:37.58 | bcrisp | i run it and it dumps junk .. not sure how to read it |
| 23:38.21 | lvlolvlo | okay, then simply do: tcpdump -s 2000 -w insert_file_name_here.pcap |
| 23:38.45 | lvlolvlo | optionally you can and port 5060 or port 5160 or portrange 5060-5160 |
| 23:39.01 | lvlolvlo | -s limits the packet size |
| 23:39.07 | lvlolvlo | -w outputs it to a file name |
| 23:39.33 | bcrisp | ok i ran it |
| 23:39.40 | bcrisp | what should i use to open .pcap? |
| 23:39.49 | lvlolvlo | okay run it and place your call that you have issues with dtmf |
| 23:39.55 | lvlolvlo | proceed with the call |
| 23:39.56 | bcrisp | just did |
| 23:40.03 | lvlolvlo | kk did you complete the call |
| 23:40.06 | lvlolvlo | hang up and all? |
| 23:40.09 | bcrisp | yep |
| 23:40.18 | *** join/#asterisk demiv (~demiv___@190.144.133.98) |
| 23:40.18 | lvlolvlo | did you call the tcpdump command? |
| 23:40.23 | lvlolvlo | kill* |
| 23:40.29 | lvlolvlo | not call, i mean kill |
| 23:40.34 | bcrisp | ya |
| 23:40.44 | bcrisp | 560 packets captured... |
| 23:40.46 | lvlolvlo | how are you connected to the system? |
| 23:40.49 | lvlolvlo | ssh? |
| 23:40.51 | demiv | hello there... it is posible to monitor or record a pickup call ? |
| 23:40.53 | bcrisp | yes |
| 23:40.59 | lvlolvlo | what is your OS? |
| 23:41.03 | bcrisp | centos |
| 23:41.08 | lvlolvlo | the computer you're using |
| 23:41.12 | bcrisp | win 7 |
| 23:41.14 | lvlolvlo | ahh.. |
| 23:41.15 | lvlolvlo | kk |
| 23:41.39 | lvlolvlo | then just use something like Filezilla and connect to the system on port 22 to transfer the file from the system to your desktip |
| 23:41.47 | lvlolvlo | and in the meantime download wireshark on your computer |
| 23:42.32 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 23:44.12 | bcrisp | cool, transferred and wireshark is installing |
| 23:45.30 | bcrisp | ran it and opened the capture file |
| 23:46.10 | lvlolvlo | okay click on Telephony and then VoIP Calls |
| 23:46.29 | bcrisp | this is an amazing tool |
| 23:46.38 | lvlolvlo | if you see the call there click on prepare filter |
| 23:46.47 | eternis | something happened :) |
| 23:46.57 | bcrisp | lvlolvlo: ok, yes i clicked prepare filter |
| 23:47.03 | lvlolvlo | click close |
| 23:47.10 | lvlolvlo | and click on Apply |
| 23:47.23 | lvlolvlo | now you should only see the packets from that one call |
| 23:47.25 | eternis | is this a good message ? --> [Oct 25 20:04:02] NOTICE[22966]: chan_sip.c:20118 handle_request_invite: Sending fake auth rejection for device |
| 23:47.38 | bcrisp | lvlolvlo: yes.. this is great |
| 23:47.39 | lvlolvlo | no that filter isn't the greatest, but it'll get the job done ehre |
| 23:47.53 | lvlolvlo | find packet which is for DTMF |
| 23:48.04 | lvlolvlo | and expand the information for RFC2833 |
| 23:48.09 | lvlolvlo | look at the event duration |
| 23:48.12 | bcrisp | no dtmf seen here |
| 23:48.16 | lvlolvlo | ruh roh |
| 23:48.18 | lvlolvlo | okay |
| 23:48.22 | lvlolvlo | clear the filter |
| 23:48.28 | lvlolvlo | and type in rtpevent |
| 23:48.34 | lvlolvlo | and click Apply |
| 23:48.45 | bcrisp | nada |
| 23:48.57 | bcrisp | maybe i need to retry this |
| 23:49.01 | lvlolvlo | okay, that just means that your system isn't sending DTMF in RFC2833 |
| 23:49.15 | lvlolvlo | it could be you're sending DTMF inband? |
| 23:49.17 | bcrisp | i will try from a cell phone, as this seems to work |
| 23:49.39 | bcrisp | if i call in from a cell phone it works, but not from these other desk voip phones (they are in a different network also) |
| 23:51.21 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 23:51.49 | nny | anyone know of any security vulnerabilities in 1.6.0.21? Looking online now, just curious.. I seem to remember a remote executable vuln |
| 23:52.11 | lvlolvlo | okay, well what phones are you using? |
| 23:53.20 | bcrisp | AltiGen IP705s |
| 23:53.33 | lvlolvlo | check the firmware to see if they are up-to-date |
| 23:54.01 | lvlolvlo | then check your config in the phone to see if they are set to send DTMF in RFC2833 or inband |
| 23:54.04 | bcrisp | i can call our IT dept for that ... the asterisk is a separate phone sys we use for a different department |
| 23:54.12 | bcrisp | i ran the tcpdump capture on the cell call |
| 23:54.18 | lvlolvlo | have your IT dept check the firmware |
| 23:54.18 | bcrisp | this time i do have rtpevents |
| 23:54.22 | lvlolvlo | sweet |
| 23:54.24 | lvlolvlo | what are they |
| 23:54.25 | lvlolvlo | ? |
| 23:54.44 | bcrisp | in the time, it says ~ 21.70 |
| 23:54.47 | bcrisp | for all of em |
| 23:54.56 | lvlolvlo | 21.7? |
| 23:55.04 | bcrisp | sorry wrong section |
| 23:55.06 | lvlolvlo | that's odd typically they're in increments on 100 |
| 23:55.06 | bcrisp | im expanding dtmf |
| 23:55.08 | lvlolvlo | kk |
| 23:55.17 | bcrisp | event duration 160 |
| 23:55.27 | bcrisp | i have one that is 760 |
| 23:55.40 | lvlolvlo | sorry excuse the typos, i'm going back and forth between work work and irc :/ |
| 23:55.42 | lvlolvlo | :P |
| 23:55.50 | lvlolvlo | anything over that? |
| 23:55.56 | bcrisp | Paylodad type=RTP Event, DTMF Eight 8 (end) |
| 23:56.05 | lvlolvlo | sorry as in value |
| 23:56.05 | bcrisp | the ends events are all 760 |
| 23:56.09 | lvlolvlo | hmm.... |
| 23:56.22 | lvlolvlo | i would imagine that to be a.ok |
| 23:56.24 | bcrisp | nope, nothing over that |
| 23:56.37 | lvlolvlo | but in some cases it might not |
| 23:56.44 | lvlolvlo | okay, here's what I would say to do in this case |
| 23:56.52 | lvlolvlo | check if there is a firmware update available |
| 23:56.57 | bcrisp | ok |
| 23:56.57 | lvlolvlo | if there is great |
| 23:57.04 | lvlolvlo | i had the same issue with my SNOM 870 |
| 23:57.08 | bcrisp | it seems to happen every once in a while |
| 23:57.09 | bcrisp | which is strange |
| 23:57.17 | lvlolvlo | and it wasn't until the latest beta firmware the DTMF fixed itself |
| 23:57.32 | bcrisp | here is the other funny thing |
| 23:57.44 | bcrisp | if i dial other numbers with these phones (like 1800 Flowers) it responds to dtmf within the IVR |
| 23:57.54 | bcrisp | just not when in the asterisk IVR |
| 23:58.16 | bcrisp | i thought if I switched asterisk to auto that it might work but that didnt solve things either |
| 23:58.21 | lvlolvlo | well the thing is that the event duration might be fine for the "pipe" it's taking to that connection, whereas this "pipe" doesn't like it. |
| 23:58.23 | lvlolvlo | does that make sense? |
| 23:58.45 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 23:58.46 | bcrisp | im not sure :/ |
| 23:58.50 | lvlolvlo | best DTMF method I would say is RFC2833 as it is out of band and DTMF isn't impacted by line quality |
| 23:58.59 | nny | anyone know of a quick way to a g729 license switched to a new box? |
| 23:59.09 | bcrisp | the in-bound sip provider swears that it is rfc2833 |
| 23:59.22 | bcrisp | and that is what has been always set in my * implementation |
| 23:59.59 | lvlolvlo | well let's say one person likes something done one way, and someone else likes something done in the other way - you gotta please both, but if you can't maintain a duration longer than ~700 the other person won't accept it. |
