| 00:00.32 | samueltc | hey, anyone here in the UK? i would like to test inbound DTMF on a mobile and landline? |
| 00:00.56 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 00:01.29 | Psykick | the inbound calls are coming in over our E1 pri |
| 00:02.07 | kizmet | Psykick, then a codec wont matter because the calls are coming in on a Digium card im assuming ? |
| 00:02.27 | Psykick | yip |
| 00:02.55 | kizmet | well you can only prioritise codecs for sip and iax connections. |
| 00:05.55 | Psykick | ok |
| 00:05.57 | Psykick | thanks |
| 00:06.14 | *** join/#asterisk poemius (n=poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
| 00:06.30 | poemius | :) I got into pocket pc development :) |
| 00:06.39 | poemius | and got confused by SIPs there :) |
| 00:07.07 | poemius | (Soft Input Panels)... I kept asking myself... what the... :) |
| 00:12.55 | *** join/#asterisk sergejfk (i=sergejfk@148.243.86.74) |
| 00:16.32 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 00:19.58 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 00:20.37 | emergion | Hello I need some help with making my server talk to my VP asterisk server I think i need my server to register with theres to receive in coming calls does that sound right? |
| 00:23.40 | SibRphrek | emergion: you need to have your server in your VP * sip.conf and the VP * in your sip.com |
| 00:24.31 | emergion | oh ok would it be more sensible to just get them to point my DID straight at my server? |
| 00:25.06 | SibRphrek | well in the VP extenstions.conf - you set your DID to Dial(EXTEN@yourserver) |
| 00:25.14 | SibRphrek | that's not the right code, but you get it |
| 00:25.52 | emergion | Oh ok becaus when I call the number they gave me it tells the person at extention number blah blah is not available |
| 00:26.01 | emergion | the extension number being my did number |
| 00:26.15 | emergion | so I would say its because I have no connected to their sip server |
| 00:26.30 | SibRphrek | make sure the VP * is passing the extention (DID) to your *, and then have yout * figure out w3hat to do |
| 00:26.37 | SibRphrek | it's not htat hard, i do it now with my server |
| 00:27.00 | *** join/#asterisk casterman (n=casterma@kasadsyed.net8.nerim.net) |
| 00:27.12 | emergion | Yeah but I just dont know how to use my asterisk to connect to there server |
| 00:27.13 | casterman | hello |
| 00:27.19 | emergion | hi |
| 00:27.44 | casterman | i have a digium dev kit but i am lost |
| 00:28.58 | casterman | i know can sound stupid but he how do i know which port should i plug the pots in the400P |
| 00:30.00 | SibRphrek | emergion you have to have their server in your sip.conf |
| 00:32.34 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 00:33.10 | *** part/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 00:35.31 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:35.56 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:40.45 | FuriousGeorge | if i were to poll 100 people who use an external db w/ asterisk, and ask them which type of db they run, what do you think the majority would say? |
| 00:40.56 | robin_sz | Mysql |
| 00:41.07 | robin_sz | or, maybe postgres |
| 00:42.13 | FuriousGeorge | robin_sz: i heard that mysql is more resource hungry than other dbs as a matter of its philosophy: it wants to use 100% of the host box's resources |
| 00:42.31 | robin_sz | ?? |
| 00:42.36 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 00:42.37 | robin_sz | nah |
| 00:42.38 | lunaphyte | SkramX: did you figure it out? |
| 00:42.47 | Abydos313 | oracle has a free database now. limited to 4gb |
| 00:42.59 | robin_sz | yeah |
| 00:43.17 | FuriousGeorge | Abydos313: any good reasons to use it? |
| 00:43.31 | Abydos313 | quality and ulitmate stablility |
| 00:43.33 | robin_sz | and i bet it only takes you a few course, a week to set it up before it runs |
| 00:43.41 | sergejfk | We use MySQL but migrating to Postgres |
| 00:43.42 | robin_sz | and I bet nothing connects to it yet |
| 00:44.02 | Abydos313 | it's an rpm format. answer 3-4 questions and all the rest is automated. configured thru browser |
| 00:44.11 | robin_sz | rpm? |
| 00:44.34 | Abydos313 | only takes 12 min or so and is made for rpm distro's.. no source |
| 00:44.37 | rt | is anybody else having strange problems with free world dialup today? |
| 00:44.38 | FuriousGeorge | i notice a lot of people use postgres with *, is there a reason for that or is it that popular in general |
| 00:44.44 | robin_sz | give me .debs or give me death! |
| 00:44.59 | lunaphyte | how can i change the folders available through voicemail? |
| 00:45.17 | robin_sz | theres little to choose between postgres and mysql .. neither is particularly quicker or hungrier than the other |
| 00:45.51 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 00:46.21 | robin_sz | mysql is easier to set up and run. postgres has a few more features ... |
| 00:46.22 | robin_sz | shurg |
| 00:46.25 | sergejfk | For applications other than * some queries run significantly faster on Postgres than MySQL - so we got hooked on Postgres |
| 00:46.35 | FuriousGeorge | robin_sz: am i correct in saying MySQL is the most popular open source db framework (or whatever it is) |
| 00:46.39 | xtrvd | Does anyone know of a dialplan validator of some sort? I'm having some troubles with my old * 1.0.4 dialplan in * 1.2 |
| 00:46.55 | robin_sz | FuriousGeorge: id says its roughly 50:50 |
| 00:47.10 | forao | it may be the most popular, but that doesn't mean its the best |
| 00:47.27 | xtrvd | Or even further, are there any substantial changes in the usage of variables from 1.0 to 1.2? It seems that as soon as I remove all the variables in my dialplan, everything runs well. |
| 00:47.38 | FuriousGeorge | robin_sz: mysql:postgress obviously, right |
| 00:48.02 | robin_sz | forao: its easier to administer ... it lacks some feature .. but if the application is written to run with either db, then its probably only using the common features between the two anyway |
| 00:48.20 | robin_sz | FuriousGeorge: yeah , those two |
| 00:49.03 | robin_sz | the two problem arease for migration between the two are usually: |
| 00:49.26 | robin_sz | postgres lacks and auto-increment facility on table rows |
| 00:49.42 | robin_sz | mysql didnt use to do sub-queries |
| 00:49.50 | robin_sz | I htink later ones do |
| 00:50.21 | Netgeeks | but migrating from mysql 4.X to mysql 5.x is no different than from mysql to postgres |
| 00:50.54 | [av]bani | \o/ |
| 00:52.14 | *** join/#asterisk kizmet (n=kizmet@freematrix/sponsor/kizmet) |
| 00:54.04 | FuriousGeorge | [av]bani: i never have any idea what your ascii cartoons mean |
| 00:55.24 | SkramX | lunaphyte: Yes'Sir :) |
| 00:57.31 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 00:58.08 | *** join/#asterisk Maxx4life (n=Maxx4lif@71-35-210-12.slkc.qwest.net) |
| 00:58.52 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 00:59.06 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 01:00.31 | casterman | good night |
| 01:00.34 | *** part/#asterisk casterman (n=casterma@kasadsyed.net8.nerim.net) |
| 01:01.54 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 01:05.35 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:05.38 | Ariel_ | hello everone |
| 01:05.40 | [av]bani | <o/ |
| 01:05.42 | [av]bani | \o> |
| 01:05.50 | Ariel_ | everone/everyone |
| 01:09.33 | poemius | hello :) |
| 01:10.28 | *** join/#asterisk Ukyo (n=Kami-Sam@68-113-211-246.dhcp.ftwo.tx.charter.com) |
| 01:11.35 | Ukyo | hey guys.. I was looking at the agentlogin/logout page on voip-info. It has a supposedly working script to autologin/login. it "assums you have the agentcode & pincode" in variables already.. how can I have teh AGI pull that? |
| 01:11.50 | Ukyo | (trying to get agent-autologin/out on a ciscophone) |
| 01:25.02 | *** join/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 01:25.12 | *** join/#asterisk acidchild (i=ashica@unaffiliated/acidchild) |
| 01:34.08 | *** join/#asterisk tfrevor (n=vrcaduc@c-67-173-209-158.hsd1.tx.comcast.net) |
| 01:34.21 | tfrevor | Hullo, all. Was hopin' I might get some assistance. |
| 01:34.48 | SkramX | tfrevor: what is the problem? |
| 01:36.10 | tfrevor | I have a D-Link DVG-1120M converted to an 1120S. The FXS ports are working beautifully. However, I'm tryinig to find out if the line in is a "true" FXO or just a passthrough to the FXS ports. I am getting a response in the help file for FXO (when telnetted into the box), but I can't find documentation on it anywhere. |
| 01:36.35 | tfrevor | I'm using the DVG-1120S as a tester to demonstrate the advantages of Asterisk to my boss. |
| 01:36.46 | trixter | its pass through |
| 01:36.50 | trixter | more or less |
| 01:37.02 | trixter | thgere is a relay that cuts it out but it cant work with that firmware as a true fxs port |
| 01:37.09 | trixter | er fxo |
| 01:37.17 | tfrevor | drat. |
| 01:37.42 | trixter | in theory they can create better firmware that cna do that if its more than just a relay and actually switched internally, but from what I have seen you cant do that |
| 01:37.58 | trixter | do you have dtmf problems if you use rfc2833 ? I can only use ulaw with the firmware I have and do inband dtmf |
| 01:37.59 | trixter | which blows |
| 01:38.04 | acidchild | :D |
| 01:38.08 | tfrevor | No worries. Just means I'll need to plunk down the money for a true FXO. |
| 01:38.44 | tfrevor | What blows is that I happen to have an Intel v.92 modem, but it's the 53_6_ instead of the 537... |
| 01:38.52 | tfrevor | one number... :) |
| 01:42.02 | tfrevor | are there any _decent_ external FXO which anyone would recommend for testing purposes? Realize that I run my department on the cheap (read "practically nonexistant budget"). |
| 01:42.36 | Ukyo | for cheap, i would spend $40ish on a linksys pap2 :) |
| 01:42.57 | Ukyo | er |
| 01:43.19 | Abydos313 | Ukyo i see them on ebay for 59.. the na model unlocked from the factory |
| 01:43.21 | tfrevor | Ukyo: Branded or unbranded? and if branded, how difficult is rewriting the firmware to make it unbranded? |
| 01:43.36 | Ukyo | I had mine unlocked in 10 minutes. |
| 01:43.44 | troyb1 | Ukyo for VoIP? |
| 01:43.45 | Ukyo | was branded form vonage ($40 including s/h) |
| 01:43.54 | xachen | heya Ukyo |
| 01:44.09 | Abydos313 | Ukyo someone in here did the same and then upgraded later to latest firmware and can't get back in :( |
| 01:44.15 | tfrevor | Troy I agree with trusting Cisco, but how much is it worth? |
| 01:44.15 | Ukyo | I use the pap2 at home for the home line ;) |
| 01:44.18 | troyb1 | Ukyo i bought mine for 50 bucks :) |
| 01:44.31 | Ukyo | i got my 7960 free :P |
| 01:44.31 | tfrevor | errr... how much does it cost? |
| 01:44.36 | troyb1 | well its worth atleast 50 to me |
| 01:44.41 | troyb1 | maybe 300~ Ukyo? |
| 01:44.52 | Ukyo | friend gave it to me. |
| 01:45.04 | Abydos313 | so what features are you guys utilizing on those cisco phones? |
| 01:45.13 | Abydos313 | that make it worth buying for home use |
| 01:45.17 | Ukyo | Abydos313: I'm just now actually setting it up |
| 01:45.18 | troyb1 | ohh *home* use |
| 01:45.27 | troyb1 | well mine is directly connected to my asterisk box.. soo |
| 01:45.33 | Ukyo | fighting with agent auto login/out |
| 01:45.38 | Ariel_ | tfrevor, fxo for testing if you do a search on ebay you can get the x100p for under 15 dollars |
| 01:45.39 | troyb1 | Ukyo do canadians say eh? a lot =P |
| 01:45.48 | Ariel_ | and the pap does not have any fxo only fxs |
| 01:45.49 | Abydos313 | heh |
| 01:45.52 | Ukyo | troyb1: aboot every 3 minutes |
| 01:45.53 | Ukyo | :P |
| 01:45.59 | troyb1 | *not me* :o |
| 01:46.07 | tfrevor | Ariel: A "true" x100p or one of the Intel modems with the resistors removed? |
| 01:46.08 | Abydos313 | sipura spa 3000 has both Ariel_ |
| 01:46.21 | Ariel_ | Abydos313, yes they work great |
| 01:46.22 | Ukyo | troyb1: been follwing http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AgentCallbackLogin |
| 01:46.33 | troyb1 | sec |
| 01:46.33 | Ukyo | it says to use a script to get the pincode & agentcode |
| 01:46.37 | Ukyo | but at that, I am like .. x.x |
| 01:46.40 | Ariel_ | for asterisk the cisco 7940 works if you have them setup for sip |
| 01:46.52 | troyb1 | yup i use SIP internally and IAX externally |
| 01:46.53 | Ukyo | Ariel_: yup, mine works great with sip :) |
| 01:46.54 | Ariel_ | but for higher end good phones I use polycom 501's |
| 01:47.00 | tfrevor | If I can get my boss to go for it, we will be buyin one of the 24-line FXO cards for the system. |
| 01:47.01 | Ukyo | really? |
| 01:47.02 | Ukyo | why ? |
| 01:47.13 | troyb1 | Ukyo who is that directed towards? |
| 01:47.15 | Abydos313 | troyb1 what ata adapters or phones do you use with iax? |
| 01:47.19 | Ukyo | Ariel_ |
| 01:47.21 | Ukyo | why the polycom |
| 01:47.32 | troyb1 | Abydos313 none i use IAX to connect to my providers. |
| 01:47.58 | Abydos313 | ahh, i thought you meant remote clients to asterisks server from outside the network |
| 01:48.03 | Ariel_ | Ukyo, cisco's make you pay for everything. I don't like there licence mess. |
| 01:48.04 | troyb1 | asterisk right now has a dialplan to route calls through about 6 different IAX providers. |
| 01:48.13 | troyb1 | Ariel_ pay for what? |
| 01:48.22 | troyb1 | if you use asterisk you just need the phone and the image |
| 01:48.22 | Ariel_ | sip firmware for cisco |
| 01:48.26 | troyb1 | Ariel_ :P |
| 01:48.37 | troyb1 | if its just for pesonal use dont worry about it. |
| 01:48.41 | troyb1 | *personal |
| 01:48.45 | Ukyo | troyb1: I am going to end up opening a guest port on my server, where anyone can connect into the phone system directly. |
| 01:48.53 | Ukyo | will make out of country customers happy i think |
| 01:49.09 | troyb1 | Ukyo yes i'd say your right, i connect directly to friends through IAX |
| 01:49.10 | Ariel_ | Ukyo, what contry are you in |
| 01:49.15 | troyb1 | US :P |
| 01:49.15 | Ukyo | us |
| 01:49.27 | Ariel_ | don't open your system up |
| 01:49.28 | troyb1 | i think he might be mexican though *kidding* |
| 01:49.33 | troyb1 | Ariel_ why not? |
| 01:49.36 | Ukyo | >,> |
| 01:49.40 | xachen | Ukyo runs a kickass NOC :P |
| 01:49.48 | Ariel_ | due to you can get sued by many different people |
| 01:49.54 | tfrevor | Well, thanks for the assistance! It's answered my question about the 1120. 'night all! |
| 01:49.55 | troyb1 | yeah i give creditz to Ukyo's old nextel phone. |
| 01:50.02 | Ukyo | nextel ? |
| 01:50.08 | Ariel_ | it's too easy to trace back to you even with the do not call list and calles like that |
| 01:50.17 | troyb1 | yeah i remember your old phone would consistently cut out. |
| 01:50.28 | Ukyo | that was you :P |
| 01:50.30 | troyb1 | Ariel_ plenty of places allow you to connect through their voice servers |
| 01:50.35 | Ukyo | anywho, big problem I am having atm |
| 01:50.42 | Ukyo | is getting this agent script tow ork |
| 01:51.12 | Nivex | cisco makes you pay, but doesn't polycom do the same thing? |
| 01:51.14 | troyb1 | Ukyo i have never used that function. |
| 01:51.15 | Ukyo | not free like that |
| 01:51.18 | Ukyo | I have a business |
| 01:51.21 | Ukyo | a free port to my main menu |
| 01:51.27 | Ukyo | to call for support, billing, sales |
| 01:51.27 | Ukyo | etc |
| 01:51.31 | Ariel_ | Nivex, firmware is free with polycom |
| 01:51.37 | Ariel_ | you just need to get it from your vendor |
| 01:51.38 | troyb1 | Ukyo yup [mainmenu] *grin* |
| 01:51.53 | troyb1 | it took me about a week to setup asterisk front to back (learning curve) |
| 01:52.14 | Ukyo | troyb1: also having a dtmf problem with the 7960 |
| 01:52.19 | xachen | dtmf problems |
| 01:52.21 | xachen | hehe |
| 01:52.25 | xachen | what codec you using Ukyo? |
| 01:52.32 | troyb1 | Ukyo i just sent allow=all |
| 01:52.36 | troyb1 | *set |
| 01:52.36 | Ukyo | same here |
| 01:52.41 | Ukyo | allow=all |
| 01:52.45 | xachen | dtmf=auto |
| 01:52.46 | troyb1 | i didnt want to bother with all the codec crap |
| 01:52.48 | xachen | make sure you have that set |
| 01:52.56 | xachen | yeah |
| 01:53.00 | xachen | that is dumb :P |
| 01:53.01 | xachen | disallow=all |
| 01:53.05 | xachen | allow=ulaw |
| 01:53.07 | xachen | works way better |
| 01:53.15 | xachen | then you don't accidentaly get lpc10 or something :O |
| 01:53.17 | troyb1 | xachen thats not a fair statement, sometimes you need allow=gsm etc. |
| 01:53.20 | Ariel_ | ulaw is best for sound. |
| 01:53.27 | Nivex | might also want to allow gsm for low bandwidth siutations |
| 01:53.29 | xachen | indeed |
| 01:53.30 | xachen | I use gsm |
| 01:53.40 | Ariel_ | I use ulaw, and g729 on my setups |
| 01:53.45 | troyb1 | indeed |
| 01:53.49 | Ukyo | nope |
| 01:53.50 | Ukyo | didnt fix it |
| 01:54.06 | troyb1 | :( Ukyo whats in your sip.conf file to register the phone? |
| 01:54.10 | lunaphyte | ~docs |
| 01:54.12 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:54.16 | troyb1 | want me to PM you my phone config? |
| 01:54.25 | Ariel_ | dtmfmode=rfc2833 if your using anything other then ulaw/alaw |
| 01:54.43 | Ukyo | my preffered codec on phone is g77ulaw |
| 01:55.15 | Ariel_ | if you use g711/ulaw or alaw dtmfmode=inband |
| 01:55.16 | troyb1 | Ukyo want to see a copy of my phone reg in sip.conf? |
| 01:55.17 | Ukyo | out of band dtmf is avt |
| 01:56.22 | troyb1 | *waits* |
| 01:56.44 | Ukyo | sure |
| 01:57.12 | troyb1 | woww this is #asterisk not wht |
| 01:57.27 | xachen | hah |
| 01:57.39 | Ukyo | Feb 26 19:52:38 WARNING[61983]: chan_sip.c:697 retrans_pkt: Maximum retries exceeded on call 000f8fe9-31dd001a-667833c0-4617b166@xx.xx.xx.xx for seqno 102 (Non-critical Response) |
| 01:57.46 | Ukyo | hmm, thats an interesting msg |
| 01:57.46 | *** part/#asterisk airwaves (n=support@c-68-50-99-18.hsd1.va.comcast.net) |
| 01:57.58 | troyb1 | PM sent. |
| 01:58.17 | *** part/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 01:58.30 | Ariel_ | Ukyo, network issues |
| 01:59.02 | Ukyo | it happens |
| 01:59.05 | Ukyo | everytime, same time |
| 01:59.12 | Ukyo | always when I go to dial |
| 01:59.36 | troyb1 | Ukyo SENT YOU MORE. |
| 01:59.38 | troyb1 | oops :( |
| 01:59.55 | troyb1 | keep in mind thats literally just a portion of my sip.conf file but should give you a basic idea |
| 02:00.15 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-149-59.nwrk.east.verizon.net) |
| 02:00.38 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 02:00.46 | troyb1 | Ukyo haha your phoneline is going to be like a spam open relay |
| 02:01.29 | xachen | yeah |
| 02:01.33 | xachen | see |
| 02:01.38 | xachen | somebody else laughed at your config Ukyo :P |
| 02:01.39 | Guest^DJ | hi guys, how do i verify that i have successfully installed h323 ? |
| 02:02.00 | troyb1 | xachen fancy seeing you here :) |
| 02:02.09 | litage | SwK: what's up bud |
| 02:02.11 | xachen | I've been here for a long time :P |
| 02:02.32 | troyb1 | heh |
| 02:02.47 | xachen | I just barely talk |
| 02:02.48 | troyb1 | xachen his phone line is *always* going to be busy. |
| 02:02.58 | xachen | why is that? |
| 02:03.03 | troyb1 | he's going to need an entire PRI just so a customer can get through :P |
| 02:03.21 | troyb1 | well if anyone can directly connect into his PBX, its just asking to have your phone slammed with phony calls |
| 02:03.33 | Ukyo | everything seems fine.. |
| 02:03.34 | xachen | he has his guest SIp account pointing tot he same context as his outbound calls? |
| 02:03.49 | troyb1 | meh, just my opinion. |
| 02:03.52 | troyb1 | Ukyo tis working? |
| 02:04.05 | Ukyo | no. |
| 02:04.10 | troyb1 | heh |
| 02:04.10 | Ukyo | dtmf = broken |
| 02:04.14 | troyb1 | :( |
| 02:04.17 | Ukyo | agent logging = broken |
| 02:04.18 | Ukyo | -_- |
| 02:04.19 | xachen | what problem are you having with dtmf? |
| 02:04.23 | Ukyo | troyb = broken |
| 02:04.23 | xachen | well Ukyo |
| 02:04.25 | xachen | update your asterisk |
| 02:04.27 | xachen | :P |
| 02:04.31 | Ukyo | >,> |
| 02:04.37 | xachen | its not overly hard |
| 02:04.38 | Ukyo | it works fine with other phones |
| 02:04.42 | troyb1 | Ukyo it took me a week to get everything working. |
| 02:05.04 | Ukyo | mines been working for months |
| 02:05.05 | Ukyo | :P |
| 02:05.16 | troyb1 | heh |
| 02:05.21 | *** part/#asterisk rhousand (n=root@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 02:05.26 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:05.29 | troyb1 | just pointing out that it might be a simple issue your just overlooking |
| 02:05.36 | xachen | alyways is |
| 02:06.09 | troyb1 | asterisk configuration isnt really straightforward, it just takes a lot of practice |
| 02:06.21 | Ukyo | so |
| 02:06.34 | xachen | asterisk if you smile at it wrong crashes |
| 02:06.37 | xachen | http://thisgeek.com/asterisk.gif |
| 02:06.42 | Ukyo | disallow=all allow=ulaw |
| 02:06.53 | Ukyo | preffered codec g711ulaw |
| 02:06.57 | Ukyo | out of band dtmf: avt\ |
| 02:07.16 | Ukyo | anything else I am missing ? |
| 02:07.19 | troyb1 | pm |
| 02:13.52 | *** join/#asterisk Garak_ (n=chris@stjhnf0111w-142162156203.dhcp-dynamic.nl.aliant.net) |
| 02:22.59 | *** join/#asterisk welles (n=welles@219.144.170.246) |
| 02:23.32 | welles | hi all |
| 02:25.37 | welles | anyone ever use ael? |
| 02:26.52 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 02:28.55 | *** join/#asterisk Chouby (i=Chouby@c207.134.244-142.clta.globetrotter.net) |
| 02:29.16 | Chouby | Anyone can help me with my Linksys PAP2 on Asterisk ?? |
| 02:30.25 | Chouby | nobody? ;) |
| 02:30.33 | mogorman | ive done it |
| 02:30.39 | mogorman | oh wait pap2 |
| 02:30.43 | mogorman | nevermind |
| 02:30.49 | Chouby | Feb 26 21:29:23 NOTICE[2905] chan_sip.c: Registration from 'test ' failed for '207.134.244.xxx' - Username/auth name mismatch |
| 02:31.02 | Chouby | I got this error all the time |
| 02:31.50 | Mavvie | too much whitespace. |
| 02:32.25 | Chouby | don'T understand |
| 02:32.47 | Mavvie | check the name in the error message |
| 02:33.42 | Chouby | 'test ' I know that the space is nowhere.. |
| 02:33.54 | Chouby | I don't have any spaces in my username in my config |
| 02:33.59 | Mavvie | asterisk doesn't agree with you. |
| 02:34.18 | Chouby | no space is used.. |
| 02:35.38 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:35.59 | mogorman | unless you dont come from same box... xachen |
| 02:36.21 | russellb | xachen: doesn't show mine :-p |
| 02:36.26 | xachen | :P |
| 02:36.36 | xachen | oh no! they know your asterisk ip russel :) |
| 02:36.44 | Chouby | my PAP2 give me: Registration State: Can't connect to login server |
| 02:48.12 | Chouby | anyone can help me to register my phone on asterisk server? |
| 02:48.59 | iaxy | Chouby: what type of phone? |
| 02:49.27 | Chouby | PAP2 from linksys unlock.. |
| 02:54.38 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
| 03:01.33 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:02.02 | jeebusroxors | oh man, xgl is awsome |
| 03:03.58 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:04.51 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@ppp-68-250-202-51.dsl.sfldmi.ameritech.net) |
| 03:05.51 | franck | Hi, I get Feb 27 15:05:16 WARNING[5570]: Unable to request echo training on channel 5 |
| 03:06.04 | franck | I have a card with echo cancellation |
| 03:06.17 | franck | What I'm doing wrong... what should be my setup? |
| 03:06.57 | litage | will a TDM400P fit into a 2U server case? |
| 03:07.09 | a1fa | with a pci raiser |
| 03:07.11 | mogorman | its full height |
| 03:07.14 | mogorman | you need reise |
| 03:07.15 | mogorman | r |
| 03:07.19 | russellb | riser |
| 03:07.22 | russellb | sheesh guys :-p |
| 03:07.49 | mogorman | sheesh yourself russellb |
| 03:07.55 | mogorman | did you stop killing drums? |
| 03:07.58 | russellb | mogorman: nope |
| 03:08.29 | russellb | mogorman: but I stopped using it as a nick :) |
| 03:10.07 | franck | litage: I have one in a 1U |
| 03:10.54 | litage | franck: did you have to do anything to the card to make it fit in a 1U case? |
| 03:11.01 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:11.34 | franck | well in fact I have a tdm2400 card, and I had nothing to do, if your server can take a full height card |
| 03:11.50 | MikeJ[Laptop] | he is now just russellb? |
| 03:11.52 | franck | usually the card is slotted flat |
| 03:11.59 | mogorman | apperantly |
| 03:12.05 | MikeJ[Laptop] | hmmm |
| 03:12.06 | mogorman | although i didnt get a memo about it |
| 03:12.08 | litage | franck: how can a full-height pci card fit into a 1U case? |
| 03:12.22 | franck | put it flat! |
| 03:12.23 | russellb | mogorman: it's on your desk |
| 03:12.23 | mogorman | a riser card litage |
| 03:12.25 | MikeJ[Laptop] | litage, sideways |
| 03:12.29 | MikeJ[Laptop] | heh |
| 03:12.29 | kizmet | litage, they have a riser and they sit flat against the motherboard usually. |
| 03:12.32 | mogorman | next to your water bottle? |
| 03:12.36 | russellb | yup |
| 03:12.40 | franck | litage: instead of putting it up |
| 03:12.45 | Guest^DJ | MikeJ[Laptop]: hi, are you familiar with h323? |
| 03:12.54 | franck | like this - and not like this | |
| 03:13.04 | litage | ah i see |
| 03:13.10 | franck | therefore you need a reiser card |
| 03:13.59 | [av]bani | reiser card? is that like reiserfs? |
| 03:14.04 | *** join/#asterisk websae (n=icechat5@h69-129-132-18.69-129.unk.tds.net) |
| 03:14.24 | litage | what happens if you have a 2U and the case doesn't have "horizontal slots" for a riser card? |
| 03:14.47 | mogorman | you can buy one |
| 03:14.51 | mogorman | for like 15 bucks |
| 03:14.59 | mogorman | otherwise you need a half height card |
| 03:15.00 | litage | mogorman: buy one....riser? |
| 03:15.04 | mogorman | yes |
| 03:15.13 | mogorman | are the pci openings not sideways? |
| 03:15.40 | litage | mogorman: they're vertical |
| 03:15.48 | mogorman | weird |
| 03:15.52 | litage | mogorman: as in perpendicular to the mobo |
| 03:15.53 | mogorman | you need a half height card |
| 03:16.08 | mogorman | sangoma has one but im not sure if it is for sale yet |
| 03:16.13 | mogorman | other wise you need a te110p |
| 03:16.15 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:16.18 | mogorman | and a channel bank |
| 03:16.24 | mogorman | or a hole in your case... |
| 03:17.06 | litage | oi vey |
| 03:17.13 | mogorman | sorries |
| 03:17.24 | mogorman | digium's only half height card at the moment is the te110p |
| 03:17.27 | litage | does Digium not make any half-height FXO cards? |
| 03:17.30 | litage | ah |
| 03:17.41 | mogorman | x100p is a half height fxo card |
| 03:17.46 | mogorman | you can get them off ebay |
| 03:17.49 | mogorman | and other odd place |
| 03:17.49 | mogorman | s |
| 03:17.54 | mogorman | but they are only one port |
| 03:17.57 | Qwell | mog!!! |
| 03:17.57 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:18.54 | mogorman | Qwell, |
| 03:19.19 | litage | shmaltz: why is that? |
| 03:19.29 | shmaltz | litage, too much bad expereince |
| 03:20.13 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:20.25 | shmaltz | litage, I now use only Adit 600 and/or Mediatrix 1204 |
| 03:20.26 | litage | shmaltz: what would you recommend if you only have 3 POTS lines to connect though? |
| 03:20.40 | shmaltz | litage, mediatrix 1204 |
| 03:20.57 | mogorman | oh litage you could also get a sipura ata |
| 03:21.07 | [av]bani | not for pots... |
| 03:21.10 | shmaltz | ~google mediatrix 1204 site:voipsupply.com |
| 03:21.16 | litage | mogorman: i need my 3 POTS lines to go into my * box |
| 03:21.21 | [av]bani | ~gateways |
| 03:21.22 | jbot | i heard gateways is at http://bani.anime.net/gateways/ |
| 03:21.28 | mogorman | yeah they have some gateways |
| 03:21.44 | franck | ok, I get some cross talk, and it is annoying, it happens from time to time... |
| 03:21.44 | shmaltz | mogorman, sipura is very good for single connection, anything more than that, and you asking for trouble |
| 03:21.58 | mogorman | bani |
| 03:21.58 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:22.08 | mogorman | did you add your link to jbot? |
| 03:22.10 | [av]bani | sipura is poo for pstn |
| 03:22.24 | [av]bani | mogorman: does that offend you? |
| 03:22.27 | mogorman | i just have digium hw |
| 03:22.37 | mogorman | no i just noticed you have it set to make money |
| 03:22.41 | mogorman | but its not obtrusive |
| 03:22.43 | mogorman | so im not upset |
| 03:22.45 | [av]bani | make money? |
| 03:22.49 | Qwell | mogorman: just a comma? |
| 03:23.00 | [av]bani | how so? |
| 03:23.12 | mogorman | the links to the store |
| 03:23.12 | mogorman | s |
| 03:23.17 | mogorman | are not referels? |
| 03:23.26 | mogorman | just direct links? |
| 03:23.26 | [av]bani | no, what makes you think they are? |
| 03:23.39 | litage | are VoiceTronix's cards supported by Asterisk? |
| 03:23.43 | mogorman | oh the 923r9joi90 on the end |
| 03:23.48 | mogorman | sorry if i jumped the gun |
| 03:23.57 | mogorman | it is a nice assortmant of whats out there |
| 03:24.10 | [av]bani | oh heh, session id... bleh |
| 03:24.17 | [av]bani | thats on one link |
| 03:24.20 | [av]bani | ill remove it |
| 03:24.37 | mogorman | i wasnt trying to imply anything |
| 03:24.39 | mogorman | just curious |
| 03:25.06 | [av]bani | you didnt imply anything, you just flatly stated :) |
| 03:25.48 | mogorman | sorry |
| 03:25.57 | [av]bani | want to verify my other table for money making? :)) |
| 03:26.17 | mogorman | nah |
| 03:26.20 | mogorman | i trust you |
| 03:26.36 | [av]bani | i dont even think any of those places have referrals |
| 03:26.43 | [av]bani | not a bad idea though :)) |
| 03:26.50 | mogorman | exactly |
| 03:26.54 | litage | is there a difference between a "half length" and "half height" pci card? |
| 03:26.55 | mogorman | ^_^ |
| 03:26.58 | mogorman | yes |
| 03:27.00 | mogorman | total |
| 03:27.09 | mogorman | one is height as in distance from mother board |
| 03:27.12 | mogorman | one is length |
| 03:27.15 | Qwell | jbot: locktopic ~qwellsprofit |
| 03:27.17 | Qwell | :D |
| 03:27.19 | mogorman | how far it goes back on motherboard |
| 03:27.23 | *** join/#asterisk MikeJ__ (n=vircuser@ppp-68-250-202-51.dsl.sfldmi.ameritech.net) |
| 03:27.34 | mogorman | most pci cards these days are half length |
| 03:27.47 | mogorman | only few ones like dialogic boards and tdm2400p are full length |
| 03:28.00 | litage | ah, thanks |
| 03:30.44 | *** join/#asterisk Manderson (n=Anderson@63.76.151.123) |
| 03:33.40 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 03:37.21 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@ppp-68-250-202-51.dsl.sfldmi.ameritech.net) |
| 03:37.28 | *** part/#asterisk Manderson (n=Anderson@63.76.151.123) |
| 03:41.42 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 03:42.03 | litecode | anybody doing faxing through asterisk with hylafax? |
| 03:53.29 | Supaplex | not me |
| 04:04.38 | litecode | Supaplex, lively group here. |
| 04:06.15 | Supaplex | oh yea |
| 04:06.16 | shido6 | get a did from NuFone. |
| 04:06.43 | shido6 | thats for inbound faxes, tho |
| 04:11.25 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 04:12.12 | x86 | anyone know of any good tutorials on setting up an IVR? |
| 04:14.05 | Nivex | x86: http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture </joke> |
| 04:14.15 | litecode | shido6, i have over 100 DID's. i am going to want to have a single (or few) IAXmodem listening that will take in faxes, and i want to email them out based on their DID. :P |
| 04:14.53 | shido6 | You can do that with a NuFone Did |
| 04:16.47 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
| 04:17.07 | Nivex | x86: actually, that link does make mention of good/bad design philosophies, so you might be able to glean something from it |
| 04:19.51 | litecode | shido6, i want to do it with hylafax. |
| 04:20.06 | litecode | i dunno where hylafax stores the DID for processing by faxrcvd |
| 04:20.19 | litecode | i haven't screwed with it much |
| 04:20.27 | sergejfk | Talking about IVR, anyone had any good/bad experiences using sipxvxml together with * for running IVR? |
| 04:23.29 | *** join/#asterisk pengyong (n=lala@222.188.132.244) |
| 04:26.38 | Winkie | litecode: i assume by DID you mean a standard POTS number, hylafax won't save these, but it will pass the modem device name onto faxrcvd |
| 04:27.16 | litecode | Winkie, hylafax should receive them because IAXmodem will send them to it. |
| 04:27.23 | Winkie | litecode: interesting, how does it send them? |
| 04:27.36 | litecode | exactly the question i have. |
| 04:27.53 | Winkie | we run hylafax production but with only 2 modems, we still handle a fair few faxes |
| 04:28.00 | litecode | presumeably via standard digital DID transimission to hylafax |
| 04:28.03 | Winkie | where did you learn iaxmodem would send it to hylafax? |
| 04:28.10 | Winkie | there's 'digital DID transmission'? |
| 04:28.11 | Winkie | :) |
| 04:28.32 | litecode | heh |
| 04:28.40 | litecode | exten => fax,1,Dial(IAX2/ttyIAX/${EXTEN}) ; replace ${EXTEN} with a meaningful string or variable |
| 04:28.40 | litecode | If you do not put ${EXTEN} there then IAXmodem will not receive DID information. However, in this example "fax" is your fax extension number. If you use the Asterisk "fax" extension given by zaptel fax detection then ${EXTEN} will lose its meaning, and you'll need to use a workaround to get the correct called-number passed on to IAXmodem. |
| 04:28.51 | litecode | http://www.voip-info.org/wiki/view/Asterisk+IAXmodem |
| 04:28.55 | Winkie | interesting |
| 04:29.13 | litecode | but, i can't find anything in hte lists for hylafax. |
| 04:29.22 | Winkie | well i'm pretty familiar with hylafax, let me have a look at IAXmodem |
| 04:29.32 | Winkie | incidentally i just realised someone sent a 118 page fax through us somehow |
| 04:29.39 | Winkie | it's emailed as PNG and PDF so that must have been 90 meg or so |
| 04:29.42 | Winkie | i bet they didn't recieve it |
| 04:31.25 | *** join/#asterisk AxelBp (n=alexlsf@200.12.40.89) |
| 04:31.46 | litecode | Winkie, that's scarey. |
| 04:32.08 | Winkie | yeah, we do have a few people who email in ridiculous things |
| 04:32.14 | Winkie | anything over about 5 meg is rejected by the mailserver |
| 04:32.15 | Winkie | for safety |
| 04:33.18 | Winkie | i disagree with the 'asterisk does not have very good fax support' |
| 04:33.27 | Winkie | from what i've tested with chan_zap and rxfax so far it's excellent |
| 04:34.00 | litecode | that's what i'm using so far. |
| 04:34.06 | litecode | and i might just use txfax too |
| 04:34.09 | Winkie | any problems with it you've noticed? |
| 04:34.12 | AxelBp | Hi gys..i just installed the asterisk system and have bad quality in sound and configured 2 extension, but just 103 can call to 102 and 102 hung up when it calls 103 |
| 04:34.27 | Winkie | AxelBp: you running asterisk with -vvv? |
| 04:34.42 | Winkie | litecode: do you have an iaxmodem and hylafax setup? |
| 04:34.47 | litecode | yes. |
| 04:35.07 | AxelBp | sorry, i'm very new...whats -vvv? |
| 04:35.22 | Winkie | litecode: have you recieved a fax through it yet? |
| 04:35.23 | litecode | level 3 verbosity |
| 04:35.38 | Winkie | AxelBp: i assume you're starting asterisk as 'asterisk -c' or 'asterisk -v'? |
| 04:35.40 | Winkie | uh, -r |
| 04:35.43 | litecode | Winkie, nope.. just set it up a few hours ago, just waiting till tmorrow to test. |
| 04:36.10 | AxelBp | i download the vmwarez asterisk@home |
| 04:36.29 | Winkie | AxelBp: ah, asterisk@home isn't supported here as far as i know |
| 04:37.01 | Winkie | litecode: i imagine then if faxgetty recieves the number, it will be passed to faxrcvd (/var/spool/hylafax/bin/faxrcvd), check /var/log/messages after you have recieved a fax, that logs all details sent to faxrcvd |
| 04:37.04 | Winkie | i'll give you an example |
| 04:37.32 | litecode | Winkie, thanks, i'll get back with this channel tomorrow after i test. |
| 04:37.39 | Winkie | Feb 26 17:38:23 a FaxGetty[32101]: RECV FAX: bin/faxrcvd "recvq/fax000005791.tif" "ttyS0" "000005789" "" "" "" |
| 04:37.44 | Winkie | there's an example of one of ours |
| 04:37.50 | AxelBp | ok...thanks |
| 04:37.52 | litecode | gotcha. |
| 04:38.25 | Winkie | the first spare parameter is for error codes that i've determined, but i have no idea about the last two, i suspect that's where you'll find the info |
| 04:38.38 | litecode | the last two are for the DID information |
| 04:38.40 | litecode | makes since now |
| 04:38.52 | litecode | thanks |
| 04:39.09 | Winkie | litecode: no problem, do you plan to email this out or whatever? |
| 04:40.16 | litecode | pstn to voip to fax to email, client to fax to voip to pstn |
| 04:40.55 | litecode | i've done fax to email with hylafax a lot, this is my first time where I will have to deal with DID (or VoIP for that matter) |
| 04:40.57 | Winkie | would you like some code i wrote to convert it into a bordered PDF and PNGs? I couldn't find any packages to accurately handle TIFF-F so i used imagemagick laffo |
| 04:41.01 | Winkie | ah right fair enough |
| 04:41.08 | Winkie | if you know hylafax then it doesn't matter :) |
| 04:41.11 | litecode | bordered? |
| 04:41.37 | Winkie | yeah, as in it has a border between it and the fax :) |
| 04:41.49 | Winkie | nothing particularly special, i'll find an advertising one to forward to you if you'd like |
| 04:42.06 | litecode | abates()batestechnology.com if you don't mind. |
| 04:42.15 | Winkie | give me a moment |
| 04:45.03 | Winkie | sent |
| 04:45.12 | Winkie | 737k, will take a moment or two |
| 04:46.50 | *** join/#asterisk alphaque (n=alphaque@219.95.36.28) |
| 04:47.31 | *** join/#asterisk tainted_ (n=identd@ppp-71-133-241-120.dsl.irvnca.pacbell.net) |
| 04:47.46 | litecode | i guess i don't see the border? |
| 04:47.51 | Winkie | in the PDF |
| 04:47.53 | Winkie | not in the PNG |
| 04:47.55 | litecode | hold up |
| 04:47.57 | Winkie | it's nothing special |
| 04:47.59 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 04:48.05 | tainted_ | how do i get the dialplan to ignore the # key in extensions |
| 04:48.08 | Winkie | like i said it's an imagemagick script because i've never found a suptable package |
| 04:48.10 | litecode | ahhh i see. |
| 04:48.16 | Winkie | suitable* |
| 04:48.21 | litecode | yah. looks good. |
| 04:48.34 | *** join/#asterisk blaylock (n=sfv100@68-69-102-120.chvlva.adelphia.net) |
| 04:48.39 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 04:48.54 | litecode | i doubt it's something that i will use.. i am surely going to try to input a bunch of info at the top though, so i'm sure i'll dive into imagemagick |
| 04:48.56 | litecode | (again) |
| 04:48.58 | litecode | thanks. |
| 04:49.12 | blaylock | anyone here have asterisk running on debian 3.1? |
| 04:49.28 | Winkie | no worries, let me know if you need anything |
| 04:49.49 | litecode | i'll email over my script when i'm done.. any critiques are always welcome |
| 04:50.22 | Winkie | no problem, mine uses temporary PNG files because i can't seem to get the TIFF-Fs generated by hylafax to convert into a PDF from imagemagick without corruption |
| 04:50.32 | Winkie | i wish i could pin it down but again TIFF-F seems almost unsupported |
| 04:50.36 | brockj49464 | Anybody know how to drop a call before it is answered and callback the callerID number? |
| 04:50.53 | Winkie | brockj49464: you can make conditional expressions in the dialplan based on CID |
| 04:51.15 | Winkie | so it could be as simple as Hangup(),Wait(5),Dial(ZAP/r1/${CALLERID}) or whatever |
| 04:51.22 | Winkie | there's bound to be info on the wiki |
| 04:51.24 | x86 | OMG! |
| 04:51.33 | x86 | i cant make my damn music on hold be random! |
| 04:51.33 | Winkie | you're an architecture? : |
| 04:51.34 | Winkie | :o |
| 04:51.39 | Winkie | x86: sure you can |
| 04:52.08 | x86 | in musiconhold.conf, in the [default] context, i changed the application line to pass -Z to mpg123, but it's not werkin! |
| 04:52.19 | Winkie | I assume -Z is the random flag? |
| 04:52.23 | x86 | i also tried the native mode, with random=yes, same results |
| 04:52.26 | brockj49464 | winkie: I have the selecting by callerid working, but when I ask it to hang up it does not and then when I dial it uses the same channel which is still connected... |
| 04:52.49 | Winkie | brockj49464: interesting, what sort of channel / got a copy of your conf? |
| 04:53.00 | Winkie | x86: i use madplay, with -z, and i believe it's working although i have no way to be absolutely sure |
| 04:53.16 | x86 | use a softphone with multiple lines |
| 04:53.17 | brockj49464 | winkie: One sec let me pastebin the code... |
| 04:53.20 | x86 | call your own extension |
| 04:53.24 | x86 | put the call on hold ;) |
| 04:53.35 | Winkie | x86: you mean each particular instance play a random one? |
| 04:53.40 | Winkie | music on hold groups together for efficiency |
| 04:53.42 | *** join/#asterisk rene- (n=rene-@dsl-201-128-115-132.prod-infinitum.com.mx) |
| 04:53.45 | Winkie | as far as i know |
| 04:53.55 | x86 | Winkie: how do i make it start with a different one? |
| 04:54.01 | litecode | i always have mpg123's left over after i kill off asterisk, any ideas? |
| 04:54.04 | Winkie | x86: i'm not entirely sure it's possible |
| 04:54.10 | Winkie | litecode: there's something on the wiki about this i believe |
| 04:54.21 | Winkie | i hate mpg123, madplay works just fine :) |
| 04:55.03 | rene- | hello, i have been setting up asterisk systems with several phones that have presence or IM support, how is people integrating support for those technologies? |
| 04:55.24 | brockj49464 | winkie: This is what I have right now http://pastebin.com/574428 |
| 04:55.27 | litecode | Winkie, resource footprint? |
| 04:55.39 | Winkie | litecode: seems low, we haven't had a chance to full scale test the E1 yet |
| 04:55.49 | Winkie | how many calls are you talking about? |
| 04:55.59 | litecode | around 50-100 |
| 04:56.03 | x86 | interesting, i put -Z on my command line, killed all mpg123 processes, reloaded asterisk, and put a new call on hold... |
| 04:56.18 | Winkie | litecode: on hold? |
| 04:56.25 | x86 | the new mpg123 process asterisk has spawned is not showing me a -Z |
| 04:56.28 | litecode | yeah. |
| 04:56.43 | litecode | right now i have no one on any of my lines on my test box, and i have 14 mpg123's sitting there. |
| 04:56.46 | trixter | startkeylogger |
| 04:57.17 | Winkie | litecode: that's a fair few, more than we handle at once infact. I imagine resource usage will be pretty low but i can't confirm it, as soon as i have a chance i will be testing 30 inbound calls on hold (or so), from Zap, so i will let you know |
| 04:57.34 | litecode | k. |
| 04:57.36 | litecode | mine are all SIP/IAX |
| 04:57.53 | litecode | we are using SIP because of the some of the issues with IAX2 timings |
| 04:57.55 | Winkie | resource usage will be the same i'd imagine, what codecs do you use? |
| 04:58.37 | Winkie | brockj49464: having a look at yours now |
| 04:59.25 | Winkie | brockj49464: ah you pass it into a specific context? still i'm not 100% sure what you're doing here, gimmie a sec |
| 05:00.30 | brockj49464 | winkie: I use amp so it is easier to goto a context than try and figure out how to get amp to do it if you know what I mean... |
| 05:00.50 | Winkie | brockj49464: ah, well i honestly don't know about amp, but i can't see where you're asking it to hangup :) |
| 05:00.57 | litecode | Winkie: i'm seriously thinking about tossing mp3 all together. just using sox. |
| 05:01.00 | Winkie | infact you're just giving it system access, from what i can tell |
| 05:01.16 | Winkie | litecode: i only use mp3 for the fact i'm testing and am using our music library instead of being bothered to rip our hold music :) |
| 05:01.21 | litecode | yeah |
| 05:01.34 | litecode | i just hate everything mp3 :) |
| 05:01.45 | Winkie | bah what's wrong with mp3 :) |
| 05:01.50 | litecode | quality is crap. |
| 05:01.54 | litecode | old technology. |
| 05:01.57 | litecode | ogg owns it. |
| 05:02.15 | litecode | all of my stuff (personal stuff) is in flac.. and if i need to transfer it to my iAudio, i use ogg |
| 05:02.19 | Winkie | try saying that when you send someone an ogg file and they go 'WHAT? |
| 05:02.31 | Winkie | hell i encoded a video as OGM/OGG once and had people complaining for days |
| 05:02.35 | litecode | Winkie, yah.. taht can be a problem ;) |
| 05:02.50 | x86 | hmm |
| 05:02.55 | Winkie | don't get me wrong, OGG is good, but mp3 is hardly obsolete and crap quality, although 96kbps rips do piss me off |
| 05:03.04 | litecode | never underestimate stupid people in large groups |
| 05:03.04 | x86 | is it possible to use SetMusicOnHold globally, instead of per-context? |
| 05:03.22 | brockj49464 | winkie: This code has been thru alot trying to get it to work. When I had a hangup early in the script it would just stop there...Let me write a complete new version and post that... |
| 05:03.28 | litecode | Winkie, i tend to notice the flat mp3 sound in my systems. |
| 05:03.44 | Winkie | litecode: perhaps you should be looking at your decoder rather than the mp3s |
| 05:03.52 | litecode | klipsch at home and h/k in the bmw |
| 05:03.57 | Winkie | i know for a fact there's no appreciable dynamic range difference between my oggs and mp3s |
| 05:04.14 | Winkie | hah, i use some crappy logitech system at home, but i try to listen with headphones when i can |
| 05:04.19 | litecode | i still like variable bitrate better. |
| 05:04.22 | Winkie | also lol harmon kardon |
| 05:04.31 | litecode | hey. they aren't too bad. |
| 05:04.32 | litecode | impressed me. |
| 05:04.36 | Winkie | also what BMW? tell me it isn't a 1 series or any of the newly styled ones |
| 05:04.49 | litecode | E46 330i ZHP |
| 05:04.55 | litecode | manual ;) |
| 05:05.00 | Winkie | lol ZHP, e46 330s are nice though |
| 05:05.12 | litecode | eh, the ZHP was worth it |
| 05:05.25 | litecode | too bad it's arrest me red. |
| 05:05.42 | Winkie | not the biggest fan of red, but the ZHP should have come with an LSD |
| 05:05.42 | litecode | i could drive so stupid in my silver 330xi |
| 05:05.51 | litecode | Winkie, agreed. |
| 05:05.52 | Winkie | i don't even drive so i shouldn't comment |
| 05:05.59 | litecode | i won't be driving soon. |
| 05:06.02 | Winkie | haha |
| 05:06.04 | litecode | selling it when i move to chicago. |
| 05:06.10 | Winkie | when are you moving to chicago? |
| 05:06.12 | litecode | no reason to have a car. |
| 05:06.16 | litecode | April 19th. |
| 05:06.26 | Winkie | sweet, i fly in on the 28th, wanna meet up for a beer? |
| 05:06.32 | litecode | hell ya. |
| 05:06.51 | litecode | some great places in chicago to hang out |
| 05:07.04 | litecode | i'm completing a contract in the quad cities now. |
| 05:07.04 | Winkie | i'm staying for a week or so, may 4th is when i'm leaving, staying with a couple of gay potheads but i am buying an E500 when i get there and i wanna go see everything in that time |
| 05:07.12 | litecode | talk about "god's country" |
| 05:07.13 | litecode | heh |
| 05:07.25 | x86 | GAH! why does my music on hold ONLY work with my default context, even though i'm SetMusicOnHold(othercontextbesidesdefault) ? |
| 05:07.33 | Winkie | haha, i'd love to get some contacts over in america, i'm 21 though with long hair and an opensolaris shirt so you can imagine what people think of me :) |
| 05:07.34 | litecode | Winkie, i would drive it out of the city and park it.. before you get killed. |
| 05:07.51 | Winkie | litecode: it's a camera, not a car ;) |
| 05:07.53 | litecode | Winkie, you'd fit in at any us LUG meeting. |
| 05:08.00 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:08.03 | Winkie | litecode: that's unfair, i'm not fat :( |
| 05:08.05 | Corydon76-home | x86: have you set up an alternative context in musiconhold.conf? |
| 05:08.07 | brockj49464 | winkie: Does this look like it should work? http://pastebin.com/574436 |
| 05:08.08 | litecode | hahaahaha |
| 05:08.21 | litecode | Winkie, gotcha... i was thinking mercedies e class? |
| 05:08.31 | litecode | why the US for an E500? |
| 05:08.37 | Winkie | because it's £200 more in the UK |
| 05:08.41 | litecode | makes since. |
| 05:08.43 | Winkie | which is about a third of the price |
| 05:08.49 | litecode | i drive to oklahoma to buy cars :P |
| 05:08.51 | Winkie | if i import it i pay 17.5% and 9% on the lenses |
| 05:09.01 | litecode | damn yo |
| 05:09.12 | litecode | my fioncee is a amatuer photographer. |
| 05:09.25 | Winkie | UK taxes heh |
| 05:09.26 | litecode | when we move to chicago i'll get her a nice digital sdl |
| 05:09.27 | litecode | slr |
| 05:09.31 | Winkie | brockj49464: what exactly are you trying to do? |
| 05:09.38 | Winkie | because i don't know if DISA is what you're after |
| 05:10.10 | Winkie | litecode: yeah this is a decent one, i was after the rebel xt, but this is slightly cheaper, i like the 4:3 frame size, 2x focal multiplier, but i'm a bit dubious about their 'supersonic wave filter' |
| 05:10.22 | Winkie | it allegedly self cleans the element with a 300khz supersonic wavefront |
| 05:10.26 | litecode | brockj49464, the whole problem lies in the fact you misspelled Incoming |
| 05:10.33 | litecode | jk ;) |
| 05:10.39 | brockj49464 | winkie: Let me get access as if I was inside when I call in with my cell, but have the call come from * with the CID set to |
| 05:10.53 | litecode | Winkie, anything that starts with "super" should be scrutinzed. |
| 05:11.11 | Winkie | brockj49464: i think you'll be wanting a Dial statement in there somewhere, i'm honestly not too sure though, there's an example somewhere on the wiki because i've seen it before |
| 05:11.17 | Winkie | i'd check that out, this is a bit beyond my range |
| 05:11.33 | Winkie | if you hit me up tomorrow i will have a better chance of trying it when i can test our phones without waking my flatmate (it's 5am) |
| 05:11.38 | Winkie | litecode: well 'supersonic' is a fair term ;) |
| 05:11.56 | Winkie | litecode: how old are you, if you don't mind me asking? |
| 05:12.09 | litecode | 23. |
| 05:12.10 | tainted_ | what's a good ipphone for a retail environment |
| 05:12.19 | Winkie | tainted_: what are your requirements? |
| 05:12.19 | tainted_ | i need something rugged and really durable |
| 05:12.27 | tainted_ | something that can take a beating |
| 05:12.38 | Winkie | what sort of beating, knocking to the floor beating or dropping 200lbs on it beating? |
| 05:12.43 | litecode | tainted_, incase it in lexan. :) |
| 05:12.50 | Mavvie | /home/edwin/asterisk/zaptel/1.2/zaptel.c:188: warning: â defined but not used |
| 05:12.50 | litecode | *encase |
| 05:12.52 | Winkie | you mean encase ;) |
| 05:12.52 | Winkie | bah |
| 05:12.53 | brockj49464 | That is always a good thing to keep flatmate happy. If I pastebin the asterisk debug might that help? |
| 05:12.59 | Mavvie | FC4 is so funny if yuo don't have your LANG set properly. |
| 05:13.07 | Winkie | Mavvie: more like LOCALE am i rite ;) |
| 05:13.16 | Mavvie | /home/edwin/asterisk/zaptel/1.2/zaptel.c:188: warning: 'fcstab' defined but not used |
| 05:13.19 | Winkie | brockj49464: it might, i need to hit the hay soon anyway |
| 05:13.20 | tainted_ | Winkie latter |
| 05:13.20 | Mavvie | that's what it wanted to say |
| 05:13.29 | Mavvie | no idea how it converts the one into the other. |
| 05:13.49 | litecode | okay, i'm finally buzzing after 4 long coronas. now i'm gonna program. |
| 05:13.49 | Winkie | tainted_: i doubt you're going to find any commercial products which can handle that sort of beating to be fair, what sort of retail is this? sledgehammer demonstration? |
| 05:13.52 | litecode | i'll bbl |
| 05:14.01 | Winkie | you should be ashamed of yourself for drinking corona |
| 05:14.04 | Winkie | D: |
| 05:14.10 | litecode | seriously. |
| 05:14.20 | litecode | they only sell "amerihecan" beer here and corona. |
| 05:14.34 | Winkie | ouch, i have some guinness in the fridge and a bottle of san miguel too |
| 05:14.48 | litecode | guiness is two steps from paint to me |
| 05:15.07 | litecode | cept I imagine paint tastes less like paint. |
| 05:15.08 | Winkie | haha, there's only apparantly one type of guinness that tastes good in the US, 'extra draft' or similar |
| 05:15.14 | litecode | :D |
| 05:15.15 | Winkie | no paint tastes pretty much like paint |
| 05:15.21 | Winkie | believe me |
| 05:15.27 | litecode | from experience i take it |
| 05:15.27 | litecode | haha |
| 05:15.39 | Winkie | hey you ever painted a ceiling? |
| 05:15.47 | litecode | point taken |
| 05:15.48 | russellb | used to eat paint chips as a kid? |
| 05:15.50 | Winkie | cause here's a hint, use a roller |
| 05:15.58 | litecode | haha |
| 05:16.15 | litecode | russellb, no, but the power lines made funny noises at night. |
| 05:16.47 | russellb | ha |
| 05:16.50 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:16.51 | litecode | Winkie, i've thought about taking a job in the UK |
| 05:17.14 | Winkie | litecode: it's a nice place to live, taxes etc will bite you but the people are friendly and there's no rampant hysteria, fundementalism or predominant gang violence |
| 05:17.14 | litecode | not exactly happy with the recent crap going on in the US. |
| 05:17.15 | Winkie | which is nice |
| 05:17.35 | litecode | two of those are really annoying here. |
| 05:17.41 | litecode | i only read the bbc: P |
| 05:17.42 | litecode | heh |
| 05:17.44 | harryvv | can hylafax operate on a asterisk box with little issues? can i direct a fax from a windows ws to the hylafax server ? |
| 05:17.45 | Mavvie | litecode: then don't come to .au neither :-( |
| 05:17.51 | Winkie | hell i live maybe half a mile from the roughest areas of Manchester, and i've walked and cycled through it in pitch black with no problems |
| 05:17.58 | Winkie | harryvv: how is the windows workstation recieving the fax? |
| 05:18.02 | litecode | Mavvie, bunch of that crap going on there? |
| 05:18.09 | Winkie | John Howard laffo |
| 05:18.17 | Mavvie | litecode: we're the US lapdog at this moment. |
| 05:18.29 | *** part/#asterisk sergejfk (i=sergejfk@148.243.86.74) |
| 05:18.34 | harryvv | winkie, its not at this moment. Personally, I would rather just install a free windows based fax application and get some of these faxes out. |
| 05:18.38 | litecode | Winkie, better than the village(world) idiot. |
| 05:18.52 | Winkie | litecode: barely, he's as crazy as Bush is i fear |
| 05:18.55 | litecode | i mean seriously, i'm no leftist... i'm a libritarian, but sheesh. |
| 05:19.16 | Winkie | harryvv: well it's up to you, hylafax is pretty simple and bar modem driver problems has been up for months now at work and i don't even bother touching it anymore |
| 05:19.34 | harryvv | bar modem driver? |
| 05:19.47 | litecode | the limiting of people rights in the name fo religion is a little exessive over here... so i'd like to try the UK or germany to get smoe perspective on life :P |
| 05:19.54 | Winkie | litecode: i'm a mix, i love guns and hate all this whiny 'second hand smoke is killing us all' crap but jesus christ, Intelligent Design? Piss off :) |
| 05:20.11 | Winkie | harryvv: yeah, one of the modems i use uses a winmodem driver which dies every week or so, it's simple enough to fix, i even scripted it |
| 05:20.17 | litecode | harryvv, i've had a hylafax server running on aclients computer for two years.. without a single login to fix ANYTHING |
| 05:20.43 | Winkie | plus the daily emails are kinda nice, and it housekeeps well, removing old fax files and logs |
| 05:20.44 | litecode | Winkie, that's top right |
| 05:20.54 | Winkie | litecode: how do you mean? |
| 05:20.57 | litecode | conservative libertarian |
| 05:20.59 | harryvv | lite, did the end users send there files from a windows box to the linux based hylfax server to send it out? |
| 05:21.05 | litecode | yes. |
| 05:21.23 | harryvv | litecode do you know of a site that talks about this arangment? |
| 05:21.30 | Winkie | ah something like that, i'm also big on freedom of speech etc, although i do think the US tends to think that because freedom of speech is in your constitution you're beyond reproach |
| 05:21.43 | Winkie | harryvv: there are tuturials on hylafax's site if i'm not mistaken |
| 05:21.49 | Winkie | tutorials* |
| 05:22.19 | litecode | think of a diamond.. you have democrats on the left, repubs on the right, statests on the bottom and libritarians on the top. |
| 05:22.28 | brockj49464 | winkie: http://pastebin.com/574460 I get the auth but it does not hangup between me calling and calling back... |
| 05:22.35 | harryvv | okay |
| 05:22.38 | litecode | you can set your political views somewhere within that diamond... which why i'm not really a member of any stupid political party. |
| 05:23.10 | Winkie | yeah, and besides my political views differ on different things |
| 05:23.26 | Winkie | i'm all for abortion, but i also would like to be able to go shooting on a weekend |
| 05:23.43 | Mavvie | Winkie: combine the two and you have best of both worlds! |
| 05:23.55 | litecode | Winkie, i agree with that |
| 05:24.02 | harryvv | ill wait latter on to install this. going to just download some kind of free fax software so i can get these forms out. |
| 05:24.20 | Winkie | harryvv: there's an online fax service too if the number is a standard POTS number |
| 05:24.24 | litecode | Winkie, i don't own a gun, but i do own a target air rifle. |
| 05:24.25 | Winkie | i wish i could remember the name of it |
| 05:24.36 | Winkie | litecode: haha i own an air rifle, pistol and a bunch of knives D: |
| 05:24.37 | litecode | Winkie, but id' like to know taht i could, if i wanted to, move up to a .22 for target practice. |
| 05:24.47 | Winkie | brockj49464: what do you mean it doesn't hangup between you calling and calling back? |
| 05:24.50 | litecode | Winkie, what kind of air rifle? |
| 05:24.53 | Winkie | it never completes the hangup? |
| 05:24.54 | litecode | Winkie, anything special? |
| 05:25.02 | Winkie | litecode: some crappy old 177, bought it for £75 when i was 14 |
| 05:25.09 | litecode | hehe |
| 05:25.10 | brockj49464 | winkie: Correct does not drop the call between |
| 05:25.19 | litecode | Winkie, sounds like my gamo, which has never failed me yet ;) |
| 05:25.26 | Winkie | it's upgraded past 12ft/lbs which is the legal limit, but still :) |
| 05:25.39 | tainted_ | how do i get my dialplan to ignore the # key |
| 05:25.55 | Winkie | brockj49464: i'm not sure why you are using DISA, although your CID looks like a manchester number? |
| 05:26.10 | brockj49464 | US cellphone |
| 05:26.24 | Winkie | ah, it's ironic because the area code for Manchester is 161 :) |
| 05:26.27 | Winkie | well 0161 |
| 05:26.56 | Winkie | still, i have no idea what the outbound-allroutes macro does, i assume that calls the EXTEN you just set, does it call you back at all? |
| 05:27.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:27.03 | litecode | Winkie, I drive 10 over whereever i go, because i know my german car is still far safer, even at that speed than a chevy. |
| 05:27.23 | litecode | Winkie, when i hit an american SUV. I _will_ survive. |
| 05:27.25 | Winkie | litecode: yeah i wouldn't be surprised, i like some american cars but ZE GERMANS know their engineering |
| 05:27.27 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 05:27.40 | Winkie | litecode: you might not if it's raised so the bumper impacts your window + B pillar instead of your axles |
| 05:27.42 | litecode | Winkie, yah and they dont' change everything on a car every two years. |
| 05:27.50 | hellop | I'm getting zap errors: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 05:28.03 | litecode | Winkie, side impact air bags and a let the damn thing roll right over me |
| 05:28.14 | litecode | seriously, i can't wait until the arabs cut us off. |
| 05:28.17 | Winkie | hellop: i assume you have zaptel installed and configured? what does cat /proc/zaptel/1 return? |
| 05:28.21 | hellop | This is my 3rd zap card. 2 others faild but didn't give this message. I'm beginning to think my MB is frying cards. |
| 05:28.28 | litecode | i've already converted my diesel jetta over to grease :P |
| 05:28.28 | Winkie | hellop: 3rd?! |
| 05:28.42 | hellop | Winkie, first I got a module not found error. maybe my HD is bad? |
| 05:28.43 | Winkie | your motherboard could be frying cards but that's unlikely |
| 05:28.46 | litecode | hellop, you are an evil, evil person. |
| 05:29.03 | litecode | poor cards. didn't even have a chance. |
| 05:29.04 | Winkie | hellop: check dmesg for hard drive errors, otherwise it sounds like you haven't installed the zaptel kernel modules correctly |
| 05:29.05 | hellop | Winkie, 3Zap cards and a modem. After I plugged in an MP3 player to recharge. |
| 05:29.25 | iaxy | did you plug power to it/ |
| 05:29.30 | Winkie | hellop: it could technically be your power supply but USB max current spec is 500mA so that's one weak ass power supply if so |
| 05:29.43 | Winkie | even so undervolting shouldn't kill a card |
| 05:29.46 | hellop | WInkie, I just got off the phone, with this brand new card. Then, couldn't dial out. Then these errors. |
| 05:30.05 | Winkie | hellop: wait, so the card was working, when did you get the module not found error? |
| 05:30.18 | hellop | Winkie, immediately when I plugged in the MP3 player, the Zap card and SB live sound card died. |
| 05:30.24 | hellop | Winkie, yes |
| 05:30.43 | Winkie | hellop: it's possible your mp3 player grounded the USB +v or so, that will kill things |
| 05:30.55 | hellop | Winkie, hung up phone, tried to dial out. Nothing happened. Closed asterisk. ./resetzap got error : No such device. |
| 05:31.05 | Winkie | the question is, why are you playing with expensive zaptel cards and using USB to charge an mp3 player |
| 05:31.11 | Winkie | also check 'dmesg' for any reports of errors |
| 05:31.12 | hellop | Swapped PCI slot, rebooted. |
| 05:31.40 | Winkie | right well if it's still bad i'd immediately swap out the board, and test the mp3 player on a different machine, a multimeter might be handy |
| 05:32.18 | Winkie | litecode: you have my email address, i'll have to figure a way to remember to set up a meet with you, but i'll be in on the 28th at 1pm and free for a couple days, minus getting high (lol arrested for terrorism) |
| 05:32.36 | harryvv | brb |
| 05:32.46 | *** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net) |
| 05:32.57 | litecode | Winkie, i'll just send you an ical appointment. |
| 05:33.04 | litecode | then you can mod it from there :P |
| 05:33.25 | Winkie | pff, the only calendar i use is on my pda, and luckily for me, synce doesn't support wm5 yet |
| 05:33.30 | Winkie | (yes i'm sad enough to use linux as a desktop) |
| 05:33.31 | litecode | Winkie, yah.. i'm totally for getting high too.. no diff from lots of booze.. and who has beat there wife while high? |
| 05:33.46 | hellop | mp3 player works fine on other machines. Either 1. Freak accident fried Zap and Sound card, 2 other zap cards coincidently died. or 2. My MB eats cards. |
| 05:33.47 | litecode | Winkie, i'm 100% linux here :P |
| 05:33.52 | Winkie | litecode: i'm sure the guys i'm staying with will be cool with that |
| 05:33.56 | Winkie | litecode: what distro? |
| 05:34.04 | litecode | gentoo (not a zealot though) |
| 05:34.08 | litecode | used to be a slack'y |
| 05:34.18 | litecode | like knowing whats going on. |
| 05:34.19 | Winkie | hellop: well that's worrying, i'd tell the card manufacturer (digium i assume) about it and see if they can diagnose what caused it |
| 05:34.25 | Winkie | Arch Linux here, i agre |
| 05:34.55 | Winkie | i used gentoo until i wanted to read an office document and i had to compile openoffice, i can install arch in 20 minutes from blank to desktop so i just did that instead :D |
| 05:35.10 | litecode | "emerge openoffice-bin" |
| 05:35.11 | litecode | ;) |
| 05:35.24 | litecode | i've never acutally compiled openoffice on gentoo |
| 05:35.27 | Winkie | hey this was before you saw -bins |
| 05:35.30 | litecode | that seems asinine. |
| 05:35.33 | litecode | gotcha |
| 05:35.44 | litecode | my buddy was the one who switched my gentoo. |
| 05:35.46 | Winkie | when GDP or whatever it was was going to be the Next Big Thing |
| 05:35.48 | litecode | i was die-hard slack |
| 05:36.00 | Winkie | you should meet moira |
| 05:36.04 | litecode | ? |
| 05:36.04 | Winkie | he drinks crap beer and likes slackware too |
| 05:36.10 | litecode | hahaha |
| 05:36.14 | litecode | my kinda guy |
| 05:36.22 | Winkie | http://www.teambarry.org |
| 05:36.25 | litecode | i'll treat you to some of Kansas City's finest bear. |
| 05:36.30 | litecode | boulevard unfiltered wheet |
| 05:36.44 | Winkie | i like the way you put 'ea' in beer and 'ee' in wheat |
| 05:36.47 | litecode | rofl |
| 05:36.49 | Winkie | still wheat beer is quite nice |
| 05:36.57 | Winkie | too frothy a lot of the time but still a good drink |
| 05:36.58 | litecode | i'm drinking heavily over here. |
| 05:37.01 | litecode | give me a break. |
| 05:37.04 | litecode | ;D |
| 05:37.06 | Winkie | also isn't that in missouri? |
| 05:37.18 | litecode | yah. i grew up in misery. |
| 05:37.24 | litecode | and wtf. |
| 05:37.30 | Winkie | wtf wtf? |
| 05:37.30 | litecode | how did you know kansas city was in missouri. |
| 05:37.31 | litecode | seriosuly. |
| 05:37.38 | litecode | everybody i talk to around here thinks i'm from kansas. |
| 05:37.49 | litecode | only thing in Kansas City, KS is trailer parks and hoes |
| 05:37.52 | Winkie | what sort of idiot would name a city after the state (lol new york) |
| 05:38.07 | litecode | Winkie, we have fair play, MO |
| 05:38.14 | litecode | and nevada, MO |
| 05:38.18 | Winkie | nah in all seriousness i was watching extreme makeover (hey i like it ok) and it was in there at the moment |
| 05:38.27 | litecode | which is pronounced "nivAda" |
| 05:38.33 | Winkie | haha yeah i know, and Birminghams, Manchesters and Londons all over the place |
| 05:38.36 | Winkie | talk about lack of originality |
| 05:38.45 | litecode | no kidding |
| 05:38.56 | litecode | i want to start a town called wtf, IL |
| 05:38.56 | Winkie | you hear about that firefighter that saved the paramedic's life when they came under fire from some nut? |
| 05:39.00 | litecode | all lowercase. |
| 05:39.09 | litecode | Winkie, naw. |
| 05:39.11 | hellop | Ok, here's te juice of my 3 dead ZAP card problem: http://pastebin.ca/43321 |
| 05:39.21 | Winkie | litecode: in KC, they redid his house, beautiful job too |
| 05:39.52 | Winkie | i love extreme makeover in a very hetro way though, i LOVE DIY, seriously love it. My biggest ambition is to build my own house, possibly in canada |
| 05:39.54 | hellop | I see something about Unable to open IAX timing interface: I wonder if that is related to my sound card timing device thingy. |
| 05:40.16 | litecode | Winkie, wierd. i'll haev to check that out |
| 05:40.23 | hellop | maybe my onboard sound is disabled.. after the SB live blew, I never enabled it now there is a timing issue. |
| 05:40.43 | litecode | Winkie, my parents built their house... very awesome. I love it.. tall ceilings, wide open areas. |
| 05:40.53 | Winkie | hellop: run `lsmod|grep zap`, and `cat /proc/zaptel/*` |
| 05:40.55 | hellop | It is my understanding * requires a sound card for timing. |
| 05:40.56 | litecode | i feel crapped in this 3bd apartment alone |
| 05:41.08 | litecode | camped |
| 05:41.10 | litecode | cramped |
| 05:41.11 | Winkie | litecode: i'm in a 3bd apartment with one other guy, and by god i'd build the biggest house you've ever seen |
| 05:41.16 | litecode | okay. i need to stop drinking |
| 05:41.23 | Winkie | or you need to drink more! |
| 05:41.30 | hellop | root@phono2:/home/user# lsmod|grep zap zaptel 223364 1 wcfxo crc_ccitt 2432 2 zaptel,irda |
| 05:41.31 | litecode | Winkie, get some of those sweet airport moving walk ways. |
| 05:41.32 | Winkie | but yeah god i would love to just own a few acres |
| 05:41.50 | Winkie | i would build the biggest goddamn garage and workshops you'd ever seen |
| 05:41.55 | litecode | i wanna own a loft in the city. |
| 05:41.57 | Winkie | seriously i want my own metalworking shop |
| 05:42.02 | litecode | hell yeah. |
| 05:42.09 | Winkie | if i move out of here i'm moving into a house, i need a garage for when i get a car |
| 05:42.15 | hellop | /proc/zaptel is empty |
| 05:42.16 | litecode | maybe later in life, i want to move to a big house with a 4 car garage to do some rally stuff. |
| 05:42.17 | Winkie | i'll always do my own shit |
| 05:42.28 | Winkie | hellop: interesting, sounds like your modules might not have unloaded fully |
| 05:42.28 | litecode | hells yeah. |
| 05:42.34 | Winkie | `rmmod wcfxo zaptel` |
| 05:42.38 | Winkie | then do the first lsmod again |
| 05:42.40 | litecode | i can't stand people working on my car. |
| 05:42.42 | litecode | even for warranty work. |
| 05:42.47 | Winkie | litecode: working on a mk2 1988 golf gti at the moment |
| 05:42.49 | Winkie | it's in BAD shape |
| 05:43.02 | *** join/#asterisk CrazyYoss (n=nobody@c-24-5-122-56.hsd1.ca.comcast.net) |
| 05:43.14 | litecode | Winkie, hehe.. my fioncee wanted a golf tdi ;) |
| 05:43.25 | litecode | she's taking my '03 jetta tdi now. |
| 05:43.31 | hellop | WInkie, yeah I have a script to rmmod zaptel and wcfxo and then modprobe them, hen run ztcfg. I have to do that alot. |
| 05:43.33 | Winkie | litecode: they're great engines, bucketloads of torque |
| 05:43.37 | litecode | Winkie, you shoudl drive that thing like there is no tomorrow ;) |
| 05:43.45 | Winkie | hellop: i saw that, but it sounds like they weren't unloaded correctly |
| 05:43.54 | CrazyYoss | Just installed asterisk on a debian box but it doesnt seem to run, starts to run then gives me this "Loading module chan_features.so failed!" Anyone have any ideas? |
| 05:43.57 | litecode | Winkie, i will give up hp for torque any day.. freakin' honda's. |
| 05:43.58 | Winkie | litecode: i have no license, and i couldn't get insured, it's a nightmare over here to be insured |
| 05:44.05 | litecode | i bet. |
| 05:44.06 | Winkie | there's no WAY i could insure a 330i even at 23 |
| 05:44.11 | litecode | i've seen those "amazing police videos" |
| 05:44.12 | Winkie | even if i'd been driving since i was 17 |
| 05:44.14 | litecode | you peopel are nuts. |
| 05:44.31 | Winkie | we just don't need bigger cars, there's 100 miles between cities instead of 800 :D |
| 05:44.58 | litecode | winkie my (USAA military) insurance on the 330i is 517USD/6mnts |
| 05:45.01 | Winkie | CrazyYoss: it could be a dependancy, do you have a modules.conf? |
| 05:45.07 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 05:45.15 | litecode | Winkie, i hate big cars.. even the bmw seems big to me. |
| 05:45.19 | Winkie | litecode: we'd be talking 1000USD/6 for me on the gti, 1.8l NA |
| 05:45.22 | litecode | i still can't stand the E90's |
| 05:45.22 | Umaro | hey guys.. anyone know if there's a SoX IRC channel here on freenode, or elsewhere? |
| 05:45.29 | Winkie | also USAA military? |
| 05:45.36 | hellop | I just htinkthe 2 pci slots on my mini-itx board are fried.. |
| 05:45.37 | litecode | my dad is ex-reserves. |
| 05:45.37 | Winkie | Umaro: you could always /list sox :) |
| 05:45.44 | Winkie | litecode: ah i see, my dad is ex-air-force |
| 05:45.47 | litecode | explosives... great 4th of july's |
| 05:45.47 | Umaro | Winkie: :/ |
| 05:45.52 | litecode | har |
| 05:45.54 | CrazyYoss | Winkie: Yes |
| 05:45.57 | Winkie | hellop: it's possible, i'd switch boards if i were you |
| 05:46.11 | litecode | Winkie, what do you do for a living? |
| 05:46.12 | Winkie | CrazyYoss: what's commented out? dependancies are relatively unmapped |
| 05:46.14 | hellop | yeah... I just have to dump it I guess... |
| 05:46.18 | litecode | voip consulting or somethin |
| 05:46.24 | Winkie | litecode: i'm a sysadmin / tech guy, working on starting my own consultancy |
| 05:46.33 | Umaro | Okay, so I have this wacky idea.. I'm sure a few people have had it before, but I'm not exactly sure what to search for.. basically, I want to mix my asterisk recordings, but I want to mix the incoming audio to the left channel, and the outgoing audio to the right channel |
| 05:46.34 | *** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 05:46.38 | propagandhi | hey people |
| 05:46.45 | litecode | hey person |
| 05:46.49 | propagandhi | I'm looking for a little advise |
| 05:46.52 | propagandhi | advice |
| 05:46.53 | Winkie | i intend to go to university within 2 years and get a degree in Electrical and Computer Engineering, so i can begin to make my own VOIP hardware |
| 05:46.55 | litecode | we are fresh out |
| 05:46.58 | Winkie | propagandhi: don't eat yellow snow |
| 05:47.02 | Winkie | don't sit on broken glass |
| 05:47.07 | Winkie | always open a can with the tab at the top |
| 05:47.07 | litecode | well winkie has a few left. |
| 05:47.10 | Winkie | anything else? |
| 05:47.17 | propagandhi | ok, asterisk specific advice if at all possible |
| 05:47.20 | litecode | okay, now we are out |
| 05:47.22 | Mavvie | Umaro: cool idea :-) |
| 05:47.24 | Winkie | propagandhi: don't install asterisk on windows |
| 05:47.32 | Umaro | Mavvie: seems like it should be doable... |
| 05:47.39 | propagandhi | Winkie: I wouldnt dream of it |
| 05:47.42 | Winkie | Umaro: certainly, it is likely going to need some code modifications though |
| 05:47.44 | litecode | propagandhi, always Answer before talking. |
| 05:47.44 | CrazyYoss | winkie: load => pbx_gtkconsole.so, noload => chan_oss.so |
| 05:47.52 | Winkie | are you recording via Monitor or Record? |
| 05:48.07 | Winkie | CrazyYoss: to be fair you're better off checking voip-info.org, there's minimal redundancy documentation on there iirc |
| 05:48.21 | propagandhi | all i want to know is if there is a way to reduce the time between when asterisk plucks the call from PSTN and then issues the ringing tone on the trunk |
| 05:48.23 | litecode | voip-info.org is so sllloooowww |
| 05:48.28 | propagandhi | or queue whatever |
| 05:48.37 | Winkie | propagandhi: show us your dialplan (pastebin) |
| 05:48.42 | CrazyYoss | Winkie: I appreciate the help, so the problem resides somewhere with the modules? |
| 05:48.43 | Winkie | litecode: really? it's ok for me |
| 05:48.44 | propagandhi | !pastebin |
| 05:48.55 | propagandhi | which pastebin can i use |
| 05:48.59 | Winkie | CrazyYoss: i would imagine you're missing a module chan_features requires, you'll often see a segfault / hard death when that happens |
| 05:48.59 | litecode | Winkie, really? it takes like 2-3 seconds to render a page for me |
| 05:49.02 | litecode | not render. |
| 05:49.03 | Winkie | propagandhi: any pastebin you like |
| 05:49.05 | propagandhi | ok |
| 05:49.08 | litecode | but "create" and serve |
| 05:49.25 | Winkie | real0m3.377s |
| 05:49.32 | Winkie | that's resolving, connecting and grabbing index.html |
| 05:50.06 | litecode | sorry about my vocabulary, but the massive influx of alcohol as stunte\d it. |
| 05:50.21 | litecode | and i have no idea how that \ got after the d |
| 05:50.28 | Winkie | i have an idea |
| 05:50.29 | Winkie | (glug) |
| 05:50.35 | litecode | good point. |
| 05:50.47 | Winkie | right, well after this J i'm going to bed |
| 05:50.52 | Winkie | i'm supposedly up in 3 hours for work (HAH) |
| 05:51.04 | hellop | dont do that |
| 05:51.23 | Winkie | don't do what, roll a J, or get up for work? |
| 05:51.31 | Winkie | cause personally i'm liking the latter |
| 05:52.18 | litecode | don't you mean former? |
| 05:52.21 | hellop | Winkie, don't not sleep. |
| 05:52.25 | x86 | hmm |
| 05:52.28 | Winkie | hellop: i am not sleeping ;) |
| 05:52.34 | litecode | i need to code. |
| 05:52.36 | Winkie | litecode: no i mean i'm liking not getting up for work ;) |
| 05:52.37 | litecode | but i don't want to. |
| 05:52.39 | propagandhi | http://pastebin.com/574479 |
| 05:52.40 | Winkie | what do you code in? |
| 05:52.42 | hellop | gnite thanks for the help everyone and Winkie.... |
| 05:52.43 | litecode | Winkie, gotcha. |
| 05:52.43 | Winkie | and what do you need to code? |
| 05:52.46 | x86 | is there a general text script to act as a teleprompter when making voice prompts for an IVR? |
| 05:52.54 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 05:52.58 | Winkie | x86: you what? |
| 05:53.02 | litecode | Winkie, python, perl, php, java, bash, c, c++, c# (if forced) |
| 05:53.08 | litecode | anything that has the application |
| 05:53.16 | Winkie | propagandhi: that's not a dialplan, that's a series of rules |
| 05:53.30 | litecode | i am finishing off an old java project that tracks inventory systems into a database for a cable television service company |
| 05:53.32 | propagandhi | Winkie: if you could elaborate I'd appreciate it |
| 05:53.32 | Winkie | litecode: hah, i write perl predominantly, i used to do VB and i have turned my hand to C more than i want |
| 05:53.40 | x86 | Winkie: i'm not sure what to say on my IVR prompts :) |
| 05:53.45 | litecode | VB is satanic. |
| 05:53.49 | x86 | Winkie: and i dont want to sound like a goon :) |
| 05:53.52 | Winkie | propagandhi: that's just a series of extension names in asterisk, i need the actual commands you're running |
| 05:53.59 | Winkie | x86: do you have stairs in your house or something? |
| 05:54.07 | litecode | x86, act like the ship on HHGG |
| 05:54.08 | Winkie | also just explain what process you want to go through and i'll help |
| 05:54.47 | franck | I have some strange crosstalk on my zaptel card |
| 05:54.59 | Winkie | crosstalk? channels can hear each other? |
| 05:55.03 | propagandhi | Winkie: basically, everything's working fine, but when asterisk answers the call, theres a pause of say 1-2 seconds before it continues in the queue and provides the continued ringing signal |
| 05:55.05 | franck | it seems when dialing outside I can listen to another channel |
| 05:55.13 | franck | not all the time but sometimes |
| 05:55.24 | Winkie | franck: that's pretty worrying, how do you acquire the channel to dial on? |
| 05:55.31 | franck | I have 8zap interface on a tdm2400 |
| 05:55.42 | franck | Winkie: sip to dial |
| 05:55.59 | Winkie | propagandhi: well i'd need to know at least some of the commands you run in that incoming context |
| 05:56.07 | Winkie | franck: do you have the Dial() line of your extensions.conf? |
| 05:56.31 | franck | Winkie: I use AMP |
| 05:56.35 | Winkie | ~amp |
| 05:56.36 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
| 05:56.36 | x86 | Winkie: i want to thank them for calling, tell them if they know their parties' 3-digit extension, enter it at any time, if not, please choose from the following menu options. press 1 for Sales, 2 for Accounts, 3 for Technical Support, or 0 to repeat this meanu |
| 05:56.41 | Winkie | franck: that's all i can say |
| 05:56.44 | x86 | Winkie: something along those lines... |
| 05:56.59 | Winkie | x86: and what, you want to record it or what exactly? |
| 05:57.03 | Winkie | (record the greeting) |
| 05:57.16 | franck | exten => 200,1,Macro(exten-vm,200,200) |
| 05:57.17 | franck | exten => ${VM_PREFIX}200,1,Macro(vm,200) |
| 05:57.17 | franck | exten => 200,hint,SIP/200 |
| 05:57.27 | propagandhi | Winkie: http://pastebin.com/574481 |
| 05:57.30 | x86 | yeah, i know HOW to record it (i setup an extension to make recordings in GSM format), but i don't know exactly how to say it :) |
| 05:57.49 | Winkie | franck: like the bot says, we don't support AMP in here because it has significantly customised configs |
| 05:58.10 | Winkie | x86: oh i see, try this "Thanks for calling XXX, if you know the number you wish to dial, please dial it now, if not please press 1 for sales etc" |
| 05:58.13 | Winkie | ;) |
| 05:58.19 | *** join/#asterisk DiscusZ (n=arrowana@dsl-lchp-1.nucleus.com) |
| 05:58.20 | Winkie | or pay alison(?) to record it for you |
| 05:58.21 | franck | Winkie: but from sip phones to sip phones there are no issues... |
| 05:58.37 | franck | is it on zap channels only |
| 05:58.46 | Corydon76-home | x86: you mean using the Background app to playback the sound file? |
| 05:58.54 | Ukyo | alrigfht. so upgraded to latest asterisk. still nothing can recognize key tones, mailbox, etc |
| 05:59.05 | x86 | Corydon76-home: no |
| 05:59.14 | Winkie | franck: like i said, please ask in #amportal, they'll have a better chance at helping than us |
| 05:59.16 | x86 | Corydon76-home: i mean i dont know how to say what i want to record ;) |
| 05:59.19 | Winkie | most people will just ~amp you and ignore you :) |
| 05:59.32 | franck | would the extensions create this problem, seems to me more a card configuration driver issue? |
| 05:59.35 | Winkie | x86: you mean you don't know how to phrase it? |
| 05:59.35 | Corydon76-home | How difficult can it be? |
| 05:59.40 | x86 | Winkie: right ;) |
| 05:59.44 | Winkie | franck: most likely a card configuration issue or a zaptel group issue |
| 05:59.50 | Winkie | x86: use what i wrote, that's the standard pretty much |
| 05:59.55 | Winkie | Corydon76-home: you'd be surprised |
| 05:59.58 | DiscusZ | I have a quick question.. and I have been readng and not getting anywhere. I am trying to call out to FWD and its not working.. the Call me on FWD works what have I screwed up? I get a recording from my Asterisk box that all circuits are busy |
| 06:00.02 | Corydon76-home | x86: my usual method is to write it down first |
| 06:00.02 | Winkie | Ukyo: what type of channel and what origin? |
| 06:00.06 | FuriousGeorge | anyone messing with the asterisk-xmpp patch? |
| 06:00.10 | franck | Winkie: yes, I use a group |
| 06:00.13 | x86 | Winkie: can i record you saying it? |
| 06:00.14 | Winkie | DiscusZ: you're using FWD, that's a start, what sort of number are you dialling? |
| 06:00.16 | franck | should I ungroup? |
| 06:00.21 | Corydon76-home | x86: then read it, playback the sound, and if I don't like it, I'll do it again |
| 06:00.30 | Winkie | x86: i'm british and i have a pretty bad voice, i doubt it's appropriate but if you want |
| 06:00.38 | DiscusZ | I was just trying the 393612 number (time) as a test |
| 06:00.40 | x86 | i have a UK phone number :) |
| 06:00.47 | Ukyo | cisco 7960 |
| 06:00.48 | Ukyo | sip |
| 06:00.54 | FuriousGeorge | ~lastspoke forao |
| 06:01.09 | Winkie | Ukyo: have you checked the dtmf mode in cisco's config? or attempted to change it in your SIP peer def |
| 06:01.09 | x86 | Winkie: a lot of my clients are female, and you know how american chicks oogle over an english man ;) |
| 06:01.13 | x86 | heh |
| 06:01.17 | FuriousGeorge | ~lastseen forao |
| 06:01.19 | Winkie | x86: haha, that's quite true |
| 06:01.23 | Winkie | i'll do it if you want but let me do it tomorrow |
| 06:01.25 | x86 | Winkie: can i /query you? |
| 06:01.29 | Winkie | go right ahead |
| 06:01.30 | x86 | hmm |
| 06:01.35 | x86 | tomorrow? |
| 06:01.38 | Winkie | yeah, it's 6am |
| 06:01.42 | Winkie | and i'm tired out |
| 06:01.49 | x86 | bedtime or just get up time? :) |
| 06:01.52 | Ukyo | winkie: tried all different combo's |
| 06:01.56 | FuriousGeorge | Winkie: portugal? |
| 06:02.00 | Winkie | FuriousGeorge: UK |
| 06:02.05 | FuriousGeorge | next guess |
| 06:02.07 | Winkie | Ukyo: what's your cisco set to? inband or info? |
| 06:02.11 | DiscusZ | I am sure its in my trunking but can not figure out what I have done wrong.. |
| 06:02.17 | Ukyo | i've tried both in the sip.conf |
| 06:02.32 | Winkie | DiscusZ: trunking? |
| 06:02.41 | Winkie | Ukyo: you tried rfc19whatever? |
| 06:02.42 | Ukyo | on the phone, ulaw, and i have tried avt, avt_always, and none for out of band |
| 06:02.44 | Ukyo | yes |
| 06:02.51 | Ukyo | i'm pulling hair out now -_- |
| 06:02.53 | DiscusZ | Outbound calling to FWD |
| 06:02.59 | Winkie | Ukyo: that is really weird |
| 06:03.01 | franck | can i upgrade the zaptel drive without upgrading asterisk? |
| 06:03.12 | Winkie | franck: kernel or userspace? |
| 06:03.30 | Winkie | DiscusZ: i can't help you at the moment, if you still haven't solved it try me in about 12-14 hours, i have a server with a working def somewhere |
| 06:03.48 | franck | well downloading the zaptel tarball from *.org |
| 06:03.50 | Ukyo | tell me about it |
| 06:03.55 | DiscusZ | ok Sounds good.. I will keep plugging away... |
| 06:03.55 | franck | and building the kernel module |
| 06:03.58 | Ukyo | what are the ideal settings for the 7960 ? |
| 06:04.05 | Ukyo | ulaw i figured, but out of band settings ? |
| 06:04.05 | troyb1 | lol Ukyo |
| 06:04.11 | Winkie | Ukyo: check voip-info.org, usually it's info or inband |
| 06:04.14 | *** part/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 06:04.14 | Winkie | codec is irrelevant |
| 06:04.23 | Ukyo | what about the out of band tho ?\ |
| 06:04.28 | Winkie | franck: you should be able to upgrade the kernel module without touching asterisk |
| 06:04.31 | Winkie | Ukyo: that's info usually |
| 06:04.35 | Winkie | IE SIP INFO |
| 06:04.39 | Ukyo | hm |
| 06:04.51 | franck | Winkie: ok I will try that... |
| 06:05.20 | Winkie | Ukyo: you have another phone other than the cisco? |
| 06:05.46 | *** join/#asterisk clive- (n=pirch@dsl-145-18-149.telkomadsl.co.za) |
| 06:06.42 | Ukyo | yes |
| 06:06.44 | franck | Winkie: thx for help |
| 06:06.49 | Ukyo | i have a pap2, it works fine wihtout any special sip settings |
| 06:06.54 | Ukyo | as far as I know |
| 06:06.56 | Winkie | franck: no problem |
| 06:07.35 | Winkie | Ukyo: according to what i've read, it should be rfc2833 (dtmfmode) |
| 06:08.00 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 06:08.05 | Winkie | if that still doesn't work, try me again in 8 hours or so, i have a friend who has a number of working ciscos, he's bound to have a 7960 |
| 06:08.31 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 06:08.41 | firestrm | hello |
| 06:08.47 | Winkie | hey |
| 06:08.49 | Ukyo | k |
| 06:08.50 | Ukyo | thanks |
| 06:08.51 | Ukyo | :/ |
| 06:09.45 | Ukyo | whats "VAD" ? |
| 06:10.00 | firestrm | any technical nuts and volts telco ppl here? I have a question about how things work.. |
| 06:10.10 | Winkie | firestrm: some of us know our way around, what's your question? |
| 06:10.27 | Winkie | Ukyo: Voice Activated Something perhaps, nothing i'm familiar with |
| 06:10.58 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 06:11.02 | firestrm | with phones that support voicemail notification, how does that work? can it be made to work with asterisk? |
| 06:12.01 | Winkie | firestrm: i'm not sure about how it works with traditional phone systems, i don't know of any POTS phone that does it over here, can you explain the 'notification', does a light come on on the phone, or do you mean the 'Message Waiting Indicator'? |
| 06:12.16 | firestrm | Ukyo, voice activated dialing.. |
| 06:12.58 | firestrm | Winkie, it usually takes the from of a blinking light on the phone or text on the display that you have X messages |
| 06:13.24 | Winkie | firestrm: yeah Message Waiting, it's a common thing, supported just fine i believe, check voip-info.org |
| 06:13.47 | Winkie | http://www.voip-info.org/wiki/view/SIP+mwi |
| 06:13.50 | Winkie | there's the SIP equivilant |
| 06:14.03 | firestrm | im trying to do it with zap |
| 06:14.11 | Winkie | to an analog phone? |
| 06:14.23 | firestrm | but thanks for the moniker.. i didnt know what the indusrty called it.. |
| 06:14.29 | firestrm | yes |
| 06:14.35 | Winkie | not going to happen i'm afraid |
| 06:14.49 | Winkie | according to the MWI page of voip-info .org it's a 90V neon lamp |
| 06:15.00 | firestrm | bummer :( |
| 06:15.00 | Winkie | and there isn't anything supporting it so far |
| 06:15.03 | emergion | Does SIP only use one port? |
| 06:15.06 | Winkie | sorry |
| 06:15.18 | Winkie | emergion: uh, one TCP and one UDP i think? |
| 06:15.26 | firestrm | thanks anyways.. at least i know what im looking for :) |
| 06:15.32 | Winkie | firestrm: no problems |
| 06:15.44 | Winkie | [root@phonetest ~]# lsof -i|grep asterisk |
| 06:15.44 | Winkie | asterisk 1603 root 6u IPv4 3362 TCP *:5038 (LISTEN) |
| 06:15.44 | Winkie | asterisk 1603 root 10u IPv4 3370 UDP *:5060 |
| 06:15.44 | emergion | hmmz |
| 06:15.56 | Winkie | i think 5038 is the manager interface |
| 06:16.02 | Winkie | yep |
| 06:16.11 | Winkie | so it only listens on 5060 UDP on my * box :) |
| 06:16.32 | emergion | iax is 4567 yeah? |
| 06:16.47 | emergion | sorry 4596 |
| 06:16.50 | Winkie | ;bindport=4569 ; bindport and bindaddr may be specified |
| 06:16.53 | Winkie | nearly right :) |
| 06:16.56 | emergion | lol |
| 06:16.59 | emergion | thats what I meant |
| 06:17.17 | Winkie | right, anyone else want help before i go to bed? |
| 06:17.23 | Winkie | no? |
| 06:17.23 | Winkie | good |
| 06:17.27 | emergion | So what is the big deal with connecting to a asterisk server behind nat with sip |
| 06:17.28 | emergion | yeah me ;-) |
| 06:17.48 | Winkie | emergion: NAT can be a tricky bugger, check out STUN for just how painful it can be |
| 06:18.00 | Winkie | it's not too bad if one side is NATted, if both are it can get annoying |
| 06:18.03 | emergion | Better off to just dmz ? |
| 06:18.15 | Winkie | if you have port forwards, there shouldn't be any issue |
| 06:18.26 | emergion | Yeah I do |
| 06:18.42 | Winkie | having any problems with one sided audio? |
| 06:19.04 | emergion | nope just timeout when trying to connect to the servetr with sip however IAX works ok |
| 06:19.13 | emergion | i only have 5060 forwarded though |
| 06:19.32 | Winkie | interesting, you've possibly forwarded 5060 on TCP not UDP? |
| 06:19.37 | emergion | because my box does not have 5038 listening |
| 06:19.47 | emergion | Nah i selected both protocols |
| 06:19.50 | Winkie | 5038 is the manager interface |
| 06:19.52 | emergion | <PROTECTED> |
| 06:20.03 | Winkie | hmm that's interesting, you can turn on 'sip debug' at the console to see if any packets are reaching the server |
| 06:20.11 | emergion | I read somewhere SIP uses like 4 ports or something and IAX is cool cause it uses one |
| 06:20.39 | emergion | No worries I will do that, just needed to get some closure on the ports which require forwarding but you would say 5060 is enough? |
| 06:21.01 | Winkie | well, asterisk is only listening on that, it may use other ports but there's a lot of docs on google and voip-info |
| 06:21.08 | Winkie | check out http://www.voip-info.org/wiki-STUN for a primer on problems |
| 06:21.19 | *** join/#asterisk sl16 (n=blah@tv.neterra.net) |
| 06:21.22 | Winkie | http://www.voip-info.org/wiki/view/NAT+and+VOIP might help too, it was in my favourites |
| 06:21.26 | sl16 | mornin |
| 06:21.33 | emergion | I think im just going to DMZ the asterisk server |
| 06:21.41 | Winkie | sl16: don't remind me |
| 06:21.58 | sl16 | Winkie: lol ;) |
| 06:22.02 | emergion | but thanks for the help check it out now |
| 06:22.10 | Winkie | no worries |
| 06:22.17 | Winkie | sl16: i haven't even slept yet, don't tell me you have a problem too? |
| 06:22.28 | sl16 | oh, no :) |
| 06:22.34 | Winkie | good good :) |
| 06:23.06 | sl16 | i'm here if something new&interesting comes up |
| 06:23.25 | Winkie | more likely something old and rehashed but it's your bandwidth ;) |
| 06:24.24 | sl16 | :))) |
| 06:24.55 | x86 | anyone in NY? |
| 06:27.45 | brookshire | jontow is from ny |
| 06:27.50 | brookshire | :) |
| 06:31.46 | Olobola | dudes, I moved to a new office which uses a microwave connection (only difference as far as I can tell) and now my calls are all jittery. |
| 06:32.17 | litecode | what kind of microwave connection? |
| 06:32.18 | brookshire | heh.. not enough bandwidth probably |
| 06:32.51 | Winkie | or wild ping deviations |
| 06:33.09 | brookshire | winkie: that doesn't matter if they are all still low |
| 06:33.10 | Winkie | Olobola: make sure you're not downloading or uploading except asterisk, try it again and if it's still bad, call your provider |
| 06:33.25 | litecode | brookshire, heh, check out orthogon :P |
| 06:33.25 | Winkie | brookshire: if they're deviating wildly they're not going to be low :) |
| 06:33.32 | litecode | that's a REAL microwave connection |
| 06:33.58 | brookshire | i had one for a while :) |
| 06:34.05 | brookshire | 802.16 |
| 06:34.22 | brookshire | only probably was i was sharing it with like 100 other people :/ |
| 06:34.31 | brookshire | probably = problem |
| 06:34.40 | litecode | you had an orthogon connection? |
| 06:34.44 | litecode | spectra or gemini? |
| 06:34.50 | Olobola | litecode: I have no idea, they're real quiet about it on their site.. they hardly mention they're even using it. |
| 06:34.54 | litecode | ahh. |
| 06:35.19 | litecode | Olobola, it's a bitch with wireless stuff. If it's isn't carrier class. you're gonna be in a hurt. |
| 06:35.25 | Winkie | Olobola: if you require the internet for your business ALWAYS ALWAYS get an SLA |
| 06:35.28 | Olobola | 100 bucks for 1Mbps connection. |
| 06:35.44 | FuriousGeorge | anyone test multiparking? i grabbed it the other day, but it kept crashing so rather than mess with it, I grabbed the xmpp branch |
| 06:36.15 | litecode | Olobola, i'm sure it's just basical 802.11b/g hardware then. |
| 06:36.56 | FuriousGeorge | which brings me to my next question: anyone using mog's patch |
| 06:37.31 | brookshire | heh.. i've seen it |
| 06:38.58 | litecode | Olobola, how big is your company |
| 06:39.15 | litecode | if you don't mind answering |
| 06:39.35 | Winkie | 'basical', oh litecode ;) |
| 06:39.38 | Winkie | anyway i'm going to bed now |
| 06:39.40 | justinu | heh |
| 06:39.41 | Winkie | ciao |
| 06:39.48 | justinu | ~netgeeks |
| 06:39.54 | litecode | Winkie, latez. |
| 06:39.54 | FuriousGeorge | Winkie: its almost 7 go to sleep! :) |
| 06:39.59 | justinu | ~seen netgeeks |
| 06:40.03 | jbot | netgeeks <n=chris@68-185-24-8.static.mdfd.or.charter.com> was last seen on IRC in channel #asterisk, 5h 49m 42s ago, saying: 'but migrating from mysql 4.X to mysql 5.x is no different than from mysql to postgres'. |
| 06:40.03 | [av]bani | ~justinu |
| 06:40.19 | Winkie | he's right you know |
| 06:40.22 | Winkie | and postgresql rocks |
| 06:40.24 | FuriousGeorge | ~seen forao |
| 06:40.25 | jbot | forao is currently on #asterisk (7h 43m 3s). Has said a total of 1 messages. Is idling for 5h 53m 15s, last said: 'it may be the most popular, but that doesn't mean its the best'. |
| 06:40.30 | Winkie | anyway yes i really am going to bed 8) |
| 06:40.38 | [av]bani | sqlite > * |
| 06:40.38 | litecode | Winkie, i'm still all mysql ;) |
| 06:40.43 | justinu | netgeeks seems to generally be right |
| 06:40.45 | Winkie | e w w |
| 06:40.55 | litecode | i'm gonna go finish watching adult swim |
| 06:40.56 | FuriousGeorge | ~seen furiousgeorge |
| 06:40.57 | jbot | furiousgeorge is currently on #asterisk (5h 41m 51s). Has said a total of 10 messages. Is idling for 1s, last said: '~seen furiousgeorge'. |
| 06:40.57 | Olobola | litecode: The sales guy set it up (!). I'm actually just squatting in a spare office, 5 employees. |
| 06:41.00 | brookshire | i like mysql :) |
| 06:41.01 | Winkie | (it's not that bad, but goddamn postgres is nicer) |
| 06:41.17 | brookshire | depends on what you call nice |
| 06:41.26 | justinu | i came from oracle, so i understand postgres better |
| 06:41.28 | *** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net) |
| 06:41.28 | justinu | seems more serious |
| 06:41.36 | litecode | Olobola, ohh.. the OS Gemini is 12 grand and the Spectra is 18 grand... so i don't think that fits your needs |
| 06:41.39 | brookshire | mysql does a lot of nice things, that make it easier to develop for |
| 06:41.41 | Winkie | well, it has a nicer set of 'enterprise grade' features and uses a more advanced auth system for a start |
| 06:41.52 | Winkie | postgresql can probably match those nice things |
| 06:42.04 | Winkie | although damnit we need arrays of compound keys |
| 06:42.04 | Winkie | D: |
| 06:42.05 | brookshire | but you can't replicated with postgres! |
| 06:42.18 | Winkie | replicated? or did you mean replicate? |
| 06:42.19 | brookshire | well.. you can, but it uses a bunch of perl scripts to do so, lol |
| 06:42.32 | brookshire | yeah.. sorry.. tired |
| 06:42.37 | litecode | replicate? |
| 06:42.48 | Winkie | there's a few ways iirc and they work just fine? |
| 06:42.49 | litecode | nevermind i don't wanna know |
| 06:42.55 | litecode | i'm gonna go fry my brain by watching adult swim. |
| 06:42.56 | litecode | latez |
| 06:42.57 | justinu | i was under the impression that postgres 8.x's replication is quite good |
| 06:42.59 | Winkie | enjoy :) |
| 06:43.05 | *** join/#asterisk arctic_import (n=jason@209-112-170-172-cdsl-rb1.nwc.acsalaska.net) |
| 06:43.16 | brookshire | when did 8 come out? |
| 06:43.24 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:43.26 | PakiPenguin | morning |
| 06:43.48 | Winkie | brookshire: like january 2005 |
| 06:43.54 | brookshire | yeah.. |
| 06:44.02 | brookshire | postgres replication is still done with perl scripts then |
| 06:44.03 | franck | I have some kind of echo when I call from SIP to zap. The echo is on the SIP side. When I call SIP2SIP no echo... |
| 06:44.04 | Mavvie | brookshire: try slony. |
| 06:44.07 | arctic_import | anyone else using a TDM400 with fxo and fxs cards? For some reson when someone calls me it rings 2 to 3 times for that person before my phone here rings. Any ideas how I can solve this? |
| 06:44.15 | franck | how can I control this echo... |
| 06:44.15 | Winkie | who cares? perl scripts are perfectly fine |
| 06:44.15 | Mavvie | brookshire: works like a charm, no perl scripts. |
| 06:44.18 | PakiPenguin | anyone here uses astguiclient? |
| 06:44.27 | Mavvie | brookshire: and it's very easy to swap between master and slaves. |
| 06:44.32 | justinu | franck: tune your gains |
| 06:45.17 | Winkie | Mavvie: actually i heard that, i don't need to replicate for another 2 months but it was recommended to me as flawless, is it really that good? |
| 06:45.19 | Winkie | also damnit i was going to bed |
| 06:45.20 | franck | justinu: are the gain global? |
| 06:45.23 | Winkie | damn you irc |
| 06:45.32 | franck | for all zap channels or you can do it by channel |
| 06:45.42 | Winkie | per span isn't it? |
| 06:45.48 | Mavvie | Winkie: flawless? not when you set it up and have no idea how it works. But when you set it up the second time, it's flawless :-) |
| 06:46.07 | justinu | franck: not sure, but I only use one rx/txgain statement in my zapata.conf |
| 06:46.09 | Winkie | Mavvie: well ignoring my inability to grasp concepts (:D) have you found any problems with it? had many outages? |
| 06:46.12 | justinu | i think you can do it by channel tho |
| 06:46.22 | justinu | or general |
| 06:46.26 | brookshire | how does it handle autonumbers then? |
| 06:46.35 | Winkie | brookshire: perfectly fine? |
| 06:46.44 | Winkie | you mean auto incrementing numbers? |
| 06:46.45 | Mavvie | Winkie: if a slave database is unreachable for a long time (weeks), then it's starting to get sluggish. |
| 06:46.48 | brookshire | yeah |
| 06:47.00 | Mavvie | Winkie: but then you just disable it as a slave and it's gone. |
| 06:47.02 | Winkie | shit man, i can't remember using postgresql where it didn't have sequences |
| 06:47.29 | Winkie | Mavvie: yeah i can understand that, i'm more concerned about the master going down and how to sync updates when it's back or handle them whilst it's down |
| 06:47.40 | Winkie | the system i'm building must be extremely fault tolerant |
| 06:47.50 | Mavvie | Winkie: we had one failure on the main database (that was a postgresql REINDEX issue) and the slave database didn't take over the problem so we used that while we repaired the first one. |
| 06:47.54 | Winkie | to the extent that we're using 4tb of disks for < 1tb data D: |
| 06:48.14 | *** join/#asterisk Eggplant (i=No@dsl-438.cascadeaccess.com) |
| 06:48.20 | brookshire | i've stopped using autonumbers for my ids.. but i was just curious :) |
| 06:48.29 | justinu | raid-0+1? |
| 06:48.30 | franck | justinu: should I use ztmonitor and adpt when it rings or when people talk? |
| 06:48.37 | justinu | franck: use your ears |
| 06:48.39 | Winkie | brookshire: there's nothing wrong with using autonumbers for IDs, and it's perfectly fine, they're really quite simple |
| 06:48.46 | Winkie | it can do some pretty nice stuff like selective indexing |
| 06:48.50 | justinu | franck: tune it as low as it can go, and not affect the volume of the call |
| 06:49.16 | Winkie | justinu: actually raid 1 but on only 400gb disk per server at the moment (4) but it may well go to 1-1.2tb |
| 06:49.19 | Mavvie | Winkie: once we knew why the master had gone crazy we remastered the system and the dropped the database on the former master and slaved that one until it was up to date, then remastered it. |
| 06:49.22 | firestrm | Winkie, i managed to get MWI working on zap.. twas easy, i just need to know what it was called so i could google for it :) |
| 06:49.31 | justinu | ie, if it starts to sound too quiet, go higher, but only till the volume is loud enough, don't go any higher than you have to |
| 06:49.36 | Winkie | firestrm: huh? that's bizarre, i wonder why the wiki page says there's no support |
| 06:49.43 | Winkie | firestrm: perhaps you should update that, it was SIP-mwi |
| 06:50.17 | Winkie | Mavvie: and it can handle updates to slaves just fine? can you promote a slave to master and have it slave to a third database which was a previous slave to the previous master, if i'm not talking bollocks? |
| 06:50.27 | firestrm | Winkie, all you need to do is stick in a mailbox= (mailbox num) in the zapata.conf |
| 06:50.39 | Mavvie | Winkie: slaves are readonly. |
| 06:50.43 | Mavvie | Winkie: only the master is writable. |
| 06:50.53 | firestrm | winkie, roger that on the SIP-wmi |
| 06:50.54 | Winkie | Mavvie: i mean when the master is down |
| 06:51.11 | Winkie | firestrm: thanks, i'm far too tired, birds are singing and it's light and this isn't good |
| 06:51.16 | Mavvie | Winkie: when the master is down, you make one of the slaves master and duplicate it to the "third". |
| 06:51.18 | Winkie | bastard birds |
| 06:51.33 | firestrm | lol been there done that.. |
| 06:51.36 | justinu | get some donuts and coffee and get back to work |
| 06:51.38 | Winkie | Mavvie: what i mean however, is can this software automatically promote a slave to master, or is it easily scriptable? |
| 06:51.44 | justinu | wussie |
| 06:51.53 | Winkie | justinu: D:, i haven't gone to bed yet and work supposedly starts in 2 hours |
| 06:51.56 | Winkie | we'll see about that though |
| 06:52.03 | Winkie | but donuts sound good, i can't argue with that |
| 06:52.07 | justinu | heh |
| 06:52.08 | Mavvie | Winkie: it's a small slony script (three lines with definitions, three lines with commands) |
| 06:52.09 | Olobola | have any of you gone through weird skin changes from sitting in front of a crt a million hours a day for several years? |
| 06:52.26 | firestrm | Winkie, you must be in TZ +000 |
| 06:52.33 | justinu | i'm off to bed... then donuts+coffee in the morning |
| 06:52.34 | justinu | night |
| 06:52.49 | Winkie | Mavvie: excellent, that's exactly what i need, i have to use some monitoring software to handle NFS and varying other daemons so it's good to know it's easier with pgsql |
| 06:52.59 | Winkie | Olobola: i have 2 TFTs, my CRT nearly burned down my room |
| 06:53.18 | Winkie | (which is why i could taste paint by the way litecode, because i had to scrub the smoke off the ceiling and paint it white because i rent this damn place D:) |
| 06:53.18 | Mavvie | Winkie: we have a special monitor table: it gets updated every minute and if the slaves don't get updated every minute it raises an alert. |
| 06:53.21 | [av]bani | yay fire |
| 06:53.31 | Ukyo | is it normal for iax to read "stopped sounds" before passing a call on |
| 06:53.32 | Ukyo | ? |
| 06:53.54 | Ukyo | cause apparently its more than dialtones that are not passing. the other end of the call cannot hear me or toones, but I hear them fine |
| 06:54.00 | Winkie | Mavvie: well i think it'll be integrated into our monitoring system but i'm just grateful it's easy with pgsql |
| 06:54.12 | Olobola | Winkie: Ha! It feels like people are looking at me like I'm a street urchin crackhead sometimes. |
| 06:54.32 | Mavvie | Winkie: like I said, set it up once so you know how it works (and what doesn't work), then you can set it up without too much hassles on the real ones. |
| 06:54.32 | Winkie | Olobola: do you stand on corners whispering 'crack' at passers by? because that would do it |
| 06:54.51 | Winkie | Mavvie: test ones? hah you wish we had that luxury, also thanks i'll pass it on to the DB design dude |
| 06:55.18 | Winkie | oh also [av]bani http://www.hardc0re.org.uk/fire/ is the URL i believe |
| 06:55.22 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 06:55.26 | firestrm | Winkie, you sure it was in SIP-mwi, i can t find any reference to zap there.. |
| 06:55.40 | Winkie | firestrm: one moment |
| 06:55.51 | Mavvie | Winkie: it works fine in combination with postgresql-relay, so that you only have to change the location of the new database in one place and all your sessions then are forwarded to the new server. |
| 06:55.54 | Winkie | i apologise it's in just MWI |
| 06:56.02 | Winkie | just before the links section at the bottom |
| 06:56.02 | firestrm | np.. thanks |
| 06:56.16 | firestrm | im off to edit :) |
| 06:56.35 | Winkie | Mavvie: that's pretty impressive, i look forward to breaking it horribly and crying |
| 06:58.03 | brookshire | winkie: what's that from? |
| 06:58.23 | Winkie | brookshire: the fire? an AOC 9KLR 19" CRT |
| 06:58.35 | Winkie | i have a rough idea what caused it too, i believe it's a design flaw at their end |
| 06:58.52 | brookshire | ouch |
| 06:58.56 | x86 | <PROTECTED> |
| 06:59.00 | x86 | what could this mean? |
| 06:59.10 | x86 | it's trying to do a SIP registration with Stanaphone |
| 06:59.33 | Ukyo | Winkie: any idea why asterisk would let you receive/send calls. hear whats being said on th eother side, but not pass what you say ? |
| 07:00.02 | Winkie | it failed partially about 2 months prior, the power supply for the plates which modify horizontal deflection failed, it whined highly (inverter or transformer i guess) and it let out a tiny bit of smoke, i hit it and it worked again so it was probably a really shitty connection inside, and i meant to replace it after christmas, but unfortunately for me it burned down before then. Lost my job over it too, that christmas sucked |
| 07:00.06 | Winkie | Ukyo: NAT |
| 07:00.16 | Winkie | i would hazard a guess at anyway |
| 07:00.33 | Winkie | also s/highly/loudly/, i'm tired ok |
| 07:01.03 | Ukyo | hmm |
| 07:01.07 | Ukyo | the server is behing a nat |
| 07:01.11 | Ukyo | with 5060 forwarded to it |
| 07:01.12 | x86 | Winkie: any ideas with my issue? :) |
| 07:01.15 | Ukyo | the phone is public internet |
| 07:01.18 | FuriousGeorge | ive noticed on my home PC when i call out via zap to iax external extensions, my side stops sending audio from time to time |
| 07:01.24 | brookshire | ukyo: that's just control |
| 07:01.31 | brookshire | you need to forward rtp too |
| 07:01.31 | Winkie | x86: dialling out via provider? |
| 07:01.32 | x86 | Ukyo: you also need ports 9000-20000 forwarded to it |
| 07:01.35 | FuriousGeorge | but i got the lastest zaptel driver installed |
| 07:01.44 | x86 | Winkie: trying to register with provider for DID service |
| 07:01.50 | FuriousGeorge | wondering if anyone had a suggestion as to how to go about troubleshooting |
| 07:01.51 | x86 | Winkie: inbound-only |
| 07:02.04 | Winkie | x86: interesting, can you PM me your register line, remove the pass if you want |
| 07:02.26 | Ukyo | x86: thats a HUGE range |
| 07:02.27 | *** join/#asterisk ptblank (n=MURDER1@yorbalnd-cuda2-68-70-91-54.lmdaca.adelphia.net) |
| 07:02.32 | Winkie | FuriousGeorge: set up a monitor on the server, so you can decide on which side is at fault |
| 07:02.37 | FuriousGeorge | cli doesnt react obviously, or i'd have a better question to ask |
| 07:02.41 | Winkie | unlikely to be the zap side but it would pin things down to start with |
| 07:02.51 | Ukyo | x86: so your saying that asterisks randomly listens on a port between 9000 & 20000 ? |
| 07:03.14 | brookshire | ukyo: that is correct :) |
| 07:03.23 | Ukyo | anywhere I can specify the range ?> |
| 07:03.45 | FuriousGeorge | Winkie: that's right there's that quazi-graphical zaptel volume analyzer |
| 07:03.48 | FuriousGeorge | know what its called? |
| 07:03.54 | brookshire | ukyo: http://www.voip-info.org/wiki-NAT+and+VOIP |
| 07:04.00 | Ukyo | thanks |
| 07:04.18 | FuriousGeorge | wait\ |
| 07:04.24 | FuriousGeorge | 9000-20000?? |
| 07:04.36 | FuriousGeorge | i thought it was 10k-20k? |
| 07:04.59 | brookshire | voip-infor says 8766 to 35000 |
| 07:05.06 | FuriousGeorge | since when |
| 07:05.18 | Winkie | FuriousGeorge: i'm sorry, i must be going crazy, what did you just say? :) |
| 07:05.20 | Ukyo | rtpstart=65520 |
| 07:05.20 | Ukyo | rtpend=65523 |
| 07:05.34 | FuriousGeorge | will that work? |
| 07:05.35 | Ukyo | and I have those forwarded to the * server |
| 07:05.41 | Ukyo | do I need to change that in the phone also ? |
| 07:05.50 | FuriousGeorge | winkie i asked about the quazi graphical volume analyzer app for zaptel |
| 07:05.54 | x86 | Ukyo: probably :) |
| 07:06.02 | Winkie | FuriousGeorge: and i was taking a roundabout way to saying 'not a clue' |
| 07:06.04 | brookshire | ukyo: you can always try first |
| 07:06.05 | x86 | Ukyo: be easier to just forward the whole range into your * server |
| 07:06.15 | x86 | Ukyo: btw, those are all UDP ports, dont need TCP |
| 07:06.24 | Winkie | sorry, i go on little textual adventures with my writing when am (a) tired or (b) stoned |
| 07:06.29 | brookshire | or buy a sip aware firewall :) |
| 07:06.40 | FuriousGeorge | zttool --help |
| 07:06.40 | FuriousGeorge | zttool: error while loading shared libraries: libnewt.so.0.51: cannot open shared object file: No such file or directory |
| 07:06.41 | FuriousGeorge | (sigh) |
| 07:06.57 | FuriousGeorge | be careful with depclean :) |
| 07:07.01 | Winkie | brookshire: ironically that's what kept me through that christmas, stupid amounts of drugs |
| 07:07.10 | Winkie | oh that reminds me i need to build a newt package for arch |
| 07:07.15 | Winkie | well libnewt |
| 07:07.37 | x86 | anyone use Stanaphone for inbound DID service? |
| 07:08.24 | brookshire | winkie: ?? |
| 07:08.47 | Winkie | brookshire: the christmas after the fire? read up man, or do you want me to repost it |
| 07:09.12 | brookshire | what were you refering? |
| 07:09.25 | brookshire | i saw the monitor caused you to lose your job |
| 07:09.34 | firestrm | winkie, wiki updated... |
| 07:09.35 | Winkie | ah yes, well i lost my job at christmas, and like i said it was depressing |
| 07:09.42 | brookshire | but how does that have to do with stupid amounts of drugs? |
| 07:09.46 | brookshire | ahh.. |
| 07:09.50 | brookshire | drugs are bad! |
| 07:09.55 | brookshire | unless they are mine |
| 07:09.56 | brookshire | :) |
| 07:10.00 | firestrm | unless its penicillin |
| 07:10.01 | Winkie | losing your job and £1000 of hardware is worse fyi :( |
| 07:10.21 | x86 | no insurance? |
| 07:10.33 | Winkie | moved in 2 months prior, and it was rented |
| 07:10.35 | brookshire | i would sue! |
| 07:10.46 | Winkie | i was going to but i couldn't prove anything, the monitor was a charred mess |
| 07:10.56 | Winkie | http://www.hardc0re.org.uk/fire/Image(39).jpg |
| 07:10.56 | x86 | no renter's insurance? |
| 07:11.03 | brookshire | how could that not prove ? |
| 07:11.05 | FuriousGeorge | that would be moeny$ |
| 07:11.05 | Winkie | x86: not 2 months in :) |
| 07:11.18 | x86 | pfft |
| 07:11.18 | Winkie | brookshire: there's no identification or proof i didn't set it on fire my self |
| 07:11.20 | x86 | i would have :) |
| 07:11.35 | Winkie | 20/20 hindsight :( |
| 07:11.36 | brookshire | winkie: where are you from? |
| 07:11.40 | Winkie | brookshire: Manchester, UK |
| 07:11.46 | brookshire | oh.. suck for you |
| 07:11.54 | brookshire | in the usa you can still sue without proof :) |
| 07:12.12 | firestrm | Winkie, looks like the keyboard survived.. |
| 07:12.17 | Winkie | still that's frivolous, and it's not 100% their fault either, it had been to 2 LAN events so it could have been me mistreating it |
| 07:12.27 | Qwell | brookshire: and boy do we |
| 07:12.32 | Winkie | firestrm: it did, the smoke damage was irrepreable |
| 07:12.37 | Winkie | or however that's spelt |
| 07:12.44 | Ukyo | k, forwarded teh whole range |
| 07:12.46 | Winkie | it's still used as a spare now and again but your hands get black when you touch it |
| 07:12.46 | brookshire | winkie: but monitors are not suppost to catch fire and burn up |
| 07:12.46 | Ukyo | still , nothing |
| 07:12.53 | brookshire | that's a safty hazard |
| 07:13.01 | brookshire | qwell!! |
| 07:13.18 | FuriousGeorge | Ukyo: externip=> and localnet=> set in sip.conf |
| 07:13.19 | Winkie | brookshire: indeed, but they contain components within which have that potential, and misuse can cause that, so if they claim you misused it, and you can't prove otherwise, you are screwed |
| 07:13.25 | Winkie | well, in sensible countries at least |
| 07:13.30 | Winkie | you want to know something even worse? |
| 07:13.31 | Qwell | brookshire! |
| 07:13.40 | firestrm | Winkie, lol.. thats a security device then.. leaves fingerprints of the person who used it.. |
| 07:13.44 | Winkie | my bed is over my desk, because it's a small room, and it had been delivered that day |
| 07:13.50 | Winkie | if i'd have put it up, the house would have gone down |
| 07:13.53 | Winkie | no question |
| 07:13.56 | brookshire | qwell: i started a secret project :) |
| 07:13.57 | Qwell | over your desk? |
| 07:14.00 | Winkie | or so say the firefighters |
| 07:14.04 | Qwell | brookshire: ooo, tell me, tell me! |
| 07:14.04 | Winkie | Qwell: it's a raised double bed |
| 07:14.11 | Qwell | Winkie: I see |
| 07:14.12 | Winkie | it's a small room and fuck single beds |
| 07:14.23 | FuriousGeorge | Qwell: didnt the rtp range for sip used to be 10K-20K? i hear the wiki is saying like 6K-30K now or something. either way could we just as easily set that to something else? |
| 07:14.25 | Winkie | so yeah, my laziness saved at least the house, and maybe our lives :) |
| 07:14.28 | FuriousGeorge | smaller range specifically |
| 07:14.28 | Qwell | brookshire: super secret, or just regular secret? |
| 07:14.36 | brookshire | qwell: it's a super secret project |
| 07:14.36 | Qwell | FuriousGeorge: use whatever you want |
| 07:14.42 | Qwell | super duper secret? |
| 07:15.03 | brookshire | so super secret it's stealth! |
| 07:15.03 | Qwell | NDA secret? Because those suck |
| 07:15.06 | Qwell | ! |
| 07:15.15 | FuriousGeorge | Qwell: so i could do, say 5062-7999, or even -6062 |
| 07:15.22 | brookshire | more like NSA secret |
| 07:15.23 | Qwell | FuriousGeorge: sure, do whatever |
| 07:15.28 | Qwell | brookshire: zomg! |
| 07:15.48 | FuriousGeorge | Qwell: then why would anyone allow 10K ports to be open? |
| 07:15.54 | Qwell | FuriousGeorge: why not? |
| 07:16.00 | brookshire | for fun! |
| 07:16.15 | brookshire | actually.. just use iax |
| 07:16.16 | FuriousGeorge | Qwell: if you do use a nat/firewall it kinda defeats some of its own purpose |
| 07:16.17 | brookshire | problem solved |
| 07:16.18 | brookshire | :) |
| 07:16.30 | brookshire | 10k ports is a sip problem |
| 07:16.30 | Qwell | firewalls are overrated |
| 07:16.30 | brookshire | lol |
| 07:16.44 | firestrm | use iax.. iax works where sip fails.. |
| 07:16.52 | Qwell | sccp works where iax fails |
| 07:16.54 | FuriousGeorge | nat isnt, unless you use sip or something that doesnt like it |
| 07:16.58 | brookshire | qwell ranch networks has a nice way to firewall all that |
| 07:17.03 | Qwell | brookshire: indeed |
| 07:17.03 | firestrm | Qwell, so true |
| 07:17.13 | Qwell | brookshire: how much do they pay you to say that? :P |
| 07:17.24 | brookshire | zero, but they do! |
| 07:17.29 | Qwell | heh |
| 07:17.42 | Qwell | I met the Ranch guys at ETel |
| 07:17.52 | brookshire | i met them last week :/ |
| 07:17.57 | Qwell | sales drone didn't know very much... |
| 07:18.23 | Qwell | "If there is a bug, who do I send a patch to?" "It's open source, you can fix it yourself." |
| 07:18.24 | firestrm | i have had zero luck with nat/firewall/proxy and SIP.. sip works great on the same subnet of the same network, but falls down anytime you have to cross anything.. |
| 07:18.30 | Qwell | yeah, I'm like...10 steps ahead of you, bud |
| 07:18.50 | FuriousGeorge | firestrm: externip and localnet set in sip.conf? |
| 07:19.03 | firestrm | oh ya.. fought with that for weeks |
| 07:19.13 | FuriousGeorge | no luck? |
| 07:19.16 | firestrm | gave up and had iax working in 5 min, never went back |
| 07:19.20 | FuriousGeorge | lol |
| 07:19.45 | FuriousGeorge | we use those snom sip phones so getting sip to work for remote clients was kinda big for me |
| 07:19.48 | firestrm | im a least resistance kind of person |
| 07:20.22 | firestrm | i'll do it if i have to, but in my case i didnt have to, and easier path became available.. |
| 07:20.28 | FuriousGeorge | i just use ipcop for my gateway, since its easy to administer and supports QoS |
| 07:20.41 | FuriousGeorge | that seems to never cause my * problems |
| 07:22.00 | firestrm | FuriousGeorge, the other reason i dont use sip is that my local internet carrier Fusks with SIP connections to try to convince you to go with their phone service.. but they dont seem to know what iax is. |
| 07:23.45 | firestrm | or they figure that if you are using iax, you know enough that they souldnt mess with you.. |
| 07:23.49 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 07:24.14 | brookshire | heck.. if you know sip.. you know too much |
| 07:24.17 | brookshire | lol |
| 07:24.50 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 07:24.59 | firestrm | brookshire, if you know how to make sip WORK you know too much |
| 07:25.08 | Qwell | brookshire: hell, if you know what sip IS, you know too much |
| 07:27.28 | brookshire | if you know what voip stands for.. you know too much |
| 07:28.12 | brookshire | omg.. so this guys is hovering over a big fan at the olympics |
| 07:29.04 | firestrm | ok.. |
| 07:29.14 | firestrm | whats the punch line? |
| 07:29.30 | brookshire | it's just crazy |
| 07:29.35 | Ahrimanes | he falls through the fan? |
| 07:29.58 | brookshire | i think it has a screen on it or something |
| 07:30.02 | firestrm | well they do need a darwin event |
| 07:30.54 | firestrm | we have that olympic crap coming here next.. yaaa. cant wait.. really.. |
| 07:31.04 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:31.17 | FuriousGeorge | firestrm: isnt that borderline ileagle |
| 07:31.22 | FuriousGeorge | *ilegal |
| 07:31.23 | firestrm | i so want to totally fusk with the olympic advertising police.. |
| 07:31.29 | brookshire | it's fun to watch in the background |
| 07:31.52 | firestrm | FuriousGeorge, Illegal? |
| 07:31.56 | brookshire | while coding |
| 07:32.32 | *** join/#asterisk P0L0 (n=tekn0@62-43-65-175.user.ono.com) |
| 07:33.15 | firestrm | im totally ready to gator any olympic advertising signs that show up here.. |
| 07:33.53 | firestrm | im thinking somthing like the olympic rings, but made with goatsee images.. |
| 07:34.24 | Ahrimanes | firestrm: bad mental image.. bad mental image.. |
| 07:34.28 | firestrm | lol |
| 07:35.00 | brookshire | omg! flashbacks! |
| 07:35.12 | firestrm | hey i miss goatsee.. |
| 07:35.24 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-43.claranet.co.uk) |
| 07:35.37 | brookshire | hehe |
| 07:35.44 | brookshire | someone made a wiki for goatse |
| 07:35.46 | brookshire | http://sam.zoy.org/goatse/ |
| 07:36.04 | firestrm | i once airpowned telus's wireless internet service at the vancouver airport.. every image that loaded was goatsee.cx |
| 07:36.29 | firestrm | it was major ROTFLMAO time. |
| 07:36.30 | FuriousGeorge | yeah, illegal, however you spell it, you might be able to complain to the FCC about that |
| 07:37.05 | firestrm | telus deserves it.. |
| 07:38.29 | firestrm | brookshire, i totally love the goatsee pumpkin.. i know whats going on my doorstep next october |
| 07:38.47 | FuriousGeorge | if someone were to make a decent sip or iax plugin for gaim my life would be better |
| 07:40.24 | FuriousGeorge | im not gonna say "complete" but it would be a step in the right direction |
| 07:41.02 | [av]bani | FuriousGeorge: you mean, you could die happy? |
| 07:41.21 | FuriousGeorge | or at least ContentGeorge |
| 07:41.26 | [av]bani | :) |
| 07:41.30 | [av]bani | it would be a change! |
| 07:41.51 | FuriousGeorge | i've been thinking, i don't wanna have kids, but i wouldnt mind having grandkids, how am i gonna work that |
| 07:42.37 | FuriousGeorge | you know i met a girl who was 22, only 2 years younger than me, and she had never heard of Curious George |
| 07:43.01 | [av]bani | :o |
| 07:43.24 | [av]bani | FuriousGeorge: adopt a pregnant teen? |
| 07:43.38 | FuriousGeorge | [av]bani: brilliant |
| 07:43.42 | [av]bani | \o/ |
| 07:43.45 | [av]bani | <- genius |
| 07:43.57 | Ahrimanes | http://sam.zoy.org/goatse/dooptats.jpg <- omg? |
| 07:44.59 | [av]bani | the thing is, everyone has been flooded with so much goatse.cx that it has no effect anymore |
| 07:45.01 | FuriousGeorge | i wouldnt wanna look at that in the mirror, but great tat nonetheless |
| 07:45.05 | [av]bani | zero, none, nada |
| 07:45.23 | [av]bani | everyones completely desensitized to it now |
| 07:46.54 | *** join/#asterisk propagandhi (n=opera@d220-236-6-192.dsl.nsw.optusnet.com.au) |
| 07:48.38 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:49.15 | FuriousGeorge | i think gaim for linux is no longer crashing ! i can stop talking to you guys in my windows xp vm (aka gaim bootloader) |
| 07:49.40 | FuriousGeorge | whoops, segfault |
| 07:50.05 | brookshire | omg.. you're still here |
| 07:50.20 | FuriousGeorge | who me? |
| 07:50.32 | gaupe | FuriousGeorge: try removing the sound device, set output device to none or something like that |
| 07:50.46 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:51.09 | FuriousGeorge | gaupe: what gaim? that could be it, it seems to happen with irc sign in primarily |
| 07:51.44 | FuriousGeorge | im trying to figure out where the core file it dumped is so i can report it like they asked |
| 07:51.55 | *** join/#asterisk davy_ (n=davy_vdm@87.215.18.236) |
| 07:51.59 | brookshire | just use xchat for irc |
| 07:52.07 | brookshire | gaim for irc stinks |
| 07:52.08 | gaupe | FuriousGeorge: most crashes I've seen seems to be attached to sound |
| 07:52.52 | FuriousGeorge | brookshire: what stinks is havign 11 different clients :) |
| 07:53.01 | brookshire | hah |
| 07:53.04 | brookshire | i only have 2 |
| 07:53.11 | brookshire | once for irc and the others in gaim |
| 07:53.18 | FuriousGeorge | i want my softphone in gaim |
| 07:53.20 | brookshire | which takes care of jabber, aim, and yahoo |
| 07:53.21 | brookshire | lol |
| 07:53.34 | brookshire | furious: phonegaim! |
| 07:53.48 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:53.49 | brookshire | http://www.phonegaim.com/ |
| 07:53.50 | tzafrir | gaim-vv ? |
| 07:53.51 | FuriousGeorge | last i looked at it, it was fugly and buggy, ill look again |
| 07:54.16 | tzafrir | gaim-vv will not be merged into gaim until after gaim 2.0 is released |
| 07:55.29 | FuriousGeorge | that reminds me of one of my issues, anyone come up with a way of sharing sip device status accross multiple * servers? |
| 07:55.40 | FuriousGeorge | im thinking API and BRI stuff? |
| 07:55.48 | FuriousGeorge | that could be a fun project |
| 07:55.56 | FuriousGeorge | phonegaim looks prettier, btw |
| 07:56.36 | x86 | PakiPenguin: hi :) |
| 07:56.40 | PakiPenguin | hey X-Rob |
| 07:56.47 | PakiPenguin | hey x86 |
| 07:56.48 | PakiPenguin | :) |
| 07:58.47 | FuriousGeorge | [av]bani: sip presence accross asterisk servers? any way to do it? |
| 07:58.53 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:59.06 | Mavvie | Ouch ... error while writing audio data: : Broken pipe |
| 07:59.06 | Mavvie | Segmentation fault |
| 07:59.09 | Mavvie | but no core dump. |
| 08:00.50 | Mavvie | aha, -g |
| 08:01.34 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:03.33 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 08:06.34 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 08:11.13 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 08:11.23 | tzafrir | Mavvie, any chance that this is the MOH program? |
| 08:12.43 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 08:13.13 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 08:13.47 | *** join/#asterisk lorinc (n=ang@caracas-1353.adsl.interware.hu) |
| 08:14.56 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 08:27.24 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:29.07 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:31.15 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 08:31.23 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-65.dialup.tiscali.it) |
| 08:31.29 | Gennaro | hi |
| 08:31.40 | Gennaro | some one can say me a thing?!? |
| 08:32.02 | FuriousGeorge | thats what annoyed me about phonegaim, it only works with sipphone |
| 08:32.10 | FuriousGeorge | as in sipphone.com |
| 08:32.58 | FuriousGeorge | Gennaro: im going to bed soon, but ill try to help |
| 08:34.19 | Gennaro | CONFIG_USB_UHCI_HCD=m |
| 08:34.21 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:34.22 | Gennaro | in kernel.. |
| 08:34.33 | FuriousGeorge | ok |
| 08:34.36 | Gennaro | my manual say that i need ztdummy |
| 08:34.42 | FuriousGeorge | yes |
| 08:34.47 | Gennaro | and i need uhci |
| 08:34.54 | Gennaro | loaded before ztd |
| 08:35.03 | FuriousGeorge | maybe not anymore |
| 08:35.08 | Gennaro | in my default kernel it is set on m |
| 08:35.15 | Gennaro | what it mean? |
| 08:35.20 | FuriousGeorge | anyone know for sure? |
| 08:35.32 | Gennaro | i need to mod probe? |
| 08:35.39 | FuriousGeorge | yeah, try it |
| 08:35.49 | Gennaro | modprobe !?!? |
| 08:36.14 | FuriousGeorge | try modprobe zaptel |
| 08:36.15 | Gennaro | what is the name of that module? USB_UCHI_HCD? |
| 08:36.17 | FuriousGeorge | if that doesnt work |
| 08:36.27 | FuriousGeorge | oh |
| 08:36.29 | FuriousGeorge | that |
| 08:36.45 | Gennaro | My manual say HUCIUSB |
| 08:36.45 | FuriousGeorge | modprobe uhciusb? |
| 08:36.56 | FuriousGeorge | i bult into kernel with '*' |
| 08:37.06 | diLLec | is there any way to get the Queue name on the extension of an outgoing Agent call ? |
| 08:37.25 | Gennaro | but isnt in kernel 2.6.11-1 1369_fc4 |
| 08:37.27 | FuriousGeorge | Gennaro: lsmod |
| 08:37.32 | FuriousGeorge | lsoci |
| 08:37.33 | Gennaro | i tried |
| 08:37.38 | Gennaro | i'm going.. |
| 08:37.40 | FuriousGeorge | paste in http://pastebin.ca |
| 08:37.44 | Gennaro | ok |
| 08:37.58 | *** join/#asterisk lithi (n=interp3@67.71.44.206) |
| 08:38.04 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:38.36 | Gennaro | lsoci?!?! |
| 08:38.57 | FuriousGeorge | i meant |
| 08:39.02 | FuriousGeorge | lsmod lspci |
| 08:39.22 | FuriousGeorge | modprobe usb_usci? |
| 08:39.30 | FuriousGeorge | usb_uhci* |
| 08:39.40 | Gennaro | uhci_hcd 35152 0 |
| 08:40.10 | FuriousGeorge | Gennaro: you have to uncomment a line in zaptel Makefile |
| 08:40.14 | lithi | I bought a X101P card, it works fine until there is alot of bandwidth going though the machine and then I get "FXO PCI Master abort" flooding the console and the computer becomes frozen until a hard reboot (stops responding to ping/keyboard). Any ideas? |
| 08:40.18 | FuriousGeorge | take out # before ZTDUMMY |
| 08:40.25 | Gennaro | so i see |
| 08:41.01 | FuriousGeorge | lilo: interrupt problems? try a different pci bay |
| 08:41.28 | FuriousGeorge | lithi: that was to you |
| 08:41.28 | lithi | FuriousGeorge: Just swap the card to a diffrent slot? |
| 08:41.34 | FuriousGeorge | ty i t |
| 08:41.36 | FuriousGeorge | try it |
| 08:41.44 | lithi | FuriousGeorge: Any picticular one? |
| 08:41.52 | FuriousGeorge | then do a cat /proc/interrupts |
| 08:41.58 | FuriousGeorge | try to get that card its own irq |
| 08:42.03 | FuriousGeorge | just a shot in the dark |
| 08:42.22 | FuriousGeorge | check bios and try to make it happen |
| 08:42.49 | FuriousGeorge | Gennaro: try modprobe usb_uhci tell me if that works |
| 08:43.26 | lithi | FuriousGeorge: yea I think its currently sharing IRQ with the SATA controller " 11: 650655 XT-PIC libata, wcfxo" |
| 08:44.56 | Gennaro | ztdummy module uses usb-uhci timers found in USB driver. You must |
| 08:45.13 | lithi | FuriousGeorge: thanks, Ill give it a try |
| 08:45.34 | Gennaro | i tried to modprobe as u said.. shell say nothing.. if i lsmod not loaded. |
| 08:46.04 | FuriousGeorge | yeah, im reading that you must load that then load ztdummy |
| 08:46.17 | heroine | hi |
| 08:46.24 | Gennaro | yeah thats true.. |
| 08:46.29 | FuriousGeorge | Gennaro: u sure u have uhci not ohci? |
| 08:46.34 | Gennaro | but module isnt usb_uchi |
| 08:46.54 | FuriousGeorge | Gennaro: what distro of linux you use |
| 08:46.55 | FuriousGeorge | ? |
| 08:46.59 | Gennaro | FC4 |
| 08:47.28 | Gennaro | i'm going to try to uncomment #zt but i'll fail. |
| 08:47.41 | FuriousGeorge | dont you have a usb mouse or anything to know if usb is working? are you sure you have uhci? |
| 08:47.52 | vgster | thought you didnt need usb with 2.6 kernel? |
| 08:48.18 | FuriousGeorge | vgster: i thought that too, but i cant find where it says so on the wiki |
| 08:48.18 | Gennaro | i an last installation that i aborted i could hear voice menu but not music_on_hold |
| 08:48.22 | FuriousGeorge | most articles say opposite |
| 08:48.37 | Gennaro | usb works.. |
| 08:48.40 | FuriousGeorge | thats something different Gennaro |
| 08:49.07 | FuriousGeorge | Gennaro: what makes you think you have uhci then? lspci -v and check it over |
| 08:49.36 | vgster | grrr MOH... |
| 08:49.59 | FuriousGeorge | Gennaro: make mpg123 in * source then cd mpg123 |
| 08:50.03 | FuriousGeorge | then make install |
| 08:50.16 | Gennaro | i already done |
| 08:50.19 | vgster | is it mp3's you are playing? |
| 08:50.22 | FuriousGeorge | Gennaro: you can use zaprtc to get it working its in the bristuff patch |
| 08:50.22 | Gennaro | and works... |
| 08:50.25 | FuriousGeorge | easy to install |
| 08:50.55 | vgster | On kernel version 2.6 ztdummy uses internal high-resolution kernel timer and does not require any USB. |
| 08:50.59 | vgster | http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 08:51.02 | Gennaro | i done lspci |
| 08:51.06 | Gennaro | and i see usb |
| 08:51.16 | Gennaro | correct model of my chipset |
| 08:51.20 | Gennaro | 845e |
| 08:51.25 | FuriousGeorge | uhci? i think nvidia chipsets usually use ohci |
| 08:51.37 | FuriousGeorge | but listen to what vgster said |
| 08:51.51 | Gennaro | just a minut pastebin |
| 08:51.53 | vgster | shouldnt need it, have you made the mod? |
| 08:53.00 | Gennaro | http://pastebin.ca/43327 |
| 08:53.19 | *** join/#asterisk astar` (n=astar@ANantes-154-1-75-158.w86-199.abo.wanadoo.fr) |
| 08:53.31 | vgster | is the problem the mod not loading? |
| 08:53.48 | Gennaro | i suppose of yes vgster |
| 08:54.03 | FuriousGeorge | vgster: i think hes just building it now |
| 08:54.07 | Gennaro | isnt a modules... |
| 08:54.25 | vgster | isnt a mod? |
| 08:54.27 | FuriousGeorge | Gennaro: did you uncomment the line and build the module? are you using kernel 2.6? |
| 08:54.43 | FuriousGeorge | then jut modprobe ztdummy after you make the module |
| 08:54.44 | vgster | uncomment the line in zaptels makefile and make linux26 |
| 08:54.53 | Gennaro | ahhh... |
| 08:54.57 | FuriousGeorge | oh thats right |
| 08:55.00 | FuriousGeorge | its linux26 |
| 08:55.03 | Gennaro | cau u explane me.. |
| 08:55.04 | Gennaro | ?? |
| 08:55.12 | Gennaro | 2.6.11-1 |
| 08:55.19 | FuriousGeorge | gzip and tar zaptel sournce |
| 08:55.29 | Gennaro | ok |
| 08:55.31 | FuriousGeorge | nano - w Makefile in zaptel source |
| 08:55.42 | Gennaro | i'm going.. |
| 08:55.49 | FuriousGeorge | take # out from line about ZT_DUMMY |
| 08:56.12 | FuriousGeorge | you can CTRL+W and search for ZT_DUMMY (at least on US keymap) |
| 08:56.14 | vgster | yes find the MODULES:= line |
| 08:56.21 | vgster | and ztdummy has a # before it |
| 08:57.28 | vgster | actually looking at my test system its built ztdummy without being asked to |
| 08:57.33 | Gennaro | just a question |
| 08:57.38 | tzafrir | there should be no need for that |
| 08:57.44 | FuriousGeorge | lol |
| 08:57.46 | tzafrir | It should be done automatically on 2.6 |
| 08:57.49 | vgster | yes it is |
| 08:57.51 | Gennaro | Voip telephony with asterisk |
| 08:57.54 | Gennaro | sayd.. |
| 08:57.59 | FuriousGeorge | its OLD |
| 08:58.07 | FuriousGeorge | ~thebook |
| 08:58.09 | jbot | rumour has it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 08:58.14 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 08:58.29 | Gennaro | the ztdummy driver is included with the MEPIS asterisk source |
| 08:58.40 | Gennaro | it is not compiled by default |
| 08:58.51 | Guest^DJ | hi, does * keep a log file when i do asterisk -vvvvvc |
| 08:58.55 | FuriousGeorge | listen to us, not old book |
| 08:58.59 | vgster | compile the zaptel drivers and that should build it anyway |
| 08:59.04 | FuriousGeorge | old book got you in this trouble |
| 08:59.07 | tzafrir | ztdummy is included with zaptel, actually. And is compiled by default with recent zaptel |
| 08:59.18 | *** part/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 08:59.20 | FuriousGeorge | gotta run |
| 08:59.28 | vgster | run forrest |
| 08:59.38 | Gennaro | to include ztdubby in asterisk edit make file in usr/src/zaptel and remove # in front of ztummy.o |
| 08:59.49 | tzafrir | Gennaro, get a more recent zaptel-source package... |
| 08:59.50 | vgster | no you actually dont need to |
| 08:59.59 | vgster | yes get a more recent zaptel |
| 09:00.05 | vgster | i cant keep up |
| 09:00.29 | Gennaro | oh.. |
| 09:00.32 | Gennaro | i'm going.. |
| 09:00.41 | Gennaro | i skip this.. |
| 09:00.42 | vgster | are you using * 1.2? |
| 09:00.49 | Gennaro | ? is right? |
| 09:00.57 | Gennaro | yes the last stable relase |
| 09:01.07 | vgster | ok and do you have the latest zaptel? 1.2.4? |
| 09:01.25 | Gennaro | 1.2.4 |
| 09:01.34 | *** join/#asterisk veepster (i=veepster@c-69-143-163-86.hsd1.va.comcast.net) |
| 09:01.43 | vgster | ok so if you go into the zaptel directory and do a make linux26 it should build the mods |
| 09:02.02 | Gennaro | ah.. |
| 09:02.12 | vgster | that should 'hopefully' build the mods |
| 09:02.19 | vgster | check for ztdummy in the mod list |
| 09:02.20 | Gennaro | i havent jet untarred file, i'm going.. |
| 09:02.24 | vgster | ok |
| 09:02.58 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:05.02 | *** part/#asterisk propagandhi (n=opera@d220-236-6-192.dsl.nsw.optusnet.com.au) |
| 09:06.20 | tzafrir | Gennaro, what kernel version do you use? |
| 09:06.29 | tzafrir | uname -r |
| 09:07.17 | vgster | 2.6.11.1 i think he said earlier |
| 09:07.31 | vgster | on FC4 |
| 09:07.37 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 09:09.06 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 09:09.12 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 09:11.22 | Guest^DJ | hi, does * keep a log file when i do asterisk -vvvvvc. screen scroll too fast and i cant read |
| 09:13.17 | vgster | check /var/log/asterisk/full |
| 09:13.42 | vgster | could be massive depending on what you have set to log |
| 09:14.00 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 09:15.08 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 09:16.00 | Guest^DJ | vgster: no suck file/directory |
| 09:17.56 | vgster | really |
| 09:18.13 | vgster | do you have a /var/log/asterisk directory at all? |
| 09:18.32 | Guest^DJ | yes, :) |
| 09:18.41 | vgster | have you checked /etc/asterisk/logger.conf? |
| 09:18.44 | vgster | see what its logging |
| 09:19.01 | Guest^DJ | i have event_log, messages, queue_log |
| 09:19.08 | vgster | check messages |
| 09:19.40 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:19.51 | iDunno | turn on full log ;) |
| 09:20.00 | vgster | full log is cool |
| 09:20.07 | Guest^DJ | iDunnoL how do i do that ? |
| 09:20.07 | festr_ | hello, is it possible in queue for member SIP not to ring SIP when it is already in use by queue? |
| 09:20.08 | vgster | quality reading for many months |
| 09:20.15 | vgster | check /etc/asterisk/logger.conf |
| 09:22.52 | Guest^DJ | hey, is cool thats what i want |
| 09:23.00 | vgster | ok |
| 09:23.40 | Gennaro | sorry waht sould i do with nano command?!? |
| 09:23.53 | Guest^DJ | tks |
| 09:24.31 | vgster | Gennaro: what stage are you at now? |
| 09:24.38 | Gennaro | oh... |
| 09:24.42 | vgster | have you unpacked zaptel? |
| 09:24.50 | Gennaro | i unpacked file in under src |
| 09:25.09 | vgster | ok, you dont need to edit anything just go into the zaptel directory and do |
| 09:25.11 | vgster | make linux26 |
| 09:25.15 | vgster | they should build |
| 09:25.19 | Gennaro | oh.. |
| 09:25.26 | Gennaro | i'm going.. |
| 09:25.29 | vgster | you should ztdummy get created |
| 09:25.51 | vgster | i didnt hjave any trouble with FC2 and 3 so i would assume FC4 would be fine too |
| 09:26.58 | Gennaro | where had u found it? |
| 09:27.14 | vgster | has zaptel built? |
| 09:27.24 | Gennaro | yes |
| 09:27.28 | vgster | it should be listed on the screen as it builds them |
| 09:27.34 | vgster | if you do a make install |
| 09:27.40 | vgster | then a modprobe ztdummy |
| 09:28.14 | Gennaro | i wrote make clean |
| 09:28.22 | Gennaro | ; make linux26 |
| 09:28.31 | Gennaro | now i'm going to make install |
| 09:28.57 | vgster | ok |
| 09:29.13 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:29.56 | Gennaro | i'm going for the others.. |
| 09:30.01 | vgster | others? |
| 09:31.55 | Gennaro | Notice: Configuration file is /etc/zaptel.conf/ line 0: Unable to open master device '/dev/zap/ctl' / 1 error(s) detected // FATAL: Error running install command for ztdummy |
| 09:32.34 | vgster | you will probably need to config /etc/zaptel.conf for ztdummy shouldnt be a big job but at least you have ztdummy installed |
| 09:33.22 | Gennaro | i cant understand |
| 09:33.51 | vgster | ok let me have a look for some docs |
| 09:34.04 | Gennaro | i dont ask too... |
| 09:34.43 | vgster | do you have /etc/zaptel.conf |
| 09:34.46 | Gennaro | i'm tinking that to let work asterisk u must buy all Digium wants... |
| 09:35.18 | vgster | no you dont, do you have a a zaptel.conf file in /etc |
| 09:35.21 | Gennaro | ok i vied |
| 09:35.28 | vgster | and its there? |
| 09:35.57 | vgster | let me try on my system |
| 09:37.06 | vgster | 1st time load of ztdummy says that |
| 09:37.09 | Gennaro | what i need to search in this configuration file!?!? |
| 09:37.15 | vgster | do a lsmod and see if ztdummy is loaded |
| 09:38.11 | Gennaro | ztdummy is loades and use 0 |
| 09:38.17 | vgster | thats ok |
| 09:38.28 | vgster | is a module called zaptel loaded too? |
| 09:38.35 | vgster | and crc_ccitt |
| 09:38.44 | Gennaro | yes and uses ztdummy |
| 09:39.07 | vgster | done then |
| 09:39.10 | vgster | its in |
| 09:39.12 | Gennaro | ok |
| 09:39.19 | Gennaro | very good.. |
| 09:39.26 | vgster | that error it has no channels |
| 09:39.38 | vgster | doing a ztcfg -vv will show no channels |
| 09:39.39 | Gennaro | but now how can i do to let system load every times this modules?!? |
| 09:40.30 | vgster | put modprobe ztdummy in /etc/rc.local on fedora |
| 09:40.36 | Gennaro | 1 channel |
| 09:42.04 | vgster | ztcfg -vv shouldnt show any channels as none should configured (you dont have any) |
| 09:42.06 | Gennaro | ZT_CHANCONFIG failed on channel 1: No such device or address (6 |
| 09:42.17 | Gennaro | thats true.. |
| 09:42.40 | Gennaro | i dont have any Card at the moment.. |
| 09:42.48 | vgster | if the module sis loaded then thats enough, i dont know too much about ztdummy as i dont use it so i dont know what else to recommend |
| 09:43.02 | Gennaro | sorry any information i need to do my study at home |
| 09:43.26 | Gennaro | so i see on digium site that is an starter kit. |
| 09:43.36 | Gennaro | do u know how it work?!? |
| 09:44.10 | vgster | what starter kit? I have no idea. You shouldnt need to buy anything really, use www.voip-info.org and mailing lists |
| 09:44.46 | Gennaro | i'd like the idea of installation support |
| 09:45.06 | Gennaro | beacuse i dont want risk money,, |
| 09:45.21 | Gennaro | in my country ups ask 80$!!! |
| 09:45.57 | Gennaro | i down belive is good to spend 100$ for a fxo x100p clone... |
| 09:46.17 | Gennaro | i'm going to test sip and iax channel.. |
| 09:46.20 | vgster | ok |
| 09:46.37 | vgster | you should be able to make calls between sip handsets |
| 09:47.00 | vgster | i need to do some work now |
| 09:47.04 | vgster | have a go |
| 09:47.08 | Gennaro | ok tanx. |
| 09:47.22 | Gennaro | see you... realy tanx.. |
| 09:48.00 | *** join/#asterisk lithi (n=interp3@67.71.44.219) |
| 09:48.06 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:48.31 | lithi | FuriousGeorge: 1 hour later with -1 network card I now have a non sharing IRQ x101p card :) |
| 09:48.47 | lithi | FuriousGeorge: and it seems to have solved the problem |
| 09:51.58 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:56.37 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:59.24 | vgster | anyone had fax receive working either through spandsp or hylafax? |
| 10:01.15 | *** join/#asterisk TallAndy (i=AndyD@82.163.6.217) |
| 10:01.47 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:02.53 | TallAndy | Is there a way of getting this message back into a variable "> Channel SIP/gamma-0de0 was never answered."? Its returned from the asterisk console, i'm using phpagi |
| 10:04.51 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 10:10.53 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 10:11.23 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:12.32 | *** join/#asterisk baahabaa (n=bbabahaa@59.144.23.226) |
| 10:13.05 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:15.10 | *** join/#asterisk MimiFleX (n=MFX@252.57.101-84.rev.gaoland.net) |
| 10:15.23 | MimiFleX | hi |
| 10:19.04 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 10:19.06 | MimiFleX | I'm searching a lowprice card to connect 2 S0 on my asterisk PBX, can someone give me some advices ? I saw the Junghanns quadBRI card but I'm not sure that's the best choice for my needs, and price can double depending the reseller.. |
| 10:19.49 | clive- | MIMIFLEX USE A HFC-S CARD FOR 1 X BRI |
| 10:19.59 | clive- | ops caps |
| 10:20.11 | clive- | oops |
| 10:20.13 | MimiFleX | clive-, ok for HFC-S chip |
| 10:20.22 | MimiFleX | seems to be the best :) |
| 10:20.30 | clive- | yes, with bristuff software |
| 10:21.10 | MimiFleX | Do you know some OEM HFS-S cards ? |
| 10:21.17 | MimiFleX | and resellers |
| 10:23.12 | clive- | mimiflex look for "billion" |
| 10:23.27 | MimiFleX | ok thanks |
| 10:23.32 | clive- | I personally threw my hfc-s cards out in favour of an eicon server card |
| 10:23.37 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 10:23.44 | MimiFleX | not the same price |
| 10:23.53 | clive- | yes, but way better |
| 10:24.04 | MimiFleX | for echo cancelation ? |
| 10:24.12 | clive- | yes |
| 10:25.32 | clive- | and voice quality |
| 10:25.39 | MimiFleX | ok |
| 10:26.13 | clive- | and more ports |
| 10:26.35 | fourcheeze | anyone got func_odbc.so to return a value? |
| 10:26.47 | fourcheeze | I'm failling to do that |
| 10:26.49 | fourcheeze | :-( |
| 10:26.56 | fourcheeze | query executes - no return value |
| 10:26.58 | vgster | anyone had fax receive working either through spandsp or hylafax? |
| 10:31.18 | *** join/#asterisk davidw (n=davidw@apache/committer/davidw) |
| 10:31.38 | davidw | hi, this is wack: ;- is not a valid comment/otherwise causes problems, but ; - works just fine |
| 10:31.41 | davidw | no bueno:-) |
| 10:32.07 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:32.14 | davidw | our asterisk dude just spent two hours chasing that down |
| 10:32.18 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:33.36 | webmind | heh :) |
| 10:34.23 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 10:39.14 | davidw | ah... it's this 'meta comment thing'... |
| 10:41.03 | davidw | definitely 'wack' :-/ |
| 10:42.27 | MGSsancho | is it mission critical? |
| 10:43.00 | Mavvie | scary.... AAPT reported one PRI in blocked state, but asterisk thought it was all happy. |
| 10:43.08 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 10:43.11 | Mavvie | an unload/reload of the zaptel driver fixed it. |
| 10:43.41 | MGSsancho | lol |
| 10:44.21 | fourcheeze | anyone know the secret to returning a value from func_odbc ? |
| 10:45.25 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 10:47.43 | davidw | mmmm no |
| 10:47.49 | davidw | ;-- starts a multi-line comment |
| 10:47.57 | MimiFleX | clive-, Eicon DIVA Server BRI-2M 2.0 - ISDN terminal ? |
| 10:48.02 | MGSsancho | since i didnt compile, i dunno and cant halp you L( |
| 10:48.18 | MGSsancho | :( |
| 10:48.30 | davidw | harumph...\ |
| 10:50.27 | MGSsancho | lol |
| 10:55.24 | *** join/#asterisk sergey (n=sergey@gw3-a22.iks.ru) |
| 10:55.59 | *** join/#asterisk _deg_ (n=deg@201.22.12.137.adsl.gvt.net.br) |
| 10:57.14 | *** join/#asterisk my007ms (n=my@213.158.171.162) |
| 10:57.18 | my007ms | hello all |
| 10:58.55 | my007ms | did any one get not worked digium card |
| 10:59.10 | my007ms | i try with my card all i can do |
| 10:59.21 | my007ms | i think it's not working card |
| 10:59.56 | my007ms | i try my new tdm2424E card all over |
| 11:00.05 | my007ms | i can not get it work |
| 11:00.21 | RoyK | call digium :) |
| 11:00.27 | RoyK | they'll help you out |
| 11:00.30 | RoyK | when they wake up |
| 11:00.36 | my007ms | they don't have phone number |
| 11:00.37 | my007ms | :) |
| 11:00.51 | RoyK | they do indeed |
| 11:00.59 | Mavvie | still wonder who is sending these congestion results on my PRI if AAPT says that the card is stuck... |
| 11:01.04 | Mavvie | must be a D-channel thing |
| 11:01.15 | my007ms | ok go to digium.com |
| 11:01.35 | my007ms | and try to find phone number and if u find it let me know :D |
| 11:01.56 | Mavvie | my007ms: found it. now your turn. |
| 11:02.07 | newl | click about, then contact or services->tech support ;) |
| 11:02.28 | newl | rehash of a conversation from a couple of days ago heh |
| 11:02.48 | MGSsancho | and send $30 to mgssancho@yahoo.com via paypal :) |
| 11:03.01 | MGSsancho | and i will give you unlimited e-hugs |
| 11:03.06 | my007ms | i call them no answer |
| 11:03.33 | MGSsancho | lol |
| 11:03.42 | MGSsancho | its 1PM where you are |
| 11:03.54 | MGSsancho | in the US, its between 3-5am |
| 11:03.59 | MimiFleX | haha |
| 11:04.01 | RoyK | my007ms: as i said wait until they wake up :) |
| 11:04.37 | my007ms | but i think they must have 24/7 support |
| 11:04.41 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:05.44 | MGSsancho | and theres no #digium chan either =/ |
| 11:06.27 | MGSsancho | speaking of 3am |
| 11:06.41 | MGSsancho | have a 7:15am class in 4 hours |
| 11:06.45 | MGSsancho | better goto bed |
| 11:06.49 | MGSsancho | ga'night |
| 11:08.10 | my007ms | did they have test team ;) |
| 11:08.41 | fourcheeze | anyone here using func_odbc and successfully returning a value from the DB? |
| 11:10.52 | fourcheeze | I can execute queries but nothing comes back |
| 11:11.21 | fourcheeze | is there another way (apart from AGI) to execute a query on a DB? |
| 11:11.26 | fourcheeze | (and get a result) |
| 11:11.38 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:14.39 | Mavvie | Core was generated by `asterisk -g'. |
| 11:15.05 | Guest^DJ | hi guys, i did a ztcfg -vv and found 31 channels, but i am getting a No channel type registered for 'Zap' |
| 11:15.11 | Guest^DJ | pls help |
| 11:16.34 | Guest^DJ | and zttool show status as OK |
| 11:18.10 | Mavvie | Guest^DJ: what is the dial command? |
| 11:19.08 | Guest^DJ | exten => s,1,Dial(Zap/g1/${EXTEN},30) |
| 11:19.39 | Guest^DJ | am now using 1.2.4 , any difference? |
| 11:20.45 | RoyK | zoa: ping |
| 11:21.07 | Mavvie | Guest^DJ: does it show up in "show channeltypes" ? |
| 11:21.21 | RoyK | ~seen zoa |
| 11:21.28 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 6d 17h 14m 3s ago, saying: '1000 interrupts per second are ok'. |
| 11:22.25 | Guest^DJ | Mavvie: did a 'show channels verbose' and show nothing |
| 11:22.35 | Mavvie | that's not what I asked for. |
| 11:22.49 | Guest^DJ | pls elaborate |
| 11:22.58 | Mavvie | <Mavvie> Guest^DJ: does it show up in "show channeltypes" ? |
| 11:23.18 | *** join/#asterisk blkremedy (n=ur3rdeye@083M13.oasis.mediatti.net) |
| 11:24.11 | Guest^DJ | u mean 'show channel' , no it does now show up |
| 11:24.43 | Mavvie | <Mavvie> Guest^DJ: does it show up in "show channeltypes" ? |
| 11:25.08 | Guest^DJ | Mavvie: please. i do not understand |
| 11:25.26 | Mavvie | which part is the problem? |
| 11:25.44 | Guest^DJ | * wont dial out |
| 11:25.51 | Mavvie | no, with my question? |
| 11:26.07 | Guest^DJ | what is 'show channel types' ? |
| 11:26.10 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-91-96.vic.westnet.com.au) |
| 11:26.14 | Mavvie | without the space. |
| 11:27.12 | Guest^DJ | yes, it does show stuff in show channeltypes |
| 11:27.32 | Mavvie | beautiful. now it's time for me to bring this little boy to bed, and I'm going to crash after that too. |
| 11:27.54 | Guest^DJ | where should i be looking at ? |
| 11:30.46 | *** join/#asterisk ibob63 (n=hp@82.111.123.141) |
| 11:30.47 | blkremedy | can anyone tell me how the spa3000 stacks up or compare to the tdm400p in terms of over all quality of service? |
| 11:32.53 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:33.11 | blkremedy | I can not get an answer on this. I've seached the internet but, can not find a good comparison. Bottom line; which is better? |
| 11:38.07 | clive- | blkremedy its apple and oranges |
| 11:39.29 | [Airwolf] | Can anyone tell me if there are any nice IAX phones ? |
| 11:39.35 | [Airwolf] | hardphones that is |
| 11:40.45 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:40.48 | blkremedy | so neither is better or worse |
| 11:44.13 | *** join/#asterisk fulgas (n=fulgas@207.226.175.2) |
| 11:50.44 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 11:58.18 | RoyK | hmmmmmmmm |
| 11:58.33 | RoyK | seems there's something really bad with 1.2.4 |
| 11:58.44 | trixter | what did you find? |
| 11:59.03 | trixter | I have had to do a few workarounds but nothing that major |
| 11:59.04 | Mavvie | congestion? |
| 11:59.07 | Mavvie | channel mixing? |
| 11:59.28 | RoyK | no incoming audio if dialling from SIP behind NAT |
| 11:59.35 | RoyK | same config on 1.0 works |
| 12:00.13 | trixter | I have had that problem |
| 12:00.27 | trixter | now that you mention it, didnt have a compare against 1.0 though so I wasnt sure why it was happening |
| 12:00.51 | RoyK | did you solve it? |
| 12:00.59 | trixter | I get a couple frames then nothing.. literally less than 0.5 seconds |
| 12:01.07 | trixter | well I worked around it I used iax2 |
| 12:01.12 | RoyK | :P |
| 12:01.15 | clive- | mavvie is "congestion" a known / common trouble ? |
| 12:01.17 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-43.claranet.co.uk) |
| 12:01.30 | trixter | its *not* a nat issue, and I only noticed it when > 20 or so sip channels concurrent off that box |
| 12:01.37 | trixter | which made me think it was thread related |
| 12:02.07 | Mavvie | clive-: on a phone network? no. on my asterisk box where I have more than enough B channels leftover? in the afternoons. |
| 12:03.01 | clive- | mavvie , no I am sometimes having a wacky thing where an * box which goes iax-sip sometimes gives a congestion to the iax side without even trying the sip side |
| 12:03.12 | *** part/#asterisk sergey (n=sergey@gw3-a22.iks.ru) |
| 12:03.18 | clive- | like auto-congestion |
| 12:03.26 | Mavvie | oh, towards SIP I have it now and then, but that is just because the reception phones are overloaded. |
| 12:06.20 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 12:06.35 | *** part/#asterisk cuco (n=diego@local.xorcom.com) |
| 12:06.49 | fourcheeze | is there a way for * a dial plan to call an ordinary sip url? |
| 12:07.06 | Mavvie | dial(sip/foo@youraddress) |
| 12:07.22 | fourcheeze | that's not a url |
| 12:07.30 | Mavvie | oh. |
| 12:07.52 | fourcheeze | sip:foo@address |
| 12:07.54 | fourcheeze | :-) |
| 12:08.07 | Mavvie | should be. |
| 12:08.11 | Mavvie | show application dial |
| 12:08.19 | fourcheeze | doesnt help |
| 12:08.26 | Mavvie | <PROTECTED> |
| 12:08.49 | fourcheeze | I wonder why dial() does use a url |
| 12:08.56 | fourcheeze | doesn't |
| 12:08.59 | fourcheeze | even |
| 12:11.23 | fourcheeze | ok, related questio |
| 12:11.23 | fourcheeze | n |
| 12:11.38 | fourcheeze | when client is behind nat |
| 12:11.55 | fourcheeze | why does asterisk sometimes store the proivate IP number as the address? |
| 12:12.05 | fourcheeze | and sometimes not |
| 12:12.17 | Mavvie | it's that what the client gives? |
| 12:12.43 | fourcheeze | if you do show peer X I thought that Reg. Contact should give you a url to contact the peer |
| 12:12.49 | fourcheeze | sip:20060033@192.168.0.2:5060 |
| 12:12.55 | fourcheeze | that doesn't really help me |
| 12:13.30 | fourcheeze | if, (for instance) I want to call the peer from a different * |
| 12:13.47 | fourcheeze | however * stores the IP address of the Peer |
| 12:13.52 | Mavvie | <PROTECTED> |
| 12:13.56 | Mavvie | <PROTECTED> |
| 12:13.59 | fourcheeze | yes |
| 12:14.09 | fourcheeze | other times it doesn't do that though |
| 12:14.17 | Mavvie | true. |
| 12:14.19 | fourcheeze | how does one dial a SIP url like that? |
| 12:14.24 | Chotaire | since this issue has still not been fixed and my bug report has been deleted, let me retry like a damn lamer, and ask the following... since update from 1.0.3 to 1.2.4 dtmf will no longer be detected on incoming SIP calls.. any idea? |
| 12:14.36 | Mavvie | even with the same kind of phones |
| 12:15.00 | fourcheeze | Mavvie: yes - I don't understand |
| 12:15.17 | fourcheeze | Chotaire: I take it you hvae looked into the inband/info/rfc setting? |
| 12:15.36 | Mavvie | only thing different they have is the grandstream OS version |
| 12:15.50 | fourcheeze | Mavvie: do you know how to initiate a dial to a url on a private IP number? |
| 12:16.26 | fourcheeze | Chotaire: also define "incoming sip calls" |
| 12:17.24 | trixter | does anyone know roughly how much a ds3->t1 mux costs? |
| 12:18.15 | trixter | its looking like its only $650 which isnt bad |
| 12:19.16 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
| 12:19.16 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 12:19.17 | *** join/#asterisk newl (n=newlook@203-59-100-129.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 12:19.17 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) [NETSPLIT VICTIM] |
| 12:21.21 | RoyK | trixter: doing sip/sip gatewaying, turning off canreinvite helped a lot :) |
| 12:22.11 | Chotaire | re |
| 12:22.19 | Chotaire | sorry, I was getting me a pack of ciggies. |
| 12:23.17 | Chotaire | fourcheeze: incoming call = someone calls me from PSTN -> SIP -> asterisk.... yes, I have checked that stuff and I am unwilling to change anything (doesn't help anyway). everything was perfect with 1.0.3, now with 1.2.4 dtmf will not be detected. |
| 12:23.38 | Chotaire | it is likely something is not ok with the sip header, and I frankly don't care since my provider will a) change NOTHING and b) it worked perfect before. |
| 12:23.50 | trixter | royk: I dont think that helped with me, that was how it was to start with :/ |
| 12:24.08 | trixter | I may play with it again, but ... I got a frame or two, enough to hear something when the call started then it dies |
| 12:24.09 | Chotaire | I use(d) rfc2833 |
| 12:24.21 | trixter | 1.2.4->1.2.4 too |
| 12:25.14 | Chotaire | oh and yes, I have already submitted a bug report to bugs.digium.com and it was deleted. still not fixed, no mentioning, no comments and nobody contacted me. |
| 12:25.36 | Chotaire | and I was by far not the only person with this problem. the only thing that got changed after my bug report is that sip info debugging was improved. |
| 12:26.03 | Chotaire | kram asked me to contact him for more info, I did and I never got a reply. |
| 12:26.37 | Chotaire | that was right after 1.2.0 release.. now this stuff still doesn't work... |
| 12:26.59 | Chotaire | great.... so I will downgrade to 1.0.3 once again?` |
| 12:27.16 | fourcheeze | Chotaire: so the pstn stuff is within asterisk? |
| 12:27.23 | Chotaire | no |
| 12:27.31 | fourcheeze | so it's just another incoming sip call? |
| 12:27.48 | Chotaire | guy calls a real phone number, which is a SIP provider and ends up in my * thru sip |
| 12:27.50 | my007ms | digium is so bad in support stuuf |
| 12:27.52 | Chotaire | so he is using PSTN |
| 12:27.53 | my007ms | digium is so bad in support stuf |
| 12:28.10 | fourcheeze | ok |
| 12:28.17 | fourcheeze | so either |
| 12:28.28 | fourcheeze | the dtmf is lost between the pstn and the sip provider |
| 12:28.30 | trixter | asterisk rfc2833 support isnt totally rfc compliant |
| 12:28.34 | Chotaire | nothing is lost. |
| 12:28.38 | fourcheeze | ok |
| 12:28.40 | Chotaire | it worked perfectly just 5 minutes before. |
| 12:28.44 | Chotaire | when I was still using 1.0.3 |
| 12:28.51 | fourcheeze | ok |
| 12:28.54 | Chotaire | (because this shit got broken soon afterwards and never got fixed) |
| 12:29.08 | fourcheeze | so it's likely that your sip provider and you are talking different dtml |
| 12:29.14 | fourcheeze | dtmf |
| 12:29.16 | fourcheeze | even |
| 12:29.29 | trixter | it also does some stuff that can lead to poor detection, in rfc2833 you have 3 start and 3 end packets, asterisk sends all starts at the same time instead of spreading them out same with end packets (3 packets are sent to survive packet loss). the sending of 'yeah I am still pressing the button' isnt really dont right either |
| 12:29.45 | fourcheeze | Chotaire: have you tried switching over to info ?> |
| 12:29.49 | Chotaire | yes |
| 12:30.01 | trixter | sip-info only works if the provider supports it too |
| 12:30.09 | Chotaire | not even inband works anymore. |
| 12:30.11 | fourcheeze | which sip provider? |
| 12:30.19 | Chotaire | 1und1 |
| 12:30.36 | trixter | inband should only work if you are doing g.711, genereally speaking it doesnt work with any other codecs |
| 12:30.49 | Chotaire | I use alaw or ulaw |
| 12:31.03 | fourcheeze | Chotaire: what about a random connection from any other sip client - does that work? |
| 12:31.14 | Chotaire | sip directly will work |
| 12:31.19 | Chotaire | but not pstn->sip |
| 12:31.25 | fourcheeze | well, that's what your sip provider are sending - just sip |
| 12:31.48 | RoyK | ~lart zoa |
| 12:31.59 | fourcheeze | if you can't believe that it's broken their end, then it must be broken your end |
| 12:32.12 | fourcheeze | so you have to find out what's different about the sip they are sending you |
| 12:32.35 | Chotaire | what was the debug command again? |
| 12:32.42 | Chotaire | I remember it was something in the header. |
| 12:32.43 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:32.45 | fourcheeze | if you switch them over to info at least you will see the incoming INFO headers |
| 12:32.53 | Chotaire | we had that before and since it got deleted, I don't find it anymore on bugs.digium.com |
| 12:33.14 | Chotaire | whatever.. it worked perfectly with 1.0.3 |
| 12:33.14 | fourcheeze | you mean "sip debug ip ip.number" ? |
| 12:33.33 | Chotaire | no I think it was something different than that. |
| 12:33.49 | fourcheeze | what do you want to do? |
| 12:33.50 | Chotaire | but let's keep one thing in mind... they will not change anything. |
| 12:34.01 | Chotaire | and they wouldn't have to anyway, it worked all perfectly with 1.0.3 |
| 12:34.09 | Chotaire | so the problem is with asterisk being too strict now or whatever. |
| 12:34.30 | Chotaire | is there any settings, compile-time-options, whatever I could change to bring back the old behaviour? |
| 12:34.31 | fourcheeze | if 1und1 is that same as 1and1 over here - I don't hold out any hope |
| 12:34.58 | Chotaire | yup it is |
| 12:36.03 | fourcheeze | are you stuck with them? |
| 12:36.21 | Chotaire | yes |
| 12:36.27 | fourcheeze | could you try out another provider in case? |
| 12:36.31 | Chotaire | anyway, this time I will not be downgrading.. I will bother the one responsible for chan_sip to death until this issue has finally been fixed. |
| 12:37.13 | fourcheeze | in your account with 1und1 can you change your dtmf type? |
| 12:37.41 | Chotaire | let me check I don't think so |
| 12:37.56 | fourcheeze | if so try using INFO |
| 12:38.05 | Chotaire | nope |
| 12:38.31 | Chotaire | I will change everything to info now |
| 12:39.00 | fourcheeze | it's not going to help your end unless you can change it at theirs too |
| 12:41.00 | fourcheeze | at least it is unlikely to help |
| 12:41.15 | Chotaire | m=audio 19058 RTP/AVP 8 0 2 99 18 110 |
| 12:41.29 | Chotaire | s=Cisco SDP 0 |
| 12:41.43 | Chotaire | a=X-cap: 1 audio RTP/AVP 110 |
| 12:41.55 | Chotaire | a=X-cpar: a=rtpmap:110 X-NSE/8000 |
| 12:41.55 | Chotaire | a=X-cpar: a=fmtp:110 192-194,200-202 |
| 12:41.55 | Chotaire | a=X-cap: 2 image udptl t38 |
| 12:44.21 | Chotaire | inband is the only one that works. |
| 12:44.23 | Chotaire | and it is UNACCEPTABLE. |
| 12:44.35 | Chotaire | for various reasons. |
| 12:45.07 | fourcheeze | ahh |
| 12:45.20 | fourcheeze | so 1und1 are sending you dtmf inband? |
| 12:45.40 | Chotaire | yup? |
| 12:45.59 | Chotaire | I will repeat... it WORKS with rfc2833 |
| 12:46.03 | Chotaire | just not with 1.2.4 |
| 12:46.06 | fourcheeze | so the old asterisk must have been able to accept inband even if you specify rfc2833 |
| 12:46.07 | Chotaire | only with 1.0.3 |
| 12:46.14 | Chotaire | no. |
| 12:46.28 | Chotaire | it was NOT inband in 1.0.3 |
| 12:46.36 | fourcheeze | so you changed your config at 1und1 to inband? |
| 12:46.43 | Chotaire | yup |
| 12:46.52 | fourcheeze | but you can't change it to info? |
| 12:47.00 | fourcheeze | only inband or rfc2833 ? |
| 12:47.00 | Chotaire | sure I can, but it won't detect |
| 12:47.11 | fourcheeze | well |
| 12:47.12 | Chotaire | rfc2833 = no worky, info = no worky, inband = worky |
| 12:47.19 | fourcheeze | if you set it to info |
| 12:47.25 | fourcheeze | and you debug the connection |
| 12:47.25 | Chotaire | I get no debug messages |
| 12:47.28 | fourcheeze | you should see info headers |
| 12:47.35 | Chotaire | I am just testing something... |
| 12:47.47 | fourcheeze | also it could be that your settings are not updated straight away |
| 12:48.10 | Chotaire | hm, I found something new that might save my ass. |
| 12:48.16 | Chotaire | inband dtmf is now filtered in 1.2.4 |
| 12:48.32 | Chotaire | in 1.0.3 I could disconnect everyone from a meetme when hitting inband "#" |
| 12:48.46 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3742202.sympatico.ca) |
| 12:49.28 | Chotaire | this no longer works.. the dtmf is filtered now... which is actually pretty good for the beginning. |
| 12:50.09 | Chotaire | which means, I might be able to live with it until rfc2833 reverse detection (whatever that stuff was called, it was removed after 1.0.3) works again.... I will do some more testing.. if it fulfills my needs, I can stop bothering you ;) |
| 12:50.13 | Chotaire | and bother ole instead ;) |
| 12:51.05 | *** join/#asterisk fugitivo (n=ajf@201.255.176.38) |
| 12:51.10 | fugitivo | hello |
| 12:52.02 | *** join/#asterisk gnosys_ (n=gnosys_@ip68-230-151-65.ri.ri.cox.net) |
| 12:52.06 | Chotaire | fourcheeze: in the meanwhile, let me say thx for your attention... I will do some more testing now... |
| 12:52.09 | Chotaire | brb |
| 12:54.08 | fourcheeze | Chotaire: NP - hope it works out for you |
| 12:55.59 | *** join/#asterisk lunaphyte_ (n=lunaphye@c-71-193-101-146.hsd1.mi.comcast.net) |
| 12:56.51 | trixter | I might have to goto kinkos and get this lata map printed poster size for the office |
| 13:00.18 | RoyK | hm |
| 13:01.38 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:02.29 | trixter | wow california has a lot of unclaimed ilec terroritory |
| 13:02.38 | trixter | I would have figured that w ould have been gone long ago |
| 13:04.35 | gnosys_ | anybody here use snom phones? |
| 13:04.57 | *** join/#asterisk ibob63 (n=hp@88-96-66-62.dsl.zen.co.uk) |
| 13:05.49 | ibob63 | can anyone recommend a uk based DID provider? |
| 13:06.28 | RoyK | fuck |
| 13:06.33 | RoyK | asterisk hangs several times a day |
| 13:06.36 | RoyK | http://bugs.digium.com/view.php?id=6445 |
| 13:08.37 | *** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk) |
| 13:08.47 | gnosys_ | [TK]D-Fender, do you use Snom phones? |
| 13:09.02 | x86 | ibob63: ipstar.us |
| 13:09.34 | [TK]D-Fender | nope |
| 13:09.38 | gnosys_ | :-( |
| 13:09.38 | *** join/#asterisk Crontibs (n=frank@office-nat.choopa.net) |
| 13:09.42 | trixter | doesnt ipstar only give you a lcfa/ncfa number? |
| 13:09.49 | x86 | trixter: yeah |
| 13:10.03 | trixter | I heard there was a free 207 provider |
| 13:10.04 | trixter | london |
| 13:10.12 | trixter | but forget who know |
| 13:10.18 | trixter | maybe sipgate.uk |
| 13:10.31 | x86 | sipgate is also ncfa |
| 13:11.26 | trixter | I know they were at one point, I thought I heard that they were doing london locals |
| 13:11.31 | trixter | calluk.co.uk is also ncfa |
| 13:13.16 | trixter | sipgate has reding numbers aparently |
| 13:13.35 | x86 | ooh? |
| 13:14.05 | x86 | i have a german ncfa from sipgate... |
| 13:14.08 | trixter | that is the rumor |
| 13:14.16 | trixter | as do I |
| 13:14.21 | trixter | well not ncfa but yeah |
| 13:14.23 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:14.26 | x86 | my german friends tell me it costs them more to call it than it would to direct dial the states ;) |
| 13:14.28 | trixter | same type of deal |
| 13:14.53 | skeffling | We have a client who really wants an 'xfersound' for sip-sip calls (they use SNOMs) It looks like this was discussed (and declined) back in April (bug id 3819). Is there any chance of this being looked in to again? |
| 13:15.08 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:15.39 | trixter | Important: Customers who sign up with sipgate for geographic phone numbers in the UK, must reside within the UK. |
| 13:15.46 | trixter | prolly part of the UK 'know your customer' laws |
| 13:18.29 | ibob63 | I live in the UK so think that I will check out sipgate. |
| 13:19.25 | trixter | I dont know if that is totally free, someone was saying sipgate.de was giving geographicals away free, its the same company, but it may have been limited time to be totally free |
| 13:19.28 | trixter | I dont know |
| 13:20.49 | gnosys_ | can anyone give me a suggestion on where to buy snom IP telephones at a good price? |
| 13:21.01 | x86 | ebay.com |
| 13:21.01 | [TK]D-Fender | gnosys_ : depends where you are |
| 13:21.07 | gnosys_ | US |
| 13:21.14 | [TK]D-Fender | www.atacomm.com has the best US prices I've seen. |
| 13:21.20 | gnosys_ | thanks, TK |
| 13:21.46 | [TK]D-Fender | Though I don't really suggest Snom, and the stronger proponents are turning around on them too.. |
| 13:22.46 | trixter | www.thevoipconnection.com isnt bad |
| 13:23.05 | trixter | they are responsive, and genereally will beat anyones price, or at least they have for me |
| 13:26.48 | trixter | You will see the link to change your number. If you are selecting an 01 or 02 number (geographic numbers) then the same requirement applies for having a British IP address (since it is a registration). |
| 13:26.54 | trixter | Ahh so it requires a proxy server :P |
| 13:27.31 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 13:27.53 | my007ms | hello all ai think my probem is with udev |
| 13:28.17 | my007ms | is there is linux without udev |
| 13:28.19 | my007ms | ?? |
| 13:28.29 | gnosys_ | [TK]D-Fender, your comment picqued my interest. Would you explain? I had the impression that Snom was a rather good choice for use with Asterisk, given the fact that the phone manufacturer supports the use of their phones with asterisk. And I had the impression that they were good quality and relatively easy to configure with asterisk. You think this is not true? |
| 13:31.14 | my007ms | hello all |
| 13:31.21 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 13:32.24 | *** join/#asterisk trelane_ (n=trelane@mail.andrewgroup.com) |
| 13:34.52 | clive- | 2.4 kernels do not use udev |
| 13:35.06 | x86 | some 2.6 kernels dont either |
| 13:35.12 | x86 | depends on how you compile it hehe |
| 13:36.21 | clive- | :) |
| 13:36.34 | clive- | thats beyond my knowledge of linux x86 :) |
| 13:37.57 | x86 | that |
| 13:38.02 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:38.02 | x86 | that's unacceptable |
| 13:38.55 | clive- | lol...well everyone has thier chance at being a newbie:) |
| 13:39.30 | iaxy | ~udev |
| 13:39.43 | jbot | [udev] at http://www.kernel.org/pub/linux/utils/kernel/hotplug/udev-FAQ , or broken and shit |
| 13:39.43 | iaxy | help udev |
| 13:39.45 | iaxy | how do you get that bot info |
| 13:39.51 | iaxy | thanks |
| 13:40.38 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.adnc.com) |
| 13:40.50 | FlyboySR22 | Good Morning Everyone |
| 13:42.28 | [TK]D-Fender | gnosys_ : Their firmware updates seem to break stuff all the time, and their audio quality is inferior to Polycom which seems to be the best value for MOST uses where quality is concerned. |
| 13:42.30 | iaxy | Isn't there a better text to speech? Festival is cool but, its really barbaric, that can't be used in production system. |
| 13:42.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:42.55 | jontow | festival works well enough, if you know how to tune it |
| 13:43.15 | [TK]D-Fender | gnosys_ : Snom's big selling points is their web admin, and their presence support. If you need a receptionist phone with BLF, then Id say Snom is the best option right now. For everything else : Polycom. |
| 13:43.25 | [TK]D-Fender | FlyboySR22 : 'mornin |
| 13:43.26 | iaxy | it works good. just the synthesized voice is a bit funny sounding |
| 13:43.45 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 13:43.49 | jontow | agreed.. snom isn't the best all-around phone.. but the 320 has a seriously nice price barrier/entry point :) |
| 13:43.57 | jontow | making it ok for budgeted places.. |
| 13:43.59 | jontow | (ie. this office) |
| 13:44.16 | *** join/#asterisk fugitivo (n=ajf@201.255.176.38) |
| 13:44.37 | iaxy | help festival |
| 13:44.54 | jontow | ~festival |
| 13:44.55 | jbot | extra, extra, read all about it, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 13:45.05 | jontow | ;) |
| 13:45.11 | trixter | ~matsuri |
| 13:45.23 | gnosys_ | thanks, [TK]D-Fender. Part of my interest in Snom stems from my inability to do customized ring-tones with the Polycom phones I have. I must've spent some 10-12 hours reading and fiddling and trying different things, all to no avail. |
| 13:45.25 | Chotaire | ehrm.. I thought the enter/leave sounds for meetme are now definable in 1.2.*? |
| 13:45.36 | [TK]D-Fender | JonR800 : its interesting, but I might still sooner go with an IP 501 in these cases as it costs the same. |
| 13:45.40 | trixter | jbot is aparently not multilingual :/ |
| 13:46.07 | [TK]D-Fender | gnosys_ : Customized how? I've got mine doing the Cisco "24" bleep, and have a small army of others handy as well... |
| 13:46.21 | gnosys_ | customized ringtones? |
| 13:46.27 | Chotaire | but the only thing I found is the has-join and has-left |
| 13:46.41 | Chotaire | is it still the damn hex-encoded stuff hidden in some .h file? |
| 13:47.05 | jontow | i spent about 15mins to get a reasonable ringtone on my SNOM 320 |
| 13:47.06 | jontow | ;) |
| 13:47.12 | jontow | "bob dylan & johnny cash - ring of fire" |
| 13:47.16 | jontow | i mentioned it last week ;) |
| 13:47.20 | [TK]D-Fender | gnosys_ : I can down-mix and sounds file I want as a ring-tone. |
| 13:47.41 | [TK]D-Fender | any* |
| 13:48.01 | jontow | i have a polycom IP600 and a snom 320 on my desk currently |
| 13:48.12 | jontow | also an old meridian phone thats hooked to a norhell pbx |
| 13:48.12 | gnosys_ | would you mind sharing one of your polycom config files for me to study? I've read the manuals and experimented every way I could think of and I just couldn't get the phone to do customized rings at all... |
| 13:48.41 | iaxy | avaya charges 20,000.00 for their voicemail. I told this company I can provide that for about 2bills. They went with avaya because of the support.....go figure |
| 13:48.44 | [TK]D-Fender | gnosys_ : in a bit, sure. |
| 13:48.52 | gnosys_ | thanks |
| 13:49.03 | iaxy | How can you justify 20,000.0 for voicemail box? thats beyond me |
| 13:49.30 | fugitivo | 20,000 what? |
| 13:50.04 | iaxy | voicemail box for the G3... |
| 13:50.40 | mut | omg i am soooooo pissed |
| 13:50.47 | Chotaire | hello? does anyone know if the enter/leave sounds for meetme are still hex-encoded or if there is any way FINALLY to customize them? |
| 13:51.06 | mut | my boss decides to move my server yesterday w/o tellin me, so he just unplug it and i find out an hour later after he;s gone and the doors are locked |
| 13:51.24 | mut | so i come in today expecting to just press a key and boot it, but no, he SMASHED the die on the processor |
| 13:51.41 | mut | so i replace the process and still no boot, so i'm guessing the mobo fried too |
| 13:52.04 | mut | boss ass kicing T minus ~14 min |
| 13:52.37 | iaxy | Chotaire: that doesn't matter... if you have sox you can custom all prompts. |
| 13:53.15 | Chotaire | iaxy: really? so how do I customize the enter/leave sounds then? |
| 13:53.55 | iaxy | Do you know which files they are? |
| 13:53.59 | Chotaire | do you? ;) |
| 13:54.19 | iaxy | hehe |
| 13:54.30 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:54.35 | Chotaire | I guess that means "no" ;) |
| 13:55.19 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:55.37 | iaxy | No I don't. I meant if they were sound files they could be negotiated. IF they are encoded in the source, which I doubt, then you can't. |
| 13:56.07 | iaxy | LEt me get my book.... brb |
| 13:56.22 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:56.57 | *** join/#asterisk oden (n=oden@ezoffice.mandriva.com) |
| 13:57.36 | *** join/#asterisk fulgas (n=fulgas@207.226.175.2) |
| 13:58.46 | Chotaire | they are encoded in the source, I find no reason for that, and yes I still wish to modify them. |
| 13:58.47 | iCEBrkr | Chotaire: You can't customize the MeetMe join/leave sounds really. |
| 13:59.17 | iCEBrkr | I suppose you could try doing a hex() on the wave file and then having to hack the offsets and shit |
| 13:59.32 | Chotaire | would anyone be willing to assist me with that? |
| 13:59.41 | Chotaire | so I can make a howto finally? |
| 14:00.05 | PakiPenguin | anyone here uses gnudialer? |
| 14:00.16 | Chotaire | and while at that.. why has nobody ever found out that the admin menu even in 1.2.4 is bullshit? conf-locked will not work and disconnect last user will flood error messages and do nothing either. |
| 14:00.27 | iaxy | Chotaire: Did you find it in the source? |
| 14:00.36 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 14:00.41 | Chotaire | somebody pointed me to the file once... it's hard to find. |
| 14:00.49 | Chotaire | I can't remember actually.. it's in some .h file. |
| 14:00.55 | Chotaire | not commented, no nothing... |
| 14:01.19 | my007ms | does redhat9 use udev |
| 14:01.20 | my007ms | ? |
| 14:01.43 | AndyCap | my007ms: not before FC2 (2.6 kernel) iirc. |
| 14:01.48 | AndyCap | or maybe fc3 |
| 14:02.09 | iaxy | my007ms: nope |
| 14:02.26 | my007ms | great so it will not make all this error with my card |
| 14:02.39 | Katty | no pain meds :>>> |
| 14:03.45 | *** join/#asterisk Rhizome (i=id@tor/session/x-7a3bc9fb3a775c3b) |
| 14:04.01 | clive- | what packages do I need to use subversion on centos 4.2 ? |
| 14:04.03 | *** join/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 14:04.12 | Rhizome | ding dong, sf cvs is dead :( |
| 14:04.18 | [TK]D-Fender | Katty: mew. |
| 14:04.24 | iaxy | Chotaire: app_meetme2.c? |
| 14:04.28 | Katty | [TK]D-Fender: :>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 14:04.30 | [TK]D-Fender | Katty: congrats.... tooka while.... but then again, that was 4 in 1 shot... |
| 14:04.32 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:05.02 | Katty | [TK]D-Fender: i'm not healed completely yet, but not need pain meds is a good start :> |
| 14:05.29 | warthawg | my make of asterisk 1.2.4 dies with cannot find -lssl. i am running ubuntu64, and have libssl installed. is that what make is looking for? |
| 14:05.41 | iDunno | Katty: yay :) |
| 14:05.43 | AndyCap | clive-: yum install subversion ? |
| 14:05.45 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 14:05.55 | clive- | andycap I tried that, ...no luck |
| 14:06.07 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:06.07 | AndyCap | clive-: odd. is it already installed perhaps? |
| 14:06.38 | clive- | andycap no....it gives me "bad filename" error |
| 14:06.59 | AndyCap | clive-: what does? |
| 14:07.14 | Chotaire | iaxy: is that a question or an answer? ;) |
| 14:07.15 | AndyCap | clive-: and "rpm -q subversion" says? |
| 14:08.03 | astvoip | hello help on sip error 408 |
| 14:08.17 | clive- | havent tried that one yet, will give it a wirl |
| 14:08.18 | astvoip | what does it mean? |
| 14:09.05 | astvoip | i look on the sip coded error on category 4xx-client error |
| 14:09.09 | [TK]D-Fender | Katty: So does their absence still freak you out? |
| 14:09.26 | RoyK | http://dilbert.com/comics/dilbert/archive/dilbert-20060226.html |
| 14:09.49 | Katty | [TK]D-Fender: what do you mean? |
| 14:09.55 | astvoip | 408 Request Timeout: Couldn't find the user in time |
| 14:11.28 | astvoip | which part on the client is it? on the termination or on the origination? |
| 14:12.49 | [TK]D-Fender | Katty: 4 teeth out all at once. I had a headach for 1 day when I had one taken out. You lost 4 all-together. And do you find yourself tonguing where the used to be and feel freaked about the fact they're "just not there"? took a while for it to feel normal for me... |
| 14:13.41 | rva | http://www.nexusuk.org/projects/VoIP/iaxcomm/missedcalls |
| 14:13.54 | rva | does anyone that uses iaxclient know this problem of lost calls? |
| 14:15.30 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:16.06 | Chotaire | asterisk/1.2.4/asterisk-1.2.4/apps |
| 14:16.06 | Chotaire | -rw-r--r-- 1 1001 users 16995 Nov 29 19:24 enter.h |
| 14:16.06 | Chotaire | -rw-r--r-- 1 1001 users 12091 Nov 29 19:24 leave.h |
| 14:16.17 | Chotaire | can anyone help me modifying this? |
| 14:16.46 | Chotaire | or know any tool/conversion script that would create such an output? |
| 14:16.55 | _Paulo_ | Chotaire, what do you want to do? |
| 14:17.07 | Chotaire | change meetme enter/leave sounds |
| 14:17.30 | fourcheeze | Chotaire: just upload new sounds |
| 14:17.36 | Chotaire | huh? |
| 14:17.59 | Chotaire | fourcheeze: excuse me? |
| 14:18.48 | *** part/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 14:19.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.58 | _Paulo_ | Chotaire, do you have the sound recorded already? |
| 14:20.07 | Katty | [TK]D-Fender: i've not tounged anything |
| 14:20.14 | Katty | [TK]D-Fender: i'm still rather paranoid of hurting myself. |
| 14:20.28 | Chotaire | paulo: I can do that within 2 minutes |
| 14:20.33 | Chotaire | so? |
| 14:21.06 | _Paulo_ | just convert every byte to exa. |
| 14:21.16 | _Paulo_ | s/exa/heaxa/ |
| 14:21.33 | _Paulo_ | s/heaxa/hex/ |
| 14:21.38 | Chotaire | paulo: sounds good, if you could tell me how to do that, that would rock. |
| 14:22.05 | Chotaire | paulo: also have you checked if any special header is included in that hex stuff? etc... |
| 14:22.12 | Chotaire | and what format was used? |
| 14:22.13 | fourcheeze | oh, isn't the sound when you enter one of those sounds that starts conf- ? |
| 14:22.29 | Chotaire | fourcheeze: scroll up ;) |
| 14:22.46 | _Paulo_ | Chotaire, think its raw. Will check in a minute... |
| 14:22.57 | Chotaire | <PROTECTED> |
| 14:22.59 | fourcheeze | oh |
| 14:22.59 | AndyCap | the header says 8-bit raw ulaw |
| 14:23.01 | iaxy | 8 bit u-law raw |
| 14:23.01 | Chotaire | * Source. enter.raw |
| 14:23.03 | fourcheeze | wtf is it like that? |
| 14:23.15 | Chotaire | fourcheeze: that's a better question. |
| 14:23.21 | [TK]D-Fender | Katty: after a week, it just seems awkward. I should have stopped bleeding a while ago. |
| 14:23.52 | Katty | i've not noticed. |
| 14:24.02 | Katty | but then again, they were never full any to start with |
| 14:24.04 | *** join/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 14:24.06 | Katty | s/any/in/ |
| 14:24.07 | Chotaire | paulo: so all I need to find out... a) how to make a raw 8-bit ulaw file (sox might probably help, dunno) and b) how to convert it to hex. |
| 14:24.17 | Katty | jbot: thank you, dear. |
| 14:24.17 | jbot | pas de quoi, Katty |
| 14:24.24 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:24.45 | _Paulo_ | Chotaire, what OS are u using? |
| 14:24.46 | AndyCap | Chotaire: sox some magic option to rate, bit and raw ulaw, | hexdump -magic here > .h |
| 14:25.36 | Chotaire | paulo: anyone you like ;) |
| 14:26.00 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 14:26.13 | Chotaire | andycap: that's some start... can I also convert it back to the original? |
| 14:26.34 | *** join/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr) |
| 14:26.55 | puzzled | afternoon |
| 14:27.41 | AndyCap | Chotaire: write some C for that |
| 14:28.36 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:29.05 | AndyCap | Chotaire: i.e "#include "enter.h" ;int main(void){ printf("%s",enter);return 0} " |
| 14:29.22 | AndyCap | except that %s probably won't work to good. :-P |
| 14:29.27 | [TK]D-Fender | Katty: in-grown molars? |
| 14:29.40 | Katty | [TK]D-Fender: sorta. |
| 14:29.43 | Chotaire | andycap: wouldn't it be possible to use a conf-enter.gsm instead? |
| 14:29.52 | Katty | [TK]D-Fender: the top ones weren't really developed too much yet. |
| 14:30.03 | Chotaire | then it would be possible to change the sounds on the fly without recompilation. |
| 14:30.04 | Katty | [TK]D-Fender: the bottom right one was really the only one giving me problems. and it was just off and on. |
| 14:30.08 | Chotaire | 0xba, 0xba, 0xb0, 0xa6, 0xa9, 0xb8, 0xfe, 0x46, 0x42, 0x46, |
| 14:30.16 | Chotaire | so that's hexdump -x? |
| 14:30.23 | Katty | [TK]D-Fender: dentist recommended they all come out before i started having complications.......there wasn't room for any of them |
| 14:30.39 | *** join/#asterisk tuxinator_linux (n=tuxinato@m010e36d0.tmodns.net) |
| 14:30.40 | AndyCap | Chotaire: I have no Idea why they did it that way |
| 14:30.47 | fourcheeze | it's mad |
| 14:31.02 | AndyCap | Chotaire: so yes, rewriting it may be possible. and probably required for sanity. :) |
| 14:31.17 | Chotaire | how long would you need? ;) |
| 14:31.19 | AndyCap | they could at least have had the courtesy of supplying the script to make them |
| 14:31.34 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 14:31.43 | AndyCap | Chotaire: me? I'm no programmer. so I'd probably break untold amounts of stuff |
| 14:32.17 | Chotaire | anyone bored? ;) |
| 14:32.30 | *** join/#asterisk qzxcd (n=heroine@gut75-1-82-67-176-238.fbx.proxad.net) |
| 14:33.03 | AndyCap | Chotaire: and besides, it could be done that way for some obscure performance or other packet timing reason. (while ingesting LSD) :-) |
| 14:33.09 | [TK]D-Fender | Katty: Similar scenario with mine,, though I only removed one. I belive its mate below is degrading with its absence (need to match in pairs to be healthy), and will likely have it removed before too long. |
| 14:33.30 | Chotaire | andycap: probably, because even in it's current form, the sounds always sound a bit different. |
| 14:33.36 | _Paulo_ | AndyCap, you can load the sounds on startup... |
| 14:33.41 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:33.46 | Chotaire | ? |
| 14:34.06 | AndyCap | Chotaire: still, other announcements are handeled in the "normal" way? |
| 14:34.13 | Chotaire | yup |
| 14:34.21 | Chotaire | the only thing that is like this is the enter/leave sounds. |
| 14:34.40 | Chotaire | even the beep (vmb) |
| 14:34.48 | Chotaire | is a gsm file. |
| 14:35.19 | AndyCap | Chotaire: bah, hexdump is freaky hexdump -e '24/1 "0x%x," "\n"' |
| 14:35.20 | trelane_ | is there a way to specify multiple e-mail addresses for a given inbox in voicemail.conf... ie a cell phone sms and a standard e-mail acct. |
| 14:36.03 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 14:36.08 | *** join/#asterisk gnosys_ (n=gnosys_@ip68-230-151-65.ri.ri.cox.net) |
| 14:36.16 | Chotaire | andycap: does that give 10 units per line? |
| 14:36.18 | Katty | [TK]D-Fender: )= don't wait too long. the older you are, the harder it is for you to recover. |
| 14:36.35 | AndyCap | Chotaire: 24. :P but there's a bug somewhere |
| 14:36.59 | Chotaire | andycap: if you can assist me get this to work completely, I will buy you something on amazon |
| 14:37.06 | Chotaire | as a little motivation ;) |
| 14:37.15 | AndyCap | Chotaire: hexdump -e '10/1 "0x%02x," "\n"' |
| 14:37.48 | FlyboySR22 | Katty, How are you feeling...? |
| 14:38.11 | Katty | FlyboySR22: better :> |
| 14:38.19 | FlyboySR22 | Katty, Good deal :-) |
| 14:38.24 | Chotaire | if that works good, I will invest time to automate this process and make a script for this. |
| 14:38.34 | AndyCap | PakiPenguin: if people don't answer about gnudialer in here, don't msg them |
| 14:38.37 | Chotaire | e.g. supply a enter.wav and leave.wav, run script, done. |
| 14:38.46 | PakiPenguin | err |
| 14:38.47 | PakiPenguin | sorry |
| 14:38.48 | PakiPenguin | <PROTECTED> |
| 14:39.14 | Chotaire | andycap: you sure that's it? ;) hexdump -e '10/1 "0x%02x," "\n"' |
| 14:39.40 | AndyCap | Chotaire: should work. sorry. you didn't get the final edit. ;) |
| 14:40.03 | AndyCap | Chotaire: got interrupted by PP, hexdump -e '10/1 "0x%02x, " "\n"' |
| 14:40.18 | AndyCap | it lacks an option to skip the last bytes.. |
| 14:40.26 | AndyCap | should figure it out. |
| 14:40.36 | Katty | i need a new song. |
| 14:40.40 | Katty | something old. |
| 14:40.43 | Katty | suggestions? |
| 14:41.03 | Chotaire | skip the last bytes? |
| 14:41.17 | Chotaire | can you explain why skip some bytes? |
| 14:41.23 | fourcheeze | Katty: how old? |
| 14:41.45 | Katty | fourcheeze: somewhere between the 60s and the 80s |
| 14:41.57 | Chotaire | I think I found something with sox |
| 14:41.58 | fourcheeze | what kind of music do you like? |
| 14:42.06 | Katty | fourcheeze: something that isn't bangbang |
| 14:42.13 | [TK]D-Fender | Katty: Style? |
| 14:42.15 | Katty | fourcheeze: or hiphop, and i'm not too fond of country. |
| 14:42.16 | _Paulo_ | zztop: bad to the bone |
| 14:42.20 | _Paulo_ | :-) |
| 14:42.27 | Katty | [TK]D-Fender: beach boys, beatles, beegees.. |
| 14:42.30 | Katty | [TK]D-Fender: the eagles. |
| 14:42.34 | *** join/#asterisk RoyK (n=roy@ti211310a080-11530.bb.online.no) |
| 14:42.35 | AndyCap | Chotaire: it will always output a full line of 10 bytes even if there should only be 5 or 9 |
| 14:42.46 | _Paulo_ | something from supertramp |
| 14:42.48 | fourcheeze | Katty: gotta love "Desperado" by the Eagles |
| 14:42.53 | Katty | [TK]D-Fender: i guess that'd be considered soft rock. |
| 14:42.55 | [TK]D-Fender | Katty: Your Mamma Don't Dance - Loggins / Messina |
| 14:43.05 | Katty | i've already got that one :P |
| 14:43.09 | [TK]D-Fender | :) |
| 14:43.11 | Chotaire | andycap: so that would be 0x00? |
| 14:43.13 | fourcheeze | Katty: how about some ELO |
| 14:43.17 | Katty | fourcheeze: ELO? |
| 14:43.22 | Katty | fourcheeze: what are you a modem? :P |
| 14:43.26 | fourcheeze | :-) |
| 14:43.26 | Chotaire | if so, I can use e.g. grep -v to get rid of them. |
| 14:43.28 | AndyCap | Chotaire: no, just 0x , |
| 14:43.30 | fourcheeze | Electric Light Orchestra |
| 14:43.34 | Katty | oh. |
| 14:43.40 | Chotaire | andycap: ok, no problem. |
| 14:43.41 | AndyCap | Chotaire: so it needs to conditionalize the rest as well. :P |
| 14:43.53 | fourcheeze | Katty: sorta rocky with a string section |
| 14:43.54 | znoG | Katty: if you don't mind me asking, what age bracket are you in? |
| 14:43.55 | AndyCap | Chotaire: to streamline it. :-P |
| 14:44.03 | [TK]D-Fender | Katty: Black Crowes - Hard To Handle |
| 14:44.06 | docelm0 | COUNTRY SUCKS! |
| 14:44.10 | docelm0 | :) |
| 14:44.11 | Chotaire | andycap: but if I strip the last bytes, stuff would work? |
| 14:44.25 | [TK]D-Fender | Katty: Bruce Springsteen - Glory Days. |
| 14:44.27 | Hmmhesays | some guy played that last night at jam night |
| 14:44.34 | _Paulo_ | Chotaire, sure will. |
| 14:44.47 | AndyCap | Chotaire: assuming we guessed the format right. ;) |
| 14:44.51 | Chotaire | andycap: could you check if app_meetme.c has any time limitation for these sounds? |
| 14:45.06 | Katty | znoG: i'm in my 20s. |
| 14:45.08 | Chotaire | in the meanwhile, I'm f*cking around with sox.... |
| 14:45.16 | fourcheeze | Katty: How about some Eppu Normaali |
| 14:45.17 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 14:45.19 | Katty | Hmmhesays: i'm off the pain meds :> |
| 14:45.41 | AndyCap | Chotaire: sox KDE_Chimes_3.ogg -t raw -r 8000 -U -b -c 1 - resample -ql | hexdump -e '10/1 "0x%02x, " "\n"' |
| 14:45.44 | Hmmhesays | grand |
| 14:45.50 | Chotaire | I am a bit confused with the comment in enter.h... it says enter.raw - but actually, sox has his own suffix for that kinda file... .uw |
| 14:46.14 | Katty | [TK]D-Fender: i think that's a bit more country than i like. |
| 14:46.19 | AndyCap | Chotaire: perhaps sox -t ogg KDE_Chimes_3.ogg -t raw -r 8000 -U -b -c 1 - resample -ql | hexdump -e '10/1 "0x%02x, " "\n"' |
| 14:46.30 | AndyCap | ahh, and hexdump compresses the output. Aargh |
| 14:46.49 | [TK]D-Fender | Katty: I figured it was just far enough. Not sure on the "crowd" I'm aiming for. How about you clue me in with what you've GOT. |
| 14:46.52 | *** part/#asterisk AxelBp (n=alexlsf@200.12.40.89) |
| 14:47.18 | AndyCap | Chotaire: perhaps sox -t ogg KDE_Chimes_3.ogg -t raw -r 8000 -U -b -c 1 - resample -ql | hexdump -ve '10/1 "0x%02x, " "\n"' |
| 14:47.28 | Chotaire | ah :) |
| 14:47.47 | Katty | [TK]D-Fender: i just did :P |
| 14:48.15 | Chotaire | andycap: can you test if the output is right? if yes, I can make the sounds asap (almost done with them) and recompile for testing ;) |
| 14:48.37 | Katty | [TK]D-Fender: bits from The Sound of Music, bowie, the muppets, dean martin, 311, sarah brightman, the eagles, nat king cole, taco |
| 14:48.47 | [TK]D-Fender | Katty: Aerosmith : Sweet Emotion. Anything by Bryan Adams ;) |
| 14:48.47 | Katty | older stuff. |
| 14:48.59 | Chotaire | andycap: if that stuff works you really make me happy... |
| 14:49.14 | [TK]D-Fender | Journey! |
| 14:49.14 | Katty | oooh, bryan adams. |
| 14:49.17 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-927134878eeac27e) |
| 14:49.20 | Chotaire | brb |
| 14:50.25 | fourcheeze | Katty: Pink Floyd ? |
| 14:50.52 | mut | http://www.record-eagle.com/2005/jul/14vat.htm |
| 14:50.54 | [TK]D-Fender | Yeah, Floyd / Dire Straits. |
| 14:50.54 | mut | what a way to go |
| 14:51.00 | _Paulo_ | Katty, killing me softly (renata flack?) |
| 14:51.19 | Katty | _Paulo_: i have that by roberta flack (= |
| 14:51.28 | Katty | fourcheeze: i have all of pink floyd's albums. |
| 14:51.32 | fourcheeze | ahh |
| 14:51.34 | Rhizome | Anyone had any experience with asterisk realtime scrambling the account code? really weird, got two sip phones that was working and now suddenly they both have accountcodes similar to lADgGe23xzp |
| 14:51.34 | Katty | fourcheeze: well, the major ones. |
| 14:52.48 | Katty | i found a song - mrs. robinson |
| 14:52.54 | fourcheeze | hehe |
| 14:53.05 | AndyCap | jesus loves you more than you will know. |
| 14:53.11 | fourcheeze | ho ho ho |
| 14:53.16 | [TK]D-Fender | "ho" |
| 14:53.20 | [TK]D-Fender | :O |
| 14:53.23 | x86 | mo ho |
| 14:53.25 | x86 | hehehe |
| 14:53.54 | fourcheeze | Katty: was just about to suggest the Beatles White Album or something |
| 14:53.57 | Rhizome | AndyCap, Are you jesus? :P |
| 14:54.10 | AndyCap | Rhizome: have you heard mrs. robinson? |
| 14:54.11 | AndyCap | :) |
| 14:54.12 | sivana | if I have a TE405P with a TDM400 and echo can turned on in zaptel, will it apply to everything? |
| 14:54.17 | x86 | i know mo ho than joe the navajo in idaho, so.... |
| 14:54.28 | Rhizome | AndyCap, uhm, now... |
| 14:54.33 | [TK]D-Fender | sivana : per-channel. |
| 14:54.35 | Rhizome | s/now/no/ |
| 14:54.56 | sivana | [TK]D-Fender: and there's no way to disable it per channel, right? |
| 14:54.58 | AndyCap | yes, thank you jbot, my regexp parser works just fine |
| 14:55.13 | [TK]D-Fender | sivana : However the algorythm is fixed for the ones that USE it, though the settings are independent IIRC |
| 14:55.16 | Chotaire | ok, sounds done... let's try converting. |
| 14:55.26 | [TK]D-Fender | sivana : You pick it PER channel. |
| 14:58.16 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-16.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:59.41 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 14:59.59 | Rhizome | ok, wtf, asterisk just deleted an account and tried to recreate it, but messed up the accountcode.. |
| 15:01.05 | x86 | anyone using ipstar.us for UK DID service? |
| 15:01.21 | x86 | or sipgate.co.uk for DE DID service? |
| 15:01.35 | x86 | nor Stanaphone for US DID service |
| 15:01.44 | x86 | IPKall DID works fine, however :) |
| 15:02.59 | x86 | anyone using those services willing to assist me set mine up? |
| 15:03.02 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:03.06 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 15:06.22 | Chotaire | I've got good news... ;) |
| 15:06.33 | Skid | i've used sipgate for DID UK |
| 15:06.38 | Skid | bag of shit really |
| 15:06.41 | jontow | you just saved a bunch of money on your car insurance? :/ |
| 15:06.52 | Skid | their "carrier" (laff) keep dropping their link |
| 15:06.55 | Chotaire | andycap: actually, it works.... just one strange thing happening... the sound will NOT be played to the person who joins... only to the other persons. however, the behaviour was different before ;) |
| 15:06.56 | Skid | according to their support staff |
| 15:07.59 | Chotaire | this is quite interesting.. I wonder why it got played before. |
| 15:08.07 | JunK-Y | ~agi api |
| 15:08.09 | jbot | from memory, agi api is at http://home.cogeco.ca/~camstuff/agi.html |
| 15:08.26 | Chotaire | andycap: also I have no trailing silence or something... so I find no reason ;) maybe it was an echo thing before... or maybe not ;) |
| 15:10.51 | AndyCap | Chotaire: hard to say. are you running the exact same version of * ? |
| 15:11.04 | Chotaire | 1.2.4, yup. same before. |
| 15:11.11 | Chotaire | i tested some more... |
| 15:11.18 | Chotaire | while the sound is playing, the caller will be silent. |
| 15:11.31 | Chotaire | so if it's a longer sound, if the caller says "hello", nobody will know. |
| 15:11.31 | AndyCap | sounds like we managed to break something. :-P |
| 15:11.39 | Chotaire | no I don't think so... |
| 15:11.44 | AndyCap | Chotaire: did he really say hello then? :-P |
| 15:11.52 | Chotaire | yes, I am testing with myself. |
| 15:11.57 | Chotaire | I think I found the reason. |
| 15:12.06 | AndyCap | Chotaire: it was a joke. ;) (schrödinger's cat) |
| 15:12.10 | Chotaire | to the caller, only like the last 100ms is playing. |
| 15:12.18 | Chotaire | or maybe 200ms |
| 15:12.40 | Chotaire | so if the sound is longer than that and ending with a fade out or even silence, the caller won't hear it and doesn't know what happened. |
| 15:12.41 | x86 | so no one uses Stanaphone? |
| 15:12.42 | Chotaire | this is interesting. |
| 15:12.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:12.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:13.05 | Chotaire | while other participants may talk during the enter sound, the joined may not. |
| 15:13.10 | Chotaire | joined=joiner |
| 15:13.33 | AndyCap | what happens if you replace the original enter.h? |
| 15:13.41 | Chotaire | well that's what I did. |
| 15:13.44 | Chotaire | I used my own sounds. |
| 15:13.48 | Chotaire | the enter sound is having a fade-out |
| 15:14.08 | AndyCap | Chotaire: I was thinking about putting the original one back. :-P not the custom |
| 15:14.13 | Chotaire | so the "joiner" only hears the very rest of the fade-out which is like unhearable. but it was enough for me to find out what's going on. |
| 15:14.29 | Chotaire | andycap: I still have it, no worries ;) |
| 15:14.32 | SibRw0rk | how do you setup * to connect on multiple ports |
| 15:14.39 | fourcheeze | x86: have some experience of using sipgate.co.uk - but wouldn't recommend it |
| 15:14.39 | SibRw0rk | like i want 5060 AND 5065 |
| 15:15.01 | AndyCap | Chotaire: but just to make sure that it is the new enter.h that triggers the bug |
| 15:15.26 | Chotaire | andycap: I can easily find out by not using a sound with fade out. |
| 15:15.36 | Chotaire | and which is probably cut to the bit |
| 15:15.52 | fourcheeze | SibRw0rk: not sure if asterisk can do that, but I think iptables can always replicate a port for you |
| 15:15.57 | Chotaire | anyway, you did it... |
| 15:16.16 | Chotaire | I have a little offer for you.. I have an account with a huge (legal) mp3 site... are you searching for any song? |
| 15:16.40 | Chotaire | I can buy it for you, just as a little thank you. |
| 15:16.53 | Chotaire | if you don't like music, what about a little book at amazon? ;) |
| 15:17.01 | x86 | fourcheeze: i'm just using their free inbound DID service to have a DID in Germany |
| 15:17.22 | x86 | fourcheeze: do you still have your configs? |
| 15:17.28 | SibRw0rk | fourcheeze: there has to be a way |
| 15:17.29 | AndyCap | Chotaire: thanks for offering but not required. :-) besides, you'd have to ship it to europe. |
| 15:17.33 | SibRw0rk | some places block 5060 |
| 15:17.48 | Chotaire | andycap: where ya from? |
| 15:17.59 | fourcheeze | x86: no, but I remember that it was very straightforward |
| 15:18.03 | AndyCap | Chotaire: norway |
| 15:18.12 | Chotaire | weee.... middle of nowhere ;) |
| 15:18.29 | fourcheeze | x86: I know they are having problems right now |
| 15:18.31 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 15:18.39 | Chotaire | andycap: then think about a song, CDM or EP... |
| 15:18.55 | Chotaire | no shipping required, legal music is digital these days (and much cheaper) |
| 15:19.09 | x86 | fourcheeze: ah... maybe that's the issue then... |
| 15:19.21 | x86 | fourcheeze: i was going to say, i followed to a tee the example on voip-info.org |
| 15:19.22 | AndyCap | Chotaire: ;) I'll let you know when I get home. |
| 15:19.31 | Chotaire | I am always here. just leave me a msg. |
| 15:19.35 | AndyCap | ok. |
| 15:19.38 | Chotaire | if I don't reply, I still log msg |
| 15:20.05 | Chotaire | andycap: I assume you use linux=? |
| 15:20.28 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 15:21.07 | Rhizome | CLI> sip show peers |
| 15:21.07 | Rhizome | Name/username Host Dyn Nat ACL Port Status |
| 15:21.08 | Rhizome | 71691/rtgIsx3kx5B+z XXX.XXX.XXX.XXX D N 5060 Unmonitored |
| 15:21.13 | Chotaire | andycap: I dunno if linux can do DRM (I don't use xwin), if not I can re-encode it for you and send it to you with the license just in case. |
| 15:21.20 | Rhizome | Anyone know where that scrabbled text is coming from? |
| 15:22.02 | fourcheeze | SibRw0rk:have you tried putting 2 ports in bindport in sip.conf ? |
| 15:22.59 | lunaphyte_ | the more i read, the more it seems like i can't modify the folders used by voicemail. is that right? |
| 15:23.18 | Tagor | I'm now using a sound file with a ringtone (exten => s,18,Playback(ring_out)). But I would like to use the standard ringtone of asterisk instead of the soundfile. Is that possible? |
| 15:23.29 | Tagor | I would like it to play for 20 seconds |
| 15:23.30 | AndyCap | Chotaire: hehe, opensource and drm is impossible. |
| 15:23.40 | Chotaire | ok no problem I will fix that for you. |
| 15:23.47 | Chotaire | so start thinking of good music ;) |
| 15:23.51 | SibRw0rk | fourcheeze: yeah we tried with a comma |
| 15:23.53 | SibRw0rk | didn't work |
| 15:23.53 | Chotaire | bbl, doing new sounds and being happy ;) |
| 15:24.12 | fourcheeze | SibRw0rk: there's port and bindport - not sure what the difference is |
| 15:24.33 | [TK]D-Fender | Tagor : Try doing a dial to LOCAL to an exten that just waits before answering.... |
| 15:24.39 | [TK]D-Fender | and let it time-out. |
| 15:24.39 | SibRw0rk | [TK]D-Fender - sup dude |
| 15:24.50 | [TK]D-Fender | SibRw0rk : Still breathing... |
| 15:24.52 | SibRw0rk | that's good |
| 15:24.54 | SibRw0rk | ? for you |
| 15:25.03 | SibRw0rk | i got a client in a country that's blocking port 5060 |
| 15:25.14 | Tagor | [TK]D-Fender >> Do you have an example of that? |
| 15:25.15 | fourcheeze | SibRw0rk: try port 5060,5065 |
| 15:25.16 | SibRw0rk | how can i config * to listen to another port as well as 5060 |
| 15:25.20 | SibRw0rk | fourcheeze: tried that |
| 15:25.26 | vgster | has anyoen had any experiences with atcom phones? |
| 15:25.57 | fourcheeze | SibRw0rk: in that case iptables is your friend |
| 15:26.14 | SibRw0rk | it's not my server that's the problem |
| 15:26.22 | SibRw0rk | is the ISP of the client |
| 15:26.37 | fourcheeze | but you want asterisk to listen on two ports? |
| 15:26.40 | SibRw0rk | yeah |
| 15:26.43 | fourcheeze | so use iptables |
| 15:26.53 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 15:27.17 | fourcheeze | use port forwarding to forward 5060 on localhost to 5070 on external ip |
| 15:27.30 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 15:27.33 | SibRw0rk | buyt then everyone would have to be on 5070 |
| 15:27.38 | [TK]D-Fender | Tagor : Just do something like "Dial(LOCAL/wastingtime@context,20) and in that exten just do wait(1000) then hangup. |
| 15:27.55 | warthawg | has anyone built 1.2.4 on ubuntu? I am getting a make error for ld not finding lssl |
| 15:28.03 | fourcheeze | SibRw0rk: what makes you think that |
| 15:28.04 | fourcheeze | ? |
| 15:28.09 | SibRw0rk | warthawg: install openssl-devel |
| 15:28.43 | fourcheeze | SibRw0rk: port forwwarding won't block the existing port |
| 15:28.56 | [TK]D-Fender | warthawg : Ubuntu doesn't come with DEV libs, its an el-cheapo chump desktop for which you'll need to pile on a bunch of packages to make workable. Try using a more "normal" distro. |
| 15:29.03 | _Paulo_ | SibRw0rk, try 2 lines with "port" |
| 15:29.03 | warthawg | SibRphrek, thank you sir |
| 15:29.11 | lunaphyte_ | my voicemail questions go as unanswered on the mailing list as they do here.. i wonder if that's because no one knows, no one cares, or it's a blindingly stupid question... ;) |
| 15:29.25 | SibRw0rk | _Paulo_: in sip.conf? so bindport=5060 on one line and port=5065 on another? |
| 15:29.26 | [TK]D-Fender | luckyduck : what is it? |
| 15:29.37 | _Paulo_ | yeps |
| 15:29.40 | SibRw0rk | ok |
| 15:30.38 | skeffling | We have a client who really wants an 'xfersound' for sip-sip calls (they use SNOMs) It looks like this was discussed (and declined) back in April (bug id 3819). Is there any chance of this being looked in to again? |
| 15:31.02 | _Paulo_ | SibRw0rk, bindport=5060 and bindport=5065 |
| 15:31.09 | fourcheeze | lunaphyte_ why not just ask the question? |
| 15:31.30 | lunaphyte_ | the more i read, the more it seems like i can't modify the folders used by voicemail. is that right? |
| 15:31.47 | *** join/#asterisk essaredee (n=srd@cpc1-lich2-0-0-cust257.brhm.cable.ntl.com) |
| 15:32.00 | fourcheeze | lunaphyte_: modify how? |
| 15:32.27 | warthawg | SibRphrek, hmmm, seems openssl-devel is not in ubuntu repositories |
| 15:32.44 | fourcheeze | warthawg: do you have universe ? |
| 15:32.51 | Nivex | Katty: May I cut in? |
| 15:32.53 | warthawg | fourcheeze, i believe so |
| 15:32.55 | Katty | Nivex: of course not. |
| 15:32.56 | essaredee | I've got a problem with incoming iax connections, even though I do set for asterisk to answer, it doesn't use the tones (ringing) as specified in indications.conf |
| 15:33.04 | Katty | but file may. |
| 15:33.21 | file | how are you doing my dearest Kitty Katty? |
| 15:33.23 | austinnichols101 | get a room! |
| 15:33.32 | Katty | file: goodly. no pain meds today :>>>> |
| 15:33.39 | file | Katty: exxxxxxxxxxcellent |
| 15:33.50 | lunaphyte_ | fourcheeze: remove work, for example - and change "old" to "saved". |
| 15:34.03 | Katty | Nivex: mommy told me to never hug strangers. |
| 15:34.09 | Nivex | Well, I am about as strange as they come :) |
| 15:34.12 | *** join/#asterisk Alric (n=nbowyer@l-69-148-121-117.stu.swau.edu) |
| 15:34.21 | fourcheeze | lunaphyte_: ahh ok |
| 15:34.21 | essaredee | might anyone know why incoming iax connections aren't using the indications I configured? |
| 15:34.23 | fourcheeze | never tried that |
| 15:34.40 | Katty | Nivex: k'then |
| 15:34.51 | gnosys_ | [TK]D-Fender, is that polycom config file with custom ringtones handy? |
| 15:35.17 | lunaphyte_ | fourcheeze: logic tells me i can't be the first person to want to do this, but everyone has been so strangely quiet. |
| 15:35.24 | warthawg | i wonder if i can just symlink lssl with libssl |
| 15:35.34 | fourcheeze | lunaphyte_: well surely it's just up to the voicemail prompt |
| 15:35.52 | fourcheeze | so record someone saying "Saved" instead of Old |
| 15:35.55 | warthawg | amazing that asterisk is easier to install on openwrt than ubuntu |
| 15:36.08 | _Paulo_ | Nivex, came to Rio. Lots of people huging nude in the streets this holiday. |
| 15:36.13 | fourcheeze | warthawg: I've never had a problem on ubuntu |
| 15:36.33 | warthawg | fourcheeze, where did you find lssl? |
| 15:36.34 | fourcheeze | warthawg: do you have openssl installed? |
| 15:36.41 | warthawg | fourcheeze, yes |
| 15:36.44 | russellb | how about openssl-dev? |
| 15:36.48 | cpm | _Paulo_, iz festival, eh? |
| 15:36.50 | lunaphyte_ | fourcheeze: wouldn't there be more than that though, since asterisk builds directories and such in /var/spool/asterisk/....? |
| 15:36.50 | Katty | file: i can't eat those yet :< |
| 15:36.54 | warthawg | no, apt doesnt find openssl-dev |
| 15:37.00 | russellb | libssl-dev? |
| 15:37.04 | file | Katty: :\ then I'll keep one waiting just for you |
| 15:37.11 | fourcheeze | warthawg: libssl-dev |
| 15:37.15 | Katty | file: kk. |
| 15:37.34 | *** join/#asterisk nothinman (i=shakey@adat202.neoplus.adsl.tpnet.pl) |
| 15:37.35 | lunaphyte_ | Katty: what'd you get your wisdom teeth ripped out or something? |
| 15:37.41 | iCEBrkr | I need a new jobbyjob |
| 15:37.46 | _Paulo_ | cpm, the damn Brazilian Carnival. |
| 15:37.57 | fourcheeze | iCEBrkr: I need a house cleaner |
| 15:38.01 | Katty | lunaphyte_: yes, all 4 of them. |
| 15:38.05 | russellb | I need some lunch |
| 15:38.06 | Katty | lunaphyte_: they were cut out. |
| 15:38.06 | Nivex | _Paulo_: how to we go from hugging to nude? |
| 15:38.09 | file | russellb: sub! |
| 15:38.10 | iCEBrkr | fourcheeze: You live in a tropical area? |
| 15:38.11 | iCEBrkr | :) |
| 15:38.13 | Nivex | *how did |
| 15:38.23 | fourcheeze | iCEBrkr: not entirely (UK) |
| 15:38.26 | iCEBrkr | fourcheeze: FOrget it |
| 15:38.30 | fourcheeze | hehe |
| 15:38.37 | lunaphyte_ | Katty: i feel. i had mine done a year or so ago. |
| 15:38.54 | _Paulo_ | Nivex, everybody is dancing (almost) nude in the streets. |
| 15:39.01 | iCEBrkr | I just need to find some 40yr old chick who recently won the house, the dog and Porche in the divorce. |
| 15:39.05 | iCEBrkr | So I can be her pool boy |
| 15:39.19 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 15:39.47 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:39.57 | [TK]D-Fender | gnosys_ : PM you e-mail addy |
| 15:39.57 | lunaphyte_ | like an idiot, i refused to take the vicodin too. |
| 15:40.06 | warthawg | now, about zlib... |
| 15:40.22 | fourcheeze | zlib1g-dev |
| 15:40.33 | nothinman | guys, girls, quick question -- how (using perl) can I get list of active calls? simplest way. pleeeease help. googling doesn't help much :/ |
| 15:41.35 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:41.36 | Nivex | nothinman: simplest way that comes to mind, do a system call to "asterisk -rx show channels" and parse its output |
| 15:41.45 | _Paulo_ | nothinman, $channels = `asterisk -r -x "show channels"` |
| 15:42.27 | ibob63 | this might not be the place. I need to get a fax-email provider and number is the US can anyone recommend a provider? |
| 15:42.36 | nothinman | oouch, I didn't know I can run the command without entering asterisk's "sub-shell" |
| 15:42.38 | fourcheeze | what's a good way to limit the amount of transcoding I do on *? |
| 15:42.47 | fourcheeze | ATM each client negotiates a codec with * |
| 15:42.54 | [TK]D-Fender | fourcheeze : Always use the same codec for everything ;) |
| 15:42.55 | fourcheeze | I realise I could force them all to be the same |
| 15:42.59 | fourcheeze | :-) |
| 15:43.04 | fourcheeze | I'd rather avoid that |
| 15:43.05 | [TK]D-Fender | well... DUH! |
| 15:43.05 | nothinman | Nivex, _Paulo_ -- thanks, I will give it a try in 5 minutes :-) |
| 15:43.27 | fourcheeze | OTOH I'd prefer if I could ensure that calls at least used the same codec if possible |
| 15:43.43 | fourcheeze | i.e. end points negotiate |
| 15:43.50 | *** join/#asterisk mopri (n=jjohn@201.192.107.58) |
| 15:44.00 | [TK]D-Fender | fourcheeze : I believe you just need to set the preference order. |
| 15:44.37 | warthawg | fourcheeze, thanks for your help, hours and hours of time you have saved me, i appreciate it |
| 15:44.37 | [TK]D-Fender | and ensure that the order is compatible |
| 15:44.38 | Tagor | Is it possible to define an own soundfile in the dial command (exten => s,3,Dial(SIP/${ARG1},20)) ? |
| 15:44.38 | warthawg | i was getting ready to replace ubuntu |
| 15:44.38 | nothinman | Nivex, _Paulo_ -- yeah, bingo. Thanks a lot! |
| 15:44.38 | fourcheeze | [TK]D-Fender: compatible with what? |
| 15:44.38 | Nivex | nothinman: You're welcome. |
| 15:44.39 | _Paulo_ | :-) |
| 15:44.41 | mopri | hi.. i'm planning to the asterisk PC, work'n as a proxy on my network, so it can do QoS at the same time. Anyone would suggest doing otherwise? |
| 15:44.45 | fourcheeze | warthawg: np - no point replacing ubuntu - you can always apt-get update to debian |
| 15:45.10 | nothinman | too long day today, simplest way is always hard to find ;-) |
| 15:45.12 | warthawg | fourcheeze, do you have zaptel running as well? |
| 15:45.18 | fourcheeze | warthawg: no |
| 15:45.18 | _Paulo_ | mopri, I think its a good idea. |
| 15:45.29 | warthawg | then my work here is done |
| 15:45.31 | mopri | Paulo: thanks |
| 15:45.33 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 15:45.48 | *** join/#asterisk lorinc (n=ang@caracas-1513.adsl.interware.hu) |
| 15:45.50 | _Paulo_ | mopri, just dont run something memory hungry as squid... |
| 15:46.23 | [TK]D-Fender | fourcheeze : that the order matches for all endpoints. though you really should force a codec. What is there to leaev to chance when you know what your gear can do? |
| 15:46.44 | iDunno | this afternoon is mostly sucking. |
| 15:47.10 | Katty | :< |
| 15:47.16 | cpm | bummer |
| 15:47.20 | fourcheeze | [TK]D-Fender: well, because end users might want to fit more down a single ADSL connection than they could with ulaw, or might want better sound quality than they get with gsm |
| 15:47.40 | Katty | ooh! |
| 15:47.42 | Katty | shredding material! |
| 15:48.02 | synthetiq | http://www.collegehumor.com/movies/1665904/ |
| 15:48.42 | fourcheeze | when storing locations for clients behind NAT, why does * sometimes store the internal IP and other times the external one ? |
| 15:49.35 | fourcheeze | I'm talking about the Reg. Contact in sip show peer |
| 15:49.48 | fourcheeze | Reg. Contact : sip:20050125@some.ip.number:1026;line=xculrlmf |
| 15:50.03 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 15:50.07 | GerbilNut | fourcheeze, do you have nat=yes for the client in the sip.conf? |
| 15:50.09 | fourcheeze | why is some.ip.number sometimes private, sometimes public |
| 15:50.15 | fourcheeze | GerbilNut: yes I have |
| 15:50.24 | [TK]D-Fender | fourcheeze : just let it transcode... you can't have people make this up as they go along. no gaurantees is the endpoints have different codec priorities. |
| 15:50.40 | fourcheeze | [TK]D-Fender: itseems apps like SER have this more sorted |
| 15:51.35 | fourcheeze | I remember someone once posting overheads for different transcodes - anyone know where that it? |
| 15:51.42 | fourcheeze | s/it/is |
| 15:51.52 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc07r.dialup.mindspring.com) |
| 15:52.02 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 15:52.44 | Egonis | I have 10+ IAX Phones, 8 of which are in the internal network, and work fantastic. The two external phones can do calls, etc with no issues.. except they cannot access VM -- 'No username but # key pressed' messages show in the logs |
| 15:54.58 | Tagor | Is it possible to define an own soundfile in this dial command (exten => s,3,Dial(SIP/${ARG1},20)) ? |
| 15:57.41 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 15:58.16 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 15:59.01 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:59.23 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:00.23 | [Airwolf] | Egonis, I can't help you with your problem. But can you tell me, what iax phones you have ? |
| 16:04.17 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 16:12.50 | *** join/#asterisk fu3 (n=kka@234-200-29-134.hcc.mnscu.edu) |
| 16:14.19 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 16:15.20 | skeffling | Tagor: it depends on your phones, some phones support it It hink |
| 16:15.54 | skeffling | Tagor: see http://www.voip-info.org/wiki/view/Asterisk+phone+snom for SNOM Custom ring tone |
| 16:15.56 | Tagor | Isn't it possible to just replace the sound? |
| 16:16.27 | skeffling | Tagor: it's not that simple |
| 16:16.38 | Tagor | Ok, ty |
| 16:16.44 | fu3 | dang it |
| 16:16.51 | fu3 | i cannot get this echo to cancel out :| |
| 16:17.08 | websae | fu3: who is your provider? |
| 16:17.14 | fu3 | this is all local |
| 16:17.16 | fu3 | just testing |
| 16:17.21 | fu3 | voip to pots :| |
| 16:17.38 | fu3 | i've tried the MG2 and MARK2 w/ AGGRESSIVE echo cancellation |
| 16:17.39 | fu3 | no dice |
| 16:17.50 | fu3 | i've messed with tx and rxgain |
| 16:18.06 | SwK[Work] | anyone know a free 411 service for north america? (preferably on a tf num) |
| 16:19.03 | wunderkin | what was it, 800-free-411? |
| 16:20.50 | websae | that's free? |
| 16:21.02 | websae | very interesting--just called it |
| 16:21.06 | *** part/#asterisk Alric (n=nbowyer@l-69-148-121-117.stu.swau.edu) |
| 16:22.00 | *** join/#asterisk tuxinator_linux (n=tuxinato@m010e36d0.tmodns.net) |
| 16:23.00 | skeffling | fu3 is this zaptel? if you do zap show channel 1 does that say that echo cancellation is on? (eg says: Echo Cancellation: 128 taps unless TDM bridged, currently ON) |
| 16:23.55 | *** join/#asterisk kremin (n=kremin@dslb-084-063-108-009.pools.arcor-ip.net) |
| 16:23.58 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:24.04 | generalhan | whats up all !? |
| 16:24.20 | *** join/#asterisk davy_ (n=davy_vdm@87.215.46.109) |
| 16:24.24 | *** join/#asterisk ManxPower (n=ewieling@stirprop-S4-0-0-21.ndcr2.datasync.net) |
| 16:25.08 | generalhan | is there a place that the # transfer options are stored ? i changed the context that the phone extensions were in and now i cant transfer calls ? |
| 16:25.14 | ManxPower | ~docs |
| 16:25.15 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:25.26 | *** join/#asterisk hyphen (n=murphyx@c-68-81-127-186.hsd1.pa.comcast.net) |
| 16:27.32 | *** join/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 16:27.40 | *** part/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 16:27.57 | *** join/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 16:28.05 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 16:31.04 | *** join/#asterisk coolhp (n=crap@modemcable240.139-203-24.mc.videotron.ca) |
| 16:31.09 | coolhp | Good day all ! |
| 16:31.21 | essaredee | bonjour |
| 16:31.32 | coolhp | bonjour ! :-) |
| 16:31.44 | essaredee | vous parlez français? |
| 16:33.00 | angler | slt! |
| 16:33.02 | coolhp | I have a very strange issue and I cant seem to find a solution for it. I have about 40 Cisco phones connected to asterisk using the 7.4 SIP image. For some reason, they lose theyre registration after a while. |
| 16:33.16 | coolhp | I see them trying to reregister but it doesnt seem to take. |
| 16:33.21 | *** join/#asterisk SibRw0rk (n=DaPhrek@std-206-231-231-48.i-2000.net) |
| 16:33.29 | generalhan | coolhp: you using 7960s ? |
| 16:33.32 | coolhp | Yeah |
| 16:33.35 | generalhan | yep |
| 16:33.39 | coolhp | 7960s and 7940s |
| 16:33.41 | generalhan | i had the same problem with the 25 that i have |
| 16:33.47 | coolhp | Oh ! |
| 16:33.56 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 16:33.57 | coolhp | And sip show peers would show them as unregistered ? |
| 16:34.04 | generalhan | go into the SIPDefaul.cnf file and change the registration time from 120 to 30,000 |
| 16:34.30 | generalhan | once i did that they never came unregistered so i never had the problem again |
| 16:34.31 | kink0 | hello, anybody knows how to return a 34 code ( no available ) when calls are incoming in determinated hours ? |
| 16:34.44 | ManxPower | coolhp, are they behind nat |
| 16:34.49 | coolhp | LOL |
| 16:34.55 | coolhp | Nope.. .no nat |
| 16:35.16 | coolhp | Its in and office environment. |
| 16:35.22 | ManxPower | kink0, you mean return an ISDN CAUSE CODE 34? |
| 16:35.33 | coolhp | looking at SIPDefault.cnf |
| 16:35.51 | kink0 | ManxPower, not sure ISDN cause, SIP return 34, to indicate to the peer we are not available at that time |
| 16:36.01 | coolhp | generalhan ? |
| 16:36.07 | generalhan | ? |
| 16:36.12 | coolhp | Was it timer_register_expires ? |
| 16:36.33 | generalhan | yea |
| 16:36.36 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 16:36.38 | coolhp | k |
| 16:36.43 | coolhp | Did you change it in sip.conf too ? |
| 16:36.50 | generalhan | umm |
| 16:37.01 | generalhan | i dont have any registration lines in my sip.conf entries |
| 16:37.15 | coolhp | Yeah |
| 16:37.21 | coolhp | LOL |
| 16:37.23 | coolhp | Sorry. |
| 16:37.25 | coolhp | My bad. |
| 16:37.52 | coolhp | Did you change the maxexpirey param in sip.conf ? |
| 16:37.57 | kink0 | ManxPower, I pretend to inform the peer when we are not available to route calls, so he can use other secondary route. |
| 16:37.58 | coolhp | Or did you just comment it out ? |
| 16:38.25 | Egonis | [Airwolf]: Gnet VP320I |
| 16:38.33 | generalhan | i dont define that in sip.conf either |
| 16:38.41 | generalhan | coolhp: do you have the split entries in sip.conf for each cisco phone ? one for "peer" and one for "user" ? |
| 16:38.51 | coolhp | Nope. |
| 16:38.54 | coolhp | I use friend. |
| 16:38.54 | ManxPower | kink0, run the Congestion application |
| 16:39.05 | generalhan | when i upgraded from 7-3 to 7-4 i had to switch it |
| 16:39.33 | coolhp | To switch from friend to peer and user ? |
| 16:39.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:39.56 | generalhan | no, hang on |
| 16:40.09 | kink0 | ManxPower, I get the congestion info when I use other asterisk to route calls and the main asterisk is out of service, but the peer told me they need a 34 code to detect our status. |
| 16:41.11 | ManxPower | kink0, Congestion should send a 34 |
| 16:41.12 | [TK]D-Fender | ISDN 34 is a deprecated "busy" signal IIRC... i get those on Allstream PRI calls sometimes... |
| 16:41.24 | ManxPower | NotFreak, 34 is No channel available |
| 16:41.33 | docelm0 | Anyone in here good with monitoring channels? Im having issues with ChanSpy |
| 16:41.33 | ManxPower | no 34 is No Channel Available |
| 16:41.48 | kink0 | ManxPower, ok, I will try now with priority 102, Congestion |
| 16:42.43 | caio1982 | tzafrir: should i use the unreleased tag in all of my .debs till someone revise them, right? |
| 16:42.52 | Tagor | Does anybody know a softphone where I can change the ringtone which the caller hears? |
| 16:43.28 | *** part/#asterisk hyphen (n=murphyx@c-68-81-127-186.hsd1.pa.comcast.net) |
| 16:43.53 | fu3 | what kind of interrupt rate should I be experiencing on my tdm400p ? Any benchmarks available? |
| 16:44.02 | ManxPower | kink0, 102 is run when something is BUSY, not when it's congested |
| 16:44.04 | fu3 | im up to like a rate of 1002 and it's idle. |
| 16:44.16 | ManxPower | fu3, that's normal |
| 16:44.23 | fu3 | of 16 million interrupts since the boot, 4 million are for the card. |
| 16:44.28 | fu3 | oh really? cool.. thats a relief. thanks. |
| 16:44.29 | ManxPower | right around 1,000 per second |
| 16:44.50 | kink0 | ManxPower, I have 1, 2 ...6 priorities, what must then I use to Congestion ? |
| 16:44.56 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 16:44.57 | kink0 | 7 ? |
| 16:44.57 | fu3 | This echo I am getting could still be hardware related.. |
| 16:45.00 | ManxPower | kink0, What verison of Asterisk |
| 16:45.05 | digime | where is the CID set in asterisk config? |
| 16:45.19 | ManxPower | digime, what protocol? Incoming or outgoing? |
| 16:45.25 | digime | out |
| 16:45.29 | digime | SIP |
| 16:45.33 | kink0 | 1.2.3 |
| 16:45.38 | ManxPower | kink0, You should not rely on priority jumping because it will not be supported after 1.2 |
| 16:45.43 | ManxPower | callerid= in sip.conf |
| 16:45.48 | digime | hmm k |
| 16:45.50 | docelm0 | Does chanspy only work with Queues? |
| 16:46.00 | ManxPower | kink0, see the macro-stdexten in extensions.conf.sample |
| 16:46.03 | ManxPower | docelm0, no., |
| 16:46.36 | digime | ManxPower: under [general]? |
| 16:46.42 | docelm0 | ManxPower, I followed the Wiki and it just scans but I never hear audio.. and it never connects to a call.. |
| 16:47.27 | ManxPower | digime, no in each friend/peer/user section |
| 16:47.33 | digime | ah k |
| 16:47.40 | digime | there is no global CID to set? |
| 16:47.42 | ManxPower | JUST LIKE IN sip.conf.sample |
| 16:47.47 | digime | ok checking |
| 16:47.56 | ManxPower | digime, I dunno. Nobody I know would use a global setting |
| 16:48.13 | digime | nod. dont have sample file, can you paste me what the line should look like (if you dont mind) |
| 16:48.25 | fu3 | Are there any good docs on the ordering of zapata.conf? I'm notcing changes depending on the order.. yet I see almost nothing online about the ordering. |
| 16:48.53 | ManxPower | fu3, nothing really to document. when you set something in zapata.conf that is set until you change it. |
| 16:48.59 | digime | Manxpower: nvrmind, found it |
| 16:49.12 | digime | callerid="name" <number> |
| 16:49.17 | fu3 | hmm.. |
| 16:49.24 | ManxPower | i.e. if you set signalling=fxo_ks then have channel =>1 and then channel =>2 then the signalling for channel 2 is fxo_ks |
| 16:49.26 | fu3 | i'll have to reexamine my conf with that in mind. |
| 16:49.34 | ManxPower | digime, DON'T USE QUOTES |
| 16:49.52 | ManxPower | some phones and service providers will reject calls with callerid with quotes in it |
| 16:50.18 | digime | the CID is working interally but when i call out it shows a strange number. weird. i want CID to show 4 digit extension internally and our 800# externally. how is this possible? |
| 16:50.42 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc07r.dialup.mindspring.com) |
| 16:50.51 | ManxPower | digime, SetCallerID(8005551515) before the call goes out (in your dialplan) |
| 16:51.10 | tzafrir | caio1982, the "UNRELEASED" tag is for changelog in the svn to note that it is not should not be built |
| 16:51.27 | digime | manxpower: thanks |
| 16:51.29 | tzafrir | (usually: because a new version was just added) |
| 16:51.58 | caio1982 | tzafrir: i thought that this tag was for NMU too, or sorta |
| 16:52.19 | digime | manxpower: does this mean i can leave quotes in sip.conf |
| 16:52.40 | ManxPower | digime, you should NEVER have quotes in callerid info for the reasons I just stated. |
| 16:52.48 | digime | okay. good |
| 16:53.02 | digime | so just callerid=Name <number> |
| 16:53.36 | ManxPower | Correct |
| 16:53.49 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 16:54.03 | digime | ManxPower: thank you!!!!! |
| 16:55.14 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 16:55.21 | shmaltz | gmail is down again :( |
| 16:55.58 | twisted[asteria] | gmail is never down |
| 16:56.21 | twisted[asteria] | in fact |
| 16:56.28 | twisted[asteria] | it's so not down right now that i'm checking it |
| 16:56.51 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 16:56.57 | astvoip | yes |
| 16:57.04 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.150.139) |
| 16:57.06 | astvoip | i do check it right now |
| 16:57.11 | shmaltz | no such thing as never, (unless you talking about the word 'never' and not the definition) |
| 16:57.15 | astvoip | its working.. fine |
| 16:57.27 | jbenson | Hi everyone - Does anyone know if there is an issue with UK CID in 1.2.4 on the TDM400P please? |
| 16:58.03 | tzafrir | jbenson, did you set it to use reverse polarity or whatever (something with v23 in zapata.conf IIRC) |
| 16:58.08 | tzafrir | ? |
| 16:58.36 | jbenson | yes, I have set cidsignalling=v23 |
| 16:58.37 | jbenson | cidstart=polarity |
| 16:59.01 | jbenson | it worked fine on 1.0 just seems to be broken on 1.2 |
| 16:59.15 | andrebarbosa | anyone can help me with SIP reinvites using realtime with asterisk 1.2.0? |
| 16:59.30 | jbenson | I get this error "Feb 27 16:44:03 WARNING[6869]: chan_zap.c:3950 zt_handle_event: Didn't finish Caller-ID spill. Cancelling." |
| 16:59.37 | shmaltz | astvoip, the server you are connected to might work fine, but the one I am connected to, doesnt' |
| 17:00.15 | andrebarbosa | i have to 2phones: * Name : xlite ; CanReinvite : Yes ; Codecs : 0x4 (ulaw) |
| 17:00.22 | shmaltz | tzafrir, http://www.xorcom.com/astribank/features.html does this offer CID? |
| 17:00.47 | shmaltz | tzafrir, Yes it does, its on that page |
| 17:01.02 | shmaltz | tzafrir, any plans on offering FXO? |
| 17:01.04 | andrebarbosa | <PROTECTED> |
| 17:01.43 | andrebarbosa | my dial string on Database: 59 | internal_ | 702 | 1 | Dial | SIP/alex |
| 17:01.48 | andrebarbosa | no dial options |
| 17:01.51 | andrebarbosa | no nat on both phones |
| 17:01.59 | andrebarbosa | and I don't get reinvite |
| 17:02.00 | andrebarbosa | :\ |
| 17:02.24 | andrebarbosa | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 17:02.27 | andrebarbosa | :( |
| 17:02.34 | ManxPower | using the same codec for all legs of the call? |
| 17:02.41 | andrebarbosa | ya |
| 17:02.42 | ManxPower | not using any dial options. |
| 17:02.51 | andrebarbosa | you can see my configs |
| 17:03.01 | ManxPower | and we don't really care what the database says, what does the CONSOLE say? |
| 17:03.01 | andrebarbosa | only G.711u on both phones |
| 17:03.08 | andrebarbosa | yah |
| 17:03.28 | andrebarbosa | that's my output from "sip show peer 'name'" |
| 17:03.38 | andrebarbosa | I have realtime cache enable btw |
| 17:03.59 | caio1982 | tzafrir: what's the best approach when the source didnt come with a Makefile: create one and put that in the orig tarball (i dont think upstream will even accept it, though) or put all Makefile magic inside debian/rules? |
| 17:06.22 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 17:06.35 | mover | why the g726 sounds very choppy? What is went wrong? |
| 17:06.44 | ManxPower | I'm still waiting for the CLI output of the Dial command. |
| 17:06.50 | ManxPower | mover, using MeetMe? |
| 17:07.19 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 17:07.27 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 17:07.30 | mover | ManxPower: the module is loaded but not in use |
| 17:07.44 | jbenson | is anyone here using a TDM400P in the UK with Caller ID please? |
| 17:08.55 | mover | ManxPower: ztdummy is loaded but what zapata.con i should uye? |
| 17:08.56 | coolhp | Out of curiosity, do most of you use type=friend ? or a combination of type=peer and type=user ( in a simple office pbx setup with cisco ip phones). |
| 17:09.28 | andrebarbosa | Executing Dial("SIP/xlite-6bd6", "SIP/alex") |
| 17:09.35 | mover | ManxPower: ztdummy is loaded but what zapata.conf should i use? |
| 17:09.36 | andrebarbosa | that's it? |
| 17:09.59 | ManxPower | mover, I only asked because MeetMe has issues with audio packet size that is not 20ms |
| 17:10.11 | ManxPower | andrebarbosa, I don't have any more suggestions |
| 17:10.19 | ManxPower | mover, I don't have any more suggestions |
| 17:10.27 | mover | k |
| 17:10.27 | andrebarbosa | :( |
| 17:10.30 | andrebarbosa | thanks! :) |
| 17:10.45 | andrebarbosa | just one more question, i have SIP debug enable |
| 17:10.53 | andrebarbosa | and i can see |
| 17:10.55 | andrebarbosa | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 17:11.17 | andrebarbosa | should i see: Allow: Re-INVITE, etc.. ? |
| 17:11.26 | fu3 | So.. what kind of coffee pots do you guys use? |
| 17:11.31 | fu3 | I got a Mr. Coffee FTX25 and it sucks. |
| 17:11.37 | mut | get bun |
| 17:11.44 | fu3 | specific model? |
| 17:11.54 | mut | buunn rather |
| 17:11.56 | fu3 | I need somethign that'll give me a solid "real" 12 cups.. not four. |
| 17:11.56 | mut | gr |
| 17:11.57 | mut | bunn |
| 17:12.04 | tzafrir | caio1982, I suppose that create a makefile if you intend to feed it to upstream |
| 17:12.29 | fu3 | I overflowed this Mr. Coffee and the LED is now destroyed :| |
| 17:12.33 | fu3 | err LCD. |
| 17:12.39 | tzafrir | caio1982, BTW: such questions could go to #debian-devel |
| 17:13.30 | caio1982 | tzafrir: okay, thanks |
| 17:13.36 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 17:14.00 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:14.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:14.19 | mut | um |
| 17:14.43 | shmaltz | tzafrir, any plans for FXO ports? |
| 17:15.28 | Chotaire | hm, any idea why SetCallerPres(prohib_not_screened) does not work with latest stable chan_capi-cm and * v1.2.4? |
| 17:15.32 | Chotaire | it will still display caller id |
| 17:15.36 | Chotaire | (outgoing) |
| 17:16.21 | tzafrir | shmaltz, hmmm.... have a look at genzaptelconf.... |
| 17:16.52 | shmaltz | tzafrir, url? |
| 17:17.40 | mut | http://housewares.hardwarestore.com/38-200-coffeemakers/12-cup-coffeemaker--612698.aspx |
| 17:17.44 | *** join/#asterisk heroine (n=heroine@m37.net81-66-58.noos.fr) |
| 17:17.45 | mut | ours is kinda like this one |
| 17:18.02 | tzafrir | http://xorcom.com/drivers/astribank/zaptel-xpp-1.2.1/xpp/genzaptelconf |
| 17:20.03 | *** join/#asterisk Rhizome (i=id@tor/session/x-d3a5b1bfcabac2cd) |
| 17:20.48 | *** join/#asterisk Zealxor (n=Zeal@ua-83-227-157-135.cust.bredbandsbolaget.se) |
| 17:20.53 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 17:21.28 | Zealxor | is asterisk free software? |
| 17:21.33 | heroine | Zealxor: yes |
| 17:21.46 | Rhizome | Depends what free means :P |
| 17:22.03 | Zealxor | is the asterisk protocol free software ? |
| 17:22.10 | Zealxor | like in gnu free software |
| 17:23.35 | ManxPower | The IAX protocols are free. |
| 17:23.39 | PoWeRKiLL | Hello |
| 17:23.50 | Zealxor | nice :D then I will try it out |
| 17:24.10 | PoWeRKiLL | Someone know how to use genmc utility to generate a certificate for the sipura |
| 17:27.04 | Chotaire | and even more gayness... callerid is also transmitted thru SIP calls... duh! |
| 17:27.46 | Chotaire | ah no, my fault... |
| 17:27.57 | Chotaire | but still, that caller-id problem with chan_capi-cm sucks. |
| 17:31.29 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 17:32.22 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 17:33.56 | Rhizome | When asterisk gives a warning and sayes to check debug info, how exactly do you do that? |
| 17:34.07 | ManxPower | dmesg | less |
| 17:34.24 | Rhizome | uhm |
| 17:34.37 | Rhizome | no :) |
| 17:34.40 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:34.47 | ManxPower | Rhizome, well be more specific |
| 17:35.09 | salviadud | yeah, what type of warning? |
| 17:36.05 | Rhizome | hang on, let me dig it out :P |
| 17:36.50 | Rhizome | Feb 27 17:31:06 WARNING[8421]: res_config_mysql.c:331 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 17:37.25 | Rhizome | And it's not the obvious, cause it can access the database most of the time, I just want to find where exactly it can't |
| 17:37.46 | salviadud | duuuuuuuuuuuuuuuuuuuuude. i don't know... |
| 17:37.51 | Rhizome | hehe |
| 17:37.55 | Rhizome | me neither :/ |
| 17:38.07 | salviadud | you see, i've never used mysql |
| 17:38.16 | salviadud | mysql and asterisk is taboo for me right now |
| 17:38.37 | salviadud | well, put together... asterisk i know, mysql, not yet |
| 17:39.13 | salviadud | well...i'm guessing asterisk could debug it for you |
| 17:39.35 | salviadud | try the help |
| 17:39.40 | Rhizome | I tried debug level filename :P |
| 17:39.56 | salviadud | rats... |
| 17:40.57 | *** join/#asterisk ToTo (n=ToTo@host72-142.pool874.interbusiness.it) |
| 17:41.01 | Rhizome | You should try realtime though, no more digging around in extensions.conf :D |
| 17:41.19 | salviadud | really? |
| 17:41.31 | [Airwolf] | Rhizome, realtime rules |
| 17:42.05 | salviadud | does realtime have a GUI? |
| 17:42.05 | [Airwolf] | Rhizome, but you can turn on debug in logger.conf and then check your /var/log/asterisk/ |
| 17:42.22 | [Airwolf] | salviadud, don't know. Have to make one myself |
| 17:43.06 | Rhizome | astbill is the closest I think :P |
| 17:43.08 | salviadud | i'm gonna read about it right now i guess... nothing better to do |
| 17:43.25 | Rhizome | thanks [Airwolf], i'll try that |
| 17:44.45 | ManxPower | Yeah, then you have to muck about in the mysql database rather than a nice clean text file. |
| 17:45.05 | ManxPower | this is asterisk, of course it doesn't have a gui |
| 17:45.49 | russellb | there are about 50 you can pay for |
| 17:45.59 | ManxPower | russellb, and they all suck |
| 17:46.04 | russellb | heh |
| 17:46.12 | russellb | I haven't personally tried any of them ... |
| 17:46.13 | Rhizome | The end :P |
| 17:46.44 | ManxPower | If it's a GUI then it sucks. |
| 17:46.50 | salviadud | i don't use gui |
| 17:46.53 | Crontibs | thats like saying gui for ios |
| 17:46.56 | Crontibs | on cisco stuff |
| 17:46.58 | salviadud | yet, i need to ask |
| 17:47.11 | Crontibs | .. manxpower hit good point gui=sucks |
| 17:47.29 | russellb | i would say they suck |
| 17:47.34 | russellb | it's just really easy to make one that sucks |
| 17:47.34 | Rhizome | I tend to agree, making a good userfriendly gui is alot harder than making something do what you want! *shiver* |
| 17:47.35 | salviadud | manxpower, what distro are you using? |
| 17:47.44 | russellb | wouldn't* say |
| 17:48.59 | ManxPower | salviadud, Mandrake |
| 17:49.11 | salviadud | no wonder you hate guis man |
| 17:49.25 | Crontibs | haha |
| 17:49.25 | ManxPower | I just don't like WIMPS (Window Icon Mouse Pointer System) Interfaces |
| 17:49.40 | ManxPower | At least for system admin |
| 17:49.49 | ManxPower | I don't mind them for just general desktop use. |
| 17:50.16 | salviadud | i'm a slackware user. guis are evil! |
| 17:50.21 | Rhizome | You don't like phpmyadmin for example? |
| 17:51.58 | Chotaire | how great.. SetCallerID and SetCallerPres will not work in some cases. anyone related here who wants to hear details? |
| 17:52.36 | russellb | Rhizome: good example |
| 17:53.01 | Rhizome | [Airwolf], thanks, found my error, unknown column somewhere :P |
| 17:54.19 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 17:54.42 | *** join/#asterisk santiagon (n=santiago@209.92.141.140) |
| 17:54.51 | ManxPower | Chotaire, not really if it's CAPI. Go talk to the CAPI people, |
| 17:55.09 | santiagon | Hi all... looking for some help with Asterisk@Home. Am I in the right place? |
| 17:55.14 | ManxPower | ~amp |
| 17:55.15 | jbot | i heard amp is NOT supported here! people using it should join #amportal |
| 17:55.49 | Chotaire | it's not really CAPI |
| 17:55.52 | Chotaire | because it's a SIP call ;) |
| 17:56.21 | Chotaire | it seems any callerid stuff will be ignored if the caller leg is a CAPI caller |
| 17:57.33 | Chotaire | to be hidden caller id will be displayed ("9xxxxx") (not as intended) |
| 17:57.33 | Chotaire | <PROTECTED> |
| 17:57.34 | Chotaire | <PROTECTED> |
| 17:57.34 | Chotaire | <PROTECTED> |
| 17:57.34 | Chotaire | caller id will not be displayed (as intended): |
| 17:57.36 | Chotaire | <PROTECTED> |
| 17:57.38 | Chotaire | <PROTECTED> |
| 17:57.55 | Chotaire | even if doing the first one exactly as the latter one, it will be bad. |
| 17:58.06 | Chotaire | the only difference is CAPI/ISDN/.... that's all. |
| 17:58.09 | russellb | ~pb |
| 17:58.11 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:58.16 | Chotaire | sorry |
| 17:58.57 | Chotaire | well, it seems if the caller leg is coming from CAPI/ISDN all the settings are frankly ignored. |
| 17:58.57 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 17:59.04 | Chotaire | although it's a sip call. |
| 18:00.18 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:00.49 | trelane_ | Chotaire, tried using the |a option in set caller id to override ANI? |
| 18:01.02 | Chotaire | yup, tried that. |
| 18:02.43 | wrench | hi asterisk'ers... question about asterisk in production environment - which is most efficient? executing advanced logic and DB related functions with AGI through a C program, or using the perl addon for builtin support |
| 18:02.58 | Chotaire | like this? |
| 18:03.02 | Chotaire | exten => s,2,SetCallerId("Anonymous"|a) |
| 18:03.10 | Hmmhesays | agi with C? |
| 18:03.58 | wrench | Hmmhesays > even though it has to execute an external program instance? |
| 18:04.12 | Hmmhesays | i'm not answering your question with that statement |
| 18:04.16 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 18:04.44 | wrench | oh, well yes AGI can call C apps |
| 18:05.18 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:05.41 | *** join/#asterisk nain (n=nain@202.59.73.36) |
| 18:06.40 | Chotaire | this is funny, the caller id presented is always "Unknown", which I haven't defined anywhere. |
| 18:06.51 | Chotaire | I tried all possibilities... so I consider it a bug. |
| 18:07.11 | denon | Chotaire: read the wiki |
| 18:07.22 | Chotaire | I did. |
| 18:07.28 | denon | in sip.conf [general], there is a directive called "callerid=" |
| 18:07.31 | Chotaire | atleast the stuff about SetCallerId |
| 18:07.41 | Chotaire | denon: so why does it work if the caller leg is a SIP call? |
| 18:07.50 | Chotaire | and it doesn't work if the caller leg is a CAPI call? |
| 18:08.02 | *** join/#asterisk stoffell (n=stoffell@d51A4D5C1.access.telenet.be) |
| 18:08.03 | denon | no clue, I never touch CAPI |
| 18:08.07 | denon | maybe capi.conf has the same |
| 18:08.19 | Chotaire | well, my config worked fine with 1.0.3 |
| 18:08.39 | Chotaire | I changed it to new format after I found out it didn't work in this one occasion... which didn't help. |
| 18:08.45 | Chotaire | reporting bug #3 today |
| 18:09.33 | russellb | chan_capi is not in Asterisk. |
| 18:09.52 | russellb | so if this bug report is about that, please report it to the chan_capi people |
| 18:10.30 | heroine | i'm trying to call from a phone to another but the call failed and i get the voicemail prompt .. i got this message in asterisk console but can't figure out why i get this as autocreatepeer=yes is set in sip.conf |
| 18:14.03 | Hmmhesays | Hey |
| 18:15.42 | *** join/#asterisk comfrey (n=comfrey@h-64-105-87-234.sttnwaho.covad.net) |
| 18:16.25 | Chotaire | I repeat, this is not a chan_capi issue. |
| 18:16.31 | Chotaire | it's a sip call. |
| 18:16.33 | Chotaire | http://bugs.digium.com/view.php?id=6607 |
| 18:16.40 | Chotaire | have fun... |
| 18:20.24 | heroine | hmm is it possible to get the asterisk console logs while not connected to it in some file or syslog ? |
| 18:21.55 | angler | logger.conf ..... verbose mode |
| 18:22.15 | heroine | hmm yes, just seen it in google .. thx :) |
| 18:26.38 | FuriousGeorge | is anyone working on a patch for sip presence accross multiple asterisk servers? |
| 18:28.23 | Ukyo | hey, is anyone here really familiar with the cisco 79xx series phones ? |
| 18:28.48 | digime | quick question about CID, when someone calls into our system it is stripping out 1, so it shows xxx-xxx-xxxx, and we can't just hit a button on our polycom to call the number back. any way to fix this? |
| 18:31.15 | jbenson | digime - how are your calling coming in to your server? |
| 18:31.37 | *** join/#asterisk distortion (i=distorti@junipero.3sheep.com) |
| 18:31.49 | russellb | Chotaire: have fun? what does that mean? |
| 18:32.19 | *** join/#asterisk hertell (n=rene@jumbo52.adsl.netsonic.fi) |
| 18:33.10 | Chotaire | russellb: because you can only test it if you have CAPI.... and I found something else, I have one phone that does never displayer CID as intended, it always displays "asterisk".. however, since update to 1.2.4 it will - when this bug is triggered - display "Unknown" instead... one more good guess it is definitely not chan_capi related. |
| 18:33.15 | hertell | good evening to all! |
| 18:34.08 | Chotaire | as if CID completely breaks on caller leg CAPI |
| 18:34.26 | digime | jbenson: ? |
| 18:34.43 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 18:35.11 | Crontibs | FuriousGeroge: couldn't you use SER for that ? |
| 18:36.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 18:37.25 | russellb | Chotaire: if callerid presentation is not set to allowed on the incoming leg of the call, chan_sip will send "Unknown". |
| 18:37.52 | *** join/#asterisk JamesVPBX_01 (n=proliant@VPN02-WLAN-0215.VPN.NYU.EDU) |
| 18:38.08 | JamesVPBX_01 | Morning |
| 18:38.20 | russellb | You will have to find a way to demonstrate your problem *without* using chan_capi if you want help from us. |
| 18:38.20 | Chotaire | russellb: hmmm.... |
| 18:38.39 | Chotaire | isn't callerid presentation allowed by default? |
| 18:38.51 | digime | <jbenson> digime - how are your calling coming in to your server - they are calling our 800# |
| 18:38.51 | russellb | depends on the channel driver |
| 18:38.52 | JamesVPBX_01 | Anyone working on any interesting projects linking asterisk to a website or web store? |
| 18:39.15 | Katty | yawn. |
| 18:39.23 | hertell | Does anyone have any links to working iptables-nat rules for an asterisk server behind a firewall? |
| 18:40.11 | Chotaire | russellb: so that means you want me to set CallerPres to allowed in order to disallow it? ;) |
| 18:40.46 | russellb | Chotaire: no, I want you to demonstrate that CallerID is not working as expected in case that does not involve chan_capi at all |
| 18:41.02 | russellb | that's the only way we can do anything about it |
| 18:41.29 | Crontibs | hertell: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 18:41.47 | digime | quick question about CID, when someone calls into our system it is stripping out 1, so it shows xxx-xxx-xxxx, and we can't just hit a button on our polycom to call the number back. any way to fix this? |
| 18:41.56 | PakiPenguin | hi , can anyone help me with this? |
| 18:41.57 | PakiPenguin | Feb 27 23:41:39 DEBUG[20701]: channel.c:1325 ast_hangup: Hanging up channel 'OutgoingSpoolFailed' |
| 18:41.58 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:42.07 | PakiPenguin | i am getting this error constantly |
| 18:42.11 | hertell | Crontibs: I have checked that out, but i still got trouble with incoming calls.. |
| 18:42.26 | *** join/#asterisk Mw3 (n=mw3@national.t-error.hu) |
| 18:44.16 | *** join/#asterisk Fiskfan (n=erikdj2@h35n3fls32o286.telia.com) |
| 18:47.06 | Chotaire | russellb: is that what you want me to do? |
| 18:47.07 | Chotaire | exten => _X.,1,SetCallerPres(allowed) |
| 18:47.07 | Chotaire | exten => _X.,2,SetCallerId("Anonymous"|a) |
| 18:47.21 | russellb | i don't know, you tell me |
| 18:47.33 | russellb | but whatever you report, it can't involve chan_capi |
| 18:47.34 | *** join/#asterisk CePe (n=jo_tjena@83.216.98.16) |
| 18:47.35 | russellb | that's all i'm saying at this point |
| 18:48.02 | FuriousGeorge | is there a solution for sip presence of peers on another * server? |
| 18:48.28 | Katty | sip presence of peers. |
| 18:48.31 | Katty | that sounds vampiric. |
| 18:48.51 | Katty | "is there a vampiric solution on another * server?" |
| 18:48.57 | *** join/#asterisk Chingetas (n=aon@201.137.207.22) |
| 18:48.59 | Katty | FuriousGeorge: sorry, fresh out! |
| 18:50.15 | Chingetas | Does anyone can help me with an issue with Asterisk as Sip UA? |
| 18:50.22 | Crontibs | Wouldn't SerSip Server |
| 18:50.31 | Crontibs | be able to do loadbalancing then have Ser |
| 18:50.35 | Crontibs | talk to asterisk |
| 18:54.00 | Chingetas | I'm trying this: OfficePBX<-->Addpac(2 fxo, 2fxs)SIP<----> ASterisk <--Trying SIP instead of Iax--> ASterisk<---> UA |
| 18:54.04 | [TK]D-Fender | Katty: :F |
| 18:54.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:54.51 | jbenson | digime - are the coming in over a Zaptel channel, SIP, IAX2? |
| 18:55.29 | FuriousGeorge | Katty: i was thinking API and manually setting say, devstate, in bripatch |
| 18:55.52 | digime | jbenson: SIP |
| 18:56.01 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
| 18:56.41 | FuriousGeorge | *bristuff patch |
| 18:59.10 | Katty | FuriousGeorge: oh ah. |
| 18:59.16 | Katty | [TK]D-Fender: don't you look at me like that! |
| 18:59.25 | [av]bani | \o> |
| 18:59.34 | MRH2 | Hi is there any way to stop asterisk being in the media path initially when canreinvite - yes ? |
| 19:00.00 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 19:00.05 | [TK]D-Fender | FuriousGeorge : You could run scripts on one to poll the other and then use DevState.... should work. |
| 19:00.08 | Darwin35 | ~amp |
| 19:00.10 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
| 19:00.24 | [TK]D-Fender | Katty: ...*LUNCH* ! |
| 19:01.27 | Katty | [TK]D-Fender: i am /not/ lunch! |
| 19:01.47 | [TK]D-Fender | ... post dinner, pre-midnight snack-meal? ;) |
| 19:02.37 | Katty | definately not. |
| 19:02.49 | [TK]D-Fender | heh |
| 19:04.13 | caio1982 | tzafrir: what's the best way to provide my stuff to review on the list? using same estructure of pkg-voip svn, just the debian/ dirs online somewhere or...? |
| 19:05.55 | *** part/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 19:07.08 | FuriousGeorge | [TK]D-Fender: i dont know if this is even possible, but i knida think it would be cool if you could ligically merge two * servers so the become aware of eachothers channels to some extent |
| 19:07.48 | *** join/#asterisk Mefios (n=Miranda@xdslt128.osnanet.de) |
| 19:08.02 | [TK]D-Fender | FuriousGeorge : what you might think of doing is to use something like SER in front. |
| 19:08.12 | Nivex | isn't that what "switch =>" is for? |
| 19:08.18 | FuriousGeorge | for instance, i heard chan_sip3 will support SLA |
| 19:08.38 | [TK]D-Fender | FuriousGeorge : Yeah.. thats for Digium's DS3 right? ;) |
| 19:08.39 | russellb | we have talked about doing some cool stuff like that using the AIS spec ... see http://developer.osdl.org/dev/openais/ |
| 19:08.46 | russellb | but it's a seriously non-trivial projecyt |
| 19:09.28 | Mefios | Hi. I need some help with sip telephony. when i call some person with my sxlite everything works finde, but when someone else calls me i don't hear him for a while |
| 19:09.29 | austinnichols101 | ~seen opsys |
| 19:09.48 | jbot | opsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 14d 13h 28m 46s ago, saying: 'betaboi" true'. |
| 19:10.01 | Mefios | after half a minute everything works fine again and we can understand each other |
| 19:11.09 | Mefios | is there someone who had the same problem? |
| 19:11.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.12 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 19:12.50 | Mefios | no answer? |
| 19:13.07 | FuriousGeorge | russellb: it appears AIS us what i was talking about. cool, and doesnt seem trivial at all :) |
| 19:13.13 | FuriousGeorge | *is what i was talking about |
| 19:13.35 | digime | jbenson: SIP |
| 19:14.00 | *** join/#asterisk sofh (i=sofh@30-150-254-84.skylogicnet.it) |
| 19:14.43 | sofh | hello everybody! |
| 19:14.50 | Mefios | hello sofh |
| 19:15.00 | sofh | i am having some problem while using oh323 |
| 19:15.11 | sofh | somebody would help me please ? |
| 19:15.43 | sofh | :| |
| 19:17.00 | santiagon | anyone know a down-and-dirty way to do a remote call forwarding via a virtual extension? |
| 19:18.32 | Mefios | could someone tell me if my problem just the fault of an asterisk noob or if it is a serious problem? |
| 19:20.07 | Mefios | mhh...i have to go now. maybe someone can answer my question tomorrow |
| 19:20.08 | Mefios | bye |
| 19:20.11 | [TK]D-Fender | santiagon : What is a "vitual" extension.... or a "real" one for that matter? |
| 19:21.11 | MRH2 | santiagon i use agent call back login for what i think you mean |
| 19:22.04 | sofh | hello..anybody can give me idea..is it good to use asterisk-oh323 to connect asterisk with voip carrier ? |
| 19:22.55 | [TK]D-Fender | sofh : H323 is *dead*. It is highly suggested you forget about it ASAP and start using SIP. |
| 19:23.00 | MRH2 | agent has number / logs in / dialing the the number reaches agent, agent logs in via a mobile, sip phone, whatever |
| 19:23.27 | sofh | i am using IAX basicaly and its working best for me in all conditions |
| 19:23.27 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:23.39 | sofh | but what if i want to connect asterisk with a carries who is only supporting h323 |
| 19:23.48 | [TK]D-Fender | sofh : So why are you even thinking about H323? |
| 19:23.51 | sofh | is ther any way to translate from h323>>>>IAX ? |
| 19:24.05 | [TK]D-Fender | sofh : I'm sure it could work, but you really DON'T wan that kind of pain... |
| 19:24.45 | sofh | Fender: work what ? h323 itself or u r telling me the way to covert the protocols |
| 19:24.58 | [TK]D-Fender | * translastes everything for you. All you need to do is take the call in from the source and spit it out to where you need it to go. Thats the point/advantage of * being a B2BUA |
| 19:25.22 | [TK]D-Fender | H.323 = trouble + Pain. Avoid if at all possible., |
| 19:25.56 | sofh | yes i am feeling , it took two days to successfuly compile asterisk-oh323 module |
| 19:26.12 | sofh | but what can i do i am forced by carrier..they want to send their calls only on h323 |
| 19:26.39 | sofh | not even on SIP,,otherwise SIP is also fine..though pesonally i like the quality of IAX |
| 19:27.17 | sofh | what iv'e to do is recieve the call from the carrier (h323) and then send to GSM GW with E1 card |
| 19:27.33 | sofh | h323 >>>asterisk >>>zap >>> GSM Gateway |
| 19:28.14 | sofh | but the problem i am facing when i or somebody else call on h323 , their voice goes fine on GSM mobile but it never comes back |
| 19:28.32 | [TK]D-Fender | no idea.... |
| 19:28.44 | sofh | means only one way voice pipe is established..no voice is coming back from gsm to asterisk |
| 19:28.49 | [av]bani | [TK]D-Fender: about 30% of the market still uses h323 only |
| 19:28.50 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 19:29.06 | sofh | whereis in the same configurations if i use h323..voice is in awesome qualiyt |
| 19:29.22 | sofh | oppps |
| 19:29.31 | sofh | i mean to say if i use IAX |
| 19:29.53 | djMax | so I have the SIP dropped registration problem back again... I have debug logs this time |
| 19:30.00 | [TK]D-Fender | [av]bani : 30% I doubt that..... |
| 19:30.34 | sofh | what if i recieve calls on h323 (asterisk box A) >>>IAX>>>aserisk box B >>>GSM gateway ? |
| 19:31.08 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 19:31.12 | Darwin35 | om amp is not bad but I dont like all the extra files |
| 19:31.38 | Darwin35 | looking at it in realtime for the first time |
| 19:31.50 | [av]bani | [TK]D-Fender: lots of people dont just throw out their existing h323 installations and buy new sip stuff |
| 19:32.09 | MikeJ[Laptop] | there is nothing wrong with h323 |
| 19:32.17 | MikeJ[Laptop] | why would somone throw it out |
| 19:32.26 | [av]bani | consider that h323 has a 10-15 year head start on sip |
| 19:32.44 | MikeJ[Laptop] | sip was just the cruel joke about making a simpler h323... |
| 19:33.23 | [av]bani | well, h323 was designed by telco engineers... sip was designed by programmers |
| 19:33.28 | MikeJ[Laptop] | yes.. |
| 19:33.35 | MikeJ[Laptop] | and they are both rediculous |
| 19:33.37 | sofh | i think h323 itself is not that bad (if network pipe is clear for tcp sessions) but for asterisk it doesn't seem to be that STABLE |
| 19:33.38 | [av]bani | :) |
| 19:33.39 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
| 19:33.54 | MikeJ[Laptop] | sofh, try using it with chan_woomera |
| 19:34.05 | [av]bani | sofh: its overly complex for what it does, but then its telco background sort of causes that |
| 19:34.16 | [av]bani | sofh: look at isdn :)) |
| 19:34.24 | [av]bani | or ss7 ... |
| 19:34.27 | MikeJ[Laptop] | from an rfc standpoint compare h323 to sip.. |
| 19:34.31 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust136.seac.broadband.ntl.com) |
| 19:34.47 | sofh | chan_woomera does support all codecs of asterisk ? like g729 |
| 19:34.47 | MikeJ[Laptop] | h323 has 3 times + the life of sip and still has much less rfc's |
| 19:34.53 | sofh | i didn't check chan_woomera |
| 19:35.03 | sofh | ss7...how i can use ss7 for GSM ? |
| 19:35.08 | [av]bani | well, h323 was also designed to solve different problems... nat wasnt really an issue in h323 because it was mainly used by ILECs with direct connections |
| 19:35.20 | MikeJ[Laptop] | chan woomera can support any of the codecs that open h323 can support |
| 19:35.35 | MikeJ[Laptop] | you can do all the nat stuff with a gatekeeper |
| 19:35.46 | MikeJ[Laptop] | which is how it is supposed to be implemented |
| 19:35.51 | [av]bani | MikeJ[Laptop]: yeah, but then h323 wasnt exactly 'extendible' like sip is... |
| 19:36.06 | justinu | level3 sure does a lot of fucking around with their NAS |
| 19:36.08 | [av]bani | tis amusing to look at sip and see it screaming http influences |
| 19:36.22 | [av]bani | in a udp based protocol |
| 19:36.52 | [av]bani | sip traces always make me lollerskate |
| 19:37.11 | justinu | 403 Forbidden |
| 19:37.43 | sofh | ok..i try tonight chan_woomera |
| 19:37.49 | sofh | hope it work for me :( |
| 19:38.02 | *** part/#asterisk ibob63 (n=hp@88-96-66-62.dsl.zen.co.uk) |
| 19:38.32 | sofh | btw is there any way to stop CLIR on Euro isdn signalling for a GSM GW ? |
| 19:39.14 | Katty | beep! |
| 19:41.08 | [av]bani | justinu: ever get echo probs on a 480i? |
| 19:42.14 | justinu | no, my 480i works pretty well |
| 19:42.51 | [av]bani | maybe its the 9112 or 9133 people are complaining about |
| 19:43.40 | generalhan | my users have the 9112is and they are great for echo cancellation |
| 19:43.46 | [av]bani | yeh, 9133 |
| 19:43.47 | generalhan | havent played with any of the others yet |
| 19:43.50 | [av]bani | http://www.voip-info.org/wiki/view/9133i |
| 19:44.07 | [av]bani | some guy having problems battling persistent echo on his 9133 |
| 19:44.44 | generalhan | i need to figure out why my cisco 7960s are doing the echo thing. it happens like 1 out of every 100 calls ... which is pretty good, but concidering that we do about 2000 or so calls a day it can get a little annoying |
| 19:45.02 | [av]bani | sure its the 7960 and not something else |
| 19:45.08 | generalhan | haha |
| 19:45.30 | generalhan | i only have the Aastra 9112i's and the 7960's |
| 19:45.43 | *** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 19:46.44 | *** part/#asterisk rva (n=rafa@200-158-80-124.dsl.telesp.net.br) |
| 19:49.15 | *** join/#asterisk backblue (n=moo@87-196-32-165.net.novis.pt) |
| 19:56.47 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:56.54 | *** join/#asterisk virterm (n=virterm@shiva.kanatek.com) |
| 19:57.04 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 19:59.01 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:02.16 | *** join/#asterisk mogorman (n=mogorman@68.62.165.220) |
| 20:03.19 | FuriousGeorge | mogorman: wadup |
| 20:03.25 | mogorman | ayer |
| 20:03.29 | mogorman | er aye |
| 20:03.45 | *** join/#asterisk Gahhef (n=ld@cpe-24-193-73-228.nyc.res.rr.com) |
| 20:04.12 | FuriousGeorge | no rattle |
| 20:04.16 | FuriousGeorge | a good sign |
| 20:04.39 | mogorman | heh |
| 20:04.43 | mogorman | i just moved |
| 20:04.48 | mogorman | let me get reall internet |
| 20:04.55 | mogorman | and my washer and dryer fixed |
| 20:05.03 | FuriousGeorge | k |
| 20:05.11 | Gahhef | any1 here have any experience getting a Cisco phone (SCCP firmware) with *? |
| 20:05.17 | *** join/#asterisk arctic_import (n=jasonj@mail.uui-alaska.com) |
| 20:05.18 | FuriousGeorge | good luck with that |
| 20:05.19 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-16.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 20:05.32 | mogorman | right now my internet providedr is linksys |
| 20:05.37 | mogorman | id like to get that fixed |
| 20:05.54 | FuriousGeorge | neighbor w/ no WEP key? |
| 20:06.23 | arctic_import | I hate echo |
| 20:06.31 | mogorman | many points |
| 20:06.35 | FuriousGeorge | ...ate echo ...echo |
| 20:06.36 | mogorman | i think i found 8 |
| 20:06.44 | FuriousGeorge | ...oints ...ts |
| 20:07.04 | arctic_import | haha |
| 20:07.21 | FuriousGeorge | mogorman: so if they left the default password on the gateway and try to foreward 4569 :) |
| 20:07.53 | FuriousGeorge | maybe you dont need ISP |
| 20:07.59 | mogorman | oh i already logged into their router |
| 20:08.02 | mogorman | and left them a note |
| 20:08.06 | FuriousGeorge | lol |
| 20:08.18 | mogorman | if i had my crossover cable for my iaxy / or a mic |
| 20:08.25 | mogorman | i could call and get comcast internet faster |
| 20:08.33 | mogorman | but now i need to call over ip relay |
| 20:10.19 | mogorman | brb have to flip over to router |
| 20:10.20 | *** join/#asterisk PakiPenguin_ (i=uppal@linuxpakistan/admin/pakipenguin) |
| 20:10.20 | mogorman | bye |
| 20:12.45 | [TK]D-Fender | I USED to get echo..... not any more...... |
| 20:13.05 | [av]bani | \o/ |
| 20:13.14 | [av]bani | echo is good for you |
| 20:13.22 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 20:15.26 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 20:16.59 | vgster | 9133i's are fine |
| 20:18.50 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:19.01 | FuriousGeorge | arctic_import: if youre using zap on one side try screwing with the gains |
| 20:19.08 | FuriousGeorge | also make sure you enabled echocancel |
| 20:20.18 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 20:21.13 | Katty | echo makes me all sad inside. |
| 20:21.18 | justinu | me too |
| 20:21.27 | file | Katty makes me happy |
| 20:21.39 | Katty | she would. |
| 20:23.04 | km- | anyone here have cisco 7905's and have the special keycode you enter in to unlock the settings? I've got a couple I have to reconfigure and can't get to the web interface |
| 20:23.17 | X-Rob_ | km-, 'cisco' |
| 20:23.36 | km- | you type that on the alpha keypad in the menu? |
| 20:24.02 | djMax | I'm seeing "Checking device state for peer xxxx" followed by a change of its state to unavailable. Any thoughts? |
| 20:24.56 | vgster | anyone got any pointers for getting inbound faxing working? my asterisk just ignores the fax tome |
| 20:25.18 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 20:26.45 | Hmmhesays | god I love it when customers say want me to know everything about the internet connection they are running their voip calls across |
| 20:27.16 | Hmmhesays | "do you think port 5060 is blocked?, do you think rtp is being dropped" blah blah blah blah blah |
| 20:27.21 | Hmmhesays | how the fark am I supposed to know |
| 20:27.35 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 20:28.18 | file | Hmmhesays: the easy answer to every one of those questions is, "42" |
| 20:29.30 | Katty | Hmmhesays: or hug. |
| 20:29.33 | Katty | Hmmhesays: that works well too. |
| 20:29.35 | Fiskfan | file, but then, what is the question? |
| 20:29.58 | Fiskfan | Hug you asterisk... |
| 20:30.17 | Hmmhesays | or a "i'm sorry i'm not able to answer that question |
| 20:30.33 | Katty | Hmmhesays: hop to it! |
| 20:30.41 | Hmmhesays | hop to what? |
| 20:30.52 | Fiskfan | it |
| 20:30.52 | Katty | 14:20 * Katty demands hugging. |
| 20:31.07 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 20:31.14 | Katty | k, all better. |
| 20:31.18 | Hmmhesays | damn I am a train wreck today, 2 hours of sleep |
| 20:31.32 | Katty | and, why, did you only get 2 hours of sleep again? |
| 20:31.52 | Katty | i sure hope it was for a good cause. |
| 20:31.55 | Katty | and not music related. |
| 20:31.57 | Hmmhesays | someone kept me up all night |
| 20:32.03 | Katty | oooh. |
| 20:32.08 | Katty | this sounds promising! |
| 20:32.19 | Hmmhesays | not really, same chicka |
| 20:32.23 | Katty | oh:< |
| 20:32.25 | Hmmhesays | the engaged one |
| 20:32.32 | file | naughty boy |
| 20:32.36 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 20:32.43 | Hmmhesays | did I ever claim to be anything else? |
| 20:32.54 | file | not really, no |
| 20:32.56 | _MartinCabrera_ | Hi |
| 20:33.00 | Hmmhesays | ok then |
| 20:34.38 | Hmmhesays | she's a cutie though |
| 20:34.43 | *** join/#asterisk [Outcast] (n=bill@222-153-72-33.jetstream.xtra.co.nz) |
| 20:35.30 | [Outcast] | does anyone have any tips on reducing the amount of bad row with rxfax? |
| 20:35.50 | MikeJ[Laptop] | [Outcast], yeah... |
| 20:35.54 | MikeJ[Laptop] | don't use on voip |
| 20:35.58 | [Outcast] | are is there are way to change the baud rate |
| 20:36.00 | [Outcast] | i am not |
| 20:36.04 | MikeJ[Laptop] | don't use on analog cards |
| 20:36.11 | Hmmhesays | don't use? |
| 20:36.22 | MikeJ[Laptop] | use newer spandsp |
| 20:36.32 | _MartinCabrera_ | I'm still getting the message "rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389)" with VAD disabled on the eyebeam. Any clue? |
| 20:36.49 | djMax | anybody know where to get the latest polycom ip500 bootrom and sip.ld? |
| 20:36.53 | [Outcast] | MikeJ: i am using the latest stable. and using it on a pri board |
| 20:37.06 | MikeJ[Laptop] | give it enough cpu... |
| 20:37.08 | Katty | file: he is naughty. |
| 20:37.20 | MikeJ[Laptop] | is it a beffy box ? |
| 20:37.21 | Katty | file: he's like the cute bad boy of north dakota. |
| 20:37.37 | [Outcast] | MikeJ: it is around 2.0 ghz |
| 20:37.43 | MikeJ[Laptop] | should do it.. |
| 20:37.53 | MikeJ[Laptop] | dunno.. |
| 20:38.00 | MikeJ[Laptop] | is it only some fax machines that do it? |
| 20:38.04 | file | hot'n'sexy Hmmhesays |
| 20:38.04 | *** join/#asterisk _DAW (n=bob@adsl-6-66-81.msy.bellsouth.net) |
| 20:38.06 | *** join/#asterisk acidchild (n=adasdasd@unaffiliated/acidchild) |
| 20:38.09 | file | Hmmhesays: that's you |
| 20:38.12 | Hmmhesays | yes file |
| 20:38.16 | [Outcast] | i am getting like 25 bad rows per fax |
| 20:38.27 | MikeJ[Laptop] | that's very high |
| 20:38.31 | [Outcast] | yea |
| 20:38.36 | MikeJ[Laptop] | talk to coppice about it.. |
| 20:38.44 | Katty | Hmmhesays: your collarbone is not sexy. |
| 20:38.50 | MikeJ[Laptop] | that being said.. I know that he is unavailable till the end of the week |
| 20:38.51 | Katty | Hmmhesays: that's my only complaint. |
| 20:38.51 | [Outcast] | i will double check the processor speed |
| 20:39.07 | Hmmhesays | sorry |
| 20:39.17 | [TK]D-Fender | djMax : You should get those from your reseller. |
| 20:39.19 | Hmmhesays | it will be, since i'm done drinking i'll gain a bunch of muscle again |
| 20:39.31 | *** part/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 20:39.34 | Katty | yay! |
| 20:39.40 | Katty | it'll be napping condusive! |
| 20:40.48 | *** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl) |
| 20:42.05 | _DAW | Hi all. |
| 20:42.16 | [Outcast] | MikeJ[Laptop]: it is a quad p4 3.0 ghz system |
| 20:42.49 | [Outcast] | MikeJ: should i use the develop branch of spandsp? |
| 20:43.08 | km- | if you're looking at a tdm04B in zttool, you should see the line wink when an incoming call occurs, right? |
| 20:43.18 | djMax | praying that somehow it fixes asterisk losing it's SIP mind. It just seems odd that nobody else is having dropped SIP sessions since the only thing I've done is update * |
| 20:43.44 | km- | I've got a pots line into one of the fxo modules, have the zaptel all setup for fxsks, but for some reason I cant seem to get a ring event to fire an answer |
| 20:43.45 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 20:44.22 | _DAW | I have a problem with a T100P. I modprobe the card in rc.local as well as start safe_asterisk, however I get a series of errors starting with zt_open: Unable to specify channel |
| 20:44.22 | _DAW | <PROTECTED> |
| 20:44.58 | [TK]D-Fender | djMax : What versions are you running now? |
| 20:45.07 | *** join/#asterisk SPoon_TSX (n=Kit@70.54.197.99) |
| 20:45.17 | _DAW | When I run ztcfg however safe_asterisk starts right up. Any ideas? |
| 20:45.18 | djMax | 1.3.4 on polycom, and "latest" on * |
| 20:45.35 | SPoon_TSX | Hi there. I got some question regarding the outgoing voice quality. |
| 20:45.49 | SPoon_TSX | Does anyone using AAstra 480 phone? |
| 20:45.53 | russellb | _DAW: run ztcfg in rc.local |
| 20:46.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:46.18 | _DAW | russellb - I have and it does not work there. |
| 20:46.23 | file | russellb: Russellllllllllll |
| 20:46.30 | djMax | (only latest because I couldn't get stables to work either) |
| 20:46.31 | russellb | _DAW: try adding a "sleep 3" after it or something |
| 20:46.44 | SPoon_TSX | I got no echo, but the remote part keep saying my voice is keep on cutting off? Any idea? |
| 20:46.46 | _DAW | russellb - OK I will try it. |
| 20:46.57 | _MartinCabrera_ | with VAD disabled on EyeBeam I'm still getting "rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389)" notice. Any clue? |
| 20:48.15 | [TK]D-Fender | djMax : You shoul bump yourself up to at least 1.5.x which runs pretty rock solid for me here. |
| 20:48.30 | djMax | k. just got my hands on 1.6.3 |
| 20:48.44 | djMax | any idea if I need to take the new sip.cfg too, or can stick with existing? |
| 20:49.00 | [TK]D-Fender | djMax : 1.6.5 is the latest out right now. |
| 20:49.11 | [TK]D-Fender | djMax : You NEED to rebuild your file... |
| 20:49.23 | [TK]D-Fender | otherwise it will panic. BAD. |
| 20:49.38 | *** join/#asterisk tuxinator_linux (n=tuxinato@m110e36d0.tmodns.net) |
| 20:49.38 | djMax | rebuild the sip.cfg, or all the phone configs? |
| 20:50.09 | [TK]D-Fender | djMax : BOTH. |
| 20:50.27 | Katty | tuxinator_linux: moo. |
| 20:50.31 | djMax | man, that sure sucks. Thanks. |
| 20:50.32 | [TK]D-Fender | 1.3 - 1.4 rand the ipmid merged format and many things changed. How many phones do you need to account for? |
| 20:50.47 | djMax | 7 |
| 20:50.54 | [TK]D-Fender | djMax : no biggie... |
| 20:51.14 | [TK]D-Fender | for $20 I'd do them all for you ;) |
| 20:52.23 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 20:52.56 | Flauto | one question |
| 20:53.02 | Flauto | i use fwd |
| 20:53.15 | Flauto | and i want to be able to dial the peering numbers |
| 20:53.43 | Flauto | like * or ** in front of the peering number |
| 20:54.06 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 20:54.27 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:55.03 | *** join/#asterisk radhios (n=roberto@200.42.64.76) |
| 20:55.20 | _DAW | russellb - Worked like a charm. Thnks.. |
| 20:55.37 | radhios | Hi all!! |
| 20:55.42 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 20:55.44 | Flauto | hi |
| 20:55.48 | Flauto | anyone would help me |
| 20:55.59 | Flauto | i setup prifix 8 for fwd |
| 20:55.59 | DeeJay[2] | when can we expect asterisk to have an SIP jitter buffer? |
| 20:56.15 | Flauto | and i want to be able to dial * and ** for peering numbers |
| 20:56.15 | Hmmhesays | zoa was working on one |
| 20:56.18 | Hmmhesays | look on mantis |
| 20:56.19 | Flauto | how would i do it |
| 20:56.35 | russellb | _DAW: you're welcome :) |
| 20:56.35 | radhios | anybody know a free web interface for asterisk?? |
| 20:56.39 | Hmmhesays | amp |
| 20:56.44 | [Outcast] | MikeJ[Laptop]: hey they were running Xwindows |
| 20:57.02 | SPoon_TSX | Hi there, I got a problem here. I got no echo, I can hear the remote party okay, but not to the remote party, they say my phone is keep on cutting out. Any idea? |
| 20:57.17 | X-Rob_ | you don't have enough outgoing bandwidth |
| 20:57.22 | Hmmhesays | sounds like you have an upload problem spp |
| 20:57.25 | Hmmhesays | SPoon_TSX |
| 20:57.33 | [TK]D-Fender | radhios : You'd be thinking of AMP, which on average will make people here shudder and the naming of, and only lead to disappointment. |
| 20:57.46 | X-Rob_ | Pfft. It's not _that_ bad. |
| 20:58.07 | generalhan | i just use the asternic FOP ... i love that and its not as huge or even close to as hard to set up |
| 20:58.36 | SPoon_TSX | Hmmhesays: ? |
| 20:58.54 | X-Rob_ | generalhan, we _are_ trying to simplify the setup, but it's not that bad at the moment. The worst big is asterisk's voicemail which always makes files chmod 600 chown asterisk. |
| 20:59.04 | X-Rob_ | which means we need to run AMP as asterisk |
| 20:59.24 | X-Rob_ | which means you need to run httpd as asterisk. |
| 20:59.59 | generalhan | i do with the FOP too ... is that bad ?? lol |
| 21:00.11 | SPoon_TSX | Any one have any idea what may goes wrong? |
| 21:00.15 | DeeJay[2] | Hmmhesays: Albert <-PSTN-> ASTERISK <-SIP-> Max is it what could make Albert hear clicks? |
| 21:00.38 | generalhan | Spoon: it could be a lot of things i had a TERRIBLE provider not too long ago that gave me the same results you are having now |
| 21:00.39 | X-Rob_ | SPoon_TSX, you've been told twice. |
| 21:00.46 | X-Rob_ | Pay attention. |
| 21:00.53 | X-Rob_ | you don't have enough outgoing bandwidth. |
| 21:01.06 | generalhan | lol on dial-up ! |
| 21:01.42 | SPoon_TSX | X-Rob_: ? I am not calling over the internet, I just do the PSTN call from my internal IP phone to the PSTN?? |
| 21:02.36 | radhios | ok, I try to install and test your sugestions and I will back with some questions soon!! thanx a lot!!! |
| 21:02.37 | *** join/#asterisk Darkcyde (i=jk@packetsexchange.net) |
| 21:02.40 | [TK]D-Fender | SPoon_TSX : Maybe its a crappy PSTN interface.... what card are you runing? |
| 21:04.17 | radhios | bye bye cu soon!!! |
| 21:04.43 | SPoon_TSX | [TK]D-Fender: TDM400 |
| 21:06.13 | *** join/#asterisk deltahex (n=m@athedsl-32548.otenet.gr) |
| 21:06.47 | deltahex | hello :) |
| 21:06.48 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 21:07.16 | [av]bani | yay zoom 5801 arrived |
| 21:07.49 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 21:08.57 | [TK]D-Fender | SPoon_TSX : Is your card on its own IRQ and are you running any other riskier apps like X / samba, etc? |
| 21:10.19 | Flauto | stupid asterisk |
| 21:10.40 | Hmmhesays | flauto whats your major malfuction |
| 21:10.53 | SPoon_TSX | No, and yes. It is on its own IRQ! |
| 21:11.11 | djMax | any way to know when a SIP registration is scheduled to expire? |
| 21:11.30 | djMax | sip show peer <peer> has an "Expire" line, but it's static, and "373" which seems like an odd time. |
| 21:12.02 | Hmmhesays | so flauto, you use fwd |
| 21:12.37 | Flauto | i am sorry, hmmhesays |
| 21:12.42 | Flauto | i really did not know |
| 21:13.10 | Hmmhesays | so you used FWD, what problem are you having? |
| 21:14.09 | Flauto | i setup 8 as prefix |
| 21:14.24 | Flauto | to go to fwd |
| 21:14.43 | Hmmhesays | so you dial 8+<fwd> number? |
| 21:14.59 | Flauto | yes |
| 21:15.02 | km- | When I try to load chan_zap I get an undefined symbol, ast_pickup_call |
| 21:15.15 | Hmmhesays | Flauto go on |
| 21:15.23 | km- | I'm trying this in 1.2.4 -- is there something special I need to change in zaptel or asterisk source tree to make everything play nice? |
| 21:15.28 | Flauto | so when i dial 8**282 to broadvoice, it gives me busy tone |
| 21:15.32 | Flauto | so |
| 21:15.54 | Hmmhesays | is **282 the right peering numbre? |
| 21:15.54 | Flauto | i set _8**.,1, in the dial plan |
| 21:15.59 | Flauto | yes |
| 21:16.03 | Flauto | for broadvoice |
| 21:16.14 | Flauto | when i dialed, it is still busy |
| 21:16.26 | _Sam-- | [av]bani : has anything gotten any better at teliax? |
| 21:16.31 | Hmmhesays | exten => _8**.,1,DIAL(SIP/${EXTEN:1}@fwd) |
| 21:16.36 | Tagor | I would like to forward a caller to another extension. Can someone tell me how to do that? This is what I would like it to be: http://www.rafb.net/paste/results/ZPflzb37.html |
| 21:16.42 | Hmmhesays | you have your dp set up like that? |
| 21:16.50 | Flauto | yes |
| 21:16.53 | Flauto | i use iax |
| 21:16.58 | Flauto | but it should be the same way |
| 21:17.02 | Tagor | So my question is how to specify the correct extension |
| 21:17.31 | Hmmhesays | Flauto: where on broadvoice are you trying to dial? |
| 21:17.40 | Hmmhesays | what number |
| 21:17.45 | Flauto | 3129620228 |
| 21:18.24 | _MartinCabrera_ | PleasE: With VAD disabled on EyeBeam I'm still getting "rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389)" notice. Any clue? |
| 21:18.36 | Hmmhesays | **282 + that number? |
| 21:18.46 | Flauto | yes |
| 21:20.06 | trixter | vad != cng |
| 21:20.21 | trixter | you may not be able to disable cng ... |
| 21:23.21 | Tagor | Anybody? :) |
| 21:24.52 | Flauto | hmmhesays, are you still there |
| 21:25.40 | _MartinCabrera_ | Thanks trixter. I will suggest asteriskguru to change that tip :) |
| 21:26.23 | Hmmhesays | flauto worked for me |
| 21:27.06 | Flauto | would you show me what you have in your dial plan |
| 21:27.29 | *** join/#asterisk kkun (n=none@198.60.73.230) |
| 21:28.31 | Flauto | exten => _8.,3,Dial(IAX2/${FWDUSERID}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60) |
| 21:28.42 | Flauto | this one is for regular fwd number dialing |
| 21:28.59 | Flauto | exten => _8*.,3,Dial(IAX2/${FWDUSERID}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60) |
| 21:29.05 | Flauto | this one is peering1 |
| 21:29.16 | Flauto | exten => _8**.,3,Dial(IAX2/${FWDUSERID}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60) |
| 21:29.23 | Flauto | this one is for peering2 |
| 21:29.59 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 21:30.21 | Flauto | but when i dial 8 |
| 21:30.23 | Flauto | it was okay |
| 21:30.32 | Flauto | *, it starts the busy tone |
| 21:30.51 | Flauto | hmmhesays, are you still there |
| 21:31.08 | Flauto | it seems that it does not take * at all |
| 21:31.34 | Hmmhesays | does it hangup when you dila? |
| 21:31.37 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
| 21:31.38 | Hmmhesays | *dial the * |
| 21:31.42 | Flauto | yes |
| 21:32.00 | Goral | hello all i need some help with asterisk... i'm trying to set up a simple sip server buti cannot get sjphone to register on my box |
| 21:32.20 | Hmmhesays | so it doesn't even try to dial out flauto? |
| 21:32.20 | GerbilNut | anyone have a motherboard recommendation for * ? |
| 21:32.30 | Flauto | not at all |
| 21:32.41 | Hmmhesays | that sounds like a problem in your softphone |
| 21:32.44 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 21:32.51 | Flauto | i am not using softphone |
| 21:32.55 | Flauto | oh |
| 21:32.56 | Hmmhesays | what are you using? |
| 21:32.56 | Flauto | got it |
| 21:33.01 | Flauto | i am so sorry |
| 21:33.08 | Hmmhesays | what is your originating endpoint? |
| 21:33.19 | Flauto | i need to setup the stupid spa3k to accept ** |
| 21:33.30 | Goral | chan_sip.c:10815 handle_request_register: Registration from '<sip:test@192.168.1.201:5060>' failed for '192.168.1.100' - Username/auth name mismatch |
| 21:33.35 | Hmmhesays | that'll $20.00 for my time thanks |
| 21:33.36 | Goral | what am i doing wrong? |
| 21:33.39 | djMax | well, poly 1.6.2 didn't fix the SIP problem. |
| 21:33.50 | Hmmhesays | Goral: something is behind nat isn't it |
| 21:33.58 | Hmmhesays | don't lie |
| 21:34.05 | Goral | no straight through my local lan |
| 21:34.23 | Hmmhesays | then you might have fat fingered something somewhere |
| 21:34.46 | Goral | no dought that one |
| 21:35.08 | Hmmhesays | i don't |
| 21:35.12 | Goral | double checked my sip.conf fild |
| 21:35.16 | Goral | file |
| 21:35.23 | Hmmhesays | did you sip reload after? |
| 21:35.30 | Goral | yes |
| 21:35.43 | Goral | infact to over shoot i restarted the box |
| 21:36.09 | Goral | my i pm you Hmmhesays? |
| 21:36.21 | *** join/#asterisk marta (n=a@host40.natpool.mwn.de) |
| 21:36.36 | djMax | and the echo cancelling has gone to complete hell in 1.6.2. Any ideas why that would be? |
| 21:37.28 | Hmmhesays | Goral: sure if you want to pay me |
| 21:37.35 | Hmmhesays | otherwise keep it in the chan |
| 21:41.41 | SPoon_TSX | Hi there, May I know what is "sidetone" it? |
| 21:42.44 | *** join/#asterisk camonz (n=camonz@200.8.20.210) |
| 21:42.52 | camonz | hi |
| 21:44.12 | djMax | yeesh, and my poly's don't have any rings anymore. |
| 21:46.49 | djMax | anybody know how to get SVN to update to a specific ver, like 1.2.1 |
| 21:47.57 | GerbilNut | anyone have a motherboard recommendation for asterisk ? |
| 21:48.12 | *** join/#asterisk Eggplant (i=No@dsl-893.cascadeaccess.com) |
| 21:49.13 | djMax | we use a SuperMicro 1U Xeon unit, haven't had any issues |
| 21:49.20 | djMax | other than asterisk driving me batty. |
| 21:49.46 | GerbilNut | how many calls can it run through it? |
| 21:49.52 | GerbilNut | and any PRIs? |
| 21:50.47 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 21:51.35 | fu3 | hey lads.. i dont have a second voip phone, so i'd like to continue testing with a soft phone.. any one in particular good for testing? |
| 21:51.42 | fu3 | i dont plan on deploying them, but I do need to test some more. |
| 21:53.27 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:53.28 | djMax | we have a single PRI card in it. |
| 21:53.35 | djMax | Haven't really stressed it, it's a small biz |
| 21:54.56 | nettie | hi guys, I have somethign like 5 sip accounts and I would like to find a smart way to decide which I shoul use before dialling. What's the dialplan prefix people use to define the line I want to use? |
| 21:56.33 | dpolitech | ungh |
| 21:56.36 | dpolitech | is it beer-thirty yet? |
| 21:57.04 | fu3 | its half past! |
| 21:57.10 | fu3 | you're late |
| 21:57.35 | fu3 | my old boss used to look at the clock, no matter what time it was and say "its goin on five.. lets get out of here" |
| 21:57.49 | fu3 | but, thats why he is my old boss. |
| 21:58.11 | fu3 | he also argued for weeks that there was a G in hexadecimal. |
| 21:58.14 | fu3 | so.. |
| 21:59.20 | mogorman | counting to G must take a long time in hex |
| 21:59.57 | *** join/#asterisk southtel (n=slester@c-67-191-211-148.hsd1.ga.comcast.net) |
| 22:00.06 | fu3 | haha |
| 22:00.35 | fu3 | now he's in a federal penitentary in Nevada for bring a minor across state lines for sex :/ |
| 22:00.42 | fu3 | so.. he's got lots of time to count to G. |
| 22:00.56 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 22:00.56 | mogorman | yikes |
| 22:01.07 | fu3 | yeah.. what a guy. |
| 22:01.27 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 22:01.55 | *** join/#asterisk QbY (n=Kelvin@cm-12-156-101-217.dhcp.geo-sc.southerncoastalcable.net) |
| 22:02.33 | PakiPenguin_ | where can i find the cheapest tollfree number |
| 22:02.34 | PakiPenguin_ | :) |
| 22:02.45 | MikeJ[Laptop] | cheapest? |
| 22:02.56 | file | I'll sell you one out of the back of my van... |
| 22:02.58 | QbY | Paki.. Broadvoice.com has 2c minute.. But they suck.. |
| 22:02.58 | MikeJ[Laptop] | you mean the one that is always broken with bad sound quality |
| 22:02.59 | file | cheap cheap |
| 22:03.01 | file | it work long time |
| 22:03.04 | MikeJ[Laptop] | ;) |
| 22:03.04 | QbY | Paki.. We use Kall8.com |
| 22:03.04 | file | high quality |
| 22:03.25 | QbY | Kall8.com has IAX termination also.. Rock solid in my opinion.. |
| 22:03.35 | PakiPenguin_ | i see |
| 22:06.13 | *** join/#asterisk southtel (n=slester@c-67-191-211-148.hsd1.ga.comcast.net) |
| 22:06.43 | QbY | Ok, SER/OpenSER Question -- I know this is the #Asterisk channel -- When a phone registers with SER and initiates a phone call.. Does SER stay in the middle during the duration of the call? (Asking for a QoS reason) |
| 22:08.03 | file | the answer can be "Yes" and "No" |
| 22:09.06 | MikeJ[Laptop] | file, have I ever mentioned that I love your answers |
| 22:09.12 | *** part/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 22:09.29 | file | well it's true |
| 22:09.45 | file | if you record route then the signalling goes through SER, if you don't it doesn't |
| 22:09.53 | file | if you proxy RTP audio goes through SER, if you don't it doesn't |
| 22:10.14 | file | so my answer is valid |
| 22:10.21 | file | therefore I win |
| 22:10.29 | SPoon_TSX | Hi there, I got a quick question regarding the fxotune. If I using the fxotune. Do I still need to adjust the rxgain and txgain in zapata.conf? |
| 22:10.46 | MikeJ[Laptop] | yes.. I know why your answer was right... |
| 22:10.54 | MikeJ[Laptop] | I just love your answers |
| 22:11.06 | file | I'm good with answering questions |
| 22:11.35 | MikeJ[Laptop] | what is the air speed volocity of..... |
| 22:11.41 | file | MikeJ[Laptop]: yes |
| 22:11.42 | generalhan | hahahaha |
| 22:12.05 | camonz | are there any other voice conferencing facilities besides meetme in *? |
| 22:12.14 | SPoon_TSX | Any one? Do you still need to adjust the zapata even if you use the fxotune? |
| 22:12.30 | Mavvie | camonz: Cisco is willing to sell you one. |
| 22:12.37 | MikeJ[Laptop] | camonz, yes... |
| 22:12.50 | MikeJ[Laptop] | well.. |
| 22:12.51 | MikeJ[Laptop] | no |
| 22:12.57 | MikeJ[Laptop] | not "in" * |
| 22:13.01 | MikeJ[Laptop] | but for *, yes |
| 22:13.18 | camonz | yep, that's what i mean, for * |
| 22:13.39 | MikeJ[Laptop] | then yes |
| 22:13.53 | camonz | i heard somewhere there was another voice conferencing utility in asterisk besides meetme |
| 22:14.05 | camonz | wich was somehow related to an .so module |
| 22:14.12 | MikeJ[Laptop] | for *, not in *, yes |
| 22:14.38 | Ikarus | hrm, one of the Dutch language voice sets for Asterisk doesn't even have correct grammer |
| 22:14.55 | MikeJ[Laptop] | ahh.. the wonders of open source.... |
| 22:15.00 | MikeJ[Laptop] | you can fix it! |
| 22:15.11 | Flauto | does fwd still offer toll free numbers in the usa? |
| 22:15.14 | Ikarus | MikeJ[Laptop]: I would have to re-record all prompts |
| 22:15.20 | MikeJ[Laptop] | yep.. |
| 22:15.22 | MikeJ[Laptop] | probably.. |
| 22:15.26 | MikeJ[Laptop] | or slice and dice them |
| 22:15.44 | MikeJ[Laptop] | Flauto, didn't know they ever did... |
| 22:16.04 | Ikarus | It is a shame, the voice itself is a very decent one (unlike the only grammer correct one, which has a horrid accent) |
| 22:16.06 | MikeJ[Laptop] | they did fwd out... but didn't think they did any inbound tollfree |
| 22:16.07 | Mavvie | Ikarus: but were they made by a native dutch person or an "ongelovelijk Mike!" persoon? |
| 22:16.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:16.25 | Ikarus | Mavvie: native dutch |
| 22:16.25 | camonz | what's the name of this module?, i'm trying to set up voice conferencing and i want to know the available options |
| 22:16.34 | MikeJ[Laptop] | app_conference |
| 22:16.40 | Ikarus | Mavvie: she has a very good voice, the translation she was reading just sucked |
| 22:16.41 | camonz | thks |
| 22:16.41 | MikeJ[Laptop] | it's on sourcefourge. |
| 22:16.57 | MikeJ[Laptop] | along with iaxclient on sourceforge I think |
| 22:17.00 | fu3 | hey whats this "voicerd" stuff all about? |
| 22:17.04 | fu3 | another open source pbx |
| 22:17.14 | MikeJ[Laptop] | voicerd? |
| 22:17.25 | fu3 | yeah |
| 22:17.32 | fu3 | its a novell-based asterisk deployment |
| 22:17.39 | fu3 | :| |
| 22:17.41 | MikeJ[Laptop] | oh.. |
| 22:17.55 | MikeJ[Laptop] | first I've seen of that mentioned |
| 22:19.32 | fu3 | yeah same here |
| 22:19.44 | fu3 | i just got a survey from the state of MN that included it as an option for campuses to explore. |
| 22:19.50 | fu3 | fuck it.. standalone asterisk beats it :) |
| 22:20.20 | SPoon_TSX | I am sorry about the trouble, but did anyone using the fxotune? |
| 22:20.46 | fu3 | hahahahhahaha "We just made Voice-Over-IP better, faster, and cheaper" |
| 22:20.56 | fu3 | what-ever. |
| 22:22.14 | camonz | MikeJ: what differences are there between meetme and app_conference? |
| 22:22.30 | MikeJ[Laptop] | app_conference does not use zap conferencing |
| 22:22.39 | camonz | ? |
| 22:22.47 | MikeJ[Laptop] | it uses another method which can be better in some situations |
| 22:22.50 | Flauto | mike, i think it is not working |
| 22:23.05 | MikeJ[Laptop] | in large conferences with few speakers it can be much more efficient |
| 22:23.11 | Flauto | do you know who else is offering toll free numbers? |
| 22:23.26 | MikeJ[Laptop] | for example.. if you have 30 people listening on g729.. it only encodes 1 frame |
| 22:23.36 | MikeJ[Laptop] | and sends it to all 30 |
| 22:24.12 | MikeJ[Laptop] | in smaller conferences it is probably less efficient.. as it has more overhead listening for who is speaking and such |
| 22:24.26 | MikeJ[Laptop] | Flauto, the company I work for does |
| 22:24.36 | camonz | is it limited to non zap channels? |
| 22:24.39 | MikeJ[Laptop] | no... |
| 22:24.50 | MikeJ[Laptop] | but it does not require zaptel like meetme does |
| 22:24.59 | Ikarus | Mavvie: "Het nummer dat u gekozen hebt is niet in gebruik" |
| 22:25.01 | Ikarus | it is such a shame |
| 22:25.05 | MikeJ[Laptop] | it does not have all the admin menu or dtmf based features in it |
| 22:25.20 | MikeJ[Laptop] | although they could be added fairly easily |
| 22:25.29 | camonz | ah..., hence the timing issue |
| 22:25.42 | camonz | there's something in the wiki saying it doesn't require a timing source |
| 22:25.43 | MikeJ[Laptop] | it's pretty good... |
| 22:25.44 | Mavvie | Ikarus: maybe it's somebody from den haag |
| 22:25.52 | MikeJ[Laptop] | camonz, correct.... |
| 22:25.54 | Ikarus | Mavvie: heh |
| 22:25.57 | MikeJ[Laptop] | it uses software timing |
| 22:26.02 | Flauto | mike, what comanpy is that |
| 22:26.05 | Mavvie | but then it would have been "Het nummer dat u gekozen heb is niet in gebruik" |
| 22:26.30 | Katty | oh. |
| 22:26.35 | Katty | s/ice/mints/ |
| 22:26.41 | camonz | couldn't it have been "De nummer dat u hebt gekozen is niet in gebruik" |
| 22:26.41 | Katty | :> |
| 22:27.34 | MikeJ[Laptop] | all this dutch.. and I don't speak a bit :P |
| 22:27.47 | camonz | :->, i speak it a lil bit |
| 22:27.50 | AndyCap | Ikarus: why not klingon while you're at it? |
| 22:27.54 | MikeJ[Laptop] | Ikarus, look at some of the voices that cepstral provides.. there are some funny ones |
| 22:27.59 | camonz | lived in belgium for about 11 months a few years back |
| 22:28.03 | MikeJ[Laptop] | like the dog... and the drill sgt. |
| 22:28.33 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 22:28.45 | Ikarus | heh, I actually want some of the samples Mark used with the talk he gave at FOSDEM :P |
| 22:29.01 | Ikarus | AndyCap: because people won't try to understand that |
| 22:29.14 | Ikarus | AndyCap: while Afrikaans is very close to Dutch |
| 22:29.35 | *** join/#asterisk buZz (n=bastiaan@213-136-16-131.adsl.bit.nl) |
| 22:29.43 | buZz | ok , so i'm ready to order a ATA device |
| 22:29.46 | camonz | btw, MikeJ the company you work for offers PSTN access to resellers? |
| 22:29.48 | buZz | should i get a ATA-1000? |
| 22:29.55 | buZz | i just need to hook up 1 phone |
| 22:29.58 | buZz | and support SIP |
| 22:30.05 | buZz | it needs to 'just work' |
| 22:30.23 | *** join/#asterisk S^Gerbitz (n=CrazyEdd@wsip-24-234-136-148.lv.lv.cox.net) |
| 22:30.25 | S^Gerbitz | hey guys |
| 22:30.42 | S^Gerbitz | Do I need any FXO cards or anything to make this work for pc - pc calls? |
| 22:30.48 | buZz | no |
| 22:30.55 | buZz | you need a pc for pc - pc calls |
| 22:30.56 | S^Gerbitz | okay |
| 22:30.57 | buZz | well |
| 22:30.59 | buZz | 2 actually |
| 22:31.09 | S^Gerbitz | okay I just want to make sure |
| 22:31.19 | Ikarus | Mavvie: You know what is worse, a friend of mine just told me, that "hebt" is correct, the Nederlandse Taalunie says so http://taaladvies.net/taal/advies/vraag/820/ |
| 22:31.51 | Mavvie | Ikarus: he's right, but it's a randstad vs boerenland thing :-) |
| 22:31.55 | S^Gerbitz | on this page: http://automated.it/guidetoasterisk.htm |
| 22:32.10 | Ikarus | Mavvie: no, the Taalunie says so, which makes it so for any official application |
| 22:32.24 | S^Gerbitz | can I just skip to RUSH HOUR III |
| 22:33.14 | Ikarus | Mavvie: well, it seems to be possible sometimes to use "heeft", but grmbl |
| 22:34.12 | zamba | if i just want to use the sip feature of asterisk.. how do i effectively disable everything else? |
| 22:34.35 | *** part/#asterisk QbY (n=Kelvin@cm-12-156-101-217.dhcp.geo-sc.southerncoastalcable.net) |
| 22:36.48 | Mavvie | aaargh. another E1 stuck. |
| 22:36.57 | Mavvie | and that's at 09:36... |
| 22:37.09 | Mavvie | so I have to wait for 12 hours before I can reset it. |
| 22:38.34 | buZz | any bad experiences with SPA-1001 or is it fine for 'just some playing with voip' |
| 22:41.34 | [av]bani | thats about right |
| 22:41.55 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 22:42.21 | buZz | k cool |
| 22:42.55 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 22:43.29 | Hmmhesays | i think i might have a couple of those in the field |
| 22:43.33 | buZz | 65 euros ^_^ |
| 22:43.39 | [av]bani | you get what you pay for |
| 22:43.46 | Hmmhesays | so true |
| 22:45.55 | AndyCap | sooo.. would you trust a firm called AstroHog? |
| 22:46.22 | Hmmhesays | they selling lube? |
| 22:46.41 | AndyCap | Hehe, probably that too. http://cgi.ebay.com/VOIP-Phone-SIP-Adapter-FXO-FXS-Port-2-Ethernet-Ports_W0QQitemZ5872868033QQcategoryZ99269QQrdZ1QQcmdZViewItem |
| 22:46.49 | *** join/#asterisk obsidian-studios (n=obsidian@c-24-129-16-31.hsd1.fl.comcast.net) |
| 22:46.56 | AndyCap | everything and the kitchen sink |
| 22:47.49 | *** part/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 22:48.23 | obsidian-studios | what up all got some ? about echo's. Which I am sure is a bottomless well, but essentially it starts with what? good hardware like fxo, then good voip phone? are * zapata settings env specific or more hardware specific? |
| 22:48.41 | Chingetas | <PROTECTED> |
| 22:48.52 | Katty | Hmmhesays: cherry flavored? |
| 22:49.35 | obsidian-studios | basically got a situation with Digium TDM400 with 4 fxo, Cisco 7960 phones, *, and echos. Here I have cheese $7 x100p clones, Cisco 7910 *, same settings in both scenarios, no echo here, initial one there? |
| 22:50.01 | kuku5 | How can I check the cisco router model number from the command line ? |
| 22:50.11 | Mavvie | "show version" |
| 22:50.13 | Mavvie | "show hardware" |
| 22:51.09 | kuku5 | 831 :) thx |
| 22:51.15 | kuku5 | is the 831 any good ? |
| 22:51.21 | Chingetas | Noone with this problem with asterisk---SIP----asterisk?? |
| 22:51.23 | kuku5 | copared with 871 |
| 22:51.29 | Mavvie | kuku5: yes, it's a beautiful box. |
| 22:51.36 | websae | chingetas: what are you issues? |
| 22:51.37 | Mavvie | kuku5: oh, the 871 is much better for its task. |
| 22:51.38 | obsidian-studios | kuku5: ethernet router, good just lacking modem |
| 22:51.40 | Hmmhesays | strawberry banana |
| 22:51.43 | Hmmhesays | thats a good flavor |
| 22:51.56 | Mavvie | but that's mostly because he can't use them right now. |
| 22:52.01 | obsidian-studios | Mavvie: same chassis and platform, just more hardware right? |
| 22:52.02 | kuku5 | Mavvie: what does 871 have that 831 doesnt |
| 22:53.01 | obsidian-studios | kuku5: http://www.cisco.com/en/US/products/hw/routers/ps380/prod_models_comparison.html |
| 22:53.15 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:53.17 | Ikarus | Actually this hunting through voices leaves me with one question, are the default asterisk voice prompts availible without compression (or with a less destructive compression) |
| 22:53.45 | obsidian-studios | so anyone got comments on good hardware etc with echo's, crappy stuff and no echo's? |
| 22:53.46 | kuku5 | obsidian-studios: 831 inst on there |
| 22:53.57 | obsidian-studios | kuku5: go down on the page |
| 22:53.58 | AndyCap | I guess there won't be an AstroHog ata in my future. :> |
| 22:54.24 | obsidian-studios | kuku5: see it? |
| 22:57.54 | Ikarus | ah, appaerently there is at 'http://www.astlinux.org/index.php?option=com_docman&task=doc_download&gid=24&Itemid=36' |
| 22:58.20 | obsidian-studios | quiet in here? |
| 22:58.25 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 22:58.27 | Ikarus | obsidian-studios: ofcourse |
| 22:58.35 | obsidian-studios | ah, after 5 est :) |
| 22:58.48 | Ikarus | obsidian-studios: and almost midnight CET |
| 22:58.50 | obsidian-studios | <PROTECTED> |
| 22:59.13 | Ikarus | obsidian-studios: it goes on and off |
| 22:59.15 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 22:59.32 | obsidian-studios | <PROTECTED> |
| 22:59.58 | obsidian-studios | just found it funny that I keep having problems with good hardware, and hardly any with crap ware, at least none that I do not have with both |
| 23:01.35 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 23:02.04 | wrench | hey all, quick q: im wondering which is a better solution for advanced logic & DB interaction in a production environment receiving high volume of brief < 1min calls: AGI calls to external C apps or using the perl builtin interpreter addon |
| 23:02.29 | Ikarus | I need to rewrite my extensions atleast partially to AEL and AGI, I need to hack up a POE component for the Manager API, (THAT ACTUALLY WORKS, unlike the current one), well, I guess reverse that order |
| 23:02.59 | fifer | I'm running into some weird behaviour with BLF / hint subscriptions |
| 23:03.14 | lesouvage | I discovered the DISA aplication (http://www.voip-info.org/wiki-Asterisk+cmd+DISA) and it is working fine except that I have to wait the full 10 seconds before dialing starts although I made adjustments on the ResponseTimeout() and DigitTimeout() values. Is there a fix for this? |
| 23:03.37 | fifer | Any one know when a device is suposed to subscribe? (may be device specific) |
| 23:04.06 | fifer | I read online that issues with subscriptions being lost after a reload were fixed, but that does not apear to be true: I'm on 1.2.4 |
| 23:05.56 | fifer | I keep using show hints to monitor the subscriptions, but it is odd when a phone reboots that subscribes to two extensions, but only one shows up as having any subscribers |
| 23:06.48 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:07.51 | harryvv | hi all, anyone here run a ip500/501 and have figured out how to switch from a current call to a cw incomming call when I hear the cw tone? |
| 23:10.05 | [av]bani | anyone configured a cisco 7970? |
| 23:12.38 | russellb | fifer: no, that has not been merged yet |
| 23:12.45 | russellb | fifer: it's in the bug tracker ... |
| 23:12.59 | Goral | defaultip=192.168.1.100 does this make the account only accessable from that ip? |
| 23:13.12 | fifer | russellb: Thanks! |
| 23:13.23 | russellb | Goral: no, that just means, if this device hasn't registered yet, give this ip a try |
| 23:13.25 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 23:13.27 | russellb | :) |
| 23:13.45 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 23:13.57 | fifer | Bug 3644? |
| 23:14.08 | russellb | i don't remember the number |
| 23:14.09 | Goral | what if i log in from another computer |
| 23:14.14 | Goral | softphone |
| 23:14.24 | Goral | well i'm using sjphone |
| 23:15.22 | fifer | russellb: It says it was commited to head Aug 05 |
| 23:15.37 | fifer | That does not mean it made it into a 1.2.x release? |
| 23:18.39 | hertell | Does anyone know what the error "Not a local SIP domain" means (*1.2.4) |
| 23:19.44 | russellb | fifer: no, that's not it ... |
| 23:19.54 | russellb | that's when the support went in for it |
| 23:20.03 | russellb | but the issue where the subscriptions get lost during a reload |
| 23:20.05 | russellb | that's something else |
| 23:20.49 | russellb | bug 6047 |
| 23:20.53 | fifer | russellib: Do you know any way to force a refresh of subscriptions? The only thing I have found is rebooting the phone, but even that does not seem to be consistant. |
| 23:21.00 | fifer | russellb: Thanks! |
| 23:21.23 | PoWeRKiLL | Someone know how to use genmc utility to generate a certificate for the sipura |
| 23:21.37 | *** join/#asterisk nanogeek (n=nanogeek@207.244.10.65) |
| 23:21.52 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 23:22.47 | [av]bani | anyone configured a cisco 7970? |
| 23:23.14 | nanogeek | Anyone have any experience configuring asterisk to pass inbound cdr to an off-net sccp device? |
| 23:24.20 | nanogeek | sorry, meant cpn, not cdr. (not enough caffine...) |
| 23:24.48 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 23:26.06 | Ikarus | Anyone been recently hacking on 16khz or lets call it "variable" frequency support in Asterisk (variable as in, any arbitary rate, not variable as in, with in one call differing) |
| 23:39.35 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 23:40.12 | brockj49464 | Is there a setting to set how long a X-Lite client can have a call on hold for? |
| 23:41.21 | Ikarus | 13 |
| 23:41.24 | Ikarus | whoops |
| 23:42.12 | [av]bani | nobody uses sccp? |
| 23:42.33 | mogorman | other than cisco ^_^ |
| 23:42.46 | [av]bani | nobody in this channel has cisco sccp phone... |
| 23:42.49 | _Sam-- | only qwell! |
| 23:42.52 | [av]bani | yay |
| 23:42.56 | [av]bani | well now i get to beat him up |
| 23:43.07 | [av]bani | _Sam--: did 1.0.2.13 fix your qualify= problem? |
| 23:43.11 | mogorman | what do you need to know [av]bani |
| 23:43.28 | [av]bani | need to know how to configure the !(*@&$ phone from the ui |
| 23:43.32 | _Sam-- | i still havent upgraded...been really busy at work, this is coming into prime season. |
| 23:43.42 | mogorman | oh he will tell you to use ftp or tftp thing |
| 23:43.45 | [av]bani | ha |
| 23:43.48 | mogorman | thats how i have configed my phones |
| 23:43.56 | [av]bani | "so the cisco ui is completely useless" |
| 23:43.59 | [av]bani | :) |
| 23:44.02 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 23:44.24 | mogorman | most ui is useless |
| 23:47.13 | *** join/#asterisk SparFux (n=player@tor/session/x-3067ff5587a702f8) |
| 23:48.26 | *** join/#asterisk Maxx4life (n=Maxx4lif@71-35-210-12.slkc.qwest.net) |
| 23:48.35 | SparFux | Hi all! |
| 23:51.02 | SparFux | Can I play ogg vorbis streams with asterisk? |
| 23:53.24 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 23:53.49 | Ikarus | SparFux: yes |
| 23:54.15 | SparFux | How that? |
| 23:54.25 | Ikarus | Hold on, I need to cluebat something first |
| 23:54.42 | SparFux | I'm 30. |
| 23:54.54 | SparFux | Why? Is that a stupid question? |
| 23:54.58 | Ikarus | SparFux: no |
| 23:55.04 | Ikarus | I am having a fight with the voip wiki |
| 23:55.12 | SparFux | I couldn't find anything about it on the wiki. |
| 23:55.22 | SparFux | What kind of fight? |
| 23:55.36 | Ikarus | It is doing wiki formatting when I don't want it to |
| 23:56.13 | SparFux | Aha. |
| 23:57.33 | *** join/#asterisk kizmet (n=kizmet@rizzo.argon.net.au) |
| 23:58.11 | SparFux | Anyway, I don't get how to play ogg vorbis on an asterisk phone. |
| 23:58.27 | Ikarus | You can as soon as I fix this |
| 23:58.50 | Ikarus | SparFux: basicly you have to tell asterisk it should handle it the same as mp3s |
| 23:58.59 | Ikarus | Only drop in ogg123 |
| 23:59.55 | SparFux | Ok, that's clear. So I have to manipulate MP3Player. |
.jpg){kind=link}
