| 00:01.25 | Druken | :) |
| 00:02.00 | Druken | where you at? |
| 00:02.00 | alephcom | How was the asterisk developers call today? I would have loved to have listened. |
| 00:02.12 | alephcom | Alberta |
| 00:02.37 | file[laptop] | the call will be put up online so you can listen |
| 00:02.54 | russellb | alephcom: it went great, IMO |
| 00:03.16 | russellb | rough starting off because of scheduling issues |
| 00:03.22 | russellb | but iwa s productive |
| 00:03.56 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:04.00 | mog_work | where is recoring file[laptop] |
| 00:04.02 | alephcom | That's good to hear. I'm going to hunt down a transcript yet. |
| 00:04.13 | russellb | i'm uploading it right now |
| 00:04.21 | russellb | ... slowly |
| 00:04.33 | linagee | russellb: get a faster connection. :) |
| 00:04.44 | russellb | linagee: pay for it :) |
| 00:05.27 | *** join/#asterisk coppice (n=chatzill@61.155.17.210.dyn.pacific.net.hk) |
| 00:05.37 | alephcom | That's what it always boils down to isn't it.... |
| 00:05.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:08.22 | russellb | uploaded, now to link it! |
| 00:09.04 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 00:10.23 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 00:10.34 | *** join/#asterisk RageMax (n=max@c-24-3-178-253.hsd1.pa.comcast.net) |
| 00:10.38 | robin_sz | linkinkg it up is is ... gcc is the way to go ;) |
| 00:11.13 | RageMax | is anyone here currently at a major university? |
| 00:11.26 | russellb | me |
| 00:11.38 | russellb | Clemson University |
| 00:11.53 | RageMax | I was seeing if we could get people together to create some asterisk peering |
| 00:12.51 | Druken | RageMax: FREE peering :) |
| 00:13.29 | coppice | don't you know peer to peer is only for thieves and pirates? :-) |
| 00:13.43 | robin_sz | he said "major" ... theres Oxford, and Cambridge ... and then theres the rest |
| 00:13.48 | Druken | arg arg arg and a bottle of rum |
| 00:14.45 | *** join/#asterisk drgalaxy (n=Brian@ppp-70-244-128-190.dsl.lbcktx.swbell.net) |
| 00:14.53 | coppice | robin_sz: I chose to go to one of the rest :-) |
| 00:15.06 | *** part/#asterisk drgalaxy (n=Brian@ppp-70-244-128-190.dsl.lbcktx.swbell.net) |
| 00:16.13 | robin_sz | coppice, most people do ... ;) |
| 00:16.57 | coppice | actually, they do. Oxford and Cambridge are so stuck up, most of us didn't want to go there. |
| 00:17.13 | robin_sz | yeajh yeah ... |
| 00:17.38 | robin_sz | same with me and ferraris ... |
| 00:17.51 | robin_sz | I just chose to drive a clapped out old peugeot ... |
| 00:17.55 | generalhan | Hahahah |
| 00:18.20 | coppice | thought the size of their engineering classes amazed me. they have really huge classes. hundreds of students |
| 00:18.35 | asterboy | <PROTECTED> |
| 00:19.09 | asterboy | yep, that's how you make money...pack em' in. |
| 00:19.42 | asterboy | like sardines |
| 00:19.54 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 00:20.04 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 00:20.24 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 00:20.58 | coppice | most good colleges in the UK have engineering classes of more like 30-40 |
| 00:21.00 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 00:21.13 | fu3 | Ok. Seriously. Who has a copy of spinrite they can send me? I'm in hell right now. |
| 00:21.41 | robin_sz | spinrite? I thought that was free? |
| 00:21.46 | asterboy | fu3, lookup hirens |
| 00:21.48 | fu3 | no, its $89 |
| 00:21.54 | fu3 | hirens eh |
| 00:21.55 | fu3 | ok :) |
| 00:22.05 | asterboy | with clusty, not google |
| 00:22.37 | fu3 | ok |
| 00:22.55 | fu3 | I take it you mean this boot CD thing? |
| 00:22.59 | robin_sz | doesnt 'doze ship with soemthing to do that anyway? |
| 00:23.42 | fu3 | well.. chkdsk is returning "Corrupt Master File Table" |
| 00:23.53 | fu3 | and it does nothign else.. from waht I can tell, my data is likely OK, but just lost.. |
| 00:23.58 | robin_sz | you'd thnk without a journalled fs, theyd at least deliver it with a tool to sort out the crap afterwards |
| 00:24.03 | fu3 | PC File Inspector doesnt work.. |
| 00:24.20 | fu3 | I could be screwed.. but i've got all my seasons of TNG and DS9 on this drive!!! |
| 00:24.21 | fu3 | :) |
| 00:24.59 | coppice | I think being screwed beat watching TV |
| 00:25.16 | Druken | hands down |
| 00:25.22 | Druken | providing it's a good screw.... |
| 00:25.23 | asterboy | while watching TV being screwed |
| 00:25.36 | fu3 | hey.. I *only* watch my TNG and DS9 |
| 00:25.39 | fu3 | I watch no other TV. |
| 00:26.18 | Druken | i make sure the tv is off... chicks don't like it when your going at it, and the tv says something funny and you start cracking up |
| 00:26.26 | robin_sz | fu3: try this http://grcsucks.com/spinrite.htm |
| 00:26.30 | Druken | bad experince in a hotel room... :( |
| 00:26.32 | asterboy | DS9, (General Hospital with actors in space) |
| 00:27.21 | asterboy | Well most Trek has good porno now. |
| 00:27.45 | fu3 | :) |
| 00:27.51 | fu3 | robin_sz.. theres no DL though |
| 00:28.07 | asterboy | Love those rub downs in Enterprise decom chamber. |
| 00:28.37 | coppice | i've never seen general hospital. is it truly the reference model for bad soap operas? |
| 00:28.49 | asterboy | for sure |
| 00:29.20 | coppice | I saw the bold and the beautiful once, and that would sure take some beating |
| 00:29.42 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 00:30.11 | asterboy | now just dress them up in space suites complete with space backdrop...add some british aliens with brochli around temples...voila! |
| 00:31.00 | coppice | the UK has/had one called crossroads, which is more amateurish in production style, but very similar in the struggle the actors have to remember their lines |
| 00:31.21 | asterboy | a perfect alien...err, well they'll have to speek good English and be able to make innuendo |
| 00:32.58 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 00:33.11 | RageMax | well, let me narrow it down a bit, is anyone in a telecom/engineering program that could use the peering b/w universities (maybe over I2?) in a lab setting |
| 00:33.25 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.175.Dial1.SanJose1.Level3.net) |
| 00:33.37 | RageMax | not actual telephone connection, just testing the various protocols |
| 00:34.28 | *** join/#asterisk orlok (i=[tcVmBC8@202-44-174-4.nexnet.net.au) |
| 00:35.15 | coppice | why is being in a telecom/engineering program relevant? are you too stuck up to peer with arts students? :-) |
| 00:36.07 | coppice | when I was at college engineering was mostly men, and languages was mostly women. we used to do our best to peer with the language students |
| 00:36.30 | fu3 | ttyl lads.. thanks anyway :) |
| 00:38.06 | dlynes_ | I've got an issue where someone gets put on hold, and when they're attempted to be taken off hold, I can hear them, but they can't hear me |
| 00:38.21 | dlynes_ | But before they were put on hold, the conversation was full duplex |
| 00:38.32 | dlynes_ | Anyone have any ideas as to what might be causing that? |
| 00:38.38 | dlynes_ | Or is it a bug in asterisk? |
| 00:40.57 | CoffeeIV_ | When I do Goto( context,s,1) it doesn't work (hangs up on me) but when I do Goto(context,s,1) it works (note the space before context) is that a bug ? I'm not using the latest asterisk |
| 00:41.44 | dlynes_ | CoffeeIV_, that would appear to be a bug, yes |
| 00:41.56 | orlok | Does anybody know exatly what a 404 not found in regards to sip means? |
| 00:41.57 | Ariel_ | dlynes, are them sip phones? do you have canreinvite=no |
| 00:42.30 | Ariel_ | CoffeeIV_, why would you have a space there for |
| 00:42.33 | generalhan | lol |
| 00:42.43 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net) |
| 00:42.52 | RageMax | coppice: because you're most likely to get funding ;) |
| 00:43.01 | generalhan | there if there is no contect called " context" then it cant Goto it .... its not a bug ... its a syntax error |
| 00:43.02 | dlynes_ | Ariel_, canreinvite is set to no on all sip devices, yes |
| 00:43.09 | RageMax | I'm sure an art department would give funding to a PBX project |
| 00:43.13 | CoffeeIV_ | Ariel: it's ok to have a space there in all the other languages, why not * |
| 00:43.16 | Ariel_ | orlok, account or context or device not there |
| 00:43.39 | orlok | Ariel_: Connecting directly to providers server with a sipura |
| 00:43.42 | Ariel_ | CoffeeIV_, becuase it's not. |
| 00:43.44 | *** join/#asterisk claude81 (n=claude_t@claudetoma.plus.com) |
| 00:43.45 | dlynes_ | generalhan, yeah, but spacing is often used for readability reasons...context identifiers shouldn't be allowed to have leading or trailing spaces |
| 00:43.48 | generalhan | its not a language thing ... its that you are telling * to Goto a context that doesnt exists |
| 00:43.53 | generalhan | exist rather |
| 00:44.21 | Ariel_ | generalhan, I know but he will find out sooner or later. |
| 00:44.30 | claude81 | anyone managed to get vmail on FC3 |
| 00:44.37 | *** part/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 00:44.54 | Ariel_ | claude81, lots of people do. But I use CentOS not Fedora |
| 00:45.16 | claude81 | i installed it with no problem |
| 00:45.21 | Ariel_ | orlok, connected directly then you have a typo |
| 00:45.22 | claude81 | i can login with no problem |
| 00:45.35 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 00:45.39 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 00:46.21 | claude81 | however it dose not display any voice mails eventhou i have mail |
| 00:46.21 | Ariel_ | claude81, are you going to the correct context |
| 00:46.21 | claude81 | yes |
| 00:46.21 | dlynes_ | claude81, asterisk-based voicemail, or telco-based voicemail? |
| 00:46.21 | Shaun2222 | could asterisk be used for dialup access? |
| 00:46.21 | Shaun2222 | i dont see why not... |
| 00:46.34 | orlok | Ariel_: so incorrect login/ph#? |
| 00:46.36 | Ariel_ | dialup access like an isp |
| 00:46.40 | Shaun2222 | ya |
| 00:46.48 | k-man | is there some sort of software fax for fax over ip? |
| 00:46.50 | Ariel_ | orlok, maybe ip address could be anything |
| 00:46.57 | Ariel_ | Shaun2222, it's a pbx |
| 00:47.09 | dlynes_ | k-man, do a voip-info.org search for spandsp |
| 00:47.11 | orlok | Ariel_: connecting via hostname, if it was a wrong hostname i wouldent be getting to a server that knew sip |
| 00:47.16 | claude81 | dluns_,voicemail |
| 00:47.18 | Ariel_ | not a ras server. But there used to be a ppras application |
| 00:47.29 | Shaun2222 | Ariel_: so... |
| 00:47.33 | orlok | Ariel_: just assuming that its like an httpd 404, except not http |
| 00:47.43 | Ariel_ | orlok, correct |
| 00:47.47 | Ariel_ | not found |
| 00:47.48 | orlok | ie, its gotten to a server that speaks the right protocol, but the requested resource isnt there |
| 00:47.55 | orlok | Hmm. Weird. |
| 00:48.54 | claude81 | i have edited the defualt contect in vmail.cgi, the login works with no proble |
| 00:49.02 | k-man | dlynes, thats interesting |
| 00:49.11 | claude81 | but when i login, it dose not dispaly any mail |
| 00:49.27 | claude81 | i chmod all the voicemails files |
| 00:49.29 | k-man | is there a way to use the voicemail, call attendant parts of asterisk with a non asterisk phone system? |
| 00:49.38 | k-man | would it be worth the effort? |
| 00:49.51 | dlynes_ | k-man, also, if you check the how-to section of the asterisk wiki, you'll see a how-to for fax, bri, and music-on-hold |
| 00:50.08 | k-man | oh |
| 00:50.08 | k-man | thanks |
| 00:50.08 | k-man | what is bri? |
| 00:50.15 | dlynes_ | k-man, basic rate ISDN |
| 00:50.24 | k-man | oh... |
| 00:50.30 | dlynes_ | k-man, it's like 64K dialup, but without the latency |
| 00:50.31 | k-man | is bri good? |
| 00:50.36 | k-man | oh, i see |
| 00:50.40 | k-man | whats the benefit? |
| 00:50.42 | dlynes_ | k-man, bri is expensive |
| 00:50.45 | ManxPower | k-man, only if you are not in the USA |
| 00:50.55 | dlynes_ | k-man, the benefit is it puts more money into the telco's pocket |
| 00:51.00 | ManxPower | BRI is CHEAP in many parts of the world. Cheaper than 2 pots lines |
| 00:51.11 | orlok | heh |
| 00:51.14 | orlok | not australia :) |
| 00:51.16 | ManxPower | In the USA BRI is expensive and nobody uses it anymore |
| 00:51.21 | k-man | im sure its more expensive in australia than 2 pots |
| 00:51.29 | dlynes_ | ManxPower, I've heard it's expensive in europe too, compared to adsl |
| 00:51.38 | k-man | im not sure, but i think its really expensive in australia too |
| 00:51.42 | ManxPower | dlynes, um, I'm talking about for voice |
| 00:51.48 | dlynes_ | ah |
| 00:51.49 | ManxPower | everyone in europe uses BRI for voice |
| 00:52.06 | ManxPower | msw, 1.0.x or 1.2.x? |
| 00:52.13 | msw | ManxPower: 1.2.x |
| 00:52.16 | dlynes_ | regular phone lines are that expensive in europe? |
| 00:52.34 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 00:52.46 | ManxPower | dlynes, not really, BRI is just cheaper than 2 pots lines in much of europe. |
| 00:52.55 | msw | SVN-trunk-r18867 |
| 00:52.56 | orlok | k-man: actually, its not that much more, but you get slugged with data tariffs |
| 00:52.58 | k-man | how is softswitch different from asterisk? |
| 00:53.00 | ManxPower | But, of course, both POTS and BRI are billed by the min even for local calls |
| 00:53.06 | dlynes_ | ah |
| 00:53.08 | k-man | orlok, oh... ok |
| 00:53.12 | dlynes_ | yeah...that would suck |
| 00:53.12 | orlok | k-man: so theres a whole thing for doing voice calls over isdn and using data over that :) |
| 00:53.21 | orlok | DoV |
| 00:53.26 | dlynes_ | that's one nice thing about NA....free local calls :) |
| 00:53.37 | orlok | so telstra flags it as a voice call |
| 00:53.41 | k-man | DoV? |
| 00:53.45 | k-man | Data over voice? |
| 00:53.45 | ManxPower | k-man, if you define "softwsitch" (everyone defines it differently) then we can give you an answer. |
| 00:53.48 | k-man | oh |
| 00:53.50 | k-man | i see |
| 00:54.12 | k-man | um.... the software talkedabout on www.soft-switch.org |
| 00:54.38 | msw | k-man: that's DSP stuff |
| 00:55.07 | dlynes_ | so i guess nobody has any ideas on why an on-hold when brought back is only sending voice back to the locally asterisk connected speakers, and not the remote caller? |
| 00:55.25 | ManxPower | k-man, yeah, that's "just" Fax DSP stuff. |
| 00:55.51 | ManxPower | I say "just" because I could never even behin to understand the math of DSP stuff |
| 00:55.54 | coppice | actually its a lot more than FAX DSP, but it isn't a full soft-switch |
| 00:56.03 | dlynes_ | ManxPower, he's wanting to do faxing on asterisk |
| 00:56.06 | *** join/#asterisk distortion (i=distorti@junipero.3sheep.com) |
| 00:56.21 | generalhan | i fax from Asterisk, but not using anything like that |
| 00:56.23 | ManxPower | dlynes, then spandsp and rx/tx fax are one good option. |
| 00:56.37 | dlynes_ | generalhan, what do you use? |
| 00:56.39 | generalhan | i fax out from our PRI lines |
| 00:56.47 | generalhan | using a TDM card |
| 00:56.57 | ManxPower | I need to put my fax2email script along with docs on the Wiki sometime, it uses spandsp as well |
| 00:57.10 | k-man | i don't really want to do fax from asterisk speficicaly |
| 00:57.18 | generalhan | i WAS using FoIP and i got soooooooooo frustrated and my boss threatened me with my life so i switched it up |
| 00:57.32 | generalhan | and now i havent had one dropped fax |
| 00:57.41 | k-man | what i was wondering is, is there a way to take a document on my computer and send it to some sort of voip provider in a form that they can just deliver it to a fax? |
| 00:57.44 | ManxPower | generalhan, FoVoIP is just asking for pain, suffering, and disapointment. |
| 00:57.58 | dlynes_ | Yeah...just curious...all my faxes drop on the second page using txfax/rxfax |
| 00:58.02 | generalhan | ohhh youre preaching to the choir Manx !! |
| 00:58.06 | Mavvie | generalhan: except if you use vegastream equipment. |
| 00:58.10 | Az_au | dlynes_: i had that problem |
| 00:58.21 | dlynes_ | Az_au, how'd you solve it? |
| 00:58.22 | ManxPower | dlynes, check your gains |
| 00:58.32 | Az_au | dlynes_: got rid of it and used iaxmodem and hylafax.. works great |
| 00:58.45 | generalhan | well first i tried faxing over our VoIP termination ... then i tried putting the fax cards in a spare server and transfering from one server to the other via IAX .. and that was only a tiny bit better |
| 00:58.56 | generalhan | so then i went and got the TDM card and havent had a problem yet |
| 00:59.12 | dlynes_ | ManxPower, rxgain/txgain is used on a pri? |
| 00:59.24 | ManxPower | in my experience txfax/rxfax problems are 1) too slow of a system, 2) timing issues on the T-1/E-1, 3) gain issues (usually caused by trying to get rid of echo on voice calls) |
| 00:59.39 | ManxPower | dlynes, yes, txgain/rxgain works on a PRI |
| 00:59.51 | dlynes_ | ManxPower, what should they be set to, then? |
| 01:00.03 | orlok | Ariel_: Hmm, managed to get it working. hostname was right, weird. |
| 01:00.05 | generalhan | set them both to 0 and then play with them till you get it perfect ! |
| 01:00.11 | dlynes_ | generalhan, lol |
| 01:00.22 | generalhan | lol ?? i was dead serious ! lol |
| 01:00.36 | dlynes_ | Oh...thought you were joking because of the exclamation point :) |
| 01:00.42 | generalhan | i played with it for a couple of days till i got it right .... but thats cause some one told me where to start it ! |
| 01:00.44 | coppice | ManxPower: echo is irrelevant to FAX, though echo cancellation must be off for it to work well. gain shouldn't matter. not sure about slow. how many have you tried to do? a dual xeon did >100 concurrent faxes the last time I tried |
| 01:00.54 | ManxPower | playing with rxgain/txgain REALLY helps with echo |
| 01:01.16 | dlynes_ | ah...yeah...i have echo cancellation enabled |
| 01:01.18 | ManxPower | coppice, Yeah. but if you set txgain=-6 to get rid of echo on voice calls...... |
| 01:01.33 | Chules | Hey, I have a quick question. In the sample sip.conf, it says the nat= setting is used "when the asterisk is on a public IP communicating with a device behind NAT". So, if the asterisk server is behind NAT and I use the externalip and localnet directives, do I still need nat=yes or something to communicate with external devices also behind nat? |
| 01:01.49 | coppice | gets rid of echo and everything else :-) |
| 01:01.51 | claude81 | <PROTECTED> |
| 01:01.53 | dlynes_ | Is there any way I can allocate so many channels on the pri to fax, and so many to voice? |
| 01:02.11 | dlynes_ | Or is asterisk not able to break it up like that? |
| 01:02.15 | generalhan | you can break it up |
| 01:02.18 | Az_au | dlynes_: you could setup different groups on the channels |
| 01:02.28 | dlynes_ | ah...ok |
| 01:02.34 | generalhan | i have 2 Voice T-1s and 7 different call groups |
| 01:03.04 | distortion | anyone play with the t38 tree? or did i come in after it was discussed.. |
| 01:03.09 | dlynes_ | It's just a little confusing when you can't break it up into contexts; I'm not sure which settings are global for the '[channels]' context, and which ones are defining a group |
| 01:03.22 | generalhan | you can change the context |
| 01:03.38 | dlynes_ | and asterisk is smart enough to know what the context is for? |
| 01:03.49 | generalhan | group=10; context=new_context; channel=10-15; |
| 01:04.03 | dlynes_ | Oh...you mean the incoming context for the dialplan |
| 01:04.08 | generalhan | so channels 10-15 will go to context new_context. |
| 01:04.43 | generalhan | well actually i meant the outgoing context |
| 01:04.46 | dlynes_ | So is the echocancel, echocancelwhenbridged, txgain, and rxgain options are localized when you used the group= option? |
| 01:04.52 | orlok | oh what the fuck! |
| 01:04.58 | generalhan | dlynes: yes |
| 01:05.04 | dlynes_ | generalhan, cool...thanks |
| 01:05.06 | orlok | one of the accounts out sip provider gave us isnt working! |
| 01:05.09 | orlok | only one out of three |
| 01:05.10 | generalhan | you can set them differently for each group |
| 01:05.38 | dlynes_ | generalhan, yeah..the sample zapata.conf file isn't very helpful for letting you know which options you can repeat, and which ones you can't |
| 01:05.52 | dlynes_ | generalhan, I've noticed some if i repeat more than once, they overwrite the old settings |
| 01:05.59 | generalhan | im just glad that some one had a question i could answer .... ive taken so much help from this chan, i want to return the favor, but i dont know enough to help many people ! lol |
| 01:06.16 | dlynes_ | heh |
| 01:06.23 | Hmmhesays | lol |
| 01:06.23 | dlynes_ | I just haven't spent much time trying to configure the pri |
| 01:06.24 | orlok | oh my god |
| 01:06.27 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 01:06.32 | orlok | i was thinking i've been going insane |
| 01:06.34 | Hmmhesays | so i'm not looking forward to jail |
| 01:06.36 | dlynes_ | I'm of the frame of mind that if it ain't broke, don't fix it |
| 01:06.40 | generalhan | dlynes: as long as you have the settings BENEATH the "group=" youll be ok ... cause then it will get to the next "group=" and start the settings all over again |
| 01:06.50 | dlynes_ | ah...ic now |
| 01:06.52 | dlynes_ | thanks |
| 01:06.59 | generalhan | Hmmhesays: jail ? |
| 01:07.17 | Hmmhesays | yeah |
| 01:07.22 | generalhan | for what ? |
| 01:07.29 | coppice | distortion: have you tried the t.38 tree? |
| 01:07.36 | Hmmhesays | i'd rather not talk about it |
| 01:07.39 | generalhan | ohh ok |
| 01:07.52 | generalhan | well at least you have time to think about it before they take you in |
| 01:07.57 | generalhan | ive never had that luxury |
| 01:08.06 | Hmmhesays | and one good lay from the gF |
| 01:08.11 | cced | why not Alerting msg in libpri? |
| 01:08.31 | generalhan | do you know how long already ? |
| 01:08.39 | Hmmhesays | its just a few days |
| 01:09.15 | generalhan | hmm ... if i knew you were in AZ i would say DUI. there arent many things you can go for a couple days for |
| 01:09.59 | dlynes_ | generalhan, thanks again...i'm off for now |
| 01:10.02 | Hmmhesays | a dui that wasn't my fault |
| 01:10.16 | generalhan | dlynes: no sweat man ! |
| 01:10.18 | dlynes_ | ManxPower, thanks for the tip about the rxgain/txgain, too |
| 01:10.29 | generalhan | i thought DUI sounded about right |
| 01:10.31 | Hmmhesays | fscking ex GF made me wreck my car that stupid grrrr |
| 01:10.50 | generalhan | convinced you to go out trashed huh ? |
| 01:10.57 | Hmmhesays | i wasn't trashed |
| 01:11.12 | generalhan | how'd you get nailed then ... |
| 01:11.12 | Hmmhesays | i only got 2 checks on my field sobriety test, out of 8 |
| 01:11.20 | generalhan | booo |
| 01:11.30 | generalhan | they test 8 different ways in your state ? |
| 01:11.31 | generalhan | wow |
| 01:11.35 | Hmmhesays | she was trashed trying to hop on me i was pushing her off, she hit the steering wheel |
| 01:11.36 | generalhan | we get 3 |
| 01:12.06 | dlynes_ | we get one...if you blow over the limit, you're screwed |
| 01:12.11 | Hmmhesays | cranked the wheel |
| 01:12.14 | coppice | they are just checking you can count as far as 8 :-) |
| 01:12.16 | dlynes_ | you blow over the limit three times, and you lose your license for life |
| 01:12.59 | *** join/#asterisk |omni| (i=cathode@216.64.178.146) |
| 01:13.12 | generalhan | hmmhesays: well man .... im sorry about that .... jail is not fun at all, but once you get out after just 2 days, you'll realize how much you NEVER want to go back and start doing things 1000 times more carefully than you did before |
| 01:13.51 | |omni| | hey..I had the same reaction :P |
| 01:14.07 | Hmmhesays | no choice i have no license for a year |
| 01:14.12 | |omni| | bunch of hoodlums in here |
| 01:14.46 | dlynes_ | damn...a year for the first offense |
| 01:15.06 | Hmmhesays | well i wrecked a car |
| 01:15.47 | dlynes_ | what about the people in it? |
| 01:15.50 | *** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net) |
| 01:15.52 | Hmmhesays | no one in it |
| 01:16.07 | dlynes_ | ah...lucky |
| 01:16.29 | Hmmhesays | no i picked that particular sidewalk because i knew everyone would be in bed |
| 01:17.03 | generalhan | hahahahaha |
| 01:17.34 | dlynes_ | but obviously you woke someone up when you hit the car |
| 01:17.38 | kamileon | jail is an interesting society |
| 01:17.40 | generalhan | well thats not too bad to lose it for a year for DUI. i got nailed WAY back in the day when i first turned 17 or something for drag racing on a public street and i lost it for a year |
| 01:17.55 | Hmmhesays | sucks doesn't it |
| 01:18.01 | generalhan | it does |
| 01:18.12 | |omni| | I got a DUI when I was 17 (32 now) refused breathalizer and in WA state that's an automatic year revocation |
| 01:18.21 | generalhan | i dont know if they do this in your state, but i got a permit to at least drive to work so that i wouldnt lose my job too |
| 01:18.21 | |omni| | it's not too bad |
| 01:18.32 | Hmmhesays | can't |
| 01:18.32 | dlynes_ | yeah...WA's almost as bad as BC |
| 01:18.44 | generalhan | why not ? |
| 01:18.47 | generalhan | they wont allow it ? |
| 01:18.50 | Hmmhesays | nope |
| 01:19.02 | Hmmhesays | good thing i have a bike, |
| 01:19.03 | generalhan | that suck ... how you gonna get to and from work ? |
| 01:19.05 | generalhan | ohh ok |
| 01:19.09 | Hmmhesays | chances of getting pulled over in that are slim |
| 01:19.14 | generalhan | Hahaha |
| 01:19.27 | dlynes_ | dood |
| 01:19.29 | Hmmhesays | its true |
| 01:19.36 | dlynes_ | you've never gotten pulled over on your schwinn before? |
| 01:19.40 | RageMax | I'm glad we have an awesome public transportation system in case that happens to me, which it probably won't ;) |
| 01:19.53 | Hmmhesays | we hoever do not |
| 01:19.56 | generalhan | well shit Hmmm, that sucks BIG TIME. |
| 01:20.01 | Hmmhesays | it services about half the city |
| 01:20.01 | generalhan | sorry bout that |
| 01:20.14 | RageMax | you can definitely get in trouble for being intoxicated on a bike |
| 01:20.19 | Hmmhesays | which means its about time to get the home business going again |
| 01:20.22 | kamileon | drive anyways |
| 01:20.25 | Hmmhesays | i meant motorcycle |
| 01:20.29 | generalhan | well gentlemen ...im outta here, i cant stay at work anymore .... i wanna kill myself ! |
| 01:20.31 | Hmmhesays | and i don't ride drunk |
| 01:20.32 | kamileon | driving without carrys only 5 days usually |
| 01:20.33 | generalhan | talk to you all later ! |
| 01:20.42 | kamileon | or ~200$ fine |
| 01:20.43 | coppice | RageMax: your public transport is only awesome when most people choose not to have a car |
| 01:20.44 | Hmmhesays | unless you're on suspended sentence |
| 01:21.05 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 01:21.07 | kamileon | you wont get violated for traffic offenses |
| 01:21.30 | kamileon | unless maybe maybe youre on federal paper, then they might violate |
| 01:21.32 | RageMax | coppice: I have a car, and so does almost everyone who rides. We have a busway, and you just park your car at the station, and you're downtown in less time than it would take to drive |
| 01:21.45 | RageMax | so the ratio doesn't really matter |
| 01:21.52 | RageMax | the parking lot is always full |
| 01:22.10 | dlynes_ | Sounds like Toronto |
| 01:22.13 | coppice | RageMax: we have a public transport system that means most people don't even consider a car a useful thing |
| 01:22.16 | dlynes_ | public transit there is freaking wicked |
| 01:22.43 | RageMax | plus I'm already paying for public transit in my tuition as a student, so I use it, heh |
| 01:22.53 | RageMax | sure saves a hell of a lot on gas |
| 01:23.56 | msw | ManxPower: so - any ideas on how app_voicemail is getting a callerid name, but ${CALLERID(name)} is ""? |
| 01:24.47 | orlok | generalhan: what sorta car? |
| 01:25.33 | dlynes_ | orlok, he never mentioned a car, and besides...he's already left |
| 01:25.42 | orlok | oh, damn :) |
| 01:26.00 | orlok | oh man |
| 01:26.08 | orlok | sip provider gave us a pile of incorrect logins |
| 01:26.20 | orlok | i've been going nuts trying to figure out why asterisk is so hard |
| 01:26.30 | orlok | "whats with all the 404's!" |
| 01:26.52 | dlynes_ | some of them work, some of them don't? |
| 01:26.56 | orlok | yeah |
| 01:26.59 | orlok | 2 work, one doesnt |
| 01:27.05 | orlok | try some numbers out of another batch, nope |
| 01:27.06 | orlok | etc etc |
| 01:27.08 | dlynes_ | and they're all configured exactly the same way? |
| 01:27.17 | orlok | but the two numbers that worked got assigned to standalone phones of course |
| 01:27.22 | msw | hmm.. show channel Zap/1-1 shows a clid="THE RIGHT NAME" <number> |
| 01:27.24 | orlok | so i'm assuming its *, cos the phones work! |
| 01:28.06 | dlynes_ | msw, what's ${CALLERIDNAME} set to? |
| 01:29.59 | msw | dlynes_: says "" |
| 01:30.20 | dlynes_ | msw, ${CALLERIDNAME}, or ${CALLERID(name)}? |
| 01:30.32 | msw | neither are set correctly |
| 01:31.02 | Qwell | both will be the same, always |
| 01:31.05 | dlynes_ | msw, are you checking the value of the variable before or after you issue an Answer()? |
| 01:31.56 | Qwell | Voicemail does an implicit answer, so... |
| 01:32.04 | dlynes_ | msw, to guarantee delivery, I would do a Wait(3), and then an Answer(), and then a Noop(Calleridname=${CALLERIDNAME}) |
| 01:32.23 | dlynes_ | Wait(3) is to give the callerid time to come in on the line |
| 01:32.29 | Qwell | ${CALLERIDNAME} shouldn't be used anymore with 1.2, and I believe will be gone in 1.4 |
| 01:32.37 | msw | dlynes: this is PRI |
| 01:32.38 | dlynes_ | Answer() is to guarantee the variable will be set |
| 01:32.47 | dlynes_ | msw, Doesn't matter |
| 01:33.16 | dlynes_ | Qwell, ah...was thinking CALLERID(name) was the 1.0 method and CALLERIDNAME was the 1.2 method :) |
| 01:33.27 | msw | dlynes_: backwards. ;-) |
| 01:33.31 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:33.51 | msw | an Answer() didn't help - trying a delay |
| 01:33.53 | dlynes_ | Yeah...my brain's been kinda fried lately...working too many hours trying to get this billing system finished |
| 01:34.34 | msw | dlynes: indeed |
| 01:34.39 | msw | a wait(3) worked |
| 01:34.46 | dlynes_ | Of course :) |
| 01:34.51 | msw | thanks |
| 01:34.51 | asterboy | don't forget the round pennies goes to your offshore account |
| 01:34.54 | msw | (is that in a FAQ? ;-)) |
| 01:35.01 | dlynes_ | msw, anyways...3 is overkill |
| 01:35.06 | dlynes_ | msw, two will probably work |
| 01:35.13 | dlynes_ | msw, I just figured 3 would guarantee it |
| 01:35.23 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:35.38 | msw | 1 worked |
| 01:36.17 | dlynes_ | msw, if it's in the faq, it'd be under the section for callerid |
| 01:36.31 | dlynes_ | msw, if not, it'll be on the net in a faq for callerid somewhere |
| 01:36.47 | dlynes_ | msw, it's a pretty common reason for why people don't get callerid to work |
| 01:36.59 | dlynes_ | callerid comes in on the 2nd ring, not the first ring |
| 01:37.10 | dlynes_ | well...just before the 2nd ring, actually |
| 01:37.59 | dlynes_ | btw...anyone know of any of those x100p clones on ebay that you can order that don't get shipped from Hong Kong or something? |
| 01:39.01 | *** join/#asterisk mmunky (n=mmunky@81-178-251-9.dsl.pipex.com) |
| 01:51.40 | Qwell | http://appleinsider.com/article.php?id=1672 |
| 01:51.41 | Qwell | :D |
| 01:54.22 | dlynes | damn, that web site's slow |
| 01:57.10 | [TK]D-Fender | DAmmittoheck... I could use a hand here.. I've got a CentoOS 4.3 server I'm working on that doesn't have the kernel sources on it. what do I do with YUM to get them installed? |
| 01:57.22 | orlok | kernel-devel |
| 01:57.28 | [TK]D-Fender | its running a 2..9smp kernel |
| 01:57.34 | [TK]D-Fender | 2.6.9 |
| 01:57.54 | orlok | there are kernel-devel packages for most kernels which are basically the source set up for that kernel |
| 01:58.06 | orlok | pissed me off as well |
| 01:58.18 | orlok | i was rebuiulding the kernel .src.rpm and stuff |
| 01:58.22 | Chules | Hey, I have a quick question. In the sample sip.conf, it says the nat= setting is used "when the asterisk is on a public IP communicating with a device behind NAT". So, if the asterisk server is behind NAT and I use the externalip and localnet directives, do I still need nat=yes or something to communicate with external devices also behind nat? |
| 01:58.38 | Qwell | Chules: on the same NAT? |
| 01:59.14 | [TK]D-Fender | Chules : Yes, * still needs to know not to trust the IP it sees in that ATA's SIP headers and instead look at the packet origin |
| 01:59.32 | Chules | Qwell: No |
| 01:59.36 | Qwell | If it's the same NAT, you don't want nat=yes. localnet will take care of it |
| 01:59.43 | Qwell | Then yes, you need nat=yes for that peer |
| 02:00.09 | Chules | Qwell: Well, the device might be on the same NAT sometimes, and other times on another NAT. |
| 02:04.50 | [TK]D-Fender | ~centosbug |
| 02:04.51 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 02:07.33 | [TK]D-Fender | wow... i'm feeling almost useful.... |
| 02:07.53 | Ariel_ | [TK]D-Fender, last centOS I update already had that fix in the release.. |
| 02:12.26 | *** join/#asterisk pengyong (n=lala@222.188.132.127) |
| 02:12.35 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 02:13.06 | [TK]D-Fender | Ariel_ : I'm finding EVERYTHING wrong on this stupid server.... missing packages EVERYWHERE!!!!! |
| 02:13.10 | [TK]D-Fender | bastards! |
| 02:13.20 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 02:13.37 | [TK]D-Fender | No libtermcap, no ncurses (can't figure how to get this one in), no "flex" |
| 02:13.45 | Chules | Debian? |
| 02:13.52 | [TK]D-Fender | Ariel_ : can you hint me for the ncurses one? |
| 02:14.00 | Chules | er wait |
| 02:14.06 | Chules | nm |
| 02:14.10 | [TK]D-Fender | Think I got it... |
| 02:14.12 | Qwell | [TK]D-Fender: apt-cache search ncurses |
| 02:14.20 | Chules | In debian it's: |
| 02:14.28 | Chules | apt-get install ncurses-dev |
| 02:14.31 | Chules | Or: |
| 02:14.38 | Chules | apt-get install libncurses5-dev |
| 02:14.57 | [TK]D-Fender | Chules : CentOS. and I got it "ncurses-devel", but now I'm missing g++ |
| 02:15.12 | Qwell | gcc-g++? |
| 02:15.30 | [TK]D-Fender | Qwell : I'm lost on this POS box... |
| 02:15.38 | Qwell | [TK]D-Fender: yum search g++ |
| 02:16.02 | [TK]D-Fender | Qwell : nada |
| 02:18.02 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 02:18.45 | *** join/#asterisk oej (n=oej@203.178.156.27) |
| 02:19.11 | Ariel_ | wow |
| 02:19.33 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 02:20.55 | Ariel_ | if you do yum info g* it will tell you what packages there are |
| 02:24.02 | [TK]D-Fender | Actually I'm a little stuck on wanpipe right now... kernel source issues |
| 02:25.19 | Ariel_ | have you don yum update |
| 02:25.22 | Ariel_ | done |
| 02:26.00 | [TK]D-Fender | not sure in this whole mess.... I hate not being on Slackware... I'd be DONE by now! |
| 02:26.57 | [TK]D-Fender | I'm running through it now.. saying a prayer... |
| 02:27.00 | Ariel_ | [TK]D-Fender, the packages should be about the same just need to do yum install blah |
| 02:28.05 | [TK]D-Fender | Ariel_ : yeah well I'm something of a Linux newb. I've been spoiled by Slackware's "just works" world.... |
| 02:28.31 | Ariel_ | [TK]D-Fender, funny that is what I say when I am not on CentOS |
| 02:29.14 | [TK]D-Fender | ok I'm missing G++ (gcc and family) |
| 02:30.11 | Ariel_ | do yum info gc* |
| 02:30.17 | Ariel_ | you will get the package names |
| 02:31.16 | Ariel_ | or yum install gcc gcc-c++ etc |
| 02:31.36 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:31.59 | [TK]D-Fender | I'm looking to do "yum intall gcc*" |
| 02:35.41 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
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| 02:38.48 | X-Rob | [TK]D-Fender, wozzup? |
| 02:39.02 | *** join/#asterisk oej (n=oej@203.178.156.27) |
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| 02:41.21 | SplasPood | [TK]D-Fender: what's the output of yum info gc* look like? could pipe to awk/xargs |
| 02:41.31 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 02:41.38 | Katty | evening. |
| 02:42.23 | Ariel_ | hello Katty |
| 02:43.28 | Katty | hey Ariel_ (= |
| 02:44.15 | docelmo | MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW |
| 02:44.53 | Katty | hey docelmo |
| 02:45.50 | docelmo | whadup? |
| 02:46.32 | Katty | not me, i'm pretty down |
| 02:49.12 | *** join/#asterisk ivanfm (n=ivanfm@201-1-167-207.dsl.telesp.net.br) |
| 02:49.20 | *** join/#asterisk Tili (i=Tili@219.136.13.134) |
| 02:52.12 | [TK]D-Fender | SplasPood : got the guy who set it up to do that and a few other things for me. Friggen SMP problems, bad kernel formats, etc... |
| 02:53.21 | [TK]D-Fender | I could use it with this frustrating dumb-ass system I'm working on.... |
| 02:53.31 | LostFrog | Ribbit Ribbit Ribbit |
| 02:55.05 | rollergrrl | exten => s,2,GotoIf($["${MYVAR}" = ""]?s|1000) |
| 02:55.18 | rollergrrl | Is there a better way to tell if a variable has been assigned than this? |
| 02:55.38 | rollergrrl | Can I remove the = ? |
| 02:55.51 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 02:56.04 | [TK]D-Fender | rollergrrl : nope, thats the way, though you don't need the "s|" |
| 02:56.23 | docelmo | new chic in #asterisk.. Move over Katty |
| 02:56.24 | docelmo | :) |
| 02:56.27 | rollergrrl | ok |
| 02:56.38 | rollergrrl | eh? |
| 02:56.42 | rollergrrl | I've been here for years |
| 02:56.42 | docelmo | haha |
| 02:56.43 | [TK]D-Fender | docelmo : thats ALEDGED |
| 02:56.56 | docelmo | [TK]D-Fender, huh? |
| 02:57.40 | russellb | docelmo: I swear, if I see "MEW MEW MEW" one more time ... |
| 02:57.46 | docelmo | hehehe |
| 02:58.22 | docelmo | 1.2.7.1? for app_page? who broke it.. :) |
| 02:58.43 | dlynes | Ciber311: you there? |
| 02:59.11 | russellb | docelmo: file[laptop] did it |
| 03:00.25 | file | yup |
| 03:00.30 | file | mee |
| 03:00.48 | russellb | just kidding file :D |
| 03:01.19 | Katty | docelmo: the more the merrier if you ask me ;) |
| 03:01.27 | [TK]D-Fender | rollergrrl : You could techincally save a few chars and do it like this : exten => s,2,GotoIf($[x${MYVAR}=x]?1000) |
| 03:01.46 | xachen | http://www.thinkgeek.com/stuff/41/1upmushroom.shtml |
| 03:01.48 | xachen | haha |
| 03:02.34 | [TK]D-Fender | xachen : Hilarity |
| 03:02.50 | *** join/#asterisk BugKham (n=BugKham@125.24.1.149) |
| 03:03.44 | BugKham | How to limit a SIP user to be able to make only outgoing calls? |
| 03:03.52 | BugKham | type=peer doesn't help |
| 03:04.05 | LostFrog | Don't forward calls to him. |
| 03:05.23 | BugKham | LostFrog: =) thought there is a particular setting for this |
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| 03:07.31 | LostFrog | I might be stupid, but if you don't have a dial for his/her extension, calls aren't getting there. |
| 03:08.39 | *** join/#asterisk Syllogism (i=Syllogis@67.14.195.195) |
| 03:09.00 | FuriousGeorge | any of the pros awake in here? i think i found a bug with the dialplan parser and nested while loops: http://pastebin.ca/49850 |
| 03:09.34 | FuriousGeorge | that loop never gets back to the original while that called it |
| 03:09.41 | dlynes | ~seen ciber311 |
| 03:09.53 | jbot | ciber311 is currently on #asterisk (6h 17m 43s), last said: '234 for overnight...'. |
| 03:10.17 | FuriousGeorge | it seems to think the endwhile that comes after i jump out to priority (finalscore) belongs to the child loop and not the parent loop as it should |
| 03:10.25 | FuriousGeorge | as such the counter increments and the loop never ends |
| 03:10.27 | FuriousGeorge | am i wrong |
| 03:10.31 | zoa | FuriousGeorge: too little blood in my alcohol |
| 03:10.34 | FuriousGeorge | http://pastebin.ca/49850 |
| 03:11.09 | FuriousGeorge | zoa: if i looked at that drunk, i'd say bug, but i was hoping someone who knows a bit more than me about this can advise as to whether or not i should file a bug report |
| 03:11.18 | dlynes | FuriousGeorge: so it was a problem with the while loop, after all? |
| 03:11.32 | zoa | i'd say post it on the mailinglist |
| 03:11.34 | FuriousGeorge | dlynes: no for me yesterday it was actually my fault |
| 03:11.38 | zoa | and see what the sober people think about it |
| 03:11.39 | dlynes | ah |
| 03:11.49 | FuriousGeorge | i used a , where i needed a : w/ random |
| 03:11.55 | FuriousGeorge | Qwell finally spotted it |
| 03:12.03 | dlynes | ah |
| 03:12.16 | FuriousGeorge | but i did find a bug w/ random where a 1% chance evaluates true as often as a 0% chance (i.e. never) |
| 03:12.23 | dlynes | yeah...I think it was you yesterday that put in a pipe symbol where there should've been a ':' too, right? |
| 03:12.36 | FuriousGeorge | and i got to file a report about that, which makes me feel better because that was breaking my code anyway yesterday |
| 03:12.41 | dlynes | i.e. for a GotoIf statement |
| 03:12.58 | FuriousGeorge | dlynes: im not making that mistake anymore :) |
| 03:13.12 | dlynes | heh...i should hope not :) |
| 03:13.28 | FuriousGeorge | dlynes: why dont you take a peek at that pastebin and see if you see my point |
| 03:13.36 | dlynes | your point? |
| 03:13.48 | FuriousGeorge | notice line 55 where the count gets set to 3 but the loop continues |
| 03:13.53 | FuriousGeorge | http://pastebin.ca/49850 |
| 03:14.04 | FuriousGeorge | this nested loop never terminates, even when it should |
| 03:15.00 | FuriousGeorge | and between count = 1 and count = 2 it should shout out that NOOP about WHY ARENT WE GETTING BACK HERE |
| 03:15.33 | dlynes | It looks like you've got a continuous loop that never exits to me |
| 03:15.41 | FuriousGeorge | when it should exit |
| 03:15.48 | dlynes | while(1) should always evaluate to true |
| 03:15.54 | dlynes | It should never exit |
| 03:15.58 | FuriousGeorge | if that random statement = true it should jump out to priority (finalscore) |
| 03:16.05 | FuriousGeorge | which is not in that while(1) loop |
| 03:16.15 | dlynes | Oh...is that what random does? |
| 03:16.28 | dlynes | I've never used the random function...I figured it just generated a random number |
| 03:16.40 | FuriousGeorge | me toop, but it dont |
| 03:16.46 | FuriousGeorge | would make my life easier |
| 03:17.01 | FuriousGeorge | random is just a goto that branches based on a probability |
| 03:17.12 | *** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
| 03:17.16 | dlynes | ah |
| 03:17.19 | FuriousGeorge | random(prob:context,exten,priority) |
| 03:17.28 | freat | hello |
| 03:17.35 | FuriousGeorge | so now you see what i mean about el bugerino |
| 03:17.49 | dlynes | prob is what? |
| 03:17.56 | dlynes | just an evaluated expression? |
| 03:18.22 | FuriousGeorge | integer |
| 03:18.29 | FuriousGeorge | in my case i set it to 33 |
| 03:18.37 | FuriousGeorge | i made that way too confusing i think |
| 03:18.41 | FuriousGeorge | with the array and the counter |
| 03:18.48 | dlynes | So it's between 0 and MATEWEIGHT[COUNT]? |
| 03:18.51 | FuriousGeorge | but its pretty simple once you get it |
| 03:19.06 | FuriousGeorge | well MATEWEIGHT1 and -2 both = 33 |
| 03:19.09 | dlynes | or 0 and MATEWEIGHT[COUNT]-1? |
| 03:19.25 | freat | can anyone tell me what defaultexpirey and maxexpirey do in sip.conf? The wiki says defaultexpirey=120 :Default length of incoming/outgoing registration... does that mean how often the registration will occur? |
| 03:20.02 | FuriousGeorge | that random says there is a 33 % chance that it will jump to finalscore |
| 03:20.05 | freat | recommended values? |
| 03:20.13 | FuriousGeorge | if not it will increment the counter and stay in that while(1) loop |
| 03:20.25 | dlynes | so MATEWEIGHT[COUNT] is 3 then? |
| 03:20.35 | ManxPower | FuriousGeorge, why not just write an AGI script? |
| 03:20.38 | FuriousGeorge | well, if it takes 3 tries it will be |
| 03:20.50 | FuriousGeorge | ManxPower: i only know pascal and asterisk :) |
| 03:21.02 | FuriousGeorge | ManxPower: im learning C as we speak but im doing this stuff to get into the mindset again |
| 03:21.05 | ManxPower | FuriousGeorge, how about AEL, that should be pretty easy. |
| 03:21.07 | FuriousGeorge | its been 7 years since i coded |
| 03:21.15 | FuriousGeorge | ManxPower: thats after i finish this cow book |
| 03:21.23 | FuriousGeorge | im just warming up now :) |
| 03:21.27 | ManxPower | FuriousGeorge, learn again. Simple Perl is not all THAT hard and neither is PHP |
| 03:21.34 | FuriousGeorge | i got a book on PHP too |
| 03:21.43 | FuriousGeorge | i agree with you |
| 03:21.44 | docelmo | PHPAGI RULES! |
| 03:21.55 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 03:22.06 | docelmo | I know.. I am one of the developers.. :P |
| 03:22.07 | FuriousGeorge | im just doing some exercises. the point is to see how much easier its gonna be to do it in C once im done |
| 03:22.10 | ManxPower | I've programmed in COBOL, FORTRAN, Modula-2, RPG-II, C, Pascal, PHP, Perl. |
| 03:22.20 | ManxPower | Oh and Logo and BASIC, of course. |
| 03:22.26 | freat | docelmo: cool... I'm an asterisk-java fan myself |
| 03:22.29 | FuriousGeorge | ManxPower: what do you think though? should i file a bug report or am i missing something about while loops |
| 03:22.29 | freat | Logo! nice |
| 03:22.31 | ManxPower | After learning 1 language, learning another is easy. |
| 03:22.50 | FuriousGeorge | ManxPower: C is comming pretty naturally |
| 03:22.51 | docelmo | FuriousGeorge, take it from someone who has done it.. PHP -> C is very simple.. I found it easy to transition |
| 03:22.52 | dlynes | C, C++, x86 assembly, z80 assembly, 65xx assembly, Java, Perl, shell scripting, ... for me |
| 03:22.59 | ManxPower | FuriousGeorge, what's the smallest number of dialplan lines can you use and still reproduce the problem? |
| 03:23.08 | docelmo | dlynes overacheiver |
| 03:23.11 | FuriousGeorge | ManxPower: FINE :) brb |
| 03:23.19 | FuriousGeorge | im down to like 8 now though |
| 03:23.27 | ManxPower | Well, I've done some shall scripting, and Java too, I guess. |
| 03:23.29 | dlynes | docelmo: well, i've learned a few other languages too...but none of them are in use anymore |
| 03:23.30 | freat | I prefer doing RTL |
| 03:23.35 | dlynes | docelmo: like pilot and logo |
| 03:23.45 | ManxPower | FuriousGeorge, 8? File a bug report if you CANNOT get it down to even fewer lines |
| 03:23.58 | ManxPower | I don't claim to have ever been any GOOD at programming. |
| 03:24.15 | FuriousGeorge | ManxPower: i was planning to, but i wanted some input as to whether it is really a bug or not |
| 03:24.29 | dlynes | I'm good at programming, but I'm kinda sick of programming after being forced to use java for four years |
| 03:24.51 | lokkju | heh |
| 03:25.08 | lokkju | try visual basic... |
| 03:25.12 | lokkju | alost worse |
| 03:25.15 | dlynes | I have |
| 03:25.19 | freat | ech |
| 03:25.25 | dlynes | I was contracted to convert a vb gui into C++ using MFC |
| 03:25.26 | file | I was forced to use visual basic for a short period of time in one of my classes |
| 03:25.38 | dlynes | that's the only time i've used vb |
| 03:25.46 | dlynes | so i never actually had to program in it...just read it :) |
| 03:26.19 | dlynes | but, ymmv which is worse...vb or mfc |
| 03:26.21 | lokkju | actually, probably the worst thing I have worked in, besides some pascal, cobol, and the like (the out there languages) is VFP (Visual Fox Pro) - my god *shivers* how horrible |
| 03:26.38 | lokkju | and lisp isn't much fun either... |
| 03:26.40 | freat | hey, can anyone recommend some kind of conference call mgmt stuff for asterisk? |
| 03:26.52 | lokkju | List in Stupid Parenthesis |
| 03:26.53 | dlynes | personally, i think vb might actually be marginally better than mfc |
| 03:27.08 | dlynes | freat: check out the MeetMe application |
| 03:27.09 | freat | even if it's just a bunch of config entries... I don't want to be putting a web server etc on the asterisk boxes |
| 03:27.10 | lokkju | dlynes, easier, at any rate |
| 03:27.28 | freat | dlynes: you mean, just meetme.conf? or something different? |
| 03:27.46 | dlynes | freat: correct, and the meetme() dialplan application |
| 03:28.09 | dlynes | freat: or you wanted a web gui to be able read/write meetme.conf? |
| 03:28.20 | freat | dlynes: meetme does what I need it to... was just wondering if anyone knew of some prebuilt extensions etc to make mgmt easy... I hunt around. no web gui! ;) |
| 03:28.31 | lokkju | hmm |
| 03:28.34 | dlynes | ah |
| 03:28.43 | dlynes | freat: seems to me it would be quite easy to write in php |
| 03:28.46 | ManxPower | fax2email.pl: http://pastebin.ca/49853 |
| 03:28.56 | lokkju | I was looking into building a xml-rpc service for scheduling conferences, but waiting for someone to fund me |
| 03:29.10 | dlynes | ManxPower: that's the one you were going to post to the wiki? |
| 03:29.13 | freat | dlynes: well, I'm OK w/ asterisk-java stuff so maybe I could do it there, and have the mgmt piece separate |
| 03:29.25 | ManxPower | dlynes, *nod* |
| 03:29.33 | dlynes | ManxPower: ah...thanks |
| 03:29.33 | ManxPower | With dialplan examples as well. |
| 03:30.03 | dlynes | ManxPower: I had some other faxemail/emailfax code that was working |
| 03:30.08 | dlynes | ManxPower: just not the way i wanted it to |
| 03:30.14 | freat | dlynes: I made an app recently for a call center. uses instant messaging to relay agent login info and call info between the web browser and asterisk |
| 03:30.24 | dlynes | ManxPower: My biggest problem was getting faxxing to stop konking out after the second page |
| 03:31.10 | dlynes | freat: cool...you could also use winpopup to do that, too, and then you don't need an instant messaging client |
| 03:31.16 | ManxPower | dlynes, spandsp has come a LONG way in the past 6 months |
| 03:31.28 | dlynes | freat: assuming all your clients are using windows machines |
| 03:31.33 | freat | dlynes: it's distributed... the agents can login from starbucks if they want |
| 03:31.46 | dlynes | freat: ah...then winpopup wouldn't work :) |
| 03:31.59 | freat | if it did... would be serious security problem hehe |
| 03:32.05 | dlynes | lol |
| 03:32.07 | dlynes | no kidding :) |
| 03:32.50 | freat | dlynes: yeah it's a custom applet actually that is in a frame in the browser app. the applet uses our IM server (lotus sametime) as a kind of transport for relaying data to / from asterisk |
| 03:33.13 | freat | dlynes: when they answer the phone, the call data gets pushed down into their browser (CID, queue time) |
| 03:33.48 | dlynes | cool |
| 03:33.55 | dlynes | i've been thinking about writing something similar |
| 03:33.58 | *** join/#asterisk nain (n=nain@202.59.90.182) |
| 03:34.03 | nain | Hi Every body |
| 03:34.07 | dlynes | an expert system that brings up all the customer info when a call comes in |
| 03:34.20 | dlynes | so you can see all their data, including all notes ever written about the client |
| 03:34.24 | nain | I am getting Bandwidth problem with g729 codec |
| 03:34.31 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 03:34.44 | ManxPower | nain, how many licenses do you have? |
| 03:34.46 | freat | basic architecture is a server side app that logs into a 'place' on the IM server and logs into asterisk. all clients connect to the same 'place' as well, and can thus relay info around. |
| 03:34.58 | nain | I got 10 licence |
| 03:35.41 | freat | nain: when you say bandwidth problem... is it really bandwidth? or is it choppy audio maybe resulting from other traffic? |
| 03:35.47 | nain | but the problem is that instead of 8 kbps, it is taking almost 30Kbps |
| 03:35.55 | freat | nain: TCP overhead |
| 03:36.02 | freat | nain: I mean IP |
| 03:36.36 | freat | nain: while the audio portion may be 8 kbps, the rest of the packet takes up a lot of space |
| 03:36.47 | nain | Actually the scenerio is that client is using xpro or eyebeam with g729 codec and they are on local net behind the nat and it's almost taking 30kbps per channel |
| 03:37.02 | LostFrog | And the packets are small which increases the overhead %. |
| 03:37.10 | nain | Then how to get rid of , rest of packet overhead |
| 03:37.18 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:37.35 | ManxPower | nain, you can increase the amount of audio in each packet. |
| 03:37.57 | ManxPower | That can cause problems with things like Meetme, but for most stuff it should decrease bandwidth usage at the cost of some latency |
| 03:38.04 | nain | How, to increse the audio in asterisk setup or at client side that is using X-Pro or eyebeam |
| 03:38.14 | *** part/#asterisk cced (n=dev2003@222.33.36.205) |
| 03:38.18 | freat | ManxPower: yes, so each packet causes even a larger drop in audio if lost... ;) |
| 03:38.33 | ManxPower | freat, modern networks don't drop many packets |
| 03:38.44 | nain | So What is the solution ? |
| 03:38.45 | freat | ahh |
| 03:38.51 | ManxPower | nain, there as some discussion about this on the asterisk-users or asterisk-dev mailing list |
| 03:38.53 | ManxPower | ~mailinglist |
| 03:38.54 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 03:39.15 | ManxPower | You change some small bit of the source code, recompile. |
| 03:39.21 | nain | Ofcourse i will also go through but i need quick fix for it |
| 03:39.23 | freat | yeah nain your numbers are right on... http://voip-info.org/wiki/index.php?page=Bandwidth+consumption |
| 03:39.39 | nain | bcz it is just eating my bandwidth and searching might take some time |
| 03:39.42 | ManxPower | The problem is not that it's using 30bkps, it's that you think that 30kbps is a lot of bandwidth in VoIP |
| 03:40.01 | freat | nain: can you get more bandwidth? That's the best solution really |
| 03:40.25 | nain | I got almost 512 kbps for 10 channel but again voice is choppy some time |
| 03:40.39 | ManxPower | nain, also remember that 30bkps could go over a 56K modem. |
| 03:40.41 | freat | nain: are you QoS'd at all? sharing the link w/ other apps? |
| 03:41.23 | nain | freat: like ?? |
| 03:41.43 | freat | nain: if you are sharing the WAN-LAN link with say, web browsers, you will have problems if you don't have some bandwidth shaping / QoS going on |
| 03:42.10 | ManxPower | freat, that really depends on a lot of things. |
| 03:42.27 | FuriousGeorge | ManxPower: http://pastebin.ca/49857 |
| 03:42.33 | FuriousGeorge | thats as short as i can make it |
| 03:42.35 | ManxPower | nain, 1) you are not going to get a quick fix. |
| 03:42.48 | FuriousGeorge | that HAS to be a bug right? nested loops shouldnt act like that |
| 03:43.02 | nain | freat: Actually, what happened in my case, User is on LAN (Which is sharing common bandwidth), User is calling from X-Pro and ASterisk is on WAN Link? is it cause bandwidth increase or over head? |
| 03:43.20 | ManxPower | nain, 2) you should look into the self timing patches and the sip jitter buffer patches. both will increase call quality when bandwidth starts to be overused. |
| 03:43.24 | freat | nain: is asterisk off-site? |
| 03:43.57 | nain | freat: yes asterisk is on remote location and user is behind the NAT mean in LAN sharing Internet ... |
| 03:44.10 | freat | nain: so your users are at home or something? |
| 03:44.29 | nain | yes like that |
| 03:44.55 | nain | But actually all of 10 users are on same location and ASterisk is on other location , |
| 03:45.03 | freat | oh man... audio quality could be all sorts of things related to the users home network. their kid could be on bittorrent for example... |
| 03:45.20 | freat | nain: oh... ok so they are at an office or something |
| 03:45.22 | LostFrog | nain: why aren't you running * at both ends with IAX? |
| 03:45.24 | FuriousGeorge | no one wants to test that for me on their dialplan, right? http://pastebin.ca/49857 |
| 03:45.37 | FuriousGeorge | to see if its replicable |
| 03:46.04 | nain | freat: Actually it's kind of call center enviorment not a home, Agents are using X-Pro to dial asterisk box and all agents are on local network sharing 512 kbps bandwidth |
| 03:46.31 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 03:46.31 | nain | Freat: I have checked individual, almost each call is taking upto 30kbps |
| 03:46.54 | freat | nain: yeah so 30x10 is still within acceptible range really |
| 03:46.56 | nain | Freat: No other application is sharing bandwidth only when call start it takes almost more then 30kbps |
| 03:46.59 | [TK]D-Fender | nain : Yup, full UDP overhead for each RTP stream... |
| 03:47.16 | nain | Is there any way to reduce overhead |
| 03:47.30 | nain | i have tried to put client on Live IP instead of behind NAT, |
| 03:47.32 | bkw__ | jitter buffers won't handle bandwidth overuse issues |
| 03:47.45 | bkw__ | NEXT!!! |
| 03:47.52 | nain | again but same results |
| 03:48.00 | freat | nain: so they have a sepparate internet connection for web browsing etc? |
| 03:48.04 | bkw__ | jitter buffers are for inter packet arrival time issues |
| 03:48.32 | nain | No, They are not doing any web browsing and i have checked with DU Meter No other application is sharing bandwidth |
| 03:48.41 | freat | nain: ahh ok. |
| 03:49.02 | freat | nain: where is the asterisk box? |
| 03:49.21 | FuriousGeorge | bkw_ can i be next? |
| 03:49.52 | freat | nain: possible bandwidth issues there? I had a site that I found out was on a wireless link and that totally was hosing VoIP |
| 03:49.55 | nain | Asterisk is on Data Center and agents are on seperate location |
| 03:51.03 | dlynes | FuriousGeorge: endless loop |
| 03:51.22 | FuriousGeorge | dlynes: explain to my why it shouldnt terminate when i set count to 2 |
| 03:51.24 | freat | nain: are you sure you don't have bandwidth contention at your Data Center? |
| 03:51.30 | FuriousGeorge | i just JIMPED OUT OF THE INFINITE LOOP |
| 03:51.33 | FuriousGeorge | why should i go back in |
| 03:51.40 | freat | FuriousGeorge: infinite fun? |
| 03:51.41 | dlynes | FuriousGeorge: don't know...looking at it |
| 03:51.53 | *** join/#asterisk bmg505 (n=leon@dsl-146-14-238.telkomadsl.co.za) |
| 03:51.59 | FuriousGeorge | freat: perhaps |
| 03:52.01 | dlynes | FuriousGeorge: but i'm testing it here, on the oej branch of asterisk, and i'm getting an infinite loop |
| 03:52.06 | *** join/#asterisk [Mojo] (n=mojito@156-147-89-200.fibertel.com.ar) |
| 03:52.15 | FuriousGeorge | oh, i thought you were saying it SHOULD be an infinite loop |
| 03:52.23 | dlynes | no |
| 03:52.44 | dlynes | I'm saying it's an infinite loop...i haven't taken a look at the dialplan to see why it is |
| 03:52.47 | FuriousGeorge | im surprised no one in asterisk dev and almost no one here seems to give a shit |
| 03:53.05 | nain | freat: NO problem is not on ASterisk side. On Data Center i have enough bandwidth but problem is on user side. they cann't afford huge bandwidth while also using g729 codec |
| 03:53.23 | FuriousGeorge | dlynes: its 7 lines. can you see any reason other than its a bug |
| 03:53.26 | dlynes | ummmm |
| 03:53.33 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 03:53.46 | dlynes | What's the s,n mean? |
| 03:53.51 | FuriousGeorge | lol |
| 03:53.53 | dlynes | Does the 'n' have a special meaning? |
| 03:54.13 | FuriousGeorge | n is so that i dont have to hardcode the numbers |
| 03:54.16 | dlynes | ah |
| 03:54.17 | dlynes | ok |
| 03:54.19 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 03:54.21 | dlynes | Never used it |
| 03:54.24 | FuriousGeorge | so if i make a change i dont have to go put in 2,3,4,5 |
| 03:54.27 | FuriousGeorge | all over again |
| 03:54.29 | FuriousGeorge | its new in 1.2 |
| 03:55.10 | nain | Also is there any suggested configration on asterisk side to force client to send number of frames per packet etc.... so by increasing little bit latency they can save bandwidth |
| 03:57.25 | FuriousGeorge | dlynes: also notice how the goto(exit) refers to a label i put on one of the n extensions |
| 03:57.28 | FuriousGeorge | pretty handy |
| 03:57.49 | dlynes | I know why |
| 03:57.55 | dlynes | When you goto out of the loop |
| 03:58.02 | dlynes | It's executing the endwhile following it |
| 03:58.06 | dlynes | Unless you have named loops |
| 03:58.15 | dlynes | It'll just go back to the last declared while loop |
| 03:58.18 | FuriousGeorge | it think the first endwhile it sees belongs to the child loop where it belongs to the parent loop |
| 03:58.25 | FuriousGeorge | names loops? |
| 03:58.33 | dlynes | asterisk doesn't support named loops |
| 03:58.40 | dlynes | therefore it's a bug in your dialplan |
| 03:58.47 | LostFrog | I named my loop bob. |
| 03:58.53 | FuriousGeorge | get the hell out of here :) |
| 03:58.54 | dlynes | Until asterisk supports named loops, you won't be able to loop that way |
| 03:59.16 | FuriousGeorge | what the heck to they expect me to do when all they give me is a while and a goto |
| 03:59.20 | FuriousGeorge | and a gotoif |
| 03:59.32 | ManxPower | dlynes, look at an extensions.conf.sample in 1.2 sometime |
| 03:59.58 | dlynes | ManxPower: ok |
| 04:00.29 | dlynes | FuriousGeorge: I was talking more like for i = 1 to 2\nprint "hello world"\nnext i |
| 04:00.36 | dlynes | FuriousGeorge: the loop in that case is called i |
| 04:00.46 | FuriousGeorge | dlynes i see what you mean |
| 04:01.21 | FuriousGeorge | otoh, this is very poor behavior if its not a bug, and i could swear ive used other nested whiles and jumped out of them |
| 04:01.39 | FuriousGeorge | or jumped from child to parent and not had this issue |
| 04:02.09 | dlynes | from a programming standpoint, it would seem like a really strange way to write it, and probably very difficult to debug, too |
| 04:02.33 | FuriousGeorge | dlynes: that whole school of never using goto's does not apply to the asterisk dialplan |
| 04:02.41 | FuriousGeorge | its not complex enough to work around it |
| 04:02.49 | FuriousGeorge | goto's make sense sometimes, even in loops, so we use em |
| 04:03.00 | FuriousGeorge | dlynes have we had this conversation before? |
| 04:03.07 | ManxPower | FuriousGeorge, ACTUALLY, with while, gosub, etc, gotos are needed less and less |
| 04:03.11 | dlynes | FuriousGeorge: no...i mean of jumping out of a whiel loop, hitting an endwhile, and expecting asterisk to know that you want the endwhile to apply to the first while |
| 04:03.26 | FuriousGeorge | ManxPower: true, but still needed sometimes |
| 04:03.39 | ManxPower | I use them all the time. |
| 04:03.45 | FuriousGeorge | thank you ManxPower |
| 04:03.50 | ManxPower | I call it "stone skipping". |
| 04:04.13 | *** join/#asterisk ivanfm (n=ivanfm@201-1-167-207.dsl.telesp.net.br) |
| 04:04.17 | FuriousGeorge | ManxPower: what do you think about all this? should i file this under bug or feature request |
| 04:04.19 | dlynes | But, still you must admit asterisk doesn't have the wherewithall to know which while he's referring to, so it assumes the last one |
| 04:04.22 | ManxPower | anything that is complex enough where stones kipping is not good should be an agi written in a real language. |
| 04:04.50 | ManxPower | dlynes, don't count on it. They did a lot of cool stuff in order to have AEL |
| 04:05.08 | dlynes | dlynes: ah...so why not just write it in ael thne? |
| 04:05.15 | dlynes | erm manx |
| 04:05.28 | ManxPower | dlynes, well, that is what FuriousGeorge *should* be doing. |
| 04:05.44 | FuriousGeorge | in the near future |
| 04:05.48 | ManxPower | or at least write it as an AGI in whatever language he wants |
| 04:05.53 | FuriousGeorge | pascal? |
| 04:05.56 | LostFrog | lol |
| 04:05.58 | ManxPower | or heck, use res_perl or res_js |
| 04:05.59 | FuriousGeorge | its been about 7 years :) |
| 04:06.00 | dlynes | because personally i don't think the extensions.conf was ever written with the intention of doing what he's trying to do |
| 04:06.18 | FuriousGeorge | nested while loops? |
| 04:06.32 | ManxPower | dlynes, no, but I do some pretty twisted dialplan things too |
| 04:06.42 | FuriousGeorge | how else would one get out of a while(1) but with a goto |
| 04:06.43 | dlynes | FuriousGeorge: no...jumping out of a nested while loop and expecting asterisk to know which while to refer to when it hits an endwhile |
| 04:06.49 | dlynes | FuriousGeorge: break? |
| 04:06.59 | FuriousGeorge | show application break and tell me what it says |
| 04:07.20 | dlynes | FuriousGeorge: or set the variable to an exit condition and then goto(endwhile)? |
| 04:07.21 | ManxPower | Just remember, AEL is translated into standard extensions.conf stuff when it's loaded. |
| 04:07.34 | ManxPower | so anything you can do in AEL you can do in non-AEL extensions.conf |
| 04:07.45 | russellb | AEL is about to get a lot more solid |
| 04:07.58 | ManxPower | russellb, YAY! |
| 04:08.01 | russellb | we're merging AEL2 in the next week or so |
| 04:08.08 | dlynes | but when you're using while(1), that's never gonna happen |
| 04:08.12 | ManxPower | russellb, what does 2 give us? |
| 04:08.30 | russellb | ManxPower: it's mainly a much more robust implementation - using flex and bison. |
| 04:08.51 | russellb | ManxPower: though it does add hint support, and #include support |
| 04:08.58 | FuriousGeorge | dlynes: look im no pro, but if thats not a bug, it should be considered one |
| 04:08.59 | russellb | and a few new other keywords ... |
| 04:09.13 | FuriousGeorge | the damn parser knows what line number has what |
| 04:09.16 | dlynes | russellb: hint support? you mean hint support has been rewritten? |
| 04:09.28 | russellb | dlynes: no, the ability to specify hints in your AEL file |
| 04:09.33 | ManxPower | With a good AEL and self timing RTP and real RTP jitter buffer, Asterisk 1.4 may be what I had hoped 1.2 would be and I can become confident in Asterisk again 8--( |
| 04:09.48 | dlynes | russellb: oh...thought you meant for dialplan hints (i.e. BLF) |
| 04:09.50 | FuriousGeorge | im looking foreward to oej's metermaid |
| 04:09.57 | russellb | dlynes: that's what i'm talking about |
| 04:10.10 | dlynes | but i'm already using hints in 1.2 |
| 04:10.10 | russellb | dlynes: but with the current AEL, you have to specify them in extensions.conf |
| 04:10.13 | dlynes | ah |
| 04:10.25 | dlynes | yeah...i don't use ael yet |
| 04:10.27 | russellb | it's just a conveneience thing, being able to completely move over to AEL |
| 04:10.45 | russellb | and it will be much better at pointing out errors at load time |
| 04:10.50 | FuriousGeorge | russellb: what do you think about this: http://pastebin.ca/49857 bug or no? |
| 04:11.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:12.16 | russellb | i'm not sure that situation was ever thought about |
| 04:12.19 | ManxPower | I must admit that using a goto from inside a while is...hinky |
| 04:12.20 | russellb | that's a pretty odd corner case |
| 04:12.38 | russellb | i totally believe you, but i have a hard time calling it a bug |
| 04:12.57 | FuriousGeorge | eh |
| 04:13.07 | ManxPower | russellb, perhaps a lack of documentation saying "you cannot goto our of a while or for loop" |
| 04:13.19 | russellb | so ... you should probably just redo that inner loop |
| 04:13.21 | ManxPower | s/ou/out |
| 04:13.24 | ManxPower | s/our/out |
| 04:13.35 | russellb | such that you set a condition telling it to quit, instead of jumping out of it |
| 04:13.53 | ManxPower | FuriousGeorge, I never use while loops 8-) |
| 04:14.06 | FuriousGeorge | ManxPower: but there is no for unless i use AEL right? |
| 04:14.09 | ManxPower | That's new in 1.2 and I'm more of a 1.0 person |
| 04:14.39 | dlynes | russellb: is there such a thing as a break statement in asterisk? |
| 04:14.50 | FuriousGeorge | dlynes again i think only in AEL |
| 04:14.55 | russellb | FuriousGeorge: you can file a bug on it if you want, but the most we'll probably do is add a note to the docs saying "don't do this." |
| 04:15.00 | dlynes | FuriousGeorge: then that's what you should be using :) |
| 04:15.14 | FuriousGeorge | russellb: fair enough, what if i call it trivial |
| 04:15.18 | file | you know, this isn't automated - I type out something witty every time my nick is mentioned |
| 04:15.39 | FuriousGeorge | file i just picked up on the pattern |
| 04:15.49 | file | russellb: 89 |
| 04:15.52 | russellb | FuriousGeorge: sure. just put it as a request that the documentation be updated to reflect this case |
| 04:15.53 | ManxPower | file, I suggest random lines from the Kama Sutra, in the original language |
| 04:15.55 | russellb | file: nice |
| 04:16.05 | file | I'm managed to gain 3 though O.o |
| 04:16.11 | file | er I've |
| 04:16.31 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 04:16.43 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 04:16.51 | russellb | FuriousGeorge: I can't think of *any* way to fix it, actually |
| 04:17.43 | russellb | it can't do some analysis on the extension to find it's matching endwhile ahead of time, because in theory, extensions can change in the middle of executing |
| 04:17.46 | russellb | that's totally valid to happen |
| 04:18.04 | ManxPower | FuriousGeorge, http://pastebin.ca/49864 |
| 04:20.28 | FuriousGeorge | http://bugs.digium.com/view.php?id=6994 |
| 04:20.42 | ManxPower | FuriousGeorge, or perhaps better.... http://pastebin.ca/49865 |
| 04:21.26 | Corydon76-home | russellb: actually, it CAN find its matching EndWhile |
| 04:21.32 | Corydon76-home | russellb: it already does |
| 04:21.57 | FuriousGeorge | ManxPower: i follow your logic but im not sure how this applies to me |
| 04:22.25 | ManxPower | You can simulate a loop using GotoIFs and SetVars |
| 04:22.38 | FuriousGeorge | Corydon76-home: i could swear i've "wentto()" out of a nested while into the parent in other stuff ive tinkered with |
| 04:22.45 | ManxPower | It's The Old Fashioned Way, Before That Newfangled 1.2 Was Around |
| 04:22.49 | Corydon76-home | russellb: It's how we made While/Endwhile act like a While loop instead of a Do/While loop |
| 04:22.56 | FuriousGeorge | ManxPower: oh yeah :) |
| 04:23.10 | FuriousGeorge | ManxPower: i didnt start doing anything too serious with the dialplan till post 1.2 |
| 04:23.59 | Corydon76-home | What I truly love are people who directly message me out of channel, even though I've never met them and I don't know who they are |
| 04:24.03 | ManxPower | FuriousGeorge, I finally started to use priority "n" when I have to renumber the priorities in an extension, as long as there are no goto/gotoifs there |
| 04:24.30 | file[laptop] | enough work for tonight |
| 04:24.39 | russellb | Corydon76-home: huh, well cool |
| 04:24.44 | ManxPower | Corydon-w, I frequently am /away What do I look like, your own Personal Tech Support Bitch? |
| 04:25.20 | Corydon76-home | Yeah, I'm getting that impression that someone thinks I'm going to help them |
| 04:25.32 | FuriousGeorge | well whatever |
| 04:25.42 | FuriousGeorge | ill just redo the loop, its not a big deal |
| 04:25.47 | FuriousGeorge | :D |
| 04:25.51 | FuriousGeorge | trivial to be exact |
| 04:25.54 | file[laptop] | russellb: !!! |
| 04:26.18 | FuriousGeorge | file |
| 04:26.28 | FuriousGeorge | i guess it really isnt automated |
| 04:26.36 | file[laptop] | nope |
| 04:26.37 | ManxPower | FuriousGeorge, don't worry, it will be only the first of many, many annoyances, disappointments, and moments of disbelief. |
| 04:26.46 | file[laptop] | I'm not at my desk either :P so no witty remark |
| 04:27.05 | dlynes | ManxPower: heh |
| 04:27.19 | FuriousGeorge | ManxPower: i spent a good 6 hours swearing there was a bug with goto when it turned out the random() on the next line was spitting out a similar complaint |
| 04:27.21 | russellb | file[laptop]: meep meep |
| 04:27.38 | FuriousGeorge | speaking of random(), how come its already listed as deprecated |
| 04:27.44 | FuriousGeorge | there is not func RAND in 1.2 |
| 04:27.54 | file[laptop] | russellb: poh, tate, oh |
| 04:28.33 | FuriousGeorge | before something is deprecated shouldnt there be a replacement? |
| 04:28.50 | Corydon76-home | Random will exist in 1.4 |
| 04:29.01 | FuriousGeorge | yeah ive heard its gonna be a func |
| 04:29.08 | FuriousGeorge | but it aint yet :) |
| 04:29.14 | Corydon76-home | No, the app will exist in 1.4 |
| 04:29.23 | Corydon76-home | The function will, as well |
| 04:30.03 | FuriousGeorge | does that mean my bug isnt gonna be fixed :( the one where feeding random an integer of 1 gives you a 0% chance |
| 04:30.22 | FuriousGeorge | or does that affect func rand in 1.4 too |
| 04:33.27 | Corydon76-home | Don't you pay attention to bugnotes? |
| 04:33.37 | Corydon76-home | That was fixed last night |
| 04:33.53 | FuriousGeorge | i must admit i dont |
| 04:34.14 | file[laptop] | you're like the vast majority or people |
| 04:34.15 | file[laptop] | er of |
| 04:34.16 | FuriousGeorge | i did notice when i filed this other thing that my issue from the other day was resolved |
| 04:34.38 | FuriousGeorge | file[laptop]: hey gimme some credit, that was my first bug |
| 04:34.58 | FuriousGeorge | usually i just think its a bug and its my failt |
| 04:34.59 | file[laptop] | well most people don't keep up with their bugs, so people ask for more details but it just sort of sits there |
| 04:35.01 | FuriousGeorge | *fault |
| 04:35.04 | Corydon76-home | The bugtracker sends you a friggin email. The least you could do is to read it |
| 04:36.06 | FuriousGeorge | Corydon76-home: back of buddy, its only been 24 hours and bugtracker isnt the only person who sends me emails, and i do other things besides * in the 18 hours im awake |
| 04:36.38 | asterboy | I just received an email that could change my life... |
| 04:36.42 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 04:36.42 | Corydon76-home | Pffft. Asterisk should run your life, the way it runs mine |
| 04:36.45 | asterboy | 500% cum volume |
| 04:37.00 | dlynes | new pr0n movie? |
| 04:37.02 | FuriousGeorge | well, now that i think about it, i am writing a dialplan to make my roomates clean the house |
| 04:37.05 | file[laptop] | another night of Asterisk After Hours... |
| 04:37.24 | Corydon76-home | asterboy: Wanna be the recipient of some bukkake? |
| 04:37.27 | linlin | Hey, having a little trouble with IAX connections, here is the error I get on the remote machine when it attempts to connact a call to my machine: |
| 04:37.34 | linlin | Apr 18 15:07:54 WARNING[1841]: chan_iax2.c:5553 socket_read: Call rejected by 67.184.230.198: No authority found |
| 04:37.36 | asterboy | ~bukkake |
| 04:37.43 | file[laptop] | linlin: configuration issue. |
| 04:37.45 | asterboy | is it 500% more? |
| 04:37.58 | linlin | thats what i thought |
| 04:38.07 | Corydon76-home | asterboy: google it |
| 04:38.09 | linlin | can you point me into the direction of where i can go to fix it |
| 04:38.09 | file[laptop] | linlin: lemme whip up a simple simple simple thing you can use |
| 04:38.12 | linlin | ok |
| 04:38.21 | linlin | appreciate it |
| 04:38.21 | asterboy | ewwww google...I'll clusty it though. |
| 04:38.32 | dlynes | wtf is clusty? |
| 04:38.52 | Corydon76-home | It's his Windows search engine |
| 04:39.14 | asterboy | ah |
| 04:39.17 | dlynes | ewww |
| 04:39.18 | asterboy | found an example: |
| 04:39.20 | asterboy | http://www.mushroomstamp.com/pictures/bukkakesm.gif |
| 04:39.53 | linlin | file[laptop]: I have this at the very bottom of my extensions.conf file, dunno if it is right, i was instructed to do so by the tech at the provider i'm tryinmg to get all this setu with |
| 04:39.57 | linlin | http://pastebin.ca/49868 |
| 04:40.02 | file[laptop] | linlin: http://pastebin.ca/49870 |
| 04:40.04 | file[laptop] | use that as a pase |
| 04:40.08 | file[laptop] | er base |
| 04:40.28 | *** join/#asterisk lorinc (n=ang@caracas-3844.adsl.interware.hu) |
| 04:40.45 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 04:40.51 | file[laptop] | if you want it dynamic... then make it host=dynamic and have the other side register using mypeer as the username and sekret as the password |
| 04:42.58 | FuriousGeorge | anyone ever sort an array with * :) |
| 04:43.14 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 04:43.17 | linlin | oops.... |
| 04:43.32 | linlin | (23:40:08) <@file[laptop]> er base << last line i cought |
| 04:43.32 | file[laptop] | linlin: get the... URL? |
| 04:43.46 | file[laptop] | if you want it dynamic... then make it host=dynamic and have the other side register using mypeer as the username and sekret as the password |
| 04:44.00 | asterboy | jbot, bukkake example here http://home.houston.rr.com/samrussell/sabrin.jpg |
| 04:44.07 | linlin | i dont have that kind of access to the machine sending outbound though |
| 04:44.19 | linlin | host claims its my problem that this isnt working |
| 04:44.46 | file[laptop] | linlin: which side is giving you that error? |
| 04:44.57 | file[laptop] | their side is sending back the rejection notice? |
| 04:45.19 | linlin | Apr 18 15:07:54 WARNING[1841]: chan_iax2.c:5553 socket_read: Call rejected by 67.184.230.198: No authority found |
| 04:45.25 | linlin | is what they see in theyre log files |
| 04:46.28 | linlin | if i turn on iax debug mode, i see something along the smae lnes |
| 04:46.28 | linlin | but nothing in the regular asterisk console |
| 04:46.28 | file[laptop] | they're trying to send a call to you? then that peer entry won't matter |
| 04:46.28 | file[laptop] | a peer in chan_iax2 is only used for outgoing calls |
| 04:46.28 | file[laptop] | unless you pasted what they have on their end |
| 04:46.59 | file[laptop] | which is entirely possible I suppose... |
| 04:46.59 | linlin | the pastebin is in my iax.conf file, the single line error is on theyre end |
| 04:47.28 | file[laptop] | there's nothing in iax.conf that will be able to authenticate them then, because that peer entry is NOT used for incoming authentication... it's only used to send calls out |
| 04:47.28 | Corydon76-home | FuriousGeorge: that's why you use quick sort |
| 04:47.47 | file[laptop] | so you probably need to setup a user entry with the same username, secret and a context |
| 04:48.01 | Corydon76-home | See qsort(3) |
| 04:48.08 | linlin | oh direct me towards how to do that? |
| 04:48.12 | linlin | they said something like that too |
| 04:48.27 | file[laptop] | [techcentric] |
| 04:48.29 | file[laptop] | type=user |
| 04:48.34 | file[laptop] | secret=your password |
| 04:48.38 | file[laptop] | context=where inbound calls should go |
| 04:48.51 | linlin | in extensons.conf ? |
| 04:48.53 | file[laptop] | iax.conf |
| 04:48.56 | linlin | okay |
| 04:49.09 | file[laptop] | replace the relevant fields with the needed info... |
| 04:49.21 | linlin | of course |
| 04:50.57 | linlin | wow, you're amazing |
| 04:51.00 | linlin | thankyou |
| 04:51.01 | linlin | :) |
| 04:51.21 | file[laptop] | meh |
| 04:51.36 | Qwell | AHHH! |
| 04:51.45 | freat | anyone know of a meetme control app that is _not_ via a web server installed on asterisk? |
| 04:51.47 | Qwell | MORON on the phone! |
| 04:51.50 | linlin | if i were of the female species i would most likely make it up to you with sexual favors, but i am not |
| 04:52.17 | Qwell | "320" "302?" "no, 320" "so 3020?" "NO! 320!" |
| 04:52.21 | file[laptop] | just support Asterisk in whatever way you can |
| 04:52.24 | FuriousGeorge | Corydon76-home: i remember quick sort, that is what im trying to do i think. isnt that when you make a = temp then a =b then b = temp |
| 04:52.35 | file[laptop] | including buying Digium hardware so we all get our pay checks, mmmkthxbye |
| 04:52.40 | file[laptop] | :D |
| 04:52.45 | freat | heh |
| 04:52.45 | Corydon76-home | FuriousGeorge: No |
| 04:52.52 | Corydon76-home | FuriousGeorge: that's bubble sort |
| 04:53.04 | freat | bubble sort! no! |
| 04:53.13 | linlin | i have an x100p ;) |
| 04:53.24 | file[laptop] | linlin: I'm sorry. |
| 04:53.30 | linlin | purchasing some iaxy's for friends soon-ish |
| 04:53.31 | linlin | haha |
| 04:53.36 | Qwell | Corydon76-home: It didn't release it before :p |
| 04:53.42 | file[laptop] | I just had to answer with that |
| 04:53.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:54.10 | linlin | works damn well for what i use it for for the 25$ it cost me |
| 04:54.11 | FuriousGeorge | Corydon76-home: oh yeah with the pivot |
| 04:54.45 | file[laptop] | I think I should get into my bed before I fall asleep |
| 04:55.26 | Ciber311 | boing |
| 04:56.34 | xachen | buy digium hardware? :O |
| 04:57.28 | FuriousGeorge | Corydon76-home: right now i wish i hadn't stopped paying attention when we got to recursion in my high school CS class |
| 04:59.26 | freat | FuriousGeorge: how much you gotta sort? |
| 04:59.47 | FuriousGeorge | it will vary but not much |
| 05:00.10 | FuriousGeorge | im using bubble sort right now, but its not working for some reason and i may as well learn how this recursive programming is supposed to work anyway |
| 05:01.13 | file[laptop] | recursively. |
| 05:01.47 | freat | FuriousGeorge: yeah bubble is ok for small sets, but much bigger and quicksort would be much better |
| 05:02.25 | Ciber311 | dlynes: ! |
| 05:02.34 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 05:02.38 | freat | also depends on if your stuff is kind of sorted to begin with or totally random |
| 05:03.05 | freat | if it's kind of sorted, I seem to remember heapsort might be better |
| 05:03.16 | FuriousGeorge | assume totally random and between 6 20 entries |
| 05:03.29 | FuriousGeorge | im not sure how im supposed to pick a good pivot |
| 05:03.31 | freat | FuriousGeorge: oh then bubble sort it |
| 05:03.49 | FuriousGeorge | i think im gonna have to stick with that, yeah |
| 05:03.50 | freat | FuriousGeorge: I seem to remember seeing something that at that small a set, bubble sort is actually faster |
| 05:04.02 | FuriousGeorge | im trying bubble sort now but its not bubble sorting and im not sure why |
| 05:04.21 | freat | code? |
| 05:04.37 | freat | pastebin it if you want I could look |
| 05:04.49 | FuriousGeorge | sure, i appreciate that |
| 05:06.39 | FuriousGeorge | http://pastebin.ca/49875 |
| 05:07.57 | FuriousGeorge | maybe if i spell my variable names right it might help |
| 05:08.38 | FuriousGeorge | i know i dont have to loop that many times, btw, but when it wasnt working i decided to test it looping the amount of times as the #s in the string |
| 05:09.31 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
| 05:09.33 | file[laptop] | Qwell: poke |
| 05:09.36 | freat | I notice count <= rows, but subcount < rows... is this what you want? |
| 05:10.10 | FuriousGeorge | freat: well shouldnt i be able to do rows-1 in at least one of those. or rows-count |
| 05:10.20 | FuriousGeorge | freat: i spelled weight wrong though |
| 05:10.25 | FuriousGeorge | my var names are misspelled :) |
| 05:10.32 | freat | uh oh |
| 05:10.51 | freat | ahh yeah MATEWIEGHT |
| 05:10.59 | FuriousGeorge | WOOT |
| 05:11.00 | FuriousGeorge | :) |
| 05:11.08 | FuriousGeorge | its working now :)) |
| 05:11.31 | FuriousGeorge | now to pick two winners and have a runoff if there is a tie :) |
| 05:11.46 | freat | cool... sure subcount shouldn't be <= rows though? since you start subcount at 1? |
| 05:12.27 | freat | well, heh glad I could help ;) |
| 05:12.41 | FuriousGeorge | cant i do while count <= rows-count and while subcount <= rows -count |
| 05:13.22 | FuriousGeorge | since i only have to sort to the end of the string - count as the last elements will be sorted alread |
| 05:13.28 | FuriousGeorge | y |
| 05:13.35 | Qwell | file[laptop]: pokepoke |
| 05:13.37 | freat | ahh ok |
| 05:13.56 | freat | yeah no need to iterate over the last element... gotcha |
| 05:14.13 | file[laptop] | Qwell: Astricon early bird registration! |
| 05:14.14 | FuriousGeorge | and on the second pass, last element - 2 |
| 05:14.17 | Qwell | file[laptop]: yeah |
| 05:14.26 | file[laptop] | Qwell: it's like omg |
| 05:15.33 | Qwell | file[laptop]: talked to my boss about it today - he wants to go...but we'll see |
| 05:15.44 | Qwell | I'll probably have a chat with him about it next week, when we travel |
| 05:15.58 | drray | where is astricon this year nova scotia? |
| 05:16.05 | Qwell | drray: many places |
| 05:16.10 | Qwell | but, Dallas, for the US one |
| 05:16.13 | freat | file[laptop]: cool yeah we have an office in Dallas so gonna try to coordinate that one |
| 05:16.24 | Qwell | heh, silly sokol-associates |
| 05:16.29 | FuriousGeorge | when is it? |
| 05:16.35 | FuriousGeorge | maybe ill go this year |
| 05:16.38 | file[laptop] | October 24th to 27th |
| 05:16.39 | Qwell | "AstriCon Europe Events :: Registration" |
| 05:16.42 | Qwell | https://www.sokol-associates.com/usa06/ |
| 05:16.57 | file[laptop] | tee he he |
| 05:17.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:18.19 | freat | yeah noticed that |
| 05:18.25 | freat | Dallas, Europe |
| 05:18.34 | Qwell | Texas, .pk |
| 05:18.52 | Qwell | "The Digium Party" - I am so going |
| 05:18.53 | file[laptop] | I hope I get to go since I've never gone |
| 05:18.58 | Qwell | file[laptop]: So are you! |
| 05:19.40 | FuriousGeorge | freat: i just figured it for the worst case scenario. its rows-1 for the parent loop, and rows- count for the child loop |
| 05:20.34 | FuriousGeorge | as in while count < rows and while subcount <= rows-count |
| 05:21.00 | freat | yeah so if the last element needs to move all the way to the front... |
| 05:21.03 | freat | makes sense |
| 05:22.18 | freat | hey is running mysql on an asterisk box much of a hit? I really don't want to load these boxes up with much. I could put another server in the rack and make it just for reporting stuff... |
| 05:22.31 | freat | I think I'm answering my own question |
| 05:24.48 | FuriousGeorge | freat: i heard someone once say that mysql (unlike other sqls) tries to use all of the system's resources by default |
| 05:24.51 | FuriousGeorge | but that hearsay |
| 05:24.58 | FuriousGeorge | ive never used it |
| 05:25.54 | freat | FuriousGeorge: thanks, yeah I think the way to go is w/ a separate reporting server. that way I could put http on it etc |
| 05:27.25 | FuriousGeorge | freat: btw, the nested loop is actually gonna be while subcount <= (rows-count) |
| 05:27.44 | FuriousGeorge | so that will trim off a bunch of cycles |
| 05:33.07 | yxa | anyone know of a callback script that when a user ringa once and hangup, and the clid is captured and * dials back to the user? |
| 05:34.06 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:36.39 | dlynes | If I was to write a wiki on setting up blf's to allow a six handset system to work like a traditional nortel keysystem, anyone have any idea where a suitable place would be to put it? |
| 05:36.45 | FuriousGeorge | yxa: i would think yu could do that with call files but you may need an agi script on top |
| 05:36.50 | FuriousGeorge | dont ask me how though cuz i dont know |
| 05:37.08 | FuriousGeorge | dlynes: i kinda tried it and you cant really do it |
| 05:37.16 | FuriousGeorge | first of all you can assign devstates to parking spots |
| 05:37.19 | dlynes | FuriousGeorge: blf? keysystem? |
| 05:37.29 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 05:37.34 | FuriousGeorge | you CANT assign devstates to parking spots |
| 05:37.47 | Qwell | actually |
| 05:37.49 | FuriousGeorge | and if you use meetme's you are gonna accidentally drop two people in the conference at some point |
| 05:37.53 | cced2 | lilo- [Global Notice] Hi all. Once again, we're still having problems with a kiddie attacking channels. You should definitely consider setting your channel to +R for now, if at all possible, until the problem is resolved. Apologies for the inconvenience. |
| 05:37.53 | cced2 | - |
| 05:37.56 | Qwell | I think oej has a patch to do that |
| 05:37.59 | dlynes | devstates? huh? who said anything about parking? |
| 05:38.04 | cced2 | how to set +r ? |
| 05:38.05 | Qwell | FuriousGeorge: ^^ |
| 05:38.07 | FuriousGeorge | and now that you mention it i think dlynes is using it |
| 05:38.10 | Qwell | cced2: don't worry about it |
| 05:38.28 | dlynes | i never mentioned anything about parking, furiousgeorge |
| 05:38.49 | FuriousGeorge | dlynes you did say you were using oel's branch |
| 05:39.03 | dlynes | Yeah, but only on my home test machine |
| 05:39.09 | dlynes | I'm not using it in any offices |
| 05:39.11 | FuriousGeorge | i assumed when you mentioned emulating a key system you wanted to be able to park a call and pick it up from another phone based on a blinkey light |
| 05:39.30 | dlynes | FuriousGeorge: correct, but i'm not parking it |
| 05:39.39 | FuriousGeorge | meetme's? |
| 05:39.41 | dlynes | FuriousGeorge: What you're talking about is BLA, not BLF |
| 05:39.43 | Qwell | FuriousGeorge: It would do exactly that |
| 05:39.56 | Qwell | woah dejavu... |
| 05:40.02 | Qwell | ~bla |
| 05:40.08 | jbot | foo |
| 05:40.08 | Qwell | dlynes: ? |
| 05:40.09 | FuriousGeorge | appearances |
| 05:40.20 | dlynes | bridged line appearance/shared line appearances |
| 05:40.36 | FuriousGeorge | refresh me on what f is again |
| 05:40.40 | Qwell | not quite shared line appearances |
| 05:40.41 | FuriousGeorge | is that sla? |
| 05:40.42 | dlynes | busy lamp field |
| 05:41.17 | dlynes | Qwell: yeah, but a lot closer to BLA than BLF is |
| 05:42.05 | dlynes | Besides...BLA is a Broadcom(?) PBX specific feature |
| 05:42.38 | dlynes | shared line appearance is another feature that does the same thing implemented by some other manufacturer, but submitted to the SIP committee to be a standard extension to SIP iirc |
| 05:43.28 | dlynes | My solution for implementing a lot of the keysystem functionality on phones that support BLF only uses BLF |
| 05:43.40 | dlynes | But it doesn't fully implement traditional keysystem functionality |
| 05:44.02 | dlynes | It does implement enough that most office workers that are used to traditional keysystems aren't left confused |
| 05:47.43 | FuriousGeorge | my dialplan hates my use of != and im not sure why exten => s,n,gotoif($[${MATEWEIGHT${COUNT}} != ${MATEWEIGHT$[${COUNT} + 1]}]?the_end) |
| 05:48.46 | FuriousGeorge | nm |
| 05:48.57 | FuriousGeorge | it was the line before that where i set count |
| 05:49.39 | cced2 | :) |
| 05:49.46 | *** join/#asterisk daurnimator (n=quae@unaffiliated/daurnimator) |
| 05:49.48 | daurnimator | yo |
| 05:49.49 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-32-50.w81-50.abo.wanadoo.fr) |
| 05:52.50 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:53.59 | dlynes | So nobody has any suggestions on where to post the blf wiki I was talking about, eh/ |
| 05:54.09 | Qwell | ~wikis |
| 05:54.13 | jbot | from memory, wikis is http://www.voip-info.org |
| 05:54.20 | kamileon | can someone help me to get my g729 to work with axvoice |
| 05:54.31 | kamileon | i have installed properly in * |
| 05:54.38 | Qwell | kamileon: Do you have a license for *? |
| 05:54.47 | kamileon | 0/0 encoders/decoders of 1 licensed channels are currently in use |
| 05:55.03 | dlynes | Qwell: Yeah, I know where the wikis are...just wondering where in the asterisk wiki would be a good place to post it...i.e. where people would look for it, and actually find it.. |
| 05:55.04 | daurnimator | hi all, i'd like to install asterisk soon |
| 05:55.06 | Qwell | kamileon: disallow=all and allow=g729 |
| 05:55.09 | kamileon | i need to buy one more though |
| 05:55.14 | Qwell | dlynes: link to it from something.. |
| 05:55.24 | kamileon | ok so i cant have allow=g729 and allow=ulaw in the same place? |
| 05:55.29 | daurnimator | the computer will be running knoppmyth (back-end) and i'm wondering how easy the install will be |
| 05:55.35 | Qwell | kamileon: You can, but ulaw will be used |
| 05:55.43 | kamileon | oooh i see |
| 05:55.58 | dlynes | Qwell: Yeah, I'm not asking how to post...just where to post :) |
| 05:56.02 | kamileon | and i have 2 channels from them so i need one more license is that correct? |
| 05:56.13 | dlynes | Qwell: i.e. to get a suggestion |
| 05:56.13 | Qwell | kamileon: yes, one license per transcoded channel |
| 05:56.15 | kamileon | what will happen if i try to use the second with only one license |
| 05:56.20 | daurnimator | whats this license for? |
| 05:56.37 | kamileon | guess ill be spending another $10 with mark real quick ;) |
| 05:57.37 | daurnimator | btw, do you need a license for g723.1? |
| 05:57.49 | boddy | hii I installed asterisk on gentoo and there is format_g729.so in "/usr/lib/asterisk/modules this mean is Ican use g729 on sip ? |
| 05:58.01 | boddy | ? |
| 05:58.38 | Qwell | boddy: If you buy licenses, or use it in passthrough mode |
| 05:58.53 | kamileon | <PROTECTED> |
| 05:58.53 | kamileon | <PROTECTED> |
| 05:59.05 | kamileon | is this because of the one license issue |
| 05:59.10 | kamileon | or the codec now working ? |
| 05:59.41 | boddy | how can I use it in passthrough mode ? |
| 06:00.27 | Qwell | boddy: By having a phone and provider that both speak g729, and not using anything like voicemail, or IVR stuff |
| 06:01.02 | dlynes | kamileon: i'm sure if you turn on sip debug, you'll see they're asking for one set of codecs, you're offering another, and the two sides don't agree |
| 06:01.47 | kamileon | ahh alright let me see |
| 06:02.37 | daurnimator | so |
| 06:02.37 | daurnimator | um |
| 06:02.48 | daurnimator | [15:54:56] <daurnimator> hi all, i'd like to install asterisk soon |
| 06:02.53 | daurnimator | [15:55:22] <daurnimator> the computer will be running knoppmyth (back-end) and i'm wondering how easy the install will be |
| 06:03.06 | Qwell | daurnimator: very. NEXT!!! |
| 06:03.11 | Qwell | </bkw__> |
| 06:03.24 | daurnimator | is there a good guide somewhere |
| 06:03.28 | Qwell | ~docs |
| 06:03.30 | jbot | hmm... 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 |
| 06:03.37 | daurnimator | btw, i'm a linux noob - be careful what you tell me |
| 06:03.54 | Qwell | daurnimator: That statement is just asking for trouble |
| 06:07.10 | daurnimator | .. |
| 06:07.17 | kamileon | im not sure how to read through the sip debug, http://pastebin.ca/49880 is the cli from my failed call |
| 06:07.20 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
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| 06:10.25 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 06:10.30 | Qwell | mog_home: ! |
| 06:10.40 | mog_home | Qwell! |
| 06:10.42 | Qwell | achika? |
| 06:11.51 | mog_home | womans lappy |
| 06:12.06 | Qwell | ahh |
| 06:12.25 | kamileon | christ |
| 06:13.23 | *** join/#asterisk _asr (i=asr@pimpbox.latency.net) |
| 06:13.26 | kamileon | hey mog_home how goes it |
| 06:17.10 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 06:17.36 | iceyp | anhyonw know how to call us numbers with voipjet? "IAX2/2513@voipjet/6508512351||rg") |
| 06:18.41 | *** join/#asterisk apardo (n=apardo@87.218.45.115) |
| 06:19.01 | iceyp | international would be 011xxxxx |
| 06:19.03 | Qwell | probably need a 1 |
| 06:19.11 | iceyp | k let me try |
| 06:19.59 | iceyp | perfect it rings |
| 06:20.53 | iceyp | great usual managed.com message mailbox thing |
| 06:20.56 | iceyp | and its full! |
| 06:20.58 | iceyp | useless bunch |
| 06:21.38 | kamileon | http://pastebin.ca/49880 <- pls someone help to read sip debug |
| 06:24.07 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 06:25.13 | Gamercjm | what is wrong |
| 06:25.43 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 06:25.55 | kamileon | i have installed one license for g729 trying to dial out through axvoice with it, when i did allow=g729 instead of allow=ulaw it will not connect now |
| 06:25.57 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:26.41 | Qwell | kamileon: Does axvoice allow g729? |
| 06:26.49 | Qwell | Is it enabled on your account? Do you have a sip debug? |
| 06:27.37 | kamileon | yes, im not sure if its enabled for me or not though, and the debug is that pastebin.ca/49880 |
| 06:28.06 | iceyp | mmmm when i try dial an 1800 number with fwd it tells me to signup with them first |
| 06:28.41 | *** part/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 06:29.16 | Qwell | kamileon: 488 not acceptable here |
| 06:29.29 | Qwell | That means it doesn't like your codec |
| 06:29.59 | kamileon | okay i guess i just need to get contact with them about it then, it says on the site they support it |
| 06:30.41 | iceyp | howcome this doesnt work, i get a message saying i need to signup with fwdexten => _*393[1-9]X.,3,Dial(IAX2/265744@fwd-gw@*18774678758,60,r) |
| 06:30.46 | kamileon | i would like a toll free incoming line from somewhere |
| 06:31.02 | boddy | do anybody know a one port digium E1 card price ? |
| 06:31.10 | Qwell | boddy: ~$600 |
| 06:31.28 | boddy | thanks Qwell |
| 06:31.37 | iceyp | <PROTECTED> |
| 06:31.39 | iceyp | woops |
| 06:31.46 | Qwell | iceyp: It's a channel mode :p |
| 06:31.51 | iceyp | ahh |
| 06:31.53 | iceyp | i getya |
| 06:31.54 | iceyp | :P |
| 06:32.04 | iceyp | any idea why i cant connect to fwd? |
| 06:32.24 | iceyp | Dial(IAX2/265744@fwd-gw@*18774678758,60,r) |
| 06:32.37 | iceyp | tells me its a FWD service and i need to signup first |
| 06:33.01 | iceyp | the lines actually like this: exten => _*393[1-9]X.,3,Dial(${FWDGW}/*${EXTEN:4},60,r) |
| 06:35.15 | yxa | If i have 3 * boxes in another location, how can I use Dial() to |
| 06:35.19 | yxa | load balance |
| 06:35.41 | litage | when dealing with headphones, do you want them to have a higher or lower impedance (in ohms)? |
| 06:35.52 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-248.claranet.co.uk) |
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| 06:38.59 | yxa | anyone? |
| 06:41.32 | dlynes | yxa: http://www.voip-info.org/wiki/view/Asterisk+at+large |
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| 06:44.08 | yxa | dlynes dont think that helps |
| 06:44.32 | dlynes | ah...no idea then...i haven't gotten big enough yet to start thinking about it |
| 06:45.02 | dlynes | But one way that could help with load balancing |
| 06:45.04 | yxa | all i want to know is instead of dialing a single box, can i group them up like zap channels and let it select a free server automatically? |
| 06:45.18 | dlynes | Is if you have say three ip addresses |
| 06:45.27 | dlynes | and give them all the same hostname in the dns server |
| 06:45.39 | dlynes | so that when you point your sip and iax devices at that hostname |
| 06:45.45 | dlynes | the dns will perform round robin on it |
| 06:45.55 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 06:46.36 | dlynes | not quite what you want, but it gives you a very crude solution |
| 06:46.49 | Altair256 | dlynes, that doesn't fix the fact that the asterisk servers themselves are not clustered (identical data) |
| 06:47.03 | dlynes | Altair256: true |
| 06:47.09 | Altair256 | yxa, I must have come in a little late... what was your question? |
| 06:47.23 | dlynes | yxa all i want to know is instead of dialing a single box, can i group them up like zap channels and let it select a free server automatically? |
| 06:47.43 | *** join/#asterisk oej (n=oej@203.178.156.27) |
| 06:47.54 | Altair256 | from the outside world, or from an internal sip client? |
| 06:48.42 | Altair256 | each internal sip client is going to need its own home server, but you couse use IAX trunks to link different physical ZAP trunks from different boxes, I believe |
| 06:49.55 | dlynes | yxa If i have 3 * boxes in another location, how can I use Dial() to |
| 06:49.55 | dlynes | yxa load balance |
| 06:50.03 | dlynes | that was the original question, altair |
| 06:50.15 | dlynes | but he seems to be afk now |
| 06:50.19 | Altair256 | thank you dlynes |
| 06:51.01 | Altair256 | I believe it would involve modifying the dialplan to route through different trunks |
| 06:51.09 | dlynes | So you're an old altair 6000 junky? |
| 06:51.21 | Altair256 | Your DNS solution would not really fix the problem I don't believe |
| 06:51.27 | FuriousGeorge | so when a macro terminates it continues down the dialplan from where you called it + 1, right |
| 06:51.33 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 06:51.38 | dlynes | Well, like I said...it was a quick solution, not a correct solution |
| 06:51.56 | Altair256 | once the asterisk box resolves the DNS name, it would cache the first response. Round robin DNS is only good for when multiple clients are querying the DNS server |
| 06:52.08 | Math` | any patches to make ooh323c compile again with the loader changes? |
| 06:52.18 | dlynes | Altair256: ah..yeah..i hadn't thought about that |
| 06:54.07 | yxa | Altair256 internal SIP client |
| 06:54.22 | Altair256 | SER frontend would be your answer them |
| 06:54.47 | Altair256 | Asterisk itself does not do that type of load balancing that I'm aware |
| 06:55.06 | Altair256 | using a pure Asterisk solution will require each SIP/user to have a home server |
| 06:55.22 | Altair256 | however, you could create a dialplan that uses another server's trunk to make outbound calls |
| 06:56.08 | Altair256 | yxa, I'm assuming you are from sip internal to zap external? And you have 3 servers in remote locations and you want some/certain calls to go through zap trunks at different locations? |
| 06:56.25 | yxa | Altair256 lets say i have 10 zap channels per machine in X country. If users want to dial that country and the lines run out on the 1st and 3rd machine, how do I get it to try the 2nd machine for example. |
| 06:56.50 | Altair256 | set a route/trunk rule |
| 06:57.05 | Altair256 | alternate routes |
| 06:58.30 | Altair256 | so for example, at location X you could have a route to use a location Y ZAP channel for calls to the Y calling area |
| 06:59.00 | Altair256 | then have a route after it that "falls back" to just using the X ZAP channel as a last resort (for calls matching a certain criteria) |
| 06:59.25 | Altair256 | does that match what you are wanting to do? |
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| 07:00.02 | Altair256 | really, a single route can have a "trunk sequence" |
| 07:01.16 | Altair256 | yxa, so at location X, you have ZAP/X channels, IAX/Y channels, IAX/Z channels all as trunks... first route checks to see if it is in Y zone, if so, attempt to use Y trunk fall back on X trunk |
| 07:01.24 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 07:01.38 | Altair256 | next route checks to see if call is to Z zone... if so, use Z trunk fall back on X trunk. |
| 07:01.55 | Altair256 | last route is your default route that just uses X trunk |
| 07:02.19 | Altair256 | you would use pattern matching on each route to decide if it was in the Y or Z zones |
| 07:02.45 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 07:02.50 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:03.37 | Snake-Eyes | Any one know how I can use g729 and fix the following error: Apr 19 17:07:52 WARNING[12240]: chan_sip.c:2527 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 07:03.37 | Snake-Eyes | Apr 19 17:07:52 WARNING[12240]: chan_sip.c:2527 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) |
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| 07:07.35 | Greek-Boy | is a licence required to use g729 with asterisk? |
| 07:08.03 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:08.56 | daurnimator | sometimes |
| 07:09.50 | mog_home | greekboy if you transcode you need it |
| 07:13.21 | Greek-Boy | but is it enforced? |
| 07:13.30 | mog_home | enforced? |
| 07:13.39 | daurnimator | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 07:13.46 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:13.49 | daurnimator | has instructions for educational use |
| 07:13.53 | mog_home | that is not a legal product |
| 07:14.09 | VeNoMouS_ | heh |
| 07:14.11 | mog_home | only legal way to do g729 in the civilized world is to buy a license from digium |
| 07:14.12 | daurnimator | read the notice on intel's site |
| 07:14.17 | mog_home | or not transcode ^_^ |
| 07:14.19 | VeNoMouS_ | u guys on full disclosure? |
| 07:14.23 | mog_home | ? |
| 07:14.25 | daurnimator | intel made it,, |
| 07:14.30 | daurnimator | they make the licenses |
| 07:14.35 | Snake-Eyes | it shouldnt be transcoding |
| 07:14.36 | mog_home | first off |
| 07:14.41 | mog_home | license isnt compattible |
| 07:14.43 | mog_home | secondly |
| 07:14.50 | Snake-Eyes | both end points have g729 |
| 07:14.51 | Altair256 | you can use it in pass-thru mode and not have to pay the license fee |
| 07:14.52 | mog_home | license doesnt cover use with something like asterisk |
| 07:14.56 | VeNoMouS_ | its like $10 a channel for g729 |
| 07:14.57 | mog_home | exactly altair |
| 07:15.08 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 07:15.11 | mog_home | thirdly like venomous said its cheap as crap |
| 07:16.11 | *** join/#asterisk Falle (n=falle@falle.se) |
| 07:16.20 | Snake-Eyes | the annoying thing is, I only get this error for g729 not for other codecs. Asterisk shouldn't be touching codec..... |
| 07:16.59 | mog_home | well force it to use g729 |
| 07:17.17 | VeNoMouS_ | maybe it just hates u |
| 07:17.26 | mog_home | hehe |
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| 07:17.34 | mog_home | g729 is for punks use speex ^_^ |
| 07:17.36 | Snake-Eyes | lol |
| 07:17.36 | yxa | Altair256 thanks for the info. i need to read up more on trunks cos I'm not too familiar |
| 07:17.37 | Greek-Boy | $10 per channel, not too bad |
| 07:17.47 | VeNoMouS_ | lol mog |
| 07:18.02 | VeNoMouS_ | Greek-Boy unless u need like 300 channels |
| 07:18.02 | Altair256 | as much as I believe some people don't like using it... |
| 07:18.02 | VeNoMouS_ | hehe |
| 07:18.07 | VeNoMouS_ | then it sucks |
| 07:18.17 | Altair256 | load up an Asterisk@Home 2.8 on a free test PC... |
| 07:18.40 | Snake-Eyes | mog_home, I have forced the phones to use g729, as for the VSP..... |
| 07:18.46 | mog_home | if you need 300 channels 3000 bucks probably isnt a big cost in the scheme of things |
| 07:18.49 | Altair256 | the dialplan is insanely complex and uses a MySQL database to store lots of the data... |
| 07:19.05 | Altair256 | but it will gives you an idea of how it "could" be done, yxa |
| 07:19.41 | Greek-Boy | mog_home, thats exactly my thoughts :) |
| 07:19.46 | VeNoMouS_ | or |
| 07:19.52 | VeNoMouS_ | u can do nice stuff like this |
| 07:19.58 | VeNoMouS_ | http://phone.turnstone.co.nz |
| 07:20.07 | VeNoMouS_ | ^^ from mysql cdr |
| 07:20.14 | VeNoMouS_ | yay for jpgraph! |
| 07:20.18 | Greek-Boy | we only have analog POTS lines connected to asterisk, should I still get a digital card to do codec processing? |
| 07:20.23 | FuriousGeorge | so i cant have any extension but s in a macro, that means i cant use background |
| 07:20.28 | Greek-Boy | i mean for g729 and other codecs |
| 07:20.28 | FuriousGeorge | can anyone verify |
| 07:20.30 | Snake-Eyes | 3000 channels = 300 calls at the same time, thats alot |
| 07:20.34 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 07:20.41 | Snake-Eyes | *300 |
| 07:20.59 | kamileon | how many channels are used for home use, 2-4 users |
| 07:21.06 | VeNoMouS_ | 2-4 |
| 07:21.12 | FuriousGeorge | meaning if i want to use background to get vars from a user i got to use a bunch of contexts and set global vars |
| 07:21.13 | Snake-Eyes | max 4 |
| 07:21.13 | VeNoMouS_ | >:P |
| 07:21.21 | VeNoMouS_ | 1 call == 1 channel |
| 07:21.27 | Snake-Eyes | yea |
| 07:21.41 | VeNoMouS_ | <Greek-Boy> we only have analog POTS lines connected to asterisk, should I still get a digital card to do codec processing? |
| 07:21.43 | VeNoMouS_ | x100p |
| 07:21.44 | Greek-Boy | there is no sure way to rule it out. I think the best way is to work out a ratio using an average of line use. |
| 07:21.53 | Greek-Boy | my ratios is 10:1 |
| 07:21.59 | Greek-Boy | for every 10 people I have one analog line |
| 07:22.00 | Snake-Eyes | VeNoMouS_, How often are all 4 ppl talking at the same time thou |
| 07:22.08 | mog_home | 1 call might be a channel it only counts when its in use |
| 07:22.10 | VeNoMouS_ | Snake-Eyes i dunno i dont live with him |
| 07:22.19 | kamileon | how do i go about debugging a fxs module that sometimes will fail calibration |
| 07:22.26 | mog_home | so like prompt transcode only takes a second |
| 07:22.30 | VeNoMouS_ | kamileon : use a hammer |
| 07:22.42 | kamileon | oh really? |
| 07:22.45 | VeNoMouS_ | yea |
| 07:22.50 | kamileon | just add it to the pile of crap digium stuff |
| 07:22.51 | VeNoMouS_ | it'll scare it itno submission |
| 07:22.57 | Snake-Eyes | kamileon, ya rly |
| 07:23.21 | kamileon | no im kidding its good stuff, ive been through alot |
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| 07:41.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:43.08 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net) |
| 07:43.42 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 07:44.07 | *** part/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net) |
| 07:51.43 | dlynes | VeNoMouS_: You know of any suppliers for x100p cards that don't require them shipping it to north america from hong kong? |
| 07:52.18 | daurnimator | <PROTECTED> |
| 07:53.11 | kamileon | dlynes: digium? |
| 07:54.00 | Shaun2222 | anybody know why when i have a macro running for a agent while why other calls in the queue will not ring other phones.. |
| 07:54.23 | Shaun2222 | the queue does attempt to send other calls to other agents until that agents macro finishes running... |
| 07:54.42 | dlynes | kamileon: digium doesn't make the x100p, or x101p anymore...you can only get clones |
| 07:54.43 | Altair256 | where does http://www.x100p.com/index.htm ship from? |
| 07:54.55 | dlynes | Altair256: hong kong...I just ordered two of those |
| 07:55.05 | Altair256 | did they arrive? |
| 07:55.12 | dlynes | Altair256: took about 1-1/2 weeks for it to arrive |
| 07:55.13 | daurnimator | i got 2 good ones from ebay |
| 07:55.17 | daurnimator | i'll find the seller |
| 07:55.31 | daurnimator | <PROTECTED> |
| 07:55.40 | dlynes | daurnimator: sure...that's where I got x100p from |
| 07:55.59 | dlynes | daurnimator: i just didn't want to take any chances, so I picked one that was recognizable from forums on voxilla |
| 07:56.03 | daurnimator | why do you want to know? |
| 07:56.20 | dlynes | daurnimator: so i can find a good source of compatible cards for asterisk |
| 07:56.31 | dlynes | most of our customers need fxo ports, not fxs |
| 07:56.41 | dlynes | and i'm getting tired of all the problems with sipura units |
| 07:56.46 | daurnimator | i dunno how many the seller has |
| 07:56.55 | daurnimator | but intel modems work |
| 07:57.01 | tasat | any app_conference users here? |
| 07:57.02 | daurnimator | just take out a resistor |
| 07:57.12 | dlynes | daurnimator: on the sipuras? |
| 07:57.21 | daurnimator | ?? |
| 07:57.21 | dlynes | daurnimator: oh...nvm |
| 07:57.33 | Altair256 | well... I'm going to call it a night. Very late here... 4am >.< |
| 07:57.38 | dlynes | How much are intel modems? |
| 07:57.38 | Altair256 | later guys |
| 07:57.42 | daurnimator | no idea |
| 07:57.43 | daurnimator | lol |
| 07:57.47 | orlok | intel modems? |
| 07:57.51 | Altair256 | like 9 bucks |
| 07:57.52 | orlok | intel or internal? |
| 07:57.52 | daurnimator | guessing $15 |
| 07:57.57 | dlynes | so i guess you don't know which resistor, eh? |
| 07:57.57 | Altair256 | $9-15 yeah |
| 07:57.59 | daurnimator | internal intel |
| 07:58.04 | orlok | depends if you want a shit software modem or a real hardwware one |
| 07:58.05 | Altair256 | anyhow... g'night :) |
| 07:58.09 | *** part/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 07:58.13 | dlynes | orlok: the intel one is a winmodem? |
| 07:58.20 | orlok | real hardware ones will Just Work |
| 07:58.22 | daurnimator | dlynes: its in a voip wiki somewhere, read it in bout august |
| 07:58.54 | dlynes | I have no interest in dealing with winmodems, anyways |
| 07:59.08 | dlynes | I had enough fun working with those with bbses |
| 07:59.15 | dlynes | I have no interest in continuing the nightmare |
| 07:59.39 | orlok | you were using bbses when winmodems were around? |
| 07:59.41 | daurnimator | a sellar that sells working x100p cards: http://search.ebay.com/_W0QQfgtpZ1QQfrppZ25QQsassZbiltmoreQ5finternational |
| 07:59.44 | dlynes | orlok: yeah |
| 07:59.47 | rkr245 | dlynes:good afternoon |
| 07:59.48 | orlok | scary |
| 07:59.53 | dlynes | heya rkr |
| 08:00.02 | dlynes | orlok: I was running a fidonet node at that time |
| 08:00.12 | daurnimator | dlynes: a sellar that sells working x100p cards: http://search.ebay.com/_W0QQfgtpZ1QQfrppZ25QQsassZbiltmoreQ5finternational |
| 08:00.35 | daurnimator | every week he puts another one for sale |
| 08:00.40 | rkr245 | dlynes:yesterday i waited for you but i dint find you |
| 08:00.43 | daurnimator | i think he has a huge stock of thenm |
| 08:00.54 | rkr245 | did you change your nick |
| 08:00.59 | dlynes | rkr245: nope |
| 08:01.06 | dlynes | rkr245: why were you waiting for me? |
| 08:01.23 | orlok | dlynes: i remember watching basic programs download at 300cps |
| 08:01.26 | orlok | thats like, reading speed |
| 08:01.31 | orlok | well, almost |
| 08:01.35 | dlynes | 300cps? |
| 08:01.41 | orlok | 300bps sorry |
| 08:01.42 | rkr245 | because i want to know from you how and what about ser |
| 08:01.49 | dlynes | first time i've ever seen anyone refer to modems and cps :) |
| 08:01.59 | dlynes | rkr245: I know nothing about ser |
| 08:02.01 | daurnimator | dlynes: did you have a look at that link? |
| 08:02.06 | orlok | and then the huge upgrade to 14.4k |
| 08:02.13 | dlynes | daurnimator: looking at it now |
| 08:02.29 | rkr245 | dlynes:can you tell any irc channel other than ser channel in thes server |
| 08:02.37 | daurnimator | a ha!: Quantity:19 available |
| 08:02.43 | dlynes | rkr245: huh? |
| 08:03.00 | rkr245 | channels |
| 08:03.10 | rkr245 | to discuss abiut ser |
| 08:03.17 | dlynes | beautiful...search and browse feature is currently unavailable |
| 08:03.24 | daurnimator | ? |
| 08:03.32 | dlynes | rkr245: try maybe #ser or #openser |
| 08:03.41 | rkr245 | o.k |
| 08:03.44 | dlynes | rkr245: if those don't work, try #ser or #openser on efnet |
| 08:03.55 | rkr245 | efnet.org? |
| 08:04.04 | dlynes | rkr245: yeah...or if you're in europe try undernet.org |
| 08:04.19 | dlynes | rkr245: undernet.org is notoriously unusable in north america, though |
| 08:04.41 | rkr245 | that is why i waited for you you got lot of knowledge on you tips |
| 08:05.37 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 08:09.33 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 08:13.17 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 08:17.34 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:17.35 | dlynes | rkr245: lol |
| 08:19.02 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 08:19.55 | rkr245 | dlynes:i dint find any channels on ser |
| 08:20.18 | implicit | #ser |
| 08:20.18 | dlynes | rkr245: what can i say? maybe they don't exist |
| 08:20.24 | implicit | :) |
| 08:20.24 | rkr245 | i wll try in undernet |
| 08:20.26 | dlynes | rkr245: Maybe you'll have to try the ser mailing list |
| 08:20.29 | implicit | oh channels in SER |
| 08:20.30 | implicit | there are none |
| 08:20.35 | implicit | its transaction stateful only |
| 08:20.40 | implicit | not dialog stateful |
| 08:20.46 | dlynes | implicit: No..he means an ser irc channel |
| 08:20.53 | implicit | #ser is one |
| 08:20.55 | implicit | i'm in there too |
| 08:21.03 | dlynes | rkr245: type /join #ser |
| 08:21.11 | implicit | if u have ser questions ask there :) |
| 08:21.22 | dlynes | implicit: yeah...he's saying he couldn't find it :) |
| 08:21.26 | rkr245 | there is ser in this freenode but nobody repies me |
| 08:21.31 | implicit | join again and ask me |
| 08:21.31 | dlynes | ah |
| 08:21.33 | implicit | i'm only around for a bit |
| 08:21.42 | rkr245 | o.k |
| 08:21.42 | implicit | but if its a quick question i can answer it for you |
| 08:24.23 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 08:24.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:27.03 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 08:30.32 | implicit | dlynes, where r u from |
| 08:31.07 | dlynes | vancouver, bc, canada |
| 08:31.42 | implicit | oh cool |
| 08:32.43 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 08:32.45 | mog_home | gnite |
| 08:32.48 | Bert- | hello there |
| 08:33.15 | Bert- | i'm working with someone using Asterisk |
| 08:33.40 | Bert- | we are sending him G723 codecs, but he says with Asterisk, he can only use GSM codecs |
| 08:34.15 | Bert- | is a way to dynamically 'translate' (on the Asterisk side) G723 to GSM plz ? |
| 08:34.36 | Bert- | i believe there is an tranlation API ? |
| 08:35.21 | {zombie} | asterisk can not transcode g723 |
| 08:35.35 | {zombie} | it only supports that codec in passthrough mode |
| 08:36.06 | {zombie} | http://voip-info.org/tiki-index.php?page=Asterisk+codecs |
| 08:40.10 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 08:41.12 | Bert- | {zombie} thanx ! |
| 08:41.33 | {zombie} | np |
| 08:42.24 | Bert- | but |
| 08:42.32 | Bert- | the way they want to use it |
| 08:42.34 | dlynes | Bert-: asterisk is also capable of g729 if the end user has purchased g729 licenses |
| 08:42.54 | Bert- | they don't want to buy any codecs license |
| 08:42.57 | Bert- | the way is : |
| 08:43.17 | Bert- | they only want to terminate VoIP calls to local pstn network |
| 08:43.30 | Bert- | so i suppose they can use G723 in this way |
| 08:43.36 | dlynes | no |
| 08:43.44 | Bert- | why ? |
| 08:43.50 | dlynes | in order to be able to do that, they'd need to transcode g723 |
| 08:43.57 | Bert- | maybe I don't understand well the 'passthrought' |
| 08:43.58 | dlynes | and asterisk isn't capable of that |
| 08:44.42 | {zombie} | eg if a user has a phone connected to asterisk, and it talks g723, they can then talk to you using g723 |
| 08:44.46 | dlynes | passthrough means client A can use g723, talks to asterisk, dials upstream B; upstream B talks g723; asterisk bridges the call - no transcoding is necessary |
| 08:44.56 | {zombie} | with asterisk in between, just passing the packets through without touching them |
| 08:45.06 | Bert- | okay |
| 08:45.41 | {zombie} | that way asterisk does not need to know how to encode / decode g723 |
| 08:46.21 | dlynes | Bert-: asterisk has a pretty big list of codecs it does support |
| 08:46.21 | dlynes | Bert-: ilbc, gsm, ulaw, alaw, g726-32, g729, speex, ... |
| 08:46.30 | dlynes | the only one of those that needs a license is g729 |
| 08:46.31 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:46.32 | boddy | thanks Qwell |
| 08:46.58 | boddy | Qwell: how can I install g729 codec ? |
| 08:47.27 | boddy | must I buy ? |
| 08:48.18 | frawd | hi all |
| 08:48.32 | dlynes | boddy: You need to buy licenses from Digium |
| 08:48.32 | frawd | just to know, what's your prefered way of dealing with dialplans: static extension.conf? realtime? ael? agi? what would you suggest for a system with lots of different permissions and services (being able to modify call permissions for any users at any time in a quite simple manner)? |
| 08:48.39 | dlynes | boddy: Go to the store at www.digium.com |
| 08:49.00 | dlynes | boddy: and then the instructions for installing your licenses after you've purchased them is at: http://www.digium.com/en/docs/G729/G729-README |
| 08:49.47 | dlynes | frawd: i don't think there is a preferred method; that is why there is three different ways of doing it; everyone has their own personal preference, depending on how they use asterisk |
| 08:50.13 | boddy | frawd:I am already planing to buy digium card |
| 08:50.37 | frawd | dlynes: how do you do it? :) |
| 08:50.40 | dlynes | boddy: the g729 license is separate from the digium card...digium doesn't include g729 codecs on the card |
| 08:50.49 | dlynes | frawd: I use extensions.conf |
| 08:50.51 | boddy | hýmm |
| 08:51.15 | dlynes | frawd: but i have it split up into a number of #included files |
| 08:51.35 | dlynes | frawd: it's all set up with a treelike directory structure under the main /etc/asterisk directory |
| 08:53.10 | *** join/#asterisk q_w_e (n=q_w_e@caiax6-092.dialup.optusnet.com.au) |
| 08:56.25 | dlynes | btw, is it just me, or is the asterisk 'make install' extremely ill-behaved when you want to install to a staging directory? |
| 08:57.43 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:57.54 | *** join/#asterisk RGi_ (i=RGi@62.97.247.44) |
| 08:57.57 | dlynes | morning, RoyK |
| 08:58.18 | RGi_ | Anyone here use Cisco ATA 18x to connect fax to asterisk ? |
| 09:00.14 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 09:00.15 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:01.12 | backblue | morning all |
| 09:01.39 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:01.43 | cced2 | o~. evening now |
| 09:03.02 | Bert- | <dlynes> Bert-: asterisk has a pretty big list of codecs it does support |
| 09:03.09 | *** join/#asterisk Bambr (n=Bambr@213-35-241-48-dsl.end.estpak.ee) |
| 09:03.27 | RoyK | dlynes: morning |
| 09:03.28 | Bert- | okay but what about codecs translation ?? from 'which to which' Asterisk is able to translate ? |
| 09:03.47 | Bert- | (without buying any license) |
| 09:04.20 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 09:04.42 | RGi_ | how does you guys solve fax issues.. outgoing faxes etc ? |
| 09:06.39 | RGi_ | do. |
| 09:07.00 | RGi_ | would be nice to get fax up and running on it.. :) |
| 09:08.08 | *** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net) |
| 09:10.06 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-200-93.telkomadsl.co.za) |
| 09:10.46 | backblue | i use iaxmodem |
| 09:11.58 | Bert- | well stupid question : i have a SIP phone and i would like to use it with GSM codecs. I have a list of codecs but i don't see 'GSM'. Does this codec has another name ?? (PCMU, PCMA, iLBC ?) |
| 09:12.23 | Bert- | of does my SIP phone just not deals with GSM codecs ... |
| 09:12.42 | Bert- | (grandstream bridge tone 100) |
| 09:13.52 | dlynes | Bert-: not all phones support gsm |
| 09:14.07 | Bert- | ok |
| 09:14.09 | dlynes | Bert-: grandstream budgetone 102 does, I think |
| 09:14.17 | dlynes | Bert-: actually very few phones support gsm |
| 09:14.18 | Bert- | so PCMU and PCMA are just other codecs |
| 09:14.28 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.44 | dlynes | Bert-: correct...pcmu is also known as g711 ulaw, or ulaw, or g711u |
| 09:14.52 | dlynes | Bert-: and similarly for pcma |
| 09:15.01 | Bert- | ok thanks |
| 09:15.54 | dlynes | Bert-: some sip phones support a very limited codec set such as the aastras...they only support g711 ulaw, g711 alaw, and g729a (nothing else) |
| 09:16.11 | backblue | you guys use misdn on quad bri, or zaphfc? |
| 09:16.15 | *** part/#asterisk q_w_e (n=q_w_e@caiax6-092.dialup.optusnet.com.au) |
| 09:16.25 | backblue | misdn it's terrible to install :( |
| 09:16.26 | dlynes | Bert-: but most support g723, g729, ulaw, alaw, and some also support g726 |
| 09:19.30 | FuriousGeorge | it finally works |
| 09:21.29 | FuriousGeorge | use asterisk to give your "weigh" your roommates and assign two of them to clean the house based on the weight ---> http://pastebin.ca/49889 |
| 09:21.41 | *** join/#asterisk BugKham (n=BugKham@125.24.4.70) |
| 09:21.43 | FuriousGeorge | *use asterisk to weight your roommates |
| 09:23.05 | FuriousGeorge | where weight is actually a number of asshole points |
| 09:30.35 | backblue | ? |
| 09:34.01 | daurnimator | lol |
| 09:34.14 | luke-jr_ | wtf? |
| 09:35.35 | luke-jr_ | you mean, Asterisk listens to conversations and recognizes things like "you get a jerk point!" and combines it with the caller Id to figure out who to increment points of? |
| 09:39.20 | BugKham | anyone knows of a good gui equivalent to CLI? |
| 09:39.43 | VxJasonxV | More specifically: AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 09:39.49 | VxJasonxV | not just for support though. check them out :P |
| 09:39.53 | BugKham | I'm using gastman now but the CLI part doesn't seem to work |
| 09:39.54 | dlynes | lol |
| 09:40.05 | dlynes | Bert-: correct |
| 09:40.41 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 09:42.24 | *** join/#asterisk stoffell (n=stoffell@54-63.240.81.adsl.skynet.be) |
| 09:44.02 | dlynes | morning, stoffell |
| 09:46.25 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 09:55.01 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 10:01.26 | tparcina | RGi, what problem do you have with Cisco ATA 186? |
| 10:02.23 | tparcina | hinting with cisco phones, does it work? |
| 10:13.01 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 10:13.13 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 10:15.06 | *** part/#asterisk daurnimator (n=quae@unaffiliated/daurnimator) |
| 10:17.42 | znoG | hey, anyone know of a decent ATA that provides 1x LAN and 1x WAN port and can bridge between them? |
| 10:18.03 | znoG | the grandstream series of ATAs are not exactly what I call decent but they do have a LAN and a WAN port |
| 10:18.35 | znoG | problem is they don't have a dialplan or anything, so you can't configure it to dial when it reaches X digits or change the type of tone if they dial a 0 (outside line) |
| 10:19.04 | znoG | ah SPA-2100 |
| 10:25.43 | *** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net) |
| 10:26.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:28.11 | rhowe | hmmm |
| 10:28.30 | rhowe | seems the office PBX doesn't transmit DTMF on calls it considers to be internal |
| 10:29.25 | *** join/#asterisk ToTo (n=cash-car@81.174.33.2) |
| 10:29.43 | ToTo | hi all |
| 10:30.22 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 10:30.23 | ToTo | i wold to use my hostname (no numerical ip address) in sip packet. is it possible? |
| 10:31.07 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 10:37.31 | *** join/#asterisk frood (n=dms@www.hal9000.org.uk) |
| 10:37.35 | frood | hey all |
| 10:38.02 | frood | is there anyone that may be able to help me with Cisco 7960s? |
| 10:39.30 | tparcina | frood, how can i help you? |
| 10:39.53 | frood | when I add a name and number to the address book, should it come up with the name i give when someone dials? |
| 10:46.06 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:53.26 | tparcina | frood, it doesn't work that way for me... |
| 10:53.37 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 10:53.46 | tparcina | have you acheve hiting on any of cisco phones? |
| 10:53.55 | tparcina | have you acheve hinting on any of cisco phones? |
| 11:06.47 | RGi_ | tparcina : well.. my fax is calling.. I can hear the oteher fax is taking the call and starts fax talk.:P but..my fax doesnt respond.. doesnt hear that the other fax is talking.. |
| 11:08.10 | boddy | must I use just digium card on asterisk are there another card works with asterisk correctly ? |
| 11:11.05 | boddy | ? |
| 11:14.00 | tparcina | RGi, sorry I can't help you. I'll try today, and if it works for me, I'll send you my configuration... |
| 11:14.55 | boddy | ? |
| 11:15.54 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:16.02 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 11:17.12 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 11:18.23 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 11:19.35 | marl | hi, can anyone help with with this? im wanting to have all calls coming in via my TDM400 try to call 2 numbers via another port on the TDM, i have the inbound route working ok to a digital assistant, i just cant seem to work out how to get that to route the calls on pressing option one to the mobile numbers, should this be done through a ring group or a queue? |
| 11:20.15 | marl | curently i have pressing option one inthe IVR transfer the calls to my SIP phone |
| 11:28.10 | hwt | what's the name of the module that provides the asterisk manager interface? |
| 11:28.12 | *** join/#asterisk saftsack (n=saftsack@p54A7E939.dip.t-dialin.net) |
| 11:28.15 | hwt | and can i manually just reload that module? |
| 11:28.16 | mut | well one of the 5 pregnant horses had their colt this mornin |
| 11:28.20 | hwt | to force it to reload manager.conf. |
| 11:33.29 | *** join/#asterisk frawd (n=francois@212.51.52.244) |
| 11:34.00 | claude81 | <PROTECTED> |
| 11:36.06 | RGi_ | tparcina : nice! :) |
| 11:37.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 11:39.19 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 11:39.21 | RGi_ | anyone got the fax to pdf system working ? |
| 11:41.19 | frawd | do incoming calls from PRI digital cards (E1/T1) go to the 's' extension in the context or to their destination extension? |
| 11:41.46 | frawd | (hello again :)) |
| 11:42.39 | boddy | hii client working properly via sip but which codec they are using ? I turn on sip debuging |
| 11:42.55 | boddy | but where I will look up ? |
| 11:44.26 | Delmar | Where Asterisk is in the middle ie. sip client -> asterisk server -> provider... can Asterisk inject audio into the conversation? |
| 11:44.35 | frawd | mboddy: in console, do a sip show peer [clientname] |
| 11:44.39 | Delmar | ie a notification or other. |
| 11:44.49 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 11:45.50 | boddy | frawd in codecs section 0x8000e |
| 11:47.55 | *** join/#asterisk shaZwaz (n=chatzill@203.81.196.167) |
| 11:50.00 | shaZwaz | hi |
| 11:50.10 | shaZwaz | ~see wasim |
| 11:50.14 | jbot | ACTION whispers to wasim "You didn't see anything...." |
| 11:50.15 | shaZwaz | ~seen wasim |
| 11:50.22 | jbot | wasim is currently on #asterisk (19h 26m 28s), last said: 'poor kiwis'. |
| 11:50.22 | marl | can anyone tell me were i can increase the gain on incoming calls on a TDM400? i need to turn the volume down to dial and back upto full to hear the person at the other end :( also, call recordings have my voice very loud and the other voice VERY quite, so i know this is an astrisk prob and not a SIP problem |
| 11:50.49 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 11:53.43 | *** join/#asterisk Aurs (n=aurs@a217-118-40-143.bluecom.no) |
| 11:57.37 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 11:58.18 | Delmar | marl, u want to mess with tx_gain and rx_gain |
| 11:58.44 | Delmar | marl, use fxotune to get the best values. |
| 11:58.53 | marl | ah thanks :) |
| 11:59.03 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:59.12 | Delmar | im about to go to bed but if u get stuck look on the wiki |
| 12:00.17 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 12:00.34 | tparcina | frawd, it goes to extension number that was diled. |
| 12:05.06 | stoffell | is this bad: Avoiding initial deadlock for ZAP/1-1 |
| 12:06.37 | *** join/#asterisk aSaDo (n=a@200.68.82.185) |
| 12:06.52 | aSaDo | hi everybody |
| 12:07.45 | aSaDo | i'm having an issue with asterisk 1.2.7 |
| 12:07.50 | aSaDo | and gnudialer |
| 12:08.05 | aSaDo | asterik crashes when gnudialer makes a call |
| 12:09.47 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:12.19 | SplasPood | aSaDo: eww, that sounds nasty... whats the console say before it dies? |
| 12:14.52 | stoffell | hi dlynes ;) (i know, a bit late :)) |
| 12:14.53 | aSaDo | SplasPood: give me a sec i give u a url so you can see it |
| 12:15.08 | dlynes | lol |
| 12:15.14 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:15.53 | aSaDo | SplasPood: take a loo kat this: http://pastebin.ca/49918 |
| 12:16.10 | aSaDo | SplasPood: that's what safe_asterisk says |
| 12:16.29 | SplasPood | hrm... |
| 12:16.34 | SplasPood | maybe it's a problem /w zaptel |
| 12:17.07 | aSaDo | i just don't know, ive already tried evetyth in my mind |
| 12:17.42 | aSaDo | zaptel version is 1.2.5 |
| 12:17.51 | aSaDo | centos |
| 12:18.00 | aSaDo | it's running in centos |
| 12:19.06 | aSaDo | if i do not run saef_asterisk, it says this: Ouch ... error while writing audio data: : Broken pipe |
| 12:19.12 | aSaDo | when crash |
| 12:19.18 | aSaDo | maybe it's zaptel |
| 12:19.26 | aSaDo | but i don't know how to fix it |
| 12:19.28 | aSaDo | any idea? |
| 12:20.05 | SplasPood | aSaDo: oh.. there was some CentOS + Zaptel issue.. bbut I thought it was a failure to compile |
| 12:20.33 | aSaDo | SplasPood: yeah, it's a failure with a datatype |
| 12:20.47 | SplasPood | aSaDo: Could be a misconfiguration /w zaptel's config too... but I'd say it really shouldn't CRASH... |
| 12:20.49 | aSaDo | SplasPood: but i ve already fix my kernel so i can compile it |
| 12:21.02 | SplasPood | aSaDo: I dunno man |
| 12:21.12 | SplasPood | Its consistent tho? Fails ALWAYS there? |
| 12:21.27 | aSaDo | SplasPood: yeah, always |
| 12:21.43 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:21.45 | aSaDo | SplasPood: i ve been testing it yesterday a mill times |
| 12:21.48 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:22.42 | SplasPood | Sadly I don't have much experience with actual zaptel hardware |
| 12:23.20 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:25.25 | [TK]D-Fender | I spent an hour beating away at CentOS for all its missing packages to allow me to compile *, Zaptel, and Wanpipe.... that was an "experience"... |
| 12:26.04 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 12:26.37 | aSaDo | SplasPood: maybe its an issue with zaptel and my FXO card |
| 12:27.15 | aSaDo | [TK]D-Fender: i ve managed to compile it, not much trouble, just a little mod to the kernel |
| 12:27.26 | SplasPood | aSaDo: You can turn on full logging in logger.conf and maybe you'll get some more info out of the log... |
| 12:27.34 | aSaDo | SplasPood: i have an X100P Clone |
| 12:27.40 | SplasPood | does 'dmesg' show anything |
| 12:27.41 | Aurs | how do i modify SIP headers from the dialplan? The docs claims that you can get/set sip headers with the function SIP_HEADER, but I got an error message that said that this function was for reading only |
| 12:27.56 | aSaDo | SplasPood: dmesg shows nothing |
| 12:28.02 | Aurs | i want to add a parameter to the TO header |
| 12:28.15 | SplasPood | Aurs: SIPAddHeader(Alert-Info:Intercom); |
| 12:29.17 | [TK]D-Fender | aSaDo : Yeah, its just that this box was missing at least a half a dozen critical packages, and had that kernel source bug to edit a few times as well as SMP issues... |
| 12:29.23 | Aurs | SplasPood: that app does not allow me to add a parameter to the To header, does it? |
| 12:30.14 | SplasPood | I dunno bout "adding" to the To: header, but it'll let you SET a header.. |
| 12:30.19 | Aurs | I don't want to add a header.. i want to modify |
| 12:30.42 | SplasPood | Well use the function to get the value, modify it, then use SIPAddHeader to write it back? |
| 12:31.15 | SplasPood | SIP_HEADER SIP_HEADER(<name>) Gets or sets the specified SIP header |
| 12:31.23 | SplasPood | hrm, looks like that's the "proper" way to set too |
| 12:31.29 | Aurs | it did not work |
| 12:31.45 | SplasPood | Set(SIP_HEADER(To)=My String) ? |
| 12:32.06 | Aurs | Apr 19 13:51:08 ERROR[7987] pbx.c: Function SIP_HEADER is read-only, it cannot b |
| 12:32.06 | Aurs | e written to |
| 12:32.25 | Aurs | yes, that is exactly what I tried |
| 12:32.27 | SplasPood | heh... I'd say 'typical', but Qwell will kick my ass :P |
| 12:32.32 | Aurs | hehe |
| 12:32.52 | Aurs | so what is the bug.. is it the function or the documentation? ;) |
| 12:32.53 | SplasPood | Aurs: what version of asterisk? |
| 12:33.10 | Aurs | 1.2.5 slightly modified |
| 12:33.13 | SplasPood | I'd say the docs since there's a message telling you it's read-only |
| 12:33.39 | SplasPood | it may have been fixed by 1.2.7 |
| 12:33.51 | aSaDo | SplasPood: i have sth here |
| 12:33.55 | Aurs | guess i could test on my 1.2.7.1 box here |
| 12:34.03 | aSaDo | SplasPood: Apr 19 09:30:23 WARNING[13506] chan_zap.c: Ignoring signalling |
| 12:34.03 | aSaDo | Apr 19 09:30:23 WARNING[13506] chan_zap.c: Ignoring rxwink |
| 12:34.04 | aSaDo | Apr 19 09:30:23 WARNING[13506] chan_zap.c: Ignoring hanguponpolarityswitch |
| 12:34.04 | aSaDo | Apr 19 09:30:23 WARNING[13506] chan_zap.c: Ignoring signalling |
| 12:34.06 | SplasPood | show functions |
| 12:34.32 | aSaDo | no much exp w/ zaptel, so don't know what's that |
| 12:34.37 | aSaDo | i'll google it |
| 12:34.38 | aSaDo | let 's |
| 12:34.43 | SplasPood | aSaDo: Same here, sorry... |
| 12:34.58 | aSaDo | SplasPood: ok tnx equally |
| 12:36.05 | Aurs | got the same error msg on 1.2.7.1 |
| 12:36.24 | Aurs | and the docs still says "gets or sets" |
| 12:36.30 | Aurs | I'll report a bug on this |
| 12:41.19 | Modcuts | I have just setup the aastra 480i and for some reason i can call out fine, But calling it goes straight to voicemail any ideas? |
| 12:41.53 | vgster | you are calling it from a different phone arent you |
| 12:43.50 | tparcina | hinting on cisco phones, has anybode done it? |
| 12:43.51 | Modcuts | hehe ofcourse calling the exten it's logged in on |
| 12:43.51 | Modcuts | but it seems to think it's busy |
| 12:43.51 | SplasPood | Aurs: cool, Qwell will be happy |
| 12:44.34 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 12:47.23 | [TK]D-Fender | Modcuts : Pastebin your sip.conf entry for it, the CLI output of "sip show peers", and your entire extensions.conf and we'll take a look...\ |
| 12:47.25 | [TK]D-Fender | ~pb |
| 12:47.31 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 12:49.36 | Modcuts | um....... ok let me see what i can do |
| 12:50.29 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 12:51.25 | Modcuts | every other exten running on gpx2000 phones like 20 of them work fine so it's not an asterisk setting, surely there is something wrong with aastra config something with the registration. |
| 12:53.02 | docelm0 | I have about 100 GXP2000's in my global network and they all work fine |
| 12:54.18 | Aurs | SplasPood: hehe, i'm sure he will be. rumors has it that there are way too few bugs in the tracker :P |
| 12:55.08 | SplasPood | Its good tho |
| 12:55.13 | SplasPood | if these things actually get fixed |
| 12:55.33 | SplasPood | ambiguity like that is one of the worst things, historically, about asterisk |
| 12:55.38 | [TK]D-Fender | And we're NOT talking about GXP's! We're talking about an Aastra with no details about how its set up.... "show me the money". |
| 12:56.32 | Modcuts | lol |
| 13:04.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.57 | docelm0 | oi |
| 13:12.43 | frawd | is there a way to change the SIP expire value on a user-basis or can it just be a [general] parameter (maxexpiry and defaultexpiry)? |
| 13:12.45 | *** join/#asterisk NSTech (n=NSTech@tokaimura.infotechsys.com) |
| 13:13.00 | NSTech | Does anyone here have any experience getting a TDM2400 to work in FreeBSD? |
| 13:13.14 | docelm0 | eeeek.. |
| 13:14.24 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 13:14.36 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 13:14.49 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 13:14.55 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 13:15.58 | koenvi | could anyone tell me the easiest way to left pad a variable in my dialplan with zeros? |
| 13:16.18 | docelm0 | koenvi what are you trying to make happen? |
| 13:17.05 | koenvi | I have a counter from 0 to 9999, but the field should always be 4 characters |
| 13:17.17 | bkw_ | _XXXX |
| 13:17.44 | bkw_ | koenvi, also you joined and didn't say Hi before asking your question. Thats awful rude |
| 13:17.45 | docelm0 | koenvi, pastebin.ca your dialplan and comment where your problem is |
| 13:18.00 | koenvi | bkw_ I thought that was only for extensions |
| 13:18.27 | frawd | (Hi!) |
| 13:18.48 | koenvi | complicated, involves multiple dialplans and an identifier traveling from one system to the other |
| 13:18.48 | bkw_ | Everyone online is so rude at times... like they forgot what their momma taught them. |
| 13:18.55 | frawd | (sorry for being rude...) |
| 13:19.01 | bkw_ | haha |
| 13:19.34 | blitzrage | koenvi: use the While() application in conjunction with the LEN() function |
| 13:19.34 | zoa | does somebody have something to save iax2 packets (sniffed) into a wav file |
| 13:19.36 | zoa | or gsm file |
| 13:19.38 | zoa | or whatever ? |
| 13:19.51 | bkw_ | zoa, not that I know of |
| 13:19.55 | zoa | k thnx |
| 13:20.01 | blitzrage | koenvi: While(${LEN(${my_var})} < 4) |
| 13:20.17 | bkw_ | blitzrage, that is uuuuuugly |
| 13:20.19 | bkw_ | :P |
| 13:20.19 | blitzrage | koenvi: Set(my_var=0${my_var}) |
| 13:20.27 | bkw_ | quick someone go write app snprintf |
| 13:20.28 | koenvi | While ... sounds good, I'll give it a try |
| 13:20.34 | blitzrage | bkw_: show me a better way |
| 13:20.37 | blitzrage | bkw_: in the dialplan |
| 13:20.55 | bkw_ | the dialplan in asterisk isn't a dialplan its a cluster |
| 13:20.56 | koenvi | bkw_ I was thinking of using printf, but how? |
| 13:21.08 | bkw_ | they should calling it scripting |
| 13:21.17 | blitzrage | bkw_: you're avoiding the question :) |
| 13:21.34 | blitzrage | koenvi: you could do it in an AGI |
| 13:21.36 | bkw_ | blitzrage, I haven't been keeping up with asterisk so chances are I would write something to do the task |
| 13:21.49 | blitzrage | bkw_: ok, that's what I thought :) |
| 13:22.16 | koenvi | oké, I know have 2 options, While and AGI |
| 13:22.19 | koenvi | thanks guys |
| 13:22.23 | blitzrage | np |
| 13:22.45 | AvoidingDeadlock | As you can see i'm avoiding deadlock |
| 13:23.03 | AvoidingDeadlock | haha |
| 13:23.05 | Deadlock | :D |
| 13:23.18 | AvoidingDeadlock | its a joke boi.. on how asterisk goes "Avoiding Deadlock" instead of finding and fixing the real problem |
| 13:23.42 | AvoidingDeadlock | lets not spend any time and fix it.. lets just use trylock and bomb if we can't lock |
| 13:24.06 | blitzrage | same ol' bkw_ |
| 13:24.13 | AvoidingDeadlock | haha |
| 13:24.17 | frawd | hHi again (thx momma), how could I catch dtmf from a user during a conversation (while Dial command is running)? |
| 13:24.53 | frawd | (like the tT option in dial cmd, but with any dtmf) |
| 13:25.21 | blitzrage | frawd: isn't there something to catch on a single digit to exit out and go to a context? then you could use something like background, or WaitExten() |
| 13:25.27 | blitzrage | frawd: show application Dial() |
| 13:26.04 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 13:26.11 | *** part/#asterisk koenvi (n=root@tech.ascom.be) |
| 13:28.11 | frawd | blitzrage: there is the t or T option to catch the pound key and do transfers, the h and H to catch star key, but I dont want to exit the conversation, but be able to catch a sequence of dtmf to transfer both parties to a meetme conference or something |
| 13:29.52 | bkw_ | haha |
| 13:29.55 | frawd | also the d option to catch one dtmf when the call is not answered |
| 13:29.57 | bkw_ | anthm did a patch to do just that |
| 13:30.09 | frawd | who? when? where? how? |
| 13:30.13 | frawd | cool!! |
| 13:30.20 | frawd | i want, i want! :p :p |
| 13:30.28 | bkw_ | you'll have to message him |
| 13:31.17 | *** join/#asterisk i_m_meen (i=i_m_meen@85.186.238.185) |
| 13:31.48 | Katty | mew. |
| 13:32.04 | aSaDo | SplasPood: issue solved :-D |
| 13:32.48 | *** part/#asterisk i_m_meen (i=i_m_meen@85.186.238.185) |
| 13:33.48 | bkw_ | we are still waiting on the last two patches we put on mantis to go in.. those have been up for a year today |
| 13:33.54 | zoa | splashpoo tss |
| 13:34.05 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:34.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:34.08 | zoa | what a name |
| 13:34.26 | NSTech | Hi, has anyone had any experience witb zaptel running on BSD? |
| 13:34.39 | anthm | hey zoa |
| 13:34.43 | frawd | hey, looking again at Dial() command, maybe I could use G(context^exten^pri) stuff with a cute agi to catch those dtmf during a call!! i'll try to study that if noone tells me that it's completely impossible |
| 13:35.57 | zoa | hi ho! |
| 13:35.59 | zoa | Ho Hi! |
| 13:36.17 | Hmmhesays | lets go |
| 13:37.05 | frawd | anyway, instead of the tTdhH options, there should be a unique option in Dial to catch DTMF and send them to a different context... anyone would use this feature? |
| 13:37.30 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 13:37.44 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:38.26 | bkw_ | frawd, we have two patches on the bug tracker that need to go in before we put anything else up |
| 13:38.36 | bkw_ | been there since Aug 05 |
| 13:38.36 | SplasPood | aSaDo: what was it? |
| 13:38.49 | bkw_ | 8 Months to get a patch in |
| 13:38.58 | bkw_ | gee that makes you want to develope don't it? |
| 13:39.14 | frawd | i would!ok |
| 13:39.28 | frawd | sed s/i would// |
| 13:39.58 | frawd | so long to get a patch in? |
| 13:40.50 | frawd | bkw: let's get together and fork asterisk with cute patches ;) |
| 13:40.55 | *** join/#asterisk Malthus (n=admin@uslec-66-255-41-2.cust.uslec.net) |
| 13:41.16 | frawd | forget it, i have stupid ideas |
| 13:41.34 | SplasPood | bkw: bug id? |
| 13:41.45 | Malthus | anyone used the vegastream 48 port SIP gateways? |
| 13:41.50 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 13:42.12 | koenvi | just another quick question |
| 13:42.34 | koenvi | do you think it's oké to use global var for system wide counters? |
| 13:42.57 | koenvi | or is there a risk to calls increment at the same time? |
| 13:43.09 | koenvi | two calls that is |
| 13:43.11 | anthm | he probably means 5161 and 5162 |
| 13:43.46 | koenvi | anthm, what are those? |
| 13:44.03 | anthm | rtp packetization and manager event hooks |
| 13:46.09 | anthm | 1 lets you send mutiple frames per RTP packet and the other lets you make internal modules that can consume manager events from inside the module so you can gateway them to say xmpp |
| 13:46.23 | Katty | hmm |
| 13:46.51 | anthm | but dunno just guessing *shrug* |
| 13:47.33 | anthm | I did however write the whole freeswitch while those patches have been "evaluated" so... =/ |
| 13:47.48 | *** join/#asterisk pengyong (n=lala@222.185.16.106) |
| 13:48.05 | anthm | i do have some more stuff |
| 13:48.08 | fenlander | what state is freeswitch in now? |
| 13:48.27 | anthm | most of the code resides in virginia |
| 13:48.32 | *** join/#asterisk crim3 (i=crim@66.207.72.229) |
| 13:48.48 | SplasPood | anthm: haha |
| 13:48.48 | [TK]D-Fender | koenvi : Vars only last as long as that call is in progress and only exist within the scop of THAT call. What are you trying to acheive? |
| 13:49.05 | anthm | jk, come ask over in the right channel dont want a flame war |
| 13:50.04 | fenlander | :) |
| 13:54.31 | Malthus | would anyone here trust vegastream for a 600 extension asterisk system? |
| 13:54.50 | Malthus | vegastream vs mediatrix vs audiocodes |
| 13:56.13 | *** join/#asterisk guigouz (n=guigouz@201-42-160-252.dsl.telesp.net.br) |
| 13:59.47 | *** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net) |
| 14:00.20 | guigouz | hello! I want to forward call from a conventional telephone line to a sip number, what do you recommend ? is there any simple hardware to do it ? |
| 14:00.45 | RoyK | an fxo interface like an x100p |
| 14:01.46 | guigouz | RoyK: do you know of any device that does it, so I don't have to use a computer ? |
| 14:02.00 | wasim | guigouz: an fxo/sip ata |
| 14:05.34 | anthm | lot's of linksys ones all locked in to vonage (creeps) |
| 14:05.54 | [TK]D-Fender | guigouz : Forward your analog line is a telco feature. Typically a * code in North America |
| 14:06.45 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:06.50 | Ariel_ | morning everyone |
| 14:06.54 | guigouz | [TK]D-Fender: I have 2 offices. |
| 14:07.22 | Hmmhesays | Morning Ariel_ |
| 14:07.29 | guigouz | [TK]D-Fender: I want to plug one internal extension into one of this converters, so people on one office can call the other one over the internet. |
| 14:07.37 | Ariel_ | Hmmhesays, morning |
| 14:07.49 | Hmmhesays | guigouz, gotta be an analog extension |
| 14:07.56 | guigouz | it is analog |
| 14:08.03 | guigouz | so i need to convert analog -> sip |
| 14:08.04 | Hmmhesays | fxo device |
| 14:08.17 | guigouz | what does fxo mean ? |
| 14:08.19 | guigouz | and fxs ? |
| 14:08.25 | [TK]D-Fender | guigouz : Ok, so * is not plugged into a real phone line, but rather into your existing PBX? |
| 14:08.39 | guigouz | [TK]D-Fender: yup. but it is analog as well. |
| 14:08.42 | [TK]D-Fender | Hmmhesays : No, you can use a line interface on a PBX for that as well. |
| 14:09.11 | [TK]D-Fender | guigouz : You you pick like "line 5" on your PBX and thats plugged into * on an analog extension port? |
| 14:09.18 | boddy | hii client working properly via sip but which codec they are using ? I turn on sip debuging |
| 14:09.28 | Hmmhesays | yeah but that is on the trunk side of the pbx |
| 14:09.31 | boddy | in peer section in codecs section 0x8000e |
| 14:09.34 | Hmmhesays | i think he's talking station side here |
| 14:09.40 | [TK]D-Fender | boddy : While a call is in progress do "sip show channels" and you'll know. |
| 14:09.46 | [TK]D-Fender | Hmmhesays : So am I.... |
| 14:10.00 | guigouz | [TK]D-Fender: on my pbx I have like 5 telco lines, and 16 extensions. when people dial "9" they get a real line |
| 14:10.16 | [TK]D-Fender | Hmmhesays : Just saying you can do it either way. I do it the reverse at my other plant just because thats what kind of port they had free and I run them off an ATA. |
| 14:10.22 | guigouz | [TK]D-Fender: i want to plug this converter on extension 10, for example. |
| 14:10.37 | guigouz | so people call "10" and it will be forwarded to a sip phone somewhere |
| 14:10.53 | boddy | showing ulaw |
| 14:10.56 | guigouz | also, from my sip phone, I'd like to be able to use a local line (dialing 9) |
| 14:11.01 | Hmmhesays | and in extension 10 you would normally have a phone plugged in? |
| 14:11.03 | [TK]D-Fender | guigouz : Ok, I follow you now. So what they'd have to do is dial ext 10, then have * answer and then send the call to the other side. |
| 14:11.08 | guigouz | Hmmhesays: yes. |
| 14:11.11 | Hmmhesays | fxo box |
| 14:11.36 | boddy | [TK]D-Fender:which one Will I use g723 or ulaw ? |
| 14:11.37 | guigouz | Hmmhesays: any brand you recommend ? |
| 14:11.49 | Hmmhesays | for a single fxo? spa-3000 will get you by |
| 14:12.27 | [TK]D-Fender | boddy : What do you mean "which will I use"? I don't know what it will pick, I've never seen your setup... |
| 14:12.48 | [TK]D-Fender | guigouz : Linksys SPA-3000. |
| 14:12.51 | boddy | which one is the more compres |
| 14:13.05 | Hmmhesays | 4 or more lines get i'd use a quintum as/ax series, just because i've found they operate with analog extensions on pbx's quite well |
| 14:13.05 | boddy | most benefit for me ? |
| 14:13.34 | [TK]D-Fender | boddy : G.723 = not really supported in *. GSM and G.729a are your real options for low-bandwidth usage |
| 14:13.44 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:14.25 | boddy | could I find free g729 codec ? |
| 14:14.37 | boddy | or I must buy from digium |
| 14:14.49 | nahirean | 729a is proprietary is it not? |
| 14:14.58 | bkw_ | its a standard |
| 14:15.05 | bkw_ | ref code is out there to build a codec with |
| 14:15.41 | mog_work | it is you need license for trancscoding boddy but pass through should work fine |
| 14:15.43 | bkw_ | but their are free g729 out there some guy builds with the intel code |
| 14:15.53 | bkw_ | not sure how legal that is |
| 14:16.13 | mog_work | its legal, in botswana |
| 14:16.18 | [TK]D-Fender | boddy : You need to pay for G729, GSM is free however. What are you planning on using it for? |
| 14:16.31 | bkw_ | its legal in the US also if its not used for commercial stuff |
| 14:16.33 | MikeJ[Laptop] | I;ve never been to botswana, is it nice |
| 14:16.35 | bkw_ | ie developers |
| 14:16.49 | boddy | client on internet connect asterisk via sip |
| 14:16.57 | bkw_ | I like ulaw |
| 14:16.59 | bkw_ | ulaw is nice |
| 14:17.01 | bkw_ | sounds great |
| 14:17.40 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 14:17.41 | OliverX | use here anyone misdn with a fritz card? |
| 14:17.48 | OliverX | i need an example :) |
| 14:17.54 | OliverX | for misdn.comf |
| 14:17.57 | OliverX | for misdn.conf |
| 14:18.03 | boddy | must I use just digium card on asterisk are there another card works with asterisk correctly ? |
| 14:18.21 | MikeJ[Laptop] | you don't "need" any cards to use asterisk. |
| 14:18.46 | OliverX | MikeJ[Laptop]: do you use misdn? |
| 14:18.47 | mog_work | ulaw for ever bkw_ |
| 14:18.49 | boddy | MikeJ[Laptop] asterisk will connect to meridian 1c via e1 |
| 14:19.08 | MikeJ[Laptop] | then you need an e1 for that |
| 14:19.23 | MikeJ[Laptop] | err.. an e1 card |
| 14:19.23 | russellb | mog_work: not after we get wideband support :-p |
| 14:19.53 | OliverX | no person here who use misdn? |
| 14:20.13 | MikeJ[Laptop] | raw pcm 64bit audio forever!! |
| 14:20.16 | MikeJ[Laptop] | :P |
| 14:20.16 | boddy | ? |
| 14:20.24 | OliverX | MikeJ[Laptop]? |
| 14:20.28 | mog_work | 16 bit ulaw forever russellb |
| 14:20.28 | MikeJ[Laptop] | ? |
| 14:21.21 | MikeJ[Laptop] | boddy, you need an e1 card, there are several different ones that are compatible, there is a list of asterisk compatible hardware on the wiki |
| 14:22.03 | mut | anyone know and telco resellers? |
| 14:22.10 | mut | lookin for a class5 telco switch |
| 14:22.16 | MikeJ[Laptop] | ebay |
| 14:22.26 | mut | i meant .. know |
| 14:22.31 | MikeJ[Laptop] | I've seen good stuff on there every once in a while |
| 14:22.32 | mut | not where to |
| 14:22.40 | MikeJ[Laptop] | :P |
| 14:22.53 | tzanger | rizzo's not around? |
| 14:23.01 | tzanger | I got a reminder to bug him about wav49 breaking on voicemail |
| 14:23.23 | GerbilWrk | mut, do not get the Lucent Compact Gateway platform, aka, Telica Plexus 9000 |
| 14:23.40 | mut | whys that |
| 14:23.56 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.225.Dial1.SanJose1.Level3.net) |
| 14:24.01 | GerbilWrk | just trust me, it is a horrible culmination of hardware flaws |
| 14:24.06 | mut | ok |
| 14:24.16 | mut | noted |
| 14:24.16 | mut | thx |
| 14:24.30 | GerbilWrk | we have one, and have had every board replaced numerous times, and still aren't running at 100% |
| 14:24.48 | mut | =\ |
| 14:25.24 | GerbilWrk | I've heard good things about Teleckt or Telect i think it was |
| 14:25.27 | GerbilWrk | just not Telica |
| 14:26.54 | mut | telect |
| 14:27.07 | mut | i'll look into |
| 14:27.14 | mut | where'd ya buy your switch? |
| 14:27.40 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 14:28.23 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 14:28.42 | mut | man quests hostnames keep gettin longer and longer |
| 14:28.59 | Nix | lol |
| 14:29.03 | mut | soon it'll have a persons gps coordinates in it |
| 14:29.52 | mut | dialup access? |
| 14:29.55 | mog_work | yes |
| 14:30.01 | mog_work | well da = yes |
| 14:30.01 | [TK]D-Fender | District Attourney? |
| 14:30.01 | Nix | hmm.. probably |
| 14:30.02 | mut | oo i'm good |
| 14:30.03 | mog_work | ^_^ |
| 14:30.22 | [TK]D-Fender | Digital>Analog? |
| 14:30.24 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.225.Dial1.SanJose1.Level3.net) |
| 14:30.59 | [TK]D-Fender | ...dumbass? |
| 14:31.07 | mut | :O |
| 14:31.09 | tzanger | [TK]D-Fender: haha |
| 14:31.20 | mut | hi, my name is tie my shoe, i will be doing some rapping for you today |
| 14:31.58 | mut | i wish today was monday so i could buy a cheeseburger for 35 cents at mcdonalds baby! |
| 14:32.08 | mut | maybe it was wednesday |
| 14:32.27 | mut | o well |
| 14:32.35 | asterboy | a real meal of champions |
| 14:32.53 | mut | o hell yea, champion meal $4.06 |
| 14:33.02 | mut | 2 cheeseburger, mcchicken and medium fry |
| 14:33.10 | asterboy | no wonder America is FAT |
| 14:33.23 | mut | and i even have 94 cents to put in the change jar |
| 14:33.29 | mut | heh |
| 14:33.32 | mut | i'm not fat |
| 14:33.36 | *** join/#asterisk heka (n=Mango@80.80.175.130) |
| 14:33.39 | asterboy | watching the Price is Right, while stuffing their cakeholes with McDicks |
| 14:33.40 | [TK]D-Fender | asterboy : "If guns kill people, then spoons make Michael Moore FAT" |
| 14:33.47 | asterboy | lol |
| 14:34.08 | [TK]D-Fender | A little cold perspective..... |
| 14:34.14 | FuriousGeorge | i dont get this: |
| 14:34.17 | asterboy | ya, he is a big fellow |
| 14:34.17 | FuriousGeorge | -- Executing GotoIf("SIP/Brian-7642", "0 | 1?winners|1:trunc|1") in new stack |
| 14:34.18 | FuriousGeorge | <PROTECTED> |
| 14:34.25 | FuriousGeorge | thats a conditional with an or |
| 14:34.26 | *** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) |
| 14:34.30 | Hmmhesays | you notice they never have retarded people on "price is right" |
| 14:34.30 | FuriousGeorge | as you see the 2nd condition is met |
| 14:34.40 | FuriousGeorge | yet it goes to trunk vs. winners |
| 14:34.42 | Pj_ | Hello ppl |
| 14:35.05 | asterboy | I love the boobie fall outs on the PIR |
| 14:35.16 | [TK]D-Fender | FuriousGeorge : Please show the CODE, not the CLI output of it alone... |
| 14:35.57 | FuriousGeorge | [TK]D-Fender: i set the vars so i know that second condition is met |
| 14:36.08 | praet | hello all. im looking to setup an asterisk server but i am not too sure about what i need |
| 14:36.11 | FuriousGeorge | ans the 1 after the | and before the ? proves it |
| 14:36.32 | nahirean | praet, You need sip/iax termination, a Linux box (or vmware, etc), and a distribution of asterisk (with all of the pre reqs) |
| 14:36.37 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:36.41 | [TK]D-Fender | FuriousGeorge : .... SHOW THE CODE.... |
| 14:36.51 | frawd | hello, anyone knows where i could find a converter between a double BNC coaxial and RJ45 to connect an E1 to a digium card? |
| 14:36.51 | FuriousGeorge | [TK]D-Fender: YOU ASKED FOR IT :) |
| 14:37.49 | [TK]D-Fender | FuriousGeorge : You're right... in duplicate as usual! People keep thinking I like the sound of my own voice or seeing myself in print... NO... people just don't LISTEN. |
| 14:37.56 | praet | nahirean: waht is sip/iax termination? a voip provider? |
| 14:38.26 | FuriousGeorge | [TK]D-Fender: what you mean im not listening, im preparing a pastebin as you asked |
| 14:38.31 | FuriousGeorge | i always listen to you |
| 14:38.32 | nahirean | praet: you got it ;) yeah, if you want to hit the PSTN through VoIP that is |
| 14:38.36 | [TK]D-Fender | :/ |
| 14:39.27 | Ariel_ | [TK]D-Fender, how did it finally go last night? |
| 14:39.43 | Aurs | what am I missing here, guys.. http://bugs.digium.com/view.php?id=5922 - my problem is that realtime members disappears on reload |
| 14:40.04 | praet | nahirean: ok so lets say i want a users to call from overseas to a landline, ill need fxo card, and a hardware phone at the calling end? |
| 14:41.23 | [TK]D-Fender | Ariel_ : Finally got it all... missing 1/2 dozen packages at least for the start, but * & Wanpipe all installed and ready. Tonight I actually build the PBX. |
| 14:41.24 | nahirean | praet, if you meant what I think you meant, why would you need an fxo card? you just want to hit a land line from a PBX registered to <x> iax provider? |
| 14:41.34 | FuriousGeorge | exten => s,n,gotoif($[${MATEWEIGHT${COUNT}} != ${MATEWEIGHT$[${COUNT} + 1]}] | $[${PRIZES} >= $[${ROWS} - ${CROP}]]?winners,1:trunc,1) |
| 14:41.34 | FuriousGeorge | <PROTECTED> |
| 14:41.39 | nahirean | in other words, you want to call out with your PBX to a <landline>? |
| 14:41.45 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 14:41.48 | FuriousGeorge | [TK]D-Fender: that was for you ^ |
| 14:41.57 | Ariel_ | [TK]D-Fender, great to hear it. |
| 14:42.25 | FuriousGeorge | [TK]D-Fender: just to give you some background, what i'm trying to do is account for the case when so many players get eliminated from the start that the remaining players have to be the winners |
| 14:42.29 | *** join/#asterisk GateWay10 (n=GateWay1@dsl-083-247-013-003.solcon.nl) |
| 14:42.38 | GateWay10 | anyone? |
| 14:42.39 | lzhang | what does sip respone 500 'Internal Server Error' mean? |
| 14:42.46 | praet | nahirean: well i am not sure of what asterisk is capable of, and where a voip provider falls in. |
| 14:42.54 | GateWay10 | anyone care to assist outgoing calls fail voipbuster |
| 14:42.59 | FuriousGeorge | therefore if prizes is >= players - eliminated players, you win |
| 14:43.29 | praet | nahirean: is it possible to connect a phone through the internet in another location to this asterisk server and then this server calls out on a regular line? |
| 14:43.40 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 14:44.18 | *** join/#asterisk kn0x (n=pinochle@71.194.235.251) |
| 14:44.35 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
| 14:44.36 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.170) |
| 14:44.37 | nahirean | praet: yes, that's possible.. but my experience with asterisk does not include working with land lines, so you may want to consult a Jedimaster in here ;) |
| 14:44.52 | FuriousGeorge | [TK]D-Fender: i should really comment that and get back to you if you still want to look for it. im just trying to account for cases that make the game fail, and im down to the last one, which is too many people have been eliminated, so the remaining players must be winners |
| 14:45.08 | [TK]D-Fender | FuriousGeorge : Staring at it funny now... |
| 14:45.11 | nahirean | praet, my experience is limited to internal extensions and strictly outgoing VoIP |
| 14:45.20 | FuriousGeorge | [TK]D-Fender: i swear it works |
| 14:45.20 | GateWay10 | has someone got a good howto for voipbuster behind nat with x-lite softphone |
| 14:45.31 | nahirean | praet, but registering an IP phone to your PBX remotely is a piece of cake |
| 14:45.31 | praet | nahirean: how do you use <snip> |
| 14:45.41 | FuriousGeorge | [TK]D-Fender: well, except in that last case i just mentioned |
| 14:45.45 | riddlebox | <PROTECTED> |
| 14:45.47 | praet | nahirean: what about firewall stuff |
| 14:46.13 | [TK]D-Fender | [10:34] <FuriousGeorge> -- Executing GotoIf("SIP/Brian-7642", "0 | 1?winners|1:trunc|1") in new stack |
| 14:46.18 | kn0x | i just recompiled the new version of asterisk... i was stilll on 1.2.2 cvs |
| 14:46.25 | kn0x | and now it wont start up |
| 14:46.27 | kn0x | http://pastebin.com/669332 |
| 14:46.31 | praet | nahirean: ok. so you have an asterisk server connected to the internet. what phones do you have internally (extensions) and how do they connect to the computer? |
| 14:46.35 | [TK]D-Fender | FuriousGeorge : You sould see just "1?" there, NOT the test... something is wrong... |
| 14:46.50 | FuriousGeorge | [TK]D-Fender: how wrong? terribly wrong? |
| 14:46.59 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 14:47.03 | [TK]D-Fender | FuriousGeorge : "0 | 1" should be EVALUATED, and not seen in CLI |
| 14:47.03 | FuriousGeorge | [TK]D-Fender: ALL MY BASE ARE BELONG TO ASTERISK!! |
| 14:47.24 | [TK]D-Fender | FuriousGeorge : coming out as a signle true/false followed by the "?" |
| 14:47.39 | FuriousGeorge | [TK]D-Fender: ok i see what you mean |
| 14:48.04 | nahirean | praet, you would use an IP phone to "register" to your pbx. Basically, (if the ip phone is on the same network) you would point the IP phone to the PBX's local IP. If it's not on the same network, or the PBX has a public IP, you'd point it to that instead. You would need to setup credentials in the SIP device and in Asterisk, but there's many examples regarding that to be found online. So basically IP Phone A) Registers to your PBX's IP |
| 14:48.08 | FuriousGeorge | [TK]D-Fender: or is | right? i mustve screwed up the syntax i guess. its my first biconditional conditional |
| 14:48.38 | GateWay10 | what softphone is best for use with asterisk |
| 14:48.43 | [TK]D-Fender | FuriousGeorge : no, "|" is the OR operator |
| 14:48.51 | *** join/#asterisk pagec (n=cpage@64-252-97-38.adsl.snet.net) |
| 14:48.54 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 14:48.56 | riddlebox | GateWay10, I use x-lite |
| 14:48.56 | FuriousGeorge | [TK]D-Fender: thats what i meant |
| 14:49.01 | [TK]D-Fender | GateWay10 : for Windows = eyeBeam |
| 14:49.06 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011872177.cable.net.co) |
| 14:49.14 | praet | so an ip phone can register from any internet connection, then once its connected to asterisk, it can place voip calls based on providers and also possible landline calls with fxo card? |
| 14:49.18 | GateWay10 | thnx |
| 14:49.27 | kn0x | BUMP |
| 14:49.31 | kn0x | i just recompiled the new version of asterisk... i was stilll on 1.2.2 cvs |
| 14:49.33 | kn0x | and now it wont start up |
| 14:49.36 | kn0x | http://pastebin.com/669332 |
| 14:49.52 | cytrak | hmm I'm getting a wierd problem I just installed the latest zaptel and now when I reboot my box it doesn't load the drivers automatically like the other version did |
| 14:49.53 | kn0x | downloaded the latest SVN |
| 14:49.59 | nahirean | praet: That's right |
| 14:50.06 | [TK]D-Fender | FuriousGeorge : I think you're missing a brace... |
| 14:50.23 | praet | nahirean: ok now im seeing the power of asterisk |
| 14:50.35 | kn0x | heh |
| 14:50.36 | nahirean | praet, Wait until you see what you can do with extensions :) |
| 14:50.41 | cytrak | So I load them manually and when I do a ztcfg -vvv even though I get the span list and the bottom I get a ZT_SPANCONFIG error ... and no device found |
| 14:50.49 | praet | nahirean: what hardware do you have for phones then? |
| 14:50.57 | praet | nahirean: you mean voicemail and such? |
| 14:51.00 | FuriousGeorge | [TK]D-Fender: oooooh, i know, the whole or statement should be in a brace, rigbht? |
| 14:51.05 | GateWay10 | ridlebox, have you got exp. with voicebuster? |
| 14:51.07 | FuriousGeorge | *right? |
| 14:51.22 | riddlebox | [TK]D-Fender, is there a way to dial a code while on a call and have asterisk park the call? |
| 14:51.32 | nahirean | praet, Way beyond voicemail. Hell, if you wanted to - you could turn off yer lights with yer phone |
| 14:51.38 | riddlebox | GateWay10, nope just x-lite, and an ATA device |
| 14:51.42 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:51.57 | *** join/#asterisk bweschke (n=bweschke@66.152.225.74) |
| 14:52.00 | GateWay10 | k thnx |
| 14:52.01 | [TK]D-Fender | FuriousGeorge : Yup... you're missing braces :) |
| 14:52.11 | Greek-Boy | exten => _9NXXNXXXXXX,1,Dial(SIP/provider/${EXTEN:1}) - do I have to dial 9 and then the long distance number with this? |
| 14:52.21 | riddlebox | nahirean, I am using my cell to call asterisk, while on the road, then I use asterisk to setup mythtv recordings |
| 14:52.31 | [TK]D-Fender | FuriousGeorge : CORRENT |
| 14:52.35 | [TK]D-Fender | CORECT even! |
| 14:52.37 | nahirean | riddlebox, Nice ;) |
| 14:52.43 | [TK]D-Fender | ... wait.. getting warmer... |
| 14:52.47 | [TK]D-Fender | CORRECT! |
| 14:52.58 | [TK]D-Fender | FuriousGeorge : You need to brace the WHOLE thing. |
| 14:53.02 | riddlebox | third times a charm |
| 14:53.20 | GateWay10 | anyone got voicebuster working behind NAT with x-lite |
| 14:53.27 | [TK]D-Fender | FuriousGeorge : So - exten => s,n,gotoif($[$[${MATEWEIGHT${COUNT}} != ${MATEWEIGHT$[${COUNT} + 1]}] | $[${PRIZES} >= $[${ROWS} - ${CROP}]]]?winners,1:trunc,1) |
| 14:53.39 | nahirean | riddlebox, I do silly things, like use my cell to ping an IP if I need to, "Consolidate" the response, and text it back to my cell.. so I can "Ping on the run" or "While at Jack n the Box" etc ;) |
| 14:53.44 | praet | nahirean: what hardware phones do you use then? |
| 14:54.03 | nahirean | praet, There are several IP phones to choose from. Many here would suggest Polycom models I'd wager ;) |
| 14:54.23 | riddlebox | nahirean, I am starting to collect some really code agi scripts, soon I will have them on a website for everyone to enjoy |
| 14:54.23 | FuriousGeorge | [TK]D-Fender: you have completed me |
| 14:54.26 | praet | k thanks |
| 14:54.31 | FuriousGeorge | [TK]D-Fender: thanks |
| 14:54.31 | [TK]D-Fender | FuriousGeorge : explains why it wasn't evaluating, now doesn't it? :) |
| 14:54.37 | FuriousGeorge | [TK]D-Fender: it sure does |
| 14:54.47 | riddlebox | nahirean, code = good haha |
| 14:54.52 | nahirean | :) haha |
| 14:55.06 | [TK]D-Fender | I just needed to bring it into an editor with syntax highlighting so I could match the braces... it was making me dizzy... |
| 14:55.19 | kn0x | anyone take a look at http://pastebin.com/669332 ? |
| 14:55.27 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.74) |
| 14:55.28 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011872177.cable.net.co) |
| 14:55.31 | riddlebox | nahirean, did you see my question earier about while on a call dial a code to park the call? |
| 14:55.36 | *** part/#asterisk GateWay10 (n=GateWay1@dsl-083-247-013-003.solcon.nl) |
| 14:55.44 | FuriousGeorge | [TK]D-Fender: i started using kate today which at least helps with the braces. it doesnt do highlighting and indenting for the asterisk dialplan, but who knows. maybe one day |
| 14:55.59 | nahirean | riddlebox, What's also entertaining is if you have a IAX provider who lets you set the CID number ;) nope, I didn't.. but I doubt I would be of much help compared the gurus who're in here |
| 14:56.06 | [TK]D-Fender | kn0x : put "noload => chan_modem.so" into your modules.conf |
| 14:56.24 | riddlebox | nahirean, broadvoice doesnt let me do that :( |
| 14:56.35 | [TK]D-Fender | FuriousGeorge : no need for anything more than brace/bracket/quote matching really... |
| 14:56.44 | praet | nahirean: those phones are more expensive than my linux box! |
| 14:57.27 | nahirean | praet, E-Bay is your friend ;) maybe look into SPAs? like the 942.. or get a VoIP adapter, that uses a analog phone |
| 14:57.38 | [TK]D-Fender | I have yet to really use Linux as a desktop for anything other than watching my DivX collection on my HDTV :) |
| 14:57.53 | FuriousGeorge | [TK]D-Fender: what do u use for the desktop |
| 14:57.58 | [TK]D-Fender | SPA-942 = waste as compared to Polycom IP 501. |
| 14:58.14 | tdonahue-laptop | FuriousGeorge, i started writing the highlighting for the extensions.conf, but never got it to work well |
| 14:58.20 | [TK]D-Fender | FuriousGeorge : My workstations are all WinXP. everything I do with LInux really is CLI.... |
| 14:58.48 | salviadud | who likes fluxbox? |
| 14:58.59 | mog_work | i used to use it |
| 14:59.02 | salviadud | yeah |
| 14:59.05 | mog_work | know im on xfce4 |
| 14:59.06 | nahirean | Fender, why do you like the 501? What advantages do you see? |
| 14:59.10 | nahirean | <-- uses fluxbox |
| 14:59.25 | salviadud | fluxbox rocks |
| 14:59.26 | FuriousGeorge | tdonahue-laptop: for kate |
| 14:59.27 | blitzrage | [TK]D-Fender: me too! |
| 14:59.37 | salviadud | pr0n on fluxbox looks even better. imho |
| 14:59.43 | blitzrage | fluxbox is ok if you like minimal -- I really like gnome |
| 14:59.54 | mog_work | gnome is sexy blitzrage |
| 14:59.57 | FuriousGeorge | [TK]D-Fender: i wont bash you for using windows, sometimes its nice when things just work. |
| 15:00.00 | mog_work | i just have mostly slow machines |
| 15:00.01 | FuriousGeorge | well until they dont |
| 15:00.09 | blitzrage | mog_work: then yah, flux makes sense |
| 15:00.12 | [TK]D-Fender | nahirean : More lines, more callls supported per key, better use of display (to show activity / controls), better audio quality on handset/speakerphone, more solid physically. |
| 15:00.16 | kn0x | TK Defender--- [chan_modem.so]Apr 19 09:59:56 WARNING[7419]: loader.c:731 __load_resource: misstng mod_data for chan_modem.so |
| 15:00.23 | kn0x | its still not working |
| 15:00.37 | blitzrage | I use WinXP, with 4-15 putty windows open to Linux CLI's each running screen :) |
| 15:00.46 | blitzrage | and IE :) |
| 15:00.46 | nahirean | D-Fender, I assume you've heard the 942? I was impressed by the crisp quality they provided.. you really beleive the 501 has the 942 beat in the quality department? |
| 15:00.56 | [TK]D-Fender | FuriousGeorge : My WinXP boxes ALWAYS "just work" :) and given my safe internet policies never really needed AV / anti-spyware... |
| 15:01.09 | blitzrage | [TK]D-Fender: same here -- we're one in the same! :) |
| 15:01.23 | [TK]D-Fender | nahirean : 942 is a 941 + PoE & Backlight, and I've owned both. |
| 15:01.24 | blitzrage | [TK]D-Fender: brothers don't shake hands... brothers hug! |
| 15:01.34 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 15:01.35 | praet | what are the Digium Wildcards used for? |
| 15:01.40 | blitzrage | praet: nothing |
| 15:01.42 | [TK]D-Fender | blitzrage : I am Siam ;) |
| 15:01.53 | praet | ok |
| 15:02.00 | FuriousGeorge | [TK]D-Fender: i dont care what anyone says, xp post sp2 is a good OS. maybe not the most exciting one in the world, but it does the job |
| 15:02.14 | Hmmhesays | I use XP as a gaming platform |
| 15:02.19 | koenvi | back again |
| 15:02.24 | praet | does everone here use asterisk@home? |
| 15:02.28 | blitzrage | LOL |
| 15:02.29 | koenvi | nope |
| 15:02.32 | blitzrage | no -- I hate A@H |
| 15:02.35 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 15:02.40 | kn0x | [TK]D-Fender heres my modules.conf http://pastebin.com/669356 |
| 15:02.41 | blitzrage | and I just got asked to write an article on it too.... |
| 15:02.45 | kn0x | its still not working |
| 15:02.48 | salviadud | a@h is terrible |
| 15:02.49 | [TK]D-Fender | FuriousGeorge : For me it gets the job done without any bitching and no trouble so what the heck. I use OSS apps ALL over it, and having installed hundreds of them am pretty good at it. |
| 15:02.59 | kn0x | hah |
| 15:03.00 | blitzrage | FuriousGeorge: :) |
| 15:03.10 | praet | can i yum install asterisk then? |
| 15:03.15 | kn0x | heh |
| 15:03.15 | [TK]D-Fender | FuriousGeorge : Works in Wine now ;) |
| 15:03.17 | kn0x | dont doo it |
| 15:03.24 | kn0x | use svn or cvs |
| 15:03.26 | [TK]D-Fender | kn0x : And? Should work... |
| 15:03.30 | kn0x | nope |
| 15:03.31 | koenvi | how can I share call variables between two legs of a call??? |
| 15:03.37 | FuriousGeorge | [TK]D-Fender: my head is already hurting just thinking about getting that running :) |
| 15:03.39 | kn0x | Asterisk Dynamic Loader Starting: |
| 15:03.39 | kn0x | <PROTECTED> |
| 15:03.42 | kn0x | Segmentation fault |
| 15:03.43 | [TK]D-Fender | praet : No, just download from digium's FTP. |
| 15:03.45 | kn0x | Asterisk Dynamic Loader Starting: |
| 15:03.47 | kn0x | <PROTECTED> |
| 15:03.50 | kn0x | Segmentation fault |
| 15:03.53 | kn0x | wjpp[s |
| 15:03.54 | [TK]D-Fender | FuriousGeorge : Well documented... |
| 15:03.55 | kn0x | SORRY |
| 15:03.58 | praet | [TK]D-Fender: thanks |
| 15:03.58 | kn0x | haha |
| 15:04.07 | [TK]D-Fender | kn0x : Fine... go delete the .so then! |
| 15:04.46 | jsharp | or noload => chan_modem.so |
| 15:05.00 | praet | one more thing. Does it make sense to look for a fax solution based on asterisk or standalone? |
| 15:05.01 | kn0x | <PROTECTED> |
| 15:05.04 | kn0x | Segmentation fault |
| 15:05.05 | [TK]D-Fender | praet : By downloading the "release" versions you never have to worry about the "picky" details of CVS/SVN, and if there is a serious problem in a "release" you'll see it here fast and a patch will come out and end up in the channel topic. |
| 15:05.08 | kn0x | still now TK :( |
| 15:05.20 | [TK]D-Fender | kn0x : DELETE THE .SO! |
| 15:05.22 | FuriousGeorge | [TK]D-Fender: ill have no reason to dual boot till i find another game i cant play on linux :) |
| 15:05.34 | praet | [TK]D-Fender: ok i will go through a new install leter today |
| 15:05.55 | kn0x | in modules.conf... i removed the .so from all of the chan_modem lines |
| 15:06.02 | kn0x | thats the message i get |
| 15:06.11 | [TK]D-Fender | I might use Linux more on my primary PC once I get the right video mode working for X on it... I need a good doc for that... |
| 15:06.24 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 15:06.27 | tzanger | ManxPower: so polycom 1.6.x SIP does separate blind/attended transfers... any way to get the old way back? i.e. transfer, if they pick up before you hang up, it's attended, if not, it's blind? |
| 15:06.29 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:06.30 | [TK]D-Fender | praet : I do it all the time... nothing to it... just DL, compile, install, configure... |
| 15:07.03 | [TK]D-Fender | tzanger : thats how they ALWAYS did it... attended first unles you hit the "blind" soft-key right after hitting "transfer" |
| 15:07.07 | praet | [TK]D-Fender: what does your asterisk usage look like? basically calls out through voip? |
| 15:07.53 | tzanger | [TK]D-Fender: oh so it *still* does that. ok. |
| 15:07.54 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:08.05 | salviadud | fender, what's your video card? |
| 15:08.06 | boddy | [TK]D-Fender:how can I learn which codec there are in asterisk |
| 15:08.12 | [TK]D-Fender | praet : I've done all sort of installs, personally, for my work, and as a consultant. I've use analog lines on a TDM400, PRI T1 on TE405P / A104d, a handful of SIP providers, and I am using my work * system + PRI AS my VoIP provider for home. |
| 15:08.21 | [TK]D-Fender | salviadud : nvidia 5500 |
| 15:08.27 | [TK]D-Fender | err.. 5200 |
| 15:08.28 | FuriousGeorge | boddy: there is no way. it isnt documented and we're not telling |
| 15:08.32 | salviadud | nice |
| 15:08.36 | [TK]D-Fender | salviadud : safe choice. |
| 15:08.49 | *** part/#asterisk koenvi (n=root@tech.ascom.be) |
| 15:08.55 | kn0x | [TK]D-Fender, any other ideas? |
| 15:08.55 | [TK]D-Fender | salviadud : It LOW ent.... wait.. I'm sorry, thats my SERVER... my "main" is a GF4-4200 |
| 15:09.00 | FuriousGeorge | ~docs |
| 15:09.02 | jbot | methinks 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 |
| 15:09.05 | [TK]D-Fender | kn0x : did you delete it and restart *? |
| 15:09.08 | tzanger | hahhaa |
| 15:09.11 | kn0x | yessum |
| 15:09.11 | tzanger | ahh ahh ~docs |
| 15:09.12 | tzanger | hahaha |
| 15:09.19 | kn0x | i cant get asterisk to start |
| 15:09.22 | tzanger | god, all over the screen and everything |
| 15:09.27 | kn0x | that is the output of asterisk -dvc |
| 15:09.39 | FuriousGeorge | me + sleep deprivation = on a comedic role |
| 15:09.40 | [TK]D-Fender | kn0x : if you can PM my SSH creds for your box I'm willing to take a look. |
| 15:09.44 | salviadud | Fender: how much resolution can you get on that card? |
| 15:09.45 | [TK]D-Fender | me* |
| 15:09.56 | lzhang | 3 |
| 15:10.07 | [TK]D-Fender | salviadud : All sorts, its just that my MONITOR uses 1440*900 which is hard to google for xconfigs for :) |
| 15:10.09 | kn0x | [TK] D-Fender thanks |
| 15:10.10 | ManxPower | tzanger, I have no idea. |
| 15:10.27 | boddy | how much bandwith ulaw uses ? |
| 15:10.32 | kn0x | query [TK]D-Fender |
| 15:10.35 | kn0x | ahh |
| 15:10.38 | salviadud | ooooh, i see |
| 15:10.43 | ManxPower | tzanger, When my users complained I told them I could give them an analog rotary phone if they preferred and they stopped complaining. |
| 15:10.45 | FuriousGeorge | lol |
| 15:11.08 | salviadud | that is a freaky resolution... is it widescreen? |
| 15:11.25 | *** join/#asterisk NSTech (n=NSTech@tokaimura.infotechsys.com) |
| 15:11.29 | tzanger | boddy: 80kbps |
| 15:11.34 | [TK]D-Fender | salviadud : YUP :D |
| 15:11.38 | RoyK | salviadud: this is 1280x854 :) |
| 15:12.00 | [TK]D-Fender | salviadud : My favourite bang/buck monitor, and #1 on my suggest list... |
| 15:12.07 | NSTech | Hello - i've just come to realize my Digum card isn't supported in BSD and am going to rebuild my Asterisk server in Linux, but would like some suggestions on which distro to go with - anyone have opinions? |
| 15:12.18 | mog_work | debian |
| 15:12.20 | RoyK | [TK]D-Fender: i've got an apple cinemadisplay 20" back home :D |
| 15:12.23 | RoyK | debian |
| 15:12.33 | salviadud | yeah, i had trouble with widescree before |
| 15:12.34 | boddy | tzanger really ? |
| 15:12.36 | NSTech | I've got 2 votes for debian... |
| 15:12.38 | boddy | or juke |
| 15:12.41 | [TK]D-Fender | Acer AL1916W 19" Wide-screen LCD 1440*900 < $300 USD retail... |
| 15:12.54 | [TK]D-Fender | RoyK : but the cost is nasty by comparison. |
| 15:12.57 | salviadud | i got a 1200 x 800 monitor on a notebook with intel 855 |
| 15:13.06 | RoyK | [TK]D-Fender: indeed... |
| 15:13.08 | tzanger | boddy: yes, 64kbps payload, but take 20ms frames and add UDP overhead and it's about 80kbps in each direction |
| 15:13.19 | RoyK | [TK]D-Fender: but apple's gear is sexy indeed |
| 15:13.25 | boddy | tzanger gsm ? |
| 15:13.28 | RoyK | tzafrir: exactly 80kbps :) |
| 15:13.43 | RoyK | boddy: gsm is 13.5kbps, overhead is 16kbps |
| 15:13.57 | boddy | hým ok so I will use gsm on sip |
| 15:13.59 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:14.07 | OliverX | ehm i answer a call with my fritz card and forward this call to sip account. i can hangup the call and i have connection |
| 15:14.10 | OliverX | but no sound |
| 15:14.14 | OliverX | have anyone here an idea? |
| 15:14.16 | tzanger | boddy: this is all on the wiki |
| 15:14.27 | boddy | ok thanks for helps |
| 15:14.30 | tzanger | yeah gsm is about 3.4kbps per call |
| 15:14.37 | tzanger | er kBps |
| 15:14.47 | tzanger | ugh, I can't get DTMF with asterlink working |
| 15:14.50 | tzanger | not reliably anyway |
| 15:14.50 | bkw_ | [TK]D-Fender, my 15.4 macbook pro does 1440x900 |
| 15:14.57 | tzanger | oh bkw_'s here |
| 15:15.24 | NSTech | anyone else like debian better than other oses? |
| 15:15.24 | praet | Does it make sense to look for a fax solution based on asterisk or go standalone? |
| 15:15.33 | praet | ive seen asterfax for example |
| 15:15.58 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 15:16.40 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 15:16.40 | websae | I am curious---what is the best way to test a business class cable modem line to see if it can handle 5 simulataneous calls |
| 15:16.59 | mut | call 5 people |
| 15:17.11 | praet | websae: there are voip tests online that ive seen |
| 15:17.23 | praet | http://www.bandwidth.com/tools/voipTest |
| 15:18.06 | websae | hrm okay |
| 15:18.22 | nahirean | i dont think faxing over VoIP is a good idea.. I mean, 2% packet loss at a hop and yer fax is gone |
| 15:18.57 | praet | what fax servers are out there that integrate with email? |
| 15:19.42 | websae | i have a 512kbps cable connection and i can only support 2 g729 calls? |
| 15:19.57 | ManxPower | websae, why do you think that? |
| 15:20.19 | websae | that's what that test told me! |
| 15:20.25 | websae | quite odd I think |
| 15:20.26 | wunderkin | you need to know the upload speed |
| 15:20.35 | ManxPower | g729 calls take about 30kbps, so you do 512/30 and you get..... |
| 15:20.49 | RoyK | g.729 takes 16+8=25kbps |
| 15:20.51 | websae | you get more than 2! |
| 15:20.53 | RoyK | 2/25/24 |
| 15:20.53 | rpm | how do i track calls with asterisk, i need to know exactly from the point the call is started until the call terminates.. so all the states need to be tracked some how.. has anything been written to do this? |
| 15:20.56 | RoyK | eh |
| 15:21.00 | ManxPower | obviously because of buffering, other data, etc. |
| 15:21.02 | RoyK | 24kbps |
| 15:21.09 | mog_work | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 15:21.13 | mog_work | dizamn |
| 15:21.13 | twisted[asteria] | rpm, omg are you kidding? CDR's |
| 15:21.21 | ManxPower | rpm, you mean like /var/log/asterisk/cdr-csv/Master.csv ? |
| 15:21.35 | RoyK | mog_work: the easiest is to just remember that with 20ms and rtp, the overhead is 16kbps |
| 15:21.39 | websae | so i should be able to get about 15concurrent calls? |
| 15:21.44 | twisted[asteria] | bkw_, your macbook pro also will burn the hair off your legs |
| 15:21.49 | websae | with that upload spead |
| 15:22.14 | OliverX | can anyone here say me what the dial syntax is for a fritz card? |
| 15:22.28 | ManxPower | websae, more like 12 but that assumes there is NO OTHER DATA ON THE LINE. |
| 15:22.41 | twisted[asteria] | such as *cough* irc |
| 15:22.44 | websae | yeah exactly |
| 15:23.10 | rpm | ManxPower: no.. cdr is not detailed enough. it only updates when the call is complete. i need to track all the call events.. QueueMemberStatus, Newchannel, Newstatus, Link, Unlink events.. |
| 15:23.27 | *** join/#asterisk razu_ (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
| 15:23.31 | pengyong | is ss7 stable for production deployment? |
| 15:23.35 | ManxPower | rpm, then you need to write a custom application using the Manager Interface (aka AMI) |
| 15:23.43 | salviadud | dude, you should get a girl |
| 15:23.48 | salviadud | redbull + girls |
| 15:23.52 | ManxPower | pengyong, We don't know. Is it? It's not part of the standard Asterisk source) |
| 15:23.55 | salviadud | = boooya |
| 15:24.20 | twisted[asteria] | salviadud, uh, thanks for telling me something I already know ;) |
| 15:25.13 | salviadud | don't mention it, remember. the sky is always blue, unless it's cloudy |
| 15:25.22 | Katty | twisted[asteria]: whyfor are you all bouncy? |
| 15:25.30 | twisted[asteria] | Katty, redbull |
| 15:25.33 | Katty | twisted[asteria]: oh, that. k |
| 15:25.36 | lzhang | salviadud: what if it's nighttime |
| 15:25.45 | websae | K is smaller than kbps? |
| 15:25.54 | salviadud | O_O |
| 15:25.55 | *** join/#asterisk Mike (n=mike@201.129.119.118) |
| 15:26.03 | salviadud | got me there |
| 15:26.21 | ManxPower | websae, K = bytes k = bits but people commonly screw that up |
| 15:26.24 | Mike | anyone knows how to force the callerid on a sip client? i set callerid=xxx but i still get his old callerid any ideas? |
| 15:26.26 | twisted[asteria] | K = 1024 bytes, kbps is 1000 bytes per second |
| 15:26.36 | nahirean | bits are smaller than bytes.. 8 bits make 1 byte |
| 15:26.43 | twisted[asteria] | err yeah, bits per seconds |
| 15:26.46 | ManxPower | Mike, you use callerid=Robert Dobbs <666> in sip.conf in the device section |
| 15:27.00 | Mike | i did |
| 15:27.08 | salviadud | 1 2 4 8 16 32 64 128 256 512 1024 2048 |
| 15:27.09 | twisted[asteria] | or something |
| 15:27.18 | ManxPower | Mike, then the call is not matching the section you set the callerid line in |
| 15:27.44 | salviadud | it's lotsa fun to count like that |
| 15:27.44 | ManxPower | Mike, put the correct context= line in each of your [sipconfsections] the in [general] put context=INVALID |
| 15:27.52 | OliverX | can anyone here say me what the dial syntax is for a fritz card? |
| 15:27.56 | ManxPower | That way calls will fail if they cannot authenticate |
| 15:28.02 | websae | so i have a 768K connection -- what is that in terms of KBPS? |
| 15:28.19 | nahirean | does the K represent kilobits or kilobytes |
| 15:28.21 | ManxPower | websae, 768Kbps |
| 15:28.45 | salviadud | should be kilobytes |
| 15:28.52 | salviadud | cause it's a capital K |
| 15:28.54 | nahirean | yeop |
| 15:28.56 | nahirean | yep* |
| 15:32.40 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-58-138.w86-217.abo.wanadoo.fr) |
| 15:33.38 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 15:34.30 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 15:35.32 | *** join/#asterisk coppice (n=chatzill@218.206.99.164) |
| 15:38.23 | sangee | how do i connect the CLI using java socket program? |
| 15:39.08 | mog_work | EEEWWWWWWWWWWWWWWWWWWWWWWWWWW |
| 15:39.30 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:39.58 | Ariel_ | sangee, well use a ssh java application I guess you can if you allow ssh. |
| 15:41.09 | sangee | accually i want to reload the configuration |
| 15:41.28 | sangee | i plan to write a java program to reload every 10min |
| 15:41.41 | lzhang | sangee: you can just call asterisk -rx 'reload' from shell |
| 15:42.04 | Ariel_ | or use the manager api |
| 15:42.08 | mog_work | manager api |
| 15:42.14 | sangee | ok |
| 15:42.18 | sangee | thanks |
| 15:42.29 | heison | */10 * * * asterisk -rx 'reload' |
| 15:42.32 | Ariel_ | what ever it is called I gess |
| 15:42.38 | Ariel_ | gess/guess |
| 15:42.41 | Ariel_ | argh |
| 15:42.50 | lzhang | nice :) |
| 15:46.18 | Ariel_ | wow just heard that china is selling organs from executed people on the black market.... hummm |
| 15:46.25 | Hmmhesays | oh i love calling mexico |
| 15:46.29 | tzanger | Ariel_: that's old news |
| 15:46.33 | Hmmhesays | i think it has to be my favorite place to call |
| 15:46.45 | Ariel_ | tzanger, really well it's still sick |
| 15:46.52 | tzanger | china is seeding clouds over beijing in order to wash away the result of a bad dust storm |
| 15:47.24 | Greek-Boy | how do i bring verbosity back down in CLI? |
| 15:47.33 | heison | set verbose # |
| 15:47.37 | Ariel_ | set verbose 3 |
| 15:47.53 | Greek-Boy | thanks |
| 15:48.02 | coppice | tzanger: beijing gets really really bad dust blowing in from the west |
| 15:48.59 | Greek-Boy | what is default linux verbosity? |
| 15:49.08 | Greek-Boy | by mistake i set verbosity in linux coz i wasn't in CLI |
| 15:49.09 | Greek-Boy | lol |
| 15:49.14 | Ariel_ | 3 |
| 15:50.07 | coppice | why are people so interesting in useless stuff like variable length DTMF? is it just because it looks a lot easier than the things which would be useful? |
| 15:50.10 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 15:50.26 | Hmmhesays | lol |
| 15:50.30 | mog_work | vldtmf is important for some sip interop crap coppice |
| 15:50.33 | Greek-Boy | would I use this for outgoing calls too? register => YOURUSERNAME:YOURPASS@host-or-ip in sip.conf |
| 15:51.08 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 15:51.08 | coppice | Its needed to fully support the broken crap in RFC2833 |
| 15:51.34 | coppice | anything which cares about the length of DTMF is totally broken |
| 15:52.22 | mog_work | <PROTECTED> |
| 15:52.48 | coppice | which ia what? |
| 15:53.43 | Greek-Boy | i c |
| 15:53.47 | *** join/#asterisk jroysdon (n=jroysdon@c-67-181-65-139.hsd1.ca.comcast.net) |
| 15:54.15 | Greek-Boy | i have setup my provider in sip.conf and also in extensions.conf but I get a 404 not found error :( |
| 15:55.11 | coppice | cellphone don't preserve the length of DTMF |
| 15:55.12 | coppice | key systems don't preserve the length of DTMF |
| 15:55.14 | coppice | various other things don't preserve the length of DTMF |
| 15:55.15 | coppice | for some brain dead reason RFC2833 tries to. |
| 15:55.17 | coppice | if you read through RFC2833 it appears to have been written by someone that had never used a telephone in their entire life |
| 15:55.23 | justinu | haha |
| 15:56.10 | Greek-Boy | gtg |
| 15:56.11 | Greek-Boy | chat later |
| 15:56.15 | tzanger | coppice: :-) |
| 15:56.48 | file | may be the case, but we can't control every other manufacturer's RFC2833 implementation so we have to go for something that gets us he best interoperability |
| 15:56.52 | ManxPower | does anyone have an example of the new style way of getting CALLERID info from inside an AGI? |
| 15:56.58 | ManxPower | Using Asterisk::AGI of course in perl |
| 15:57.32 | coppice | what interoperability problems do you have related to DTMF? (there are plenty related to other tones) |
| 15:57.47 | *** join/#asterisk delox99 (n=delox@209-87-229-62.storm.ca) |
| 15:57.53 | delox99 | hi all |
| 15:58.03 | delox99 | i have a problem with incoming calls from a DID |
| 15:58.32 | zoa | coppice |
| 15:58.40 | zoa | do you have any dtmf talkoff test tapes ? |
| 15:58.44 | delox99 | the calls are forwarded to my * box as ext@domain.com |
| 15:58.47 | *** join/#asterisk lokkju_ (n=lokkju@unaffiliated/lokkju) |
| 15:59.20 | delox99 | the problem is that when i call my did from certain pstn phones it goes through |
| 15:59.43 | coppice | I have clean wave files from the Bellcore tapes. I think kram or Jim Dixon has the original tapes |
| 15:59.47 | delox99 | on some other pstn phones, my server rejects the call because of bad authentication |
| 16:00.08 | file | coppice: Ciscos for example will generate the tone based on the actual time elapsed between received RFC2833 packets and not the contents... so because our current implementation sends the packets all in one burst, you get clipped DTMF |
| 16:00.20 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 16:00.24 | delox99 | probably something to do with CLI forwarding that mess up the auth process |
| 16:00.29 | zoa | coppice, could you email them to me ? I bought that other test tape |
| 16:00.33 | zoa | still waiting for it to come |
| 16:00.39 | zoa | mitel test tape |
| 16:00.42 | zoa | iirc |
| 16:01.00 | coppice | file: sending in a burst is obviously wrong, but a fixed time will be fine |
| 16:01.08 | coppice | you can't buy the Mitel tape any more |
| 16:01.13 | delox99 | is it possible to tell asterisk to not authenticate certain incoming sip connections based on source ip or something? |
| 16:01.42 | coppice | the bellcore tapes are copyright material |
| 16:01.42 | zoa | coppice, i bought a second hand copy in NY |
| 16:01.49 | kamileon | i just want * to give me slow head. |
| 16:02.13 | coppice | it probably useless if its second hand. they stretch really bady |
| 16:02.18 | zoa | ah shit |
| 16:02.21 | zoa | shit shit shit |
| 16:03.01 | coppice | they use some kind of super stretchy crap cassettes. I used to have terrible trouble with those tapes |
| 16:03.08 | brettnem | I have horrible DTMF problems on my network.. :( |
| 16:03.40 | zoa | we paid 600 euro for such a tape, hope its not going to be broken :( |
| 16:04.01 | brettnem | http://www.dsprelated.com/showmessage/49733/1.php |
| 16:04.23 | coppice | someone told me if you talk nicely to the people at zarlink they will send you a clean wave file of the mitel tape, but i've never tried |
| 16:04.40 | zoa | brettnem: thats the guy i bought it from |
| 16:04.44 | brettnem | ah |
| 16:04.49 | brettnem | wow.. expensive |
| 16:04.58 | zoa | well its the only place i could find it |
| 16:05.07 | mog_work | whats on this tape? |
| 16:05.10 | zoa | zarlink.com ? |
| 16:05.15 | mog_work | just pure dtmf? |
| 16:05.17 | Qwell | mog_work: secret spy stuff |
| 16:05.18 | zoa | it contains sounds close to dtmf |
| 16:05.21 | zoa | but not dtmf |
| 16:05.25 | coppice | the mitel tape is fairly easy to pass. the bellcore is much tougher |
| 16:05.27 | mog_work | why not just make one? |
| 16:05.35 | zoa | because its a standard test |
| 16:05.44 | mog_work | ahh |
| 16:05.52 | coppice | zarlink.com is the semi division of Mitel spun out as an a separate company |
| 16:05.55 | mog_work | do we pass either? |
| 16:06.06 | brettnem | coppice, they don't actually have voices on the test, it's just tones, right? |
| 16:06.08 | tzanger | yeah I ignored the bot in #shadowmite |
| 16:06.12 | tzanger | er wrong channel heh |
| 16:06.19 | justinu | treo? |
| 16:06.31 | brettnem | heh shadowmite |
| 16:06.44 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 16:06.53 | coppice | spandsp generates all the tone sequences on the mitel tape. zoa wants the second side which is voice |
| 16:07.23 | brettnem | DTMF is one of my biggest problems with asterisk right now.. |
| 16:07.42 | zoa | false detection of dtmf ? |
| 16:07.48 | brettnem | nah |
| 16:07.51 | brettnem | bad detection |
| 16:08.03 | brettnem | I use a PSTN GW that requires inband DTMF.. |
| 16:08.08 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:08.36 | brettnem | so sending calls asterisk->ser->asterisk, I have to send inband because it *might* goto the GW from SER.. and inband detection seems sketchy at best |
| 16:08.40 | tzanger | coppice: yeah but he could record it into the PC, and use knowledge of what the known frequency is supposed ot be to speed up the recording to get it back ot normal |
| 16:08.48 | blitzrage | brettnem: yes, it's a known issue |
| 16:08.57 | Peaceful | Is there a recommended 'best practice' for handling all DID's that you have but aren't actually using at the moment? |
| 16:09.27 | blitzrage | Peaceful: send them to tt-weasels |
| 16:09.32 | brettnem | Peaceful: 404 is a good thing to do. |
| 16:09.41 | coppice | the DTMF detector in * doesn't work as well as when I gave it to *, but it still seems to be pretty good, from some input I had recently |
| 16:09.41 | brettnem | blitzrage: is there anything that can be done to make it any better? |
| 16:09.45 | justinu | 404 or inband SIT + disco message |
| 16:10.19 | blitzrage | brettnem: send a 404 Not Found or something like that is probably fine |
| 16:10.19 | coppice | tzanger: we used a variable speed cassette deck to get the recordings from the bellcore tapes spot on |
| 16:10.34 | brettnem | the inband detection almost always fails for me.. not sure why... using all dedicated connections and G.711 everywhere. |
| 16:10.53 | brettnem | blitzrage: no, I meant about the DTMF issues.. anything I can do to improve it? |
| 16:10.53 | tzanger | coppice: exactly |
| 16:11.19 | blitzrage | brettnem: make sure there is no jitter on your network |
| 16:11.41 | blitzrage | brettnem: other than that, it's hit and miss depending on the network conditions and the device you're sending them to until we can get VLDTMF |
| 16:11.43 | brettnem | blitzrage: there shouldn't be much.. it's all hi-cap circuits and dedicated equipment. |
| 16:11.55 | brettnem | VLDTMF? |
| 16:11.56 | blitzrage | brettnem: rfc2833 I assume? |
| 16:12.03 | brettnem | blitzrage: no, all inband |
| 16:12.05 | blitzrage | brettnem: bugs.digium.com and check bug 6667 |
| 16:12.16 | blitzrage | brettnem: oh -- I don't use inband, so I'd be surprised you're having problems |
| 16:12.27 | blitzrage | inband is not known to have issues afaik |
| 16:12.43 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 16:12.51 | brettnem | yeah, well I use all inband.. all G.711.. all dedicated connections and it doesn't work 90% of the time |
| 16:13.23 | blitzrage | brettnem: I'm not sure using inband, even with G.711, is ideal, but I haven't tested extensively |
| 16:13.42 | blitzrage | unless you're talking about TDM |
| 16:13.43 | blitzrage | of course :) |
| 16:14.00 | brettnem | blitzrage: nah, it's all SIP to SIP |
| 16:14.19 | brettnem | the SIP to TDM stuff actually works fine |
| 16:14.39 | blitzrage | brettnem: hrmmm... yah, SIP to TDM makes sense... not sure about the SIP to SIP other than the network |
| 16:14.57 | blitzrage | latency and jitter and packet loss would be prime candidates to cause issues |
| 16:15.37 | brettnem | yeah, like I said, they are pretty good dedicated connections.. so it shouldn't be an issue. |
| 16:16.18 | coppice | brettnem: what does bad detection mean? missing digits? doubling them? |
| 16:16.23 | blitzrage | brettnem: hrmmm... good test would be to actually listen to the DTMF to see if you hear any issues that Asterisk might perceive as a problem |
| 16:16.49 | blitzrage | but I'm sure I'm just spouting off all sorts of things you already know :D |
| 16:17.09 | zoa | yes you are repeating yourself blitzrage! |
| 16:17.11 | zoa | :p |
| 16:17.16 | zoa | go sit in the corner now! |
| 16:17.21 | brettnem | yeah.. I had to install dundi and IAX to circumvent the problem. :( |
| 16:18.33 | blitzrage | zoa: nevah! |
| 16:19.30 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011872177.cable.net.co) |
| 16:20.08 | coppice | the DTMF decoder in * was designed to work in a system where EC removes dial tone. If you don't remove dial tone, and you have a fairly high spill of dial tone back to the DTMF receiver it badly desensitises it |
| 16:20.29 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:21.06 | ManxPower | one of these days I'm going to post a bounty to make STDERR from AGI scripte PRINT ON THE DAMN TTY I'm ON |
| 16:21.35 | *** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 16:22.43 | mut | 63-227-209-67.ptld.qwest.net |
| 16:22.43 | mut | whats ptld? |
| 16:22.48 | coppice | another bug just died in T.38 gateway operation. shame * seems like it will never get its act together and let me plug T.38 in |
| 16:22.52 | Peaceful | ManxPower: Eyeball <=> Screen Detection? Sounds advanced. ;-) |
| 16:24.05 | coppice | mut: ptld, or pldt? |
| 16:25.51 | mut | ptld |
| 16:25.52 | brad_mssw | coppice: thought oej created a t.38 branch for testing |
| 16:25.52 | brad_mssw | i think we'd all really like to see T.38 |
| 16:26.09 | coppice | oej's stuff seems to be going nowhere |
| 16:26.44 | [TK]D-Fender | coppice : Careful with that line of yours... the "Man" is watching ;) |
| 16:27.12 | coppice | is he big and scary? |
| 16:27.24 | mut | .. |
| 16:27.35 | brad_mssw | V150 would be nice too :) |
| 16:27.53 | coppice | i wonder how much v.150 would really be used |
| 16:28.25 | coppice | mut: i think that should be pldt, and its the main telco in the philipinnes |
| 16:28.44 | brad_mssw | heh, well, we'd use it ... for testing modems to credit card processing institutions ... but you're right, there's probably not much need |
| 16:29.09 | zoa | v.150 is for modems, right ? |
| 16:29.15 | brad_mssw | yeah |
| 16:29.22 | coppice | there's a lot of interest in getting V.22bis into spandsp for credit card processing via * |
| 16:29.41 | zoa | does someone have ata's with support for that ? |
| 16:29.43 | zoa | :) |
| 16:29.44 | coppice | zoa: V.150 is like T.38 for normal data modems |
| 16:29.47 | zoa | aha |
| 16:29.53 | zoa | i read something about it i think |
| 16:29.57 | *** join/#asterisk Skarmeth (n=Skarmeth@201009035218.user.veloxzone.com.br) |
| 16:29.58 | zoa | i googled for it a while ago |
| 16:30.05 | zoa | but its just another draft, no ? |
| 16:30.10 | mut | coppice: o, well the ip reverses to that hostname |
| 16:30.16 | coppice | The only thing I've seen supporting V.150 are cisco boxes |
| 16:30.20 | mut | so i dunno |
| 16:30.45 | coppice | mut: 63 is the philipinnes, so i think ptld is just a typo somewhere |
| 16:30.46 | Skarmeth | hi all |
| 16:30.56 | mut | ah |
| 16:30.56 | mut | k |
| 16:31.11 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:31.40 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
| 16:32.22 | *** join/#asterisk ramo (n=ramo@59.92.155.33) |
| 16:32.37 | Skarmeth | I'm about to install an asterisk system on a 64bits machine (Xeon EM64T), and I need to hear some experiênces of people alerdy using it. My system it's formed by a Digium Wildcard TE110P, a TDM04B and a Dell PowerEdge 1800 (Xeon 3.0 EM64T) |
| 16:33.00 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 16:33.14 | coppice | zoa: V.150 was a draft for a long time, but i think its a standard now. there's V.151 and V.152 as well. can't remember what they are |
| 16:33.31 | vlrk | how to enable the offset for polycom 501 any ideas ? |
| 16:33.38 | vlrk | gmt offset |
| 16:33.41 | jsharp | They're all patent encumbered though, right? |
| 16:34.32 | coppice | I can't imagine what they could patent |
| 16:34.44 | coppice | but that's never stopped anyone |
| 16:35.00 | coppice | i saw something the other day about possible patents on T.38 |
| 16:35.25 | zoa | people have patents on everything |
| 16:35.26 | zoa | its a shame |
| 16:36.15 | Hmmhesays | thats cause people want money |
| 16:36.26 | coppice | you can look in the ITU patent declaration database for who claims something, but that database doesn't tell you what they claim |
| 16:36.40 | coppice | the only winners are lawyers |
| 16:37.39 | zoa | indeed |
| 16:38.09 | docelm0 | ZOA! |
| 16:38.17 | coppice | when I have finished my current rework of unicall, i'm going to plug my T.38 gateway support into it, patents or not :-) |
| 16:38.52 | ManxPower | Does anyone happen to have an asterisk-perl example of getting the Caller*ID info for 1.2 (it changed between 1.0 and 1.2) |
| 16:40.32 | key2 | !seen kram |
| 16:40.36 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 16:40.53 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
| 16:42.00 | vlrk | does any body have idea on provisiong of polycom |
| 16:42.28 | ManxPower | *grumble* I'm going to stop helping people unless I start to get some help back |
| 16:43.20 | Hmmhesays | no doubt |
| 16:43.26 | zoa | docelm0! |
| 16:43.32 | OliverX | Have anyone an idea, why i have connection but no sound? |
| 16:43.42 | zoa | look the midget is talking! |
| 16:43.45 | zoa | :p |
| 16:43.47 | OliverX | handy->isdn dritz card over misdn -> sip |
| 16:43.51 | [TK]D-Fender | vlrk : you should be setting it in your DHCP server |
| 16:44.35 | vlrk | i am trying for documentation for that but no hope till now |
| 16:46.51 | [TK]D-Fender | What are you using for your DHCP server? |
| 16:47.17 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 16:48.50 | [TK]D-Fender | vlrk : An you may want to check out section 4.6.1.10.2 of your admin guide..... it looks pretty obvious there... |
| 16:49.20 | vlrk | thanks |
| 16:49.33 | vlrk | is that you mean to say polycom admin guide |
| 16:49.41 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 16:50.04 | DoktorGreg | have a question about sip.conf |
| 16:50.07 | tzanger | vlrk: polycom provisioning is very easy once you get the hang of it |
| 16:50.14 | tzanger | the wiki has a VERY good introduction/tutorial on it |
| 16:50.23 | OliverX | anyone here who user misdn with fritz? |
| 16:50.26 | tzanger | "polycom provisioning" is the search term, and will lead you to a specific example for the 501s |
| 16:50.30 | DoktorGreg | I have a server with two external facing ip addresses |
| 16:51.06 | [TK]D-Fender | vlrk : Yes, the Polycom SoundPoint IP Admisitrator's Guide.... |
| 16:51.13 | vlrk | thx |
| 16:51.30 | [TK]D-Fender | vlrk : Yes, the Polycom SoundPoint IP Administrator's Guide.... |
| 16:51.38 | [TK]D-Fender | can't type today... |
| 16:52.11 | vlrk | can you please give the url |
| 16:52.22 | vlrk | dont mind for this dependency |
| 16:52.44 | DoktorGreg | how do i tell sip.conf wich ip address to use on a given connection? |
| 16:53.26 | [TK]D-Fender | vlrk : www.polycom.com |
| 16:53.38 | vlrk | i tried there |
| 16:53.42 | vlrk | but not with much success |
| 16:53.52 | [TK]D-Fender | vlrk : The guides are all there nice and easy to get to... |
| 16:54.13 | [TK]D-Fender | vlrk : You're not trying very hard... |
| 16:54.27 | vlrk | ok fine |
| 16:54.34 | vlrk | i will try hard and very hard |
| 16:54.41 | DoktorGreg | anyone? |
| 16:54.44 | vlrk | thx for the update |
| 16:54.57 | [TK]D-Fender | vlrk : Here... found from scratch in 10 seconds flat... http://www.polycom.com/resource_center/0,1454,pw-26-482-10533,FF.html |
| 16:55.22 | OliverX | anyone here who user misdn with fritz? |
| 16:57.36 | Qwell[] | DoktorGreg: ie; use a different IP for each peer? |
| 16:57.46 | Qwell[] | DoktorGreg: You don't.. |
| 16:57.59 | blitzrage | anyone got an extra IP501 they want to sell to me? |
| 16:58.37 | blitzrage | and that the phone is already in Canada (it's not worth it to ship across the border and pay for duty) |
| 16:58.40 | OliverX | No person here who use MISDN? |
| 16:59.23 | DoktorGreg | oh from the jfg files |
| 16:59.43 | DoktorGreg | nat=route |
| 16:59.56 | DoktorGreg | sorry for question |
| 16:59.59 | shido6 | blitz new or sed |
| 17:00.03 | shido6 | used |
| 17:00.08 | [TK]D-Fender | blitzrage : Mine ships on the 24th... tired of only running a 301 at home... |
| 17:00.39 | [TK]D-Fender | blitzrage : You could just buy one like everyone else you know... |
| 17:01.29 | Hmmhesays | anyone ever have asterisk just stop accepting regisrations? |
| 17:01.34 | [TK]D-Fender | blitzrage : I did. Mind you I'm going to try to write off every penny of my consulting income where at all possible.... |
| 17:01.43 | FuriousGeorge | is there any known issue where asterisk just kinda occupies zap channels and leaves them off the hook? |
| 17:01.53 | FuriousGeorge | using 1.2.5 zaptel here |
| 17:02.28 | Qwell[] | ariel_: free as in libre |
| 17:02.39 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 17:02.56 | FuriousGeorge | luckily i had the telco roll over to my soon to be non existent DiD and no calls were lost |
| 17:03.52 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 17:05.13 | FuriousGeorge | ive heard that cron-ing asterisk to restart periodically is a sign of paranoia, but im starting not to think so |
| 17:05.29 | FuriousGeorge | granted i 0.1.1 behind |
| 17:05.32 | FuriousGeorge | *im |
| 17:06.16 | [TK]D-Fender | FuriousGeorge : 1.0.6 ? My server stays up for weeks at a time easy |
| 17:06.38 | [TK]D-Fender | FuriousGeorge : Upgrade before the obsolescence grows any further.... |
| 17:06.53 | [TK]D-Fender | FuriousGeorge : And you run out of people to help you ;) |
| 17:07.01 | FuriousGeorge | no |
| 17:07.07 | FuriousGeorge | 1.2.6.0 |
| 17:07.17 | backblue | someone using misdn? what means red leds blinking? |
| 17:07.17 | FuriousGeorge | + 1.1 = 1.2.7.1 |
| 17:07.31 | Qwell[] | 1.2.7.1 + 0.0.0.0 == 1.2.7.1 |
| 17:07.37 | Qwell[] | man, that almost looks like an IP |
| 17:07.38 | FuriousGeorge | i knew it was a bad idea |
| 17:08.05 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 17:08.09 | FuriousGeorge | to try to do arithmetic with versions |
| 17:08.40 | [TK]D-Fender | FuriousGeorge : They only just introduced a 4th decimal position... its not the norm.... |
| 17:09.06 | blitzrage | shido6: used is perfectly fine (and desirable) since it's just for at home |
| 17:09.43 | Qwell[] | blitzrage: Next time you come here, just smuggle one home :p |
| 17:09.53 | FuriousGeorge | yeah, i was wondering why, but then i just figured it was a minor detail :) |
| 17:09.55 | [TK]D-Fender | blitzrage : how much are you ecpecting to get one for? |
| 17:10.10 | blitzrage | [TK]D-Fender: yah, but I'm poor, so I'm looking for a deal :) |
| 17:10.10 | FuriousGeorge | re: the 4th decimal, that is |
| 17:10.42 | [TK]D-Fender | blitzrage : Stop spending all your money on ! ! ! |
| 17:11.10 | blitzrage | [TK]D-Fender: I wouldn't be so stressed out if I was getting ! ! ! :) |
| 17:11.29 | tzanger | ha |
| 17:11.38 | [TK]D-Fender | blitzrage : Having been publicised and touring for it I'm surprised you're not pulling in lots of $ because of *. I know I am now... |
| 17:11.49 | *** join/#asterisk alexis101 (n=alexis@70.54.204.92) |
| 17:11.59 | alexis101 | Hello |
| 17:12.08 | tzanger | blitzrage: you didn't reply to my messages :-( |
| 17:12.15 | blitzrage | tzanger: yes I did |
| 17:12.23 | blitzrage | tzanger: I sent you several messages |
| 17:12.31 | blitzrage | [TK]D-Fender: yah, I wish I was too, but I'm not |
| 17:12.55 | tzanger | no you didn't |
| 17:13.10 | blitzrage | yes I did |
| 17:13.17 | blitzrage | I just pasted them again to you |
| 17:13.22 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 17:13.23 | blitzrage | if you're blocking me, it's not my fault :) |
| 17:13.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:13.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:14.29 | tzanger | I'm not |
| 17:14.34 | alexis101 | i have a little question about rtp port ... is there a prefered number of ports ??? Because I did make samples for config files and by default they put me 10 000 to 20 000 but this is a lot of port to foward on my firewall ! |
| 17:14.38 | blitzrage | tzanger: do you not see my messages to you?> |
| 17:14.55 | Qwell[] | alexis101: You only need a few, really |
| 17:15.12 | tzanger | blitzrage: nope |
| 17:15.15 | tzanger | and I am not ignoring you |
| 17:15.17 | tzanger | are you registered? |
| 17:15.23 | blitzrage | yes |
| 17:15.44 | Qwell[] | alexis101: Don't msg me |
| 17:15.53 | tzanger | blitzrage: ok I am getting tons of "hi" now |
| 17:16.02 | alexis101 | like how mutch ? |
| 17:16.12 | ariel_ | alexis101, I change mine from 10,000 - 20,000 to 10,001 to 11,000 and it works fine for me. But I only have a few phones on my setup. |
| 17:16.30 | blitzrage | tzanger: I just re-authed with nickserv, so that must have fixed it |
| 17:16.31 | Katty | hi lads. |
| 17:16.37 | Katty | i've come to bug you |
| 17:16.39 | justinu | herro |
| 17:16.42 | Katty | and ask a million and one questions |
| 17:16.47 | Katty | k, well, maybe not |
| 17:17.02 | Katty | but i sure will hug you! |
| 17:17.30 | Katty | ! |
| 17:17.30 | Katty | yay! |
| 17:17.30 | Katty | :>>> |
| 17:17.32 | justinu | :) |
| 17:17.51 | [TK]D-Fender | justinu : Put your pants back on! |
| 17:17.52 | alexis101 | ok thanks i have about 11 phones on my setup so i should i stay with 10 000 ports ? |
| 17:17.53 | *** part/#asterisk guigouz (n=guigouz@201-42-160-252.dsl.telesp.net.br) |
| 17:17.58 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 17:17.59 | [TK]D-Fender | ;) |
| 17:18.02 | justinu | i work from home |
| 17:18.05 | justinu | no pants requires ;) |
| 17:18.10 | justinu | required |
| 17:18.15 | [TK]D-Fender | justinu : just referring to your "ring" :D |
| 17:18.50 | Katty | >.< |
| 17:18.54 | Katty | at least it's after lunch. |
| 17:19.01 | b00mer_ | How do I define an outbound trunk in my extensions to use a pri? So that it uses the next available channel? |
| 17:19.02 | Katty | i can handle that sort of mental imagery this time of day |
| 17:19.07 | ariel_ | alexis101, 10,000 is the starting point not how many ports you need. about 200 or 300 is ok I use 1000 |
| 17:19.14 | justinu | haha |
| 17:19.29 | alexis101 | well i know but 20 000 -10 000 = 10 000 no ? |
| 17:19.39 | b00mer_ | I have OUTBOUNDTRUNK=Zap/1 |
| 17:19.46 | [TK]D-Fender | Katty : You wouldn't want to be queasy on an empty stomach, now would you? :) |
| 17:19.47 | *** join/#asterisk holaa (n=sergio@85.137.87.134) |
| 17:19.50 | ariel_ | alexis101, yes and I have from 10,001 to 11,000 on mine only 999 |
| 17:19.51 | b00mer_ | but then it only uses the first channel |
| 17:19.57 | Katty | [TK]D-Fender: course not ;) |
| 17:20.02 | alexis101 | ok thanks a lot :) |
| 17:20.08 | b00mer_ | what is the terminology to specify first available? |
| 17:20.12 | ariel_ | b00mer_, how about a group like =Zap/g0 |
| 17:20.19 | b00mer_ | ahh |
| 17:20.21 | b00mer_ | group |
| 17:20.22 | holaa | Using X-Lite and A@H everything is ok except when I hang up any phone call before establishing connection. If I hang up before that, there is no "Hungup" command (i can see that in logs) and the line stays busy. I mean phoning from outside, cell phone or whatever to any extension. |
| 17:20.32 | b00mer_ | where do I define the group? |
| 17:20.38 | b00mer_ | oh wait... I think I already did!!! |
| 17:20.39 | [TK]D-Fender | holaa : Please read the channel topic... |
| 17:20.40 | b00mer_ | n/m |
| 17:20.42 | b00mer_ | thanks! |
| 17:20.49 | bkw_ | RUN FOREST RUN!!! |
| 17:21.11 | [TK]D-Fender | "Gonna get me a shotgun and kill all the whiteys I seeeeeee!!!!!" |
| 17:21.19 | nahirean | A@H configuration files are like a extensions.hurricane |
| 17:21.19 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:21.38 | holaa | [TK]D-Fender... I think that in this case, using A@H is irrelevant. |
| 17:22.13 | blitzrage | anyone else find asterisk will drop a call after exactly 5 minutes with canreinvite=yes ? |
| 17:22.15 | brettnem | A@H conf files are rediculously complicated. |
| 17:22.16 | b00mer_ | ariel : so if I have group 1 setup in my zapata.conf... then it should be Zap/g1 ? |
| 17:22.21 | nahirean | brettnem, Yepo |
| 17:22.35 | brettnem | nahirean: try phonecall instead |
| 17:22.43 | blitzrage | A@H is stupid imho |
| 17:22.47 | ariel_ | holaa, correct your zap port might not have the correct configuration |
| 17:22.52 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:22.53 | blitzrage | nothing but problems trying to interconnect them |
| 17:22.54 | nahirean | I dont use A@H, I use standard distribution |
| 17:22.57 | ariel_ | blitzrage, it's asterisk |
| 17:23.06 | brettnem | nahirean: http://www.vecsector.com/phonecall |
| 17:23.06 | nahirean | will never use amp to configure anything ;) |
| 17:23.08 | blitzrage | ariel_: my statement re: problems still stands :) |
| 17:23.28 | brettnem | first...the ears.. bwahahahaha |
| 17:23.29 | [TK]D-Fender | blitzrage : No, I'd sooner say "trying to DEBUG it" is. When functional it keeps trolls out of here :) |
| 17:23.29 | ariel_ | blitzrage, yes but it's still plain asterisk with fancy rules you can go around them. |
| 17:23.33 | nahirean | brett, I like to confirugre things by hand ;) |
| 17:23.42 | nahirean | configure* |
| 17:23.47 | brettnem | nahirean: phonecall is not much different than by hand.. |
| 17:24.04 | holaa | ariel_ what exactly do you think could be wrong with the zap port? |
| 17:24.23 | brettnem | hmmm.. somebody stuck a beer in my lunchbox |
| 17:24.28 | Hmmhesays | awesome |
| 17:24.32 | ariel_ | it's not hanging it up or does not know when you hang up. |
| 17:24.38 | nahirean | one beer is but a tease |
| 17:24.42 | brettnem | haha |
| 17:24.43 | ariel_ | holaa, how did you configure it. |
| 17:24.54 | brettnem | a beer and a container of sour cream.. what the heck |
| 17:24.56 | brettnem | :) |
| 17:25.17 | Hmmhesays | nice lunch |
| 17:25.22 | brettnem | mmm |
| 17:25.23 | *** join/#asterisk MrChimpy (n=MrChimpy@smtp-gw.amplefuture.com) |
| 17:25.46 | MrChimpy | hello * people. long time no /join |
| 17:25.55 | holaa | ariel_ i just let default configuration with A@H. When I finish an "usual" conversation "Hungup" is in the logs and everithing ok. But if the caller or me hangs up before establishing any "contact"... no "hungup" in logs and line stays busy. |
| 17:26.01 | brettnem | hello chimpy person |
| 17:26.15 | brettnem | holaa: try /join #freepbx |
| 17:26.39 | holaa | brettnem.. I think that in this case, using A@H is irrelevant. |
| 17:26.44 | MrChimpy | i had a demo of a 3G video IVR type thing today |
| 17:26.57 | brettnem | holaa: then why do you say that you are using it? ;0 |
| 17:27.01 | brettnem | er ;) |
| 17:27.03 | MrChimpy | whole lotta kit for doing not too much |
| 17:27.20 | MrChimpy | is asterisk moving towards taking 3g calls? |
| 17:27.23 | holaa | brettnem.. just to let you know more about the situation :) |
| 17:27.24 | brettnem | 3G video IVR? wtf? sounds like shockwave on a phone |
| 17:27.25 | klasstek | Are there any digium account reps present. If so please /msg me. Please no resellers/developers/contractors/etc... |
| 17:27.30 | ariel_ | holaa, lets go over to the other channel. But your correct his is an issue with the zap channel not a@h it's just people here don't like it so they don't want to help. I use both ways. The both work great. |
| 17:27.31 | [TK]D-Fender | holaa : Based on what? What CLI output and script debbing have you presented us with? How about zapata paramteres and a confirmation about the kind of line / country its being run on? |
| 17:27.35 | Katty | scooby do |
| 17:27.38 | Katty | where are you |
| 17:27.38 | brettnem | holaa: but... it's irrelelvant? |
| 17:27.54 | MrChimpy | brettnem: same as normal IVR, but playing 3gps rather than audio |
| 17:28.08 | MrChimpy | menu screens and stuff are done by sending a 1 frame 3gp file :) |
| 17:28.21 | brettnem | MrChimpy: bah |
| 17:28.24 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-88-155-214-97.red.bezeqint.net) |
| 17:28.28 | [TK]D-Fender | brettnem : Because if it were relevent he'd have to face that virtually nobody here will want to waste that kind of time helping him.... |
| 17:28.50 | brettnem | ;) |
| 17:29.09 | [TK]D-Fender | Thank you russellb! |
| 17:29.11 | brettnem | I don't use A@H anymore.. I've switched to phonecall if I need a GUI |
| 17:29.23 | holaa | [TK]D-Fender... ok, i get the point. Thank you anyways. |
| 17:29.30 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 17:29.33 | ManxPower | The problem with Asterisk@Home is that it's config files are SO complex, nobody here COULD help with it even if they wanted to. |
| 17:29.34 | PakiPenguin | evening |
| 17:29.35 | holaa | ariel_ lets go to the other channel |
| 17:29.41 | [TK]D-Fender | holaa : I'm willing to bend if you'll show us some effort on your part... |
| 17:29.58 | brettnem | ManxPower: exactly. the configs are an absolute mess.. |
| 17:30.03 | MrChimpy | 3g calls turn up as an E1 voice channel which you then yank out the video and audio from |
| 17:30.13 | [TK]D-Fender | holaa : But so far you look like every other "I just did it in the web GUI and 'it doesn't work'" |
| 17:30.23 | MrChimpy | this solution has a load of hardware in it to do the extraction etc which seems excessive |
| 17:30.52 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:30.59 | holaa | [TK]D-Fender... I understand, siriously, you are right. I move to the other channel. |
| 17:31.06 | brettnem | most A@H questions originate from people who 1) don't know how to use asterisk 2) don't care how to use asterisk and 3) are looking for someone to fix it for them. This might not be you but it happens alot. |
| 17:31.10 | MrChimpy | and for the price of a box we could probably get digium to do 3gp support :) |
| 17:31.46 | MrChimpy | yeah and A@H questions get in the way of me blathering about all the cool crap I'm doing ;) |
| 17:31.51 | ariel_ | brettnem, so people like me that uses both and have been help for many years here should no longer help people here... |
| 17:32.17 | [TK]D-Fender | ariel_ : His statement OBVIOUSLY excludes you :) |
| 17:32.29 | [TK]D-Fender | ariel_ : We're talking about the OPTHER 99.999% ;) |
| 17:33.53 | brettnem | ariel_: hey, I help a lot of people in here.. I don't care if they use A@H or not, but tell me what you've done and lets fix it.. right? |
| 17:34.02 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:34.02 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 17:34.08 | brettnem | lots of people with A@H don't know what they've done or how to do things to fix stuff. |
| 17:34.46 | Hmmhesays | which is good for those of us who know what we're doing $$ |
| 17:34.51 | ariel_ | ok but just plain telling them right off the bat to go away without us trying to see what the problem is. Is just plain wrong. |
| 17:35.24 | lzhang | the price of free support. |
| 17:35.28 | *** join/#asterisk tuck3r (n=tucker@24.137.146.210) |
| 17:35.30 | brettnem | a question was presented that would have been more appropriately served in another channel |
| 17:35.33 | brettnem | ~gwypf |
| 17:35.37 | jbot | i guess gwypf is Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 17:35.37 | brettnem | !! |
| 17:35.50 | ariel_ | We all started from not knowning anythigns and some of the people that still ask here don't even know what a conf file are so it's not that much of a problem. |
| 17:35.54 | justinu | fender helps a lot of n00bs... i think he's allowed to tell people to go away if the attitude isn't right |
| 17:36.03 | MrChimpy | has there been any pondering on the subject of an XML i/f for IVR apps? |
| 17:36.06 | [TK]D-Fender | yup.... I've paind my dues |
| 17:36.08 | brettnem | justinu: amen! |
| 17:36.10 | justinu | damn straight |
| 17:36.23 | brettnem | MrChimpy: Mark hates XML (tm) |
| 17:36.39 | MrChimpy | brett: don't blame him. I did. |
| 17:36.46 | brettnem | oh whatever.. :P |
| 17:36.51 | nahirean | No no, you guys are clearly here to be at everyone's beck and call. In fact, you need to /topic yer personal DIDs so you can be called. 24/7 on site.. |
| 17:36.52 | Katty | yes. mister fender should be hugged. |
| 17:36.53 | brettnem | CPL would probably look funny in asterisk |
| 17:36.58 | Katty | [TK]D-Fender: let's hug! |
| 17:37.07 | justinu | group hug! |
| 17:37.11 | brettnem | wahoo! |
| 17:37.12 | MrChimpy | thing is, if i do this interface I can have our java people do all the implementation, just passing AGI commands back and forth using xml |
| 17:37.14 | brettnem | party blob! |
| 17:37.18 | justinu | hehe |
| 17:37.33 | Katty | party blob? |
| 17:37.37 | Katty | this sounds dangerous. |
| 17:37.48 | brettnem | only for those on the bottom |
| 17:37.50 | Katty | just tell me that's not the lap dancer's nickname |
| 17:37.55 | MrChimpy | "blob" as in menstruation, and that sounds so wrong. |
| 17:37.56 | justinu | katty: resistance is futile, you will be assimilated |
| 17:38.06 | *** join/#asterisk Damin_PDA (n=pocketir@62.sub-70-221-3.myvzw.com) |
| 17:38.13 | Katty | justinu: into a hug? :> |
| 17:38.21 | justinu | into the party blob :) |
| 17:38.36 | MrChimpy | "Hey gina, what's with all the tampons, twinkies, chocolate and paracetamol?" "me and the girls are having a BLOB PARTY!" |
| 17:38.59 | Katty | MrChimpy: move along. |
| 17:39.02 | justinu | lol |
| 17:39.04 | MrChimpy | whereas infact they'd probably just spend the evening screaming at the nearest available male |
| 17:39.06 | Katty | next! |
| 17:39.09 | MrChimpy | :) |
| 17:39.12 | brettnem | wtf?! |
| 17:39.25 | Katty | that's so revolting. |
| 17:39.27 | brettnem | ok, that's not the blob to which I was referring to. |
| 17:39.33 | Katty | and i'm a female! |
| 17:39.45 | iDunno | Katty: are you?! woah! |
| 17:39.46 | iDunno | ;) |
| 17:39.55 | justinu | lol |
| 17:40.03 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 17:40.12 | Katty | yes, RUN |
| 17:40.24 | brettnem | ok.. well, it's been fun |
| 17:40.29 | Katty | before i, uhh...do something dangerous. |
| 17:40.29 | Katty | right. |
| 17:40.44 | MrChimpy | well, you learn something every day on #asterisk. when in the UK buy "jam rags" when you're "on the blob". |
| 17:40.55 | Katty | oh god. make it stop |
| 17:41.02 | MrChimpy | i'll stop now :) |
| 17:41.13 | ariel_ | argh |
| 17:41.23 | tzanger | ok what the fuck is mkinitrd trying to use ash for. jesus |
| 17:41.28 | tzanger | and it throws no errors if it can't find it |
| 17:42.10 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 17:42.20 | tzanger | oh no ash is there ok |
| 17:42.24 | tzanger | so wtf |
| 17:42.47 | holaa | is it x100p "so shitty" as I have read? |
| 17:42.55 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:43.03 | Qwell[] | holaa: yes, it's quite shitty |
| 17:43.51 | holaa | but... in what ways? sound quality? it hangs up? ... |
| 17:44.03 | ariel_ | holaa, the x100p have many problems with just that and echo as well |
| 17:44.17 | ariel_ | but depending on your system it's something most can live with and fix. |
| 17:44.32 | *** join/#asterisk kn0x (n=pinochle@71.194.235.251) |
| 17:44.38 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:44.39 | ariel_ | irq's sharing is a nono |
| 17:44.40 | Katty | ariel_: :> |
| 17:44.47 | kn0x | is D-Fender still here |
| 17:44.49 | kn0x | ? |
| 17:44.57 | ariel_ | Katty, did you ever get pictures from the other night? |
| 17:45.16 | ariel_ | I have new ones from my dauthers 3 year old sears sitting.... |
| 17:45.33 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 17:46.02 | *** part/#asterisk tuck3r (n=tucker@24.137.146.210) |
| 17:46.42 | blebleble | i'm trying to reboot a polycom phone remotely and am trying 'sip notify polycom-check-cfg $ext' yet not comming up, anyone have a suggestion on how to do this or if i'm doing something incorrectly |
| 17:46.55 | [TK]D-Fender | yup |
| 17:47.10 | [TK]D-Fender | justinu : See, another one I helped... today even ;) |
| 17:47.11 | tzanger | blebleble: there's a script ot reboot polycom phones... |
| 17:47.23 | tzanger | blebleble: make sure you touch the config file so it sees that it's newer than what it has internally |
| 17:47.26 | tzanger | that's basically what it does |
| 17:47.34 | [TK]D-Fender | tzanger : I'm well scripted to pulling the plug on my PoE switches :D |
| 17:47.34 | NewSole | Question... Has Anyone used VegaStream Equipment |
| 17:47.45 | [TK]D-Fender | kn0x : 'sup? |
| 17:47.49 | kn0x | ahh thanks |
| 17:47.54 | lzhang | you can also lynx to the phone's web interface and just hit submit on any page, should reboot it |
| 17:47.54 | blebleble | tzanger: yah i got it off polycoms site but couldnt get it to work |
| 17:48.07 | kn0x | something exploeded [TK]D-Fender |
| 17:48.24 | kn0x | i rebooted and started asterisk and every single module errored |
| 17:48.41 | tzanger | blebleble: touch the /path/to/ftp/dir/[macaddress_of_phone].cfg |
| 17:48.43 | [TK]D-Fender | kn0x : .... |
| 17:48.44 | tzanger | blebleble: then send the notify |
| 17:48.50 | kn0x | is this svn related? |
| 17:49.27 | kn0x | lol, i didnt mess with anything i swear.. haha :) |
| 17:49.35 | blebleble | tzanger: can this work with tftp setup? |
| 17:49.43 | [TK]D-Fender | kn0x : PM me the new pass. |
| 17:50.35 | tzanger | blebleble: yes, but touch /tftpboot/[macid_of_phone].cfg BEFORE sending the notify |
| 17:51.44 | blebleble | ok thank you |
| 17:51.54 | tzanger | man that send notify is so much nicer than the script |
| 17:51.54 | tzanger | wow |
| 17:52.02 | tzanger | very cool, who added that, olle? |
| 17:52.05 | justinu | heh |
| 17:52.07 | tzanger | probably. thank you again, olle |
| 17:53.06 | Assid | worth upgrading from 1.2.5 ? |
| 17:54.08 | blebleble | tzanger: like a charm, thanks a lot |
| 17:55.02 | tzanger | blebleble: the notify is literally telling the polycom to see if the server config is newer than the config it has, if it hasn't changed, why would the phone reboot? :-) |
| 17:57.44 | blebleble | yah, is there a better way than echoing somthing stupid like "<!-- stupid -->" into the config file for it to update the time etc etc ? |
| 17:57.45 | [TK]D-Fender | Hey, kn0x here has just developed a NAST crash on his box and spews out errors like this on almost all .so's being loaded : [res_musiconhold.so]Apr 19 12:56:27 WARNING[9815]: loader.c:726 __load_resource: new style res_musiconhold.so (0x30) loaded RTLD_LOCAL |
| 17:57.56 | Assid | alrite something def. wrong |
| 17:58.11 | Assid | my options.. for voicemail.. is being used as an email address |
| 17:58.12 | [TK]D-Fender | and it repeats for most modules. Any clue why that is, given that it was working just fine right before being rebooted? |
| 17:58.37 | file | that's not a problem, just some debugging info |
| 17:59.02 | justinu | blebleble: just run "touch" on the config file |
| 17:59.17 | [TK]D-Fender | file : Well EVERYTHING is throwing off warnings now... |
| 17:59.17 | blebleble | oh i thought that would overwrite it |
| 17:59.38 | tzanger | blebleble: just touch it |
| 17:59.43 | file | loader is undergoing changes |
| 17:59.44 | blebleble | justinu: thanks |
| 17:59.46 | tzanger | literlly "touch /tftpboot/004355889933.cfg |
| 17:59.57 | tzanger | blebleble:all it does is update the last modified time wihtout actually changing the file |
| 18:00.04 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D12B.access.telenet.be) |
| 18:00.27 | ariel_ | with all these touch going around I now have the stupid song I touch my self in my head.... |
| 18:00.44 | blebleble | yah got it, thanks |
| 18:01.18 | [TK]D-Fender | kn0x : your modules folder looks like an older version.... |
| 18:01.41 | kn0x | should i delete it? |
| 18:01.51 | kn0x | i upgraded from 1.2.2 cvs |
| 18:03.04 | kn0x | is this svn related? |
| 18:03.07 | kn0x | ahh |
| 18:03.59 | [TK]D-Fender | kn0x : I blew away your modules folder before doing a make clean, and then a new make... the timestamps don't match the .o file dates.... |
| 18:04.04 | *** join/#asterisk crochat (i=crochat@84-74-147-2.dclient.hispeed.ch) |
| 18:04.13 | [TK]D-Fender | kn0x : What did I tell you about CVS/SVN? |
| 18:05.36 | [TK]D-Fender | kn0x : I just flushed the folder and am redoing the "make install" |
| 18:05.50 | justinu | heh |
| 18:06.13 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 18:07.17 | [TK]D-Fender | UGH... |
| 18:09.02 | Assid | okay im getting tons of dtmf issues |
| 18:09.23 | Assid | using the gsm codec |
| 18:09.26 | Assid | softphone |
| 18:10.51 | [TK]D-Fender | ? |
| 18:11.47 | bkw_ | don't do inband |
| 18:11.48 | bkw_ | duh |
| 18:11.50 | bkw_ | ;) |
| 18:11.54 | bkw_ | rfc2833 |
| 18:14.05 | Damin_PDA | bkw: Giordanos deep dish..need I say more? |
| 18:14.40 | cytrak | has any configuration files changed from 1.2.5 to 1.2.7 ? |
| 18:14.49 | bkw_ | Damin_PDA, OMG no |
| 18:15.34 | Damin_PDA | I'm at a pizzeria uno...close, but not quite the same...;) |
| 18:17.15 | Katty | you know what i want? |
| 18:17.17 | Katty | david gilmour |
| 18:17.20 | Katty | back in the late 60s |
| 18:17.37 | Damin_PDA | katty: me too. |
| 18:17.56 | Katty | Damin_PDA: i'd take his strat too |
| 18:18.00 | Katty | Damin_PDA: that was a /nice/ strat |
| 18:18.13 | Damin_PDA | katty: not sure what my wife would think.... |
| 18:18.47 | Katty | who cares? |
| 18:18.51 | Damin_PDA | I am full! |
| 18:18.53 | Katty | this is david gilmour we're talking about! |
| 18:18.53 | b00mer_ | anybody know of voip providers who will let me set my own callerid with SIP or IAX? |
| 18:19.01 | b00mer_ | prefer IAX |
| 18:19.23 | Damin_PDA | b00mer: callerid name? none.. |
| 18:19.28 | tzanger | b00mer_: both asterlink and nufone |
| 18:19.52 | Damin_PDA | b00mer: what tzanger said... |
| 18:20.03 | b00mer_ | tzanger : you used either? recommendations? |
| 18:20.25 | Katty | Damin_PDA: the cover of ummagumma is yum too |
| 18:20.30 | tzanger | I use both, both work well, both ahve almost identical rates |
| 18:20.35 | Damin_PDA | b00mer: name is set in national cnam....not by voip provider.. |
| 18:20.42 | Katty | anthm: can you invite david gilmour to cluecon please :P |
| 18:20.55 | anthm | k |
| 18:20.57 | Katty | thx. |
| 18:21.02 | bkw_ | Katty, who is he? |
| 18:21.07 | Damin_PDA | katty: drop some acid and listen to that album.... |
| 18:21.21 | Katty | bkw_: http://www.clapton.addr.com/PinkFloyd/DavidGilmour/BlackStrat.jpg |
| 18:21.27 | bkw_ | ya ya |
| 18:21.29 | b00mer_ | Damin_PDA : i just care about the number |
| 18:21.41 | Katty | Damin_PDA: i don't need to take acid to listen to the album |
| 18:21.47 | Damin_PDA | b00mer: what tzanger said...then |
| 18:21.51 | Qwell[] | wow, that's the second time I've seen addr.com today |
| 18:21.54 | b00mer_ | cool thanks! |
| 18:22.02 | Qwell[] | The first time, was to represent a a fake email address |
| 18:23.16 | Assid | umm.. is it worth upgrading from 1.2.5 -> current? |
| 18:23.38 | Damin_PDA | yes |
| 18:23.44 | [TK]D-Fender | Assid : So you can have all the latest bugs, instead of the same old ones! |
| 18:24.42 | cytrak | any config files change though ? |
| 18:24.51 | Assid | hehe |
| 18:25.15 | cytrak | i got all my files in subversion and I would hate to redo all of them (-: |
| 18:25.39 | Assid | hrmm .. touching less than 1MB/sec from digium.. must be some load there |
| 18:25.47 | Damin_PDA | cytrak: minor changes... |
| 18:26.04 | salviadud | what would i need to change in the source so instead of extension => bla bla bla i do "yo => bla bla bla" ? |
| 18:26.41 | Assid | Damin_PDA: the changelog doesnt really provide that does it ? |
| 18:27.42 | Assid | arrgh |
| 18:27.50 | Assid | i didnt load the kernel with unloading features |
| 18:28.20 | salviadud | ¬¬ |
| 18:28.38 | salviadud | i'll do it i tell ya! i will change extension to yo |
| 18:29.08 | OloBola | nufone is up again: |
| 18:29.13 | OloBola | http://www.nufone.net/press/ |
| 18:29.27 | OloBola | yeah |
| 18:29.32 | Assid | i thougth they havingissues cause of that carrier |
| 18:32.06 | cytrak | Damin: minor changes like "if I use my old ones it won't cause any issues" type of minor |
| 18:32.10 | OloBola | 800 numbers got yanked from 3PM est yesterday until sometime today. |
| 18:32.25 | Damin_PDA | yes |
| 18:32.25 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 18:32.46 | vlrk | [TK]D-Fender : |
| 18:33.02 | vlrk | i went through the provision section of the admin guide |
| 18:33.18 | [TK]D-Fender | vlrk : I told you exactly what section #..... |
| 18:33.33 | vlrk | 4.6.1.10 |
| 18:34.12 | vlrk | 4.6 is total of the configuration files |
| 18:34.58 | ariel_ | OloBola, my 800 numbers did not go down at all. They are still working fine here. |
| 18:35.01 | [TK]D-Fender | vlrk : 4.6.1.10.2 is the section on SNTP.... have you considered using a text search for it? |
| 18:36.25 | Assid | err.. i read this somewhere a while back.. on how to compile GSM with cFLAGS or something |
| 18:37.05 | tainted- | anyone have problems with sip peers that lose registration? |
| 18:37.14 | vlrk | [TK]D-Fender : with your direction i gone through the whole section |
| 18:37.22 | [TK]D-Fender | tainted- : only those w/o "qualify" |
| 18:37.34 | [TK]D-Fender | vlrk : Do a text search for SNTP..... |
| 18:37.38 | Assid | #K6OPT = -DK6OPT |
| 18:37.48 | Assid | does that affect pentium4 as well? |
| 18:38.57 | tzanger | Assid: the codecs all use the correct cflags as set in the toplevel makefile |
| 18:40.09 | brettnem | ~seen elg |
| 18:40.24 | jbot | elg <n=fugalh@hfugal.NMSU.Edu> was last seen on IRC in channel #utah, 46m 6s ago, saying: 'righteous indignation is ok'. |
| 18:40.24 | Assid | so by just setting PROC=i686 is that all i gotta do ? |
| 18:40.59 | Assid | isnt K6 for the amd machines? |
| 18:41.19 | brettnem | ugh.. I just erased 38 baby prank calls from my VM box! |
| 18:41.31 | Qwell[] | brettnem: undo! undo! |
| 18:41.47 | brettnem | undo? no. delete! delete! |
| 18:41.54 | brettnem | 761,761,761,761.. |
| 18:42.35 | brettnem | "bah bah bah bah.. hello glue.. stupid glue... hey what do you want for a snack.. don't do that to your sister.. put that phone down..." |
| 18:42.56 | tzanger | Assid: yes, but I'd say "PROC=pentium4" |
| 18:43.12 | Assid | <PROTECTED> |
| 18:43.13 | Assid | okay |
| 18:44.24 | Assid | so i dont need to enable #K6OPT = -DK6OPT ? |
| 18:44.53 | Assid | having it as pentium4 should take care of itself ? |
| 18:45.17 | X-Gen | hey freaks |
| 18:45.38 | brettnem | ~seen freaks |
| 18:45.41 | jbot | freaks <~freaks@23.pool1.ftthtokyo.att.ne.jp> was last seen on IRC in channel #debian, 528d 9h 28m 39s ago, saying: 'it's somewhere else, i suspect it'saa in /etc/ somewhere but i just dunno which file ...'. |
| 18:45.54 | brettnem | freaks dosen't appear to be in here |
| 18:46.11 | brettnem | ftthtokyo eh? |
| 18:46.18 | X-Gen | brettnem: id check again. since when do u believe a bot ? |
| 18:46.34 | brettnem | bots never lie |
| 18:52.43 | vlrk | [TK]D-Fender : i have a small doubt |
| 18:53.22 | [TK]D-Fender | vlrk : Good... those are easier to ignore :) |
| 18:53.22 | vlrk | is polycom while rebooting every time loads the firm ware |
| 18:53.52 | [TK]D-Fender | vlrk : if you're upgrading expect it to do that a few times, then to stabilize |
| 18:53.58 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
| 18:54.54 | vlrk | does it checks it self that the firm ware which it has already is same as the one in the sntp server so it wont download it . Is this kind of check there in the Polycom? |
| 18:55.55 | Assid | tzanger: dont you have to do anything for zaptel? |
| 18:56.16 | Netgeeks | Hey, any thoughts on this issue: I have a 1.2.6 install, VERY basic dialplan, three test extensions that 1,Answer 2,Wait(1) and either 3,Echo(), 3,VoiceMailMain(), or 3,MusicOnHold()..... |
| 18:56.32 | mmlj4 | oh, this is rich: "i don't know anything about VoIP, i can't even make asterisk work on my box here at home, but I want to start a company, please do it for me" |
| 18:56.53 | Qwell[] | mmlj4: You'd be surprised how common that is |
| 18:56.55 | brettnem | mmlj4: here's my paypal account, please wire $3,000,000 to me |
| 18:57.06 | [TK]D-Fender | vlrk : it does a version check. |
| 18:57.22 | Netgeeks | When calling from an attached sip ua (reinvite=no for future reference), the Echo works perfect, but the voicemailmain and musiconhold result in nothing but silence on the SIP ua... |
| 18:57.42 | vlrk | do we need to enable any flag in configuration file to do that or does it automatically do that ? |
| 18:57.42 | [TK]D-Fender | vlrk : Keep in mind if you upgrade from 1.5 to 1.6 the config files may no longer be compatible and its suggested to rebuild |
| 18:57.44 | brettnem | Netgeeks: do you see the apps actually running on the CLI? |
| 18:57.45 | tzanger | Assid: ? |
| 18:57.47 | Netgeeks | Two extensions on the system can also call each other just fine. There are no CLI errors |
| 18:57.51 | Netgeeks | brett yes |
| 18:57.55 | Netgeeks | CLI looks perfect |
| 18:58.00 | tzanger | [TK]D-Fender: the are in fact incompatible. :-) |
| 18:58.05 | Qwell[] | Netgeeks: So...I got only about 2500 concurrent calls |
| 18:58.16 | ariel_ | reinvite hummm should it not be canreinvite=no |
| 18:58.24 | Netgeeks | Qwell: 5000 legs on what SunFire? |
| 18:58.27 | brettnem | Netgeeks: are they all on the same network |
| 18:58.33 | Qwell[] | 2500 legs.. |
| 18:58.35 | [TK]D-Fender | Netgeeks : incompatible codec? |
| 18:58.39 | Netgeeks | sorry, it is canreinvite=no, my bad for being short |
| 18:58.45 | brettnem | Qwell: couldn';t be asterisk eh? |
| 18:58.50 | Netgeeks | sip UA's are using ulaw only |
| 18:59.00 | brettnem | Netgeeks: do you allow ulaw on the server? |
| 18:59.11 | brettnem | [TK]D-Fender: woudn't that result in a 488 |
| 18:59.11 | ariel_ | Netgeeks, one way audio is normally nat issues. |
| 18:59.24 | ariel_ | you do have nat=yes on the sip accounts settings |
| 18:59.35 | Qwell[] | Netgeeks: was using -sf uac_pcap.xml, from my box to the sunfire...the sunfire hit about 90% cpu |
| 18:59.37 | brettnem | [TK]D-Fender: were you telling me about that UT-Starvox phone? |
| 18:59.43 | Qwell[] | I think it was like 80% kernel |
| 18:59.47 | Qwell[] | 10% or so user |
| 18:59.52 | twisted[asteria] | omg becky |
| 19:00.07 | Qwell[] | and I saw similar results when using localhost |
| 19:00.08 | Netgeeks | one is a 7960 and one is a Poly 601. ulaw is enabled on both. both phones share a private rfc1918 address space with the server on a dumb 100Mbps switch |
| 19:00.17 | Netgeeks | nat=no is set for both phones |
| 19:00.33 | brettnem | does the server have multiple IPs? Have you checked the SDP? |
| 19:00.46 | Netgeeks | Qwell: did you look at the cpu load breakdown on the server? |
| 19:00.48 | Qwell[] | Netgeeks: How did you know that it was pci/interrupt stuff that was causing your bottleneck? |
| 19:00.57 | Qwell[] | Netgeeks: I was just watching it in top |
| 19:01.08 | Assid | tzanger: nevermind.. found it.. config_mmx for zaptel |
| 19:01.09 | Qwell[] | was at like 20 load I believe |
| 19:01.25 | Netgeeks | brettnem: the server has two ethernet interfaces, eth0 has 1918 address, eth1 is set down and the ip adress for eth0 is set as the bind address in sip.conf |
| 19:01.32 | Qwell[] | it was doing a steady 100 calls per second too |
| 19:01.54 | Netgeeks | Qwell: no, I mean did you see what the breakdown on cpu load what, i.e. user, system, interrupt, etc. |
| 19:01.57 | Qwell[] | Netgeeks: How can I see the cpu load breakdown? |
| 19:02.43 | [TK]D-Fender | brettnem : Yes, UT Stracom 6700 PPC |
| 19:02.48 | [TK]D-Fender | Starcom* |
| 19:02.49 | *** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-26-93.w83-114.abo.wanadoo.fr) |
| 19:02.50 | Netgeeks | use the latest version of top, select show all cpus and in the cpu % load section, you should see it broken out by user, system, idle, and interrupts |
| 19:03.07 | Qwell[] | Netgeeks: I only saw user, kernel, and system I think |
| 19:03.15 | Qwell[] | didn't show interrupts |
| 19:03.40 | Qwell[] | will figure out how to show all CPUs, and perhaps upgrade |
| 19:03.53 | Qwell[] | with 6 cores, should 100% be the max still? |
| 19:03.54 | Netgeeks | brettnem: any thoughts, I'm befuddled.. especially since echo works fine (rtp going both ways). Only seems when I'm having the sserver source the audio I get none |
| 19:04.11 | Hmmhesays | hey file |
| 19:04.12 | Netgeeks | 600% should be the max for the asterisk process in top |
| 19:04.22 | Qwell[] | it was dying at about 90% |
| 19:04.25 | justinu | Netgeeks: your issue sounds like the async rtp problem |
| 19:04.30 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:04.38 | justinu | Netgeeks: are your UAs sending RTP to asterisk to "kick off" the call? |
| 19:04.39 | Qwell[] | only like 20mbit |
| 19:04.44 | Netgeeks | justinu: how could that be if echo works fine? |
| 19:04.49 | brettnem | Netgeeks: I'd inspect the SDP |
| 19:04.55 | brettnem | justinu: no NAT here.. |
| 19:05.11 | justinu | Netgeeks: echo seems to work differently, IME |
| 19:05.12 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:05.21 | justinu | i never really bothered to look into it tho |
| 19:05.36 | brettnem | echo should work differently.. |
| 19:05.42 | Netgeeks | UA sends invite to server, server answers, server waits 1 second, server runs app (either echo, musiconhold, or voicemailmain) |
| 19:05.49 | *** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-26-93.w83-114.abo.wanadoo.fr) |
| 19:06.06 | Netgeeks | CLI indicates no errors and everything looks good from that view |
| 19:06.18 | Netgeeks | SIP debug looks right on all three calls |
| 19:06.23 | justinu | right, but does RTP start flowing? |
| 19:06.25 | [TK]D-Fender | Netgeeks : While in echo, I'd confirm the codec in use... |
| 19:06.31 | Netgeeks | okay, brb |
| 19:06.53 | Netgeeks | will run tcpdump and see if rtp is going both ways |
| 19:06.58 | [TK]D-Fender | Since its the only thing returning audio I don't trust that it doesn't just pump the packet back as-is |
| 19:06.58 | justinu | or rtp debug |
| 19:07.49 | *** join/#asterisk Creathir (n=Creathir@207.71.17.206) |
| 19:07.52 | ariel_ | you did say the phones and the asterisk box are all on the same network or are they being connected via a vpn. |
| 19:08.04 | Creathir | question about g729 codecs |
| 19:08.32 | Creathir | I am trying to use the opensource codec for testing purposes (before we go production and get the lics) and I am getting an ELF class error |
| 19:08.36 | Creathir | any ideas? |
| 19:09.24 | Netgeeks | ariel_ they are all on the same physical network |
| 19:09.48 | justinu | what type of UAs |
| 19:09.54 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 19:10.49 | Hmmhesays | anyone know how long it takes iaxtel to turn a number on? |
| 19:11.40 | ariel_ | hummm are they still working |
| 19:12.12 | Netgeeks | justinu: one is a cisco 7960 and the other is a Polycom 601 |
| 19:12.27 | justinu | hmm, those phones should send RTP as soon as they go off hook |
| 19:12.36 | Qwell[] | Netgeeks: off the top of your head, do you know what the interrupt usage looks like? |
| 19:12.39 | Netgeeks | and rtp debug as well as tcpdump show no rtp traffic from the server to the ua when the app is either voicemailmain or musiconhold |
| 19:12.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:13.19 | Netgeeks | Qwell: Cpu0 : 2.3% us, 0.0% sy, 97.7% ni, 0.0% id, 0.0% wa, 0.0% hi, 0.0% si |
| 19:13.20 | ariel_ | you did say your using ulaw |
| 19:13.21 | justinu | netgeeks how about rtp from UA to asterisk? |
| 19:13.29 | Qwell[] | Netgeeks: Which one is interrupts? |
| 19:13.46 | Netgeeks | Qwell: that is one of my top lines, hi = hardware interrupts si = software interrupts |
| 19:13.58 | Qwell[] | so hi is the important one? |
| 19:14.12 | Netgeeks | Justinu: there are the expected amount of rtp packets flowing from the UA's to the server |
| 19:14.22 | Dandan | is sip encryption still not implemented? |
| 19:14.43 | justinu | Netgeeks: that's odd then... |
| 19:14.54 | Netgeeks | Qwell: a high load on either hi or si is indicitive of interrupt related bottleneck... either of them hitting close to 80% or more on any cpu means you have an interrupt bottleneck |
| 19:14.56 | justinu | it's on the right ports, etc? |
| 19:15.14 | justinu | ie, the RTP being sent from the UA is being sent to the port that * wants to get it on? |
| 19:15.17 | Qwell[] | Netgeeks: and all those combined are "cpu usage"? |
| 19:15.33 | Netgeeks | justinu: the rp ports appears to be those agreed upon during setup |
| 19:15.38 | Netgeeks | rtp ports, that is |
| 19:15.48 | key2 | anyone has experiances with cirpack ? |
| 19:15.49 | Qwell[] | will install procps when I get home then...hopefully there is something workable |
| 19:16.25 | Netgeeks | asterisk makes no attempt to send even one rtp packet for voicemailmain or musiconhold so I can't tell if it's trying the right rtp port, but for echo it does |
| 19:16.35 | justinu | netgeeks, not sure what the deal with voicemail is, but I remember needing ztdummy to get MoH working |
| 19:16.44 | justinu | does playback work? |
| 19:17.05 | Netgeeks | there is a functional te410p card working with functioning zaptel channels via a pri, so I know we have a timing device |
| 19:17.09 | ariel_ | ztdummy is needed for meetme and iax trunking not moh |
| 19:17.12 | justinu | ic |
| 19:17.16 | Netgeeks | let me test playback, one moment |
| 19:17.50 | [TK]D-Fender | Netgeeks : you don't even need a timing source for enerything you've mentioned so far... |
| 19:17.52 | ariel_ | Netgeeks, have you stopped asterisk and restarted it? |
| 19:18.07 | justinu | the only other thing I can think of is to set the debug level high and turn on full logging |
| 19:18.17 | justinu | if you haven't yet |
| 19:18.25 | tainted- | i don't remember needing anything for moh except mpg321 and asterisk tar ball |
| 19:18.37 | LostFrog | mpg321?? |
| 19:18.38 | ariel_ | mpg123 not mpg321 |
| 19:18.40 | justinu | oh... |
| 19:18.41 | LostFrog | You mean mpg123 |
| 19:18.43 | tainted- | yea that one |
| 19:18.50 | ariel_ | besides format_mp3 is better |
| 19:18.51 | tainted- | .51r or something |
| 19:18.54 | justinu | the reason I needed ztdummy for MoH was because I was doing it asyncronously |
| 19:18.55 | Creathir | nvm... found that it is my EMT64 CPU not liking the binary I downloaded... going to have to recompile |
| 19:19.03 | justinu | playing out MoH during early media phase with no incoming RTP :) |
| 19:19.47 | *** join/#asterisk Maxxed (n=whyman@65.59.245.122) |
| 19:19.47 | tainted- | that's crazytalk |
| 19:19.56 | justinu | lol, not quite |
| 19:20.01 | tainted- | ;) |
| 19:20.25 | justinu | i can offer my users custom ringback tones |
| 19:20.31 | justinu | when people call their numbers |
| 19:20.56 | tainted- | that is going to a bane to voip |
| 19:21.01 | justinu | all the kiddies want to have little jon as their ringback tone |
| 19:21.01 | Netgeeks | ariel: stopped asterisk, restarted, upgraded (from 1.2.4 to 1.2.6), restarted asterisk, deleted astdb, sprinkled the server with star dust, called in a occultist to check for ghosts, etc. |
| 19:21.06 | tainted- | i hate ringback |
| 19:21.20 | Netgeeks | Playback acts just like the other non-functioning apps |
| 19:21.32 | tainted- | thinking about ringback makes me want to punch babies! |
| 19:21.32 | [TK]D-Fender | Netgeeks : And the codec check on echo? |
| 19:21.34 | LostFrog | Netgeeks: did you call your mother for advice |
| 19:21.40 | tainted- | lol |
| 19:21.47 | justinu | netgeeks: rtp debug shows you the RTP packets being sent from the UA to *? |
| 19:22.01 | Netgeeks | justinu: yes on the rtp being sent from ua to server |
| 19:22.17 | justinu | you checked it with both tcpdump, and rtp debug from the CLI? |
| 19:23.14 | Netgeeks | both rtp debug and tcpdump show the same thing, rtp going both ways for echo, and single from ua to server for other apps |
| 19:23.17 | justinu | ok |
| 19:23.21 | justinu | just making sure ;) |
| 19:23.27 | justinu | weird shit |
| 19:23.33 | *** join/#asterisk boch (n=fran@unirc.com.ar) |
| 19:23.49 | Netgeeks | Apr 19 11:08:04 WARNING[4514]: app_queue.c:3102 queue_exec: Unable to join queue 'admin' |
| 19:23.57 | Netgeeks | ack, wrong past |
| 19:24.01 | boch | hi mates, simple question: is it possible to force a sip ep to unregister from the cli ? |
| 19:24.14 | Netgeeks | anyway, yeah, I'm befuddled |
| 19:24.37 | justinu | you've got full logging enabled? |
| 19:24.38 | Netgeeks | yes |
| 19:24.54 | Netgeeks | can't find any errors anywhere during the bad calls |
| 19:25.17 | Netgeeks | debug looks good, no warning or errors, no warning or errors in the dialplan |
| 19:25.25 | [TK]D-Fender | Netgeeks : So far you get audio from the only app that doesn't have to PROCESS it... I'd still like to hear what codec the call comes up as... becuase if Echo works, then clearly RTP is flowing.... |
| 19:25.37 | justinu | you don't have anything like "externip" set, do you? |
| 19:25.43 | [TK]D-Fender | Netgeeks : just nothing NEW.... which for me only leaves Codec to blame.... |
| 19:25.46 | Maxxed | im trying to monitor all outbound calls from any extent that starts with a 25xx, but not monitor anybody else, you guys know of an easy way to do this |
| 19:25.51 | Netgeeks | one moment, dialing the echo app and capturing rtp data |
| 19:25.52 | Maxxed | i can think of a few ways |
| 19:26.01 | Maxxed | using wacky agi scriptage |
| 19:26.05 | [TK]D-Fender | Netgeeks : forget capturing it, just do a sip show channels! |
| 19:26.08 | justinu | GotoIf |
| 19:26.12 | Maxxed | but cant i like use if's in the exentions .conf |
| 19:26.13 | Maxxed | ah! |
| 19:26.18 | Maxxed | gotoif sweet |
| 19:26.30 | Maxxed | thx :) |
| 19:26.33 | justinu | np |
| 19:26.53 | Netgeeks | rtp during echo = ulaw, rtp during others = ulaw |
| 19:27.16 | [TK]D-Fender | GotoIf($["${EXTEN:0:2}] |
| 19:27.41 | justinu | Netgeeks: all I can think of is that you over looked something very simple/basic, or your compile broke horribly somehow. |
| 19:27.41 | [TK]D-Fender | GotoIf($["${EXTEN:0:2}"="25"]?somewhere) |
| 19:27.46 | *** join/#asterisk strom (i=storm@jihad.carbomb.org) |
| 19:27.57 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:28.10 | justinu | Netgeeks: either way, I'm curious what the ultimate fix is. |
| 19:28.30 | Netgeeks | justinu: yeah, and I've triple checked everything. I'm about to tell the client to wipe the server clean and rebuild it from scratch |
| 19:29.18 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:30.21 | ariel_ | Netgeeks, well another suggestion is to upgrade to 1.2.7.1 and rebuild everything and try again. |
| 19:31.21 | justinu | Netgeeks: cli command "show translation" looks ok? |
| 19:31.27 | justinu | ulaw codec module loaded? |
| 19:31.55 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 19:31.56 | asterboy | where do I get digg.com headlines only? |
| 19:32.02 | Qwell[] | asterboy: digg.com? |
| 19:32.16 | asterboy | just headlines? |
| 19:32.25 | Netgeeks | codec_ulaw.so and format_slin.so both loaded |
| 19:32.26 | Qwell[] | dunno, some rss feed? |
| 19:32.40 | Netgeeks | format_sln.so actually |
| 19:32.50 | Netgeeks | show translation looks fine |
| 19:32.55 | dpryo | Is there any good ways to debug PRI ties? I'm trying to connect an asterisk with avaya.. |
| 19:33.23 | justinu | app_echo simply just spits back any RTP it recieves |
| 19:33.58 | asterboy | well I guess this is close enough: http://digg.com/rss/index.xml |
| 19:34.26 | Netgeeks | okay, I'm going to wipe the modules directory clean and rebuild from scratch |
| 19:34.30 | justinu | netgeeks: asterisk thinks the channel is "up" after you've answered, right? |
| 19:34.32 | Netgeeks | will be back in 10 minutes with results |
| 19:34.33 | tzanger | dpryo: what are you trying to debug? |
| 19:35.17 | dpryo | tzanger: First of all, the link state.. |
| 19:35.22 | Hmmhesays | hmm suddenly i'm gettting choppy audio on my ivr menues |
| 19:35.26 | dpryo | tzanger: I'm not sure the cable is correct |
| 19:36.02 | Assid | err.. what the hell is up with this: Apr 20 01:05:12 WARNING[14531]: file.c:509 ast_openstream_full: File /var/lib/asterisk/mohpublic/beethoven__violin_sonata_n does not exist in any format |
| 19:36.09 | Assid | the file exists.. |
| 19:36.14 | Assid | thats how it got the name |
| 19:36.24 | *** join/#asterisk david-c (n=david@monkey.n2net.net) |
| 19:36.33 | GerbilWrk | looksin var/lib/asterisk/mohpublic? |
| 19:36.42 | tzanger | dpryo: zap show span 1 ? |
| 19:37.51 | GerbilWrk | -looks |
| 19:38.38 | b00mer_ | if I wanted to set the callerid for an outbound call to the 212555+extension of the phone making the call... how might I do that? |
| 19:40.24 | b00mer_ | if I use the ${EXTEN} won't I be using the phone number being called? |
| 19:40.49 | ariel_ | Set(callerid(21555${EXTEN}) |
| 19:41.01 | Assid | can someone help me with moh |
| 19:41.03 | ariel_ | wait I formatted the line incorrect |
| 19:41.09 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:41.32 | Hmmhesays | anyone ever had asterisk just start play files in a choppy fashion? |
| 19:41.48 | justinu | probably jitter on the inbound stream |
| 19:41.51 | justinu | or packet loss |
| 19:41.54 | Netgeeks | no joy on the asterisk rebuild... same thing going on, and I deleted all files in /usr/lib/asterisk/modules |
| 19:41.56 | Hmmhesays | thats what iw as thinking |
| 19:41.58 | asterboy | cool...I now have digg.com in my [SERVICES] button on the Polycom Microbrowser. |
| 19:42.07 | Hmmhesays | but it doesn't happen on when you're actually in a call, just on the ivr menu's |
| 19:42.09 | asterboy | just used this after wget |
| 19:42.11 | asterboy | cat index.xml |grep title |sed -e "s/<\/title>/<br\/>/g" -e "s/<title>//g" >digg.html |
| 19:42.13 | blaylock | anyone have a working TDM400P card with an FXS module? |
| 19:42.28 | b00mer_ | ariel_: I'm confused... if I use the ${EXTEN} don't I reference the number being dialed? |
| 19:42.32 | asterboy | yes |
| 19:42.36 | justinu | Hmmhesays: in a call, i believe asterisk simply proxies the RTP, it doesn't rely on it as a timing source |
| 19:42.47 | Assid | Apr 20 01:12:30 WARNING[14570]: res_musiconhold.c:225 ast_moh_files_next: Unable to open file '/var/lib/asterisk/mohpublic/beethoven__violin_sonata_n': No such file or directory |
| 19:43.09 | Assid | it keeps syaing that error.. the file is there |
| 19:43.16 | Qwell[] | Assid: ls -l /var/lib/asterisk/mohpublic/beethoven__violin_sonata_n.* |
| 19:43.19 | asterboy | blaylock, yes |
| 19:43.30 | wunderkin | assid, and that's what it is named? hmm.. sure about that? :P rename the file |
| 19:43.36 | b00mer_ | Hmmhesays : I had lots of chopiness until I upraded from i386 kernel to i686 kernel |
| 19:43.40 | Assid | -rw-r--r-- 1 root root 2723839 2006-01-31 01:21 /var/lib/asterisk/mohpublic/beethoven__violin_sonata_n.mp3 |
| 19:43.49 | Qwell[] | and do you have format_mp3 loaded? |
| 19:43.55 | Assid | hrmm |
| 19:43.59 | Qwell[] | Didn't think so |
| 19:44.01 | Assid | i think i forgot that |
| 19:44.13 | wunderkin | oh, yeah |
| 19:44.19 | Assid | addons? |
| 19:44.21 | Qwell[] | yes |
| 19:44.23 | ariel_ | b00mer_, yes looking for the correct setting just a sec |
| 19:44.30 | *** join/#asterisk Curi (n=la@pc-69-70-104-200.cm.vtr.net) |
| 19:44.31 | Qwell[] | Assid: hope you aren't running trunk |
| 19:44.36 | b00mer_ | ariel : cool thanks! |
| 19:44.39 | *** join/#asterisk Greek-Boy (n=fusion@193.220.93.162) |
| 19:44.40 | Assid | nah.. stable |
| 19:44.49 | Assid | i stopped trunk a long time ago |
| 19:44.53 | ariel_ | b00mer_, I did say format was in correct remember |
| 19:44.55 | Assid | too much headache to keep updating |
| 19:45.02 | Curi | is there a way to create a new uniqueid for a call? |
| 19:45.10 | Greek-Boy | what settings will a typical sip provider require in sip.conf? |
| 19:45.19 | justinu | lol |
| 19:45.24 | Greek-Boy | never mind |
| 19:45.26 | Greek-Boy | i'll search the wiki again |
| 19:45.27 | Greek-Boy | lol |
| 19:45.30 | b00mer_ | ariel : yes... I am thanking you for getting the correct format for me |
| 19:45.47 | b00mer_ | ariel : heck... i'm waiting for it |
| 19:45.47 | justinu | Greek-Boy: i'm just laughing because it depends a lot on the provider |
| 19:45.56 | b00mer_ | :) |
| 19:46.06 | Qwell[] | s/a lot/entirely/ |
| 19:46.22 | Curi | any one? some way to create a uniqueid? |
| 19:46.54 | *** join/#asterisk Ciber311 (i=Ciber@static-71-249-235-160.nycmny.east.verizon.net) |
| 19:47.05 | Qwell[] | Curi: Why not use the uniqueid from the cdr? |
| 19:47.13 | Ciber311 | hey guys, how can i increase the parking time? |
| 19:47.17 | wunderkin | curi, try asking again in 2 minutes, but it sounds like you will need to do whatever you are trying to do another way |
| 19:47.25 | LostFrog | Put quarters in the meter, Ciber311? |
| 19:47.27 | Ciber311 | right now it seems calls go back to the parker after like 30 seconds |
| 19:49.08 | Assid | thanks Qwell |
| 19:50.55 | Curi | Qwell[]: is just that i'm trying to do a call through type of service, like a calling card, and when someone dials in to the asterisk gets a IVR, this leg of the call has a uniqueid, but then i want the other leg, outgoing leg to have another uniqueid |
| 19:51.00 | b00mer_ | ariel : would this work? exten => _91NXXNXXXXXX,1,SetCallerID(<212555${CALLERIDNUM}>) |
| 19:51.11 | Curi | thats why i'm trying to figure a way to create another uniqueid |
| 19:51.21 | justinu | curi: i think you'd need to use a local channel to do that |
| 19:51.38 | boch | is it possible to force a sip friend to unregister from the cli ? |
| 19:51.43 | Curi | justinu: and how can i do that? |
| 19:51.50 | ariel_ | b00mer_, something like Set(CallerID(number)=2155${CALLERID(number)}) might just do it. |
| 19:51.53 | justinu | by dialing out on a channel type of Local |
| 19:52.18 | justinu | try reading up a bit on local channels, you'll understand it better that way |
| 19:52.50 | Curi | thnx justinu |
| 19:53.10 | Hmmhesays | voicemail messages are still saved as gsm? |
| 19:53.12 | Hmmhesays | can I change that? |
| 19:54.05 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:54.10 | ariel_ | Hmmhesays, yes you have the options for WAV49, wav or gsm |
| 19:54.13 | Greek-Boy | justinu, i thought a basic config could work with my provider but i guess it doesn't work that way in the real world |
| 19:54.14 | Greek-Boy | lol |
| 19:54.14 | Greek-Boy | :) |
| 19:54.20 | Hmmhesays | is this in voicemail.conf? |
| 19:54.26 | ariel_ | Hmmhesays, yes |
| 19:54.36 | Hmmhesays | thats right |
| 19:54.37 | ariel_ | see the voicemail.conf.sample |
| 19:54.38 | justinu | Greek-Boy: there's usually gonna be some amount of tweaking required |
| 19:54.57 | Qwell[] | Hmmhesays: I think it's a per peer option now too |
| 19:55.04 | *** join/#asterisk guingooc (n=nospam@moua7.com) |
| 19:55.05 | guingooc | hi |
| 19:55.52 | guingooc | Is there any pci-x digium card working with asterisk? |
| 19:56.16 | Qwell[] | guingooc: I think a few cards that work with pci also work with pcix |
| 19:56.23 | Qwell[] | call Digium sales... |
| 19:57.10 | guingooc | oh isn't it different slots? |
| 19:57.18 | Qwell[] | yes, but pci fits |
| 19:57.37 | Qwell[] | call and ask...they'll tell you which cards will work |
| 19:58.02 | guingooc | right :) |
| 19:58.12 | guingooc | thank you btw |
| 20:01.25 | *** join/#asterisk kn0x (n=root@71.194.235.251) |
| 20:02.17 | lzhang | hey guys, I'm getting one way audio on a Polycom 601, there is no NAT involved, and it happens a number of times a day. how would I go about debugging this? I'd like to know what to look for next time it happens |
| 20:02.22 | Netgeeks | Whoah! okay, I was able to get things to work on that buggy system.... and the fix is .... well, I'm even more bedfuddled |
| 20:02.27 | wunderkin | guingooc, yes it will work in pci-x |
| 20:02.51 | Qwell[] | lzhang: ask Netgeeks :P |
| 20:02.51 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:02.57 | justinu | heh |
| 20:03.25 | lzhang | Netgeeks helpppp meeeeeeee |
| 20:03.33 | Netgeeks | justinu: the fix was to unload wct4xxp, and load ztdummy |
| 20:03.52 | file | ooh blood |
| 20:04.02 | justinu | Netgeeks: that sounds pretty broken... |
| 20:04.08 | [TK]D-Fender | Netgeeks : OMG, I forgot about that... someone else had that problem earlier and I couldn't make sense for WHY... |
| 20:04.25 | justinu | in your setup, you shouldn't need either |
| 20:04.33 | justinu | at least for what you described |
| 20:04.52 | *** join/#asterisk mmmmmToop (n=chatzill@dsl-146-126-36.telkomadsl.co.za) |
| 20:05.13 | Netgeeks | lzhang: make sure that in your polycom config you have the parameter: tcpIpApp.port.rtp.filterByIp set to 0 |
| 20:05.25 | LostFrog | Has anyone set up Raffi's Bananaphone as a ringtone? :) |
| 20:05.31 | Netgeeks | I'll go unload ztdummy and see if everything still works |
| 20:05.39 | oelewapperke | do you need to have a module loaded to use asteriskù's interface ? |
| 20:05.48 | Qwell[] | oelewapperke: What interface? |
| 20:05.51 | oelewapperke | I cannot control my asterisk using asterisk -rvvvvv |
| 20:05.59 | oelewapperke | yet the process is running |
| 20:06.17 | oelewapperke | but /var/run/asterisk.ctl exists, it is owned by the correct user, but it is zero bytes |
| 20:06.25 | Qwell[] | zero bytes is fine |
| 20:06.25 | Netgeeks | yes, it does, but loading up wct4xxp with no pri plugged in seems to cause the apps to not work |
| 20:06.28 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:06.40 | oelewapperke | Qwell[]: is it supposed to be a pipe or some such ? |
| 20:06.49 | Qwell[] | dunno |
| 20:07.05 | Qwell[] | are you trying to connect as the same user who started asterisk? |
| 20:07.20 | oelewapperke | no |
| 20:07.21 | oelewapperke | as root |
| 20:07.39 | oelewapperke | I tried with the asterisk user, same result |
| 20:07.43 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 20:07.47 | justinu | netgeeks: i'd consider filing a bug report |
| 20:07.51 | opus_ | is there a way to specify DIAL Zap/gXXX where it goes *down* from the top instead of from the bottom? |
| 20:07.55 | Qwell[] | You're positive asterisk is running? Check ps |
| 20:08.00 | justinu | opus_: try capital G |
| 20:08.01 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
| 20:08.09 | opus_ | justinu thanks |
| 20:08.12 | oelewapperke | asterisk 9507 0.0 1.1 13088 6060 ? Ssl 22:03 0:00 /usr/sbin/asterisk -p -U asterisk |
| 20:09.19 | oelewapperke | Qwell[]: so yes, Iù'm sure it's running |
| 20:11.32 | Netgeeks | yep, interesting enough, with wct4xxp loaded, the apps don't work, without it, they do |
| 20:11.57 | justinu | is that card known to be ok? |
| 20:11.58 | ariel_ | Netgeeks, make a loop back t1 cable |
| 20:14.36 | jsharp | Make sure your TDM card is generating interrupts. |
| 20:15.20 | jsharp | I've seen a brokeded card that didn't generate interrupts screw everything up. |
| 20:16.14 | justinu | good to know |
| 20:17.09 | mmmmmToop | we are also struggeling to get a wct4xxp card working...using exactly same configs as before...just on diff Asterisk version & server...mmm... |
| 20:17.34 | mmmmmToop | not sure if card broken....both lights are flashing on card... |
| 20:17.46 | justinu | some people here have had problems making those cards work on certain mobos |
| 20:17.46 | Netgeeks | having the guy on-site make some loopback cables, will test with them in, I'll check interrupts at that time |
| 20:17.47 | alexis101 | i have a question about sip and nat ... I already ask it a lot of time but i never add a good anwser ! |
| 20:18.24 | mmmmmToop | justinu: interesting...any way to check what prob is? |
| 20:18.33 | alexis101 | i have a sip extentions behind a nat router but the asterisk server is not behind nat |
| 20:18.41 | *** part/#asterisk strom (i=storm@jihad.carbomb.org) |
| 20:18.47 | tainted- | why doesn't your service work when i dial '80055512345'!!!1! YOU GUYS SUCK!!1!11~ |
| 20:19.05 | alexis101 | when a call the extentions there is no audio when the extention behind nat call me i get audio |
| 20:19.08 | tainted- | ^ that was a customer just now |
| 20:19.16 | alexis101 | when i call ** |
| 20:19.22 | justinu | mmmmmToop: i don't think so... you might have to try it in a different server |
| 20:19.26 | wunderkin | mmmmmToop, 2 port card? i think if the lights are blinking then the module isn't loaded |
| 20:19.54 | justinu | actually, the cards seem to work, but people can't keep their D channel up |
| 20:20.05 | tzanger | which cards are those |
| 20:20.09 | alexis101 | is it a codec problem ... port fowarding or other asterisk configuration ?? |
| 20:20.09 | mmmmmToop | yes...that sounds like it... |
| 20:20.19 | mmmmmToop | when loading I get some D-Channel can't load |
| 20:20.39 | justinu | the wct4xxp cards |
| 20:20.49 | wunderkin | ok yeah then the module isn't loaded, so something else is wrong |
| 20:20.50 | jsharp | alexis101: Port forwarding issue. |
| 20:20.58 | ariel_ | alexis101, do you have nat=yes on the sip settings |
| 20:21.06 | alexis101 | yes nat=yes is set |
| 20:21.06 | jsharp | And what he said. |
| 20:21.44 | alexis101 | What port i need to foward 5060 ? and all the rtps port ?? |
| 20:21.51 | jsharp | Look in dmesg for errors loading the module. |
| 20:21.52 | mmmmmToop | zaptel module & wct4xxp loads...Asterisk won't though...mmmm.... |
| 20:22.04 | lzhang | so what does tcpIpApp.port.rtp.filterByIp in the sip.cfg mean? |
| 20:22.08 | ariel_ | alexis101, 5060 is for registration |
| 20:22.14 | alexis101 | i know :) |
| 20:22.18 | mmmmmToop | jsharp: Tried that already...looks fine I think... |
| 20:22.36 | ariel_ | if your firewall allows outbound fine you don't need to forward any ports from your location the asterisk box is on the outside network correct |
| 20:22.51 | mmmmmToop | wondering if it is an old version of firmware on card...mmmm.... |
| 20:22.52 | justinu | mmmmmToop: what do you mean "asterisk won't"? |
| 20:22.57 | ariel_ | unless you have a firewall that is not configered correctly |
| 20:23.20 | Netgeeks | lzhang: tcpIpApp.port.rtp.filterByIp 0, 1 1 If set to 1, reject RTP packets |
| 20:23.20 | Netgeeks | arriving from (sent from) |
| 20:23.20 | Netgeeks | a non-negotiated (via SDP) |
| 20:23.20 | Netgeeks | IP address. |
| 20:23.20 | Netgeeks | tcpIpApp.port.rtp.filterByPort 0, 1 0 If set to 1, reject RTP packets |
| 20:23.21 | Netgeeks | arriving from (sent from) |
| 20:23.23 | Netgeeks | a non-negotiated (via SDP) |
| 20:23.26 | Netgeeks | grrr |
| 20:23.32 | mmmmmToop | justinu: wct4xxp and zaptel load...when I run asterisk -vv I get a chan_zap.c won't load... |
| 20:23.34 | Netgeeks | synbergy must die |
| 20:23.50 | Netgeeks | sorrry for multi-line paste, my synergy client is all fubard |
| 20:23.57 | justinu | mmmmmToop: in that case, make sure that your /etc/zaptel.conf and /etc/asterisk/zapata.conf are configured correctly |
| 20:24.00 | alexis101 | yes the asterisk box is outside the network ... |
| 20:24.12 | *** join/#asterisk TheGreyElf (n=tracy@72.14.24.2) |
| 20:24.34 | lzhang | interesting, mine was set to 1, so hopefully this change will fix the prob |
| 20:24.37 | justinu | mmmmmToop: in my experience, if the two config files don't match properly, chan_zap.so will crash/not load. |
| 20:24.47 | mmmmmToop | sure... |
| 20:25.10 | mmmmmToop | when I run ztcfg -vv I get: ZT_CHANCONFIG failed on channel 49: No such device or address (6) |
| 20:25.23 | justinu | ok, then square away your zaptel.conf |
| 20:25.38 | mmmmmToop | let me give that a bash... |
| 20:25.45 | Netgeeks | lzhang: tcpIpApp.port.rtp.filterByIp 0, 1 1 If set to 1, reject RTP packets arriving from (sent from) a non-negotiated (via SDP) IP address. |
| 20:25.50 | Netgeeks | there, much better |
| 20:26.00 | lzhang | Netgeeks: thanks hehe |
| 20:26.47 | Netgeeks | beyond what the manual says, I have no idea why the polycom thinks some small portion of calls are coming from a non-negotiated IP address |
| 20:27.13 | mmmmmToop | justinu: cant believe that works....just gives "0 Channels configured"... |
| 20:28.16 | justinu | mmmmmToop: please read this document if you haven't already: http://www.digium.com/en/docs/misc/configuring_zaptel.pdf |
| 20:28.39 | *** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com) |
| 20:29.00 | mmmmmToop | not to worry...done this many times before ;) & read doc...maybe I will read it again |
| 20:29.06 | justinu | ok |
| 20:29.53 | [TK]D-Fender | bbaib |
| 20:30.09 | fndude | I'm trying to get my rules to dial an area code before the number. I think I am a little slow. I figure I should be able to insert this into the dial command before the $exten. Whats the proper way to do this? |
| 20:30.20 | justinu | you're right |
| 20:30.22 | Qwell[] | 555${EXTEN} |
| 20:30.40 | fndude | ahhhh. |
| 20:30.42 | fndude | no + |
| 20:31.38 | fndude | wow that works. I thought I was retarded. |
| 20:32.02 | *** join/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 20:35.02 | b00mer_ | how do I debug some chopiness on a phone call to a IAX provider? I tried g726 and then gsm... and its still got some delay. is there any procedure for investigating this? |
| 20:35.15 | file | who is the provider? |
| 20:35.27 | b00mer_ | asterlink |
| 20:35.36 | Qwell[] | file: :( |
| 20:35.56 | file | Qwell[]: whyfor unhappy? |
| 20:36.00 | Qwell[] | you |
| 20:36.19 | file | Qwell[]: FINE! blame me |
| 20:36.29 | Qwell[] | no, you unhappy |
| 20:36.38 | file | I am? I never got that memo |
| 20:36.43 | Qwell[] | yes. :( |
| 20:36.45 | Assid | argh.. very difficult to find a nice provider.. at 2c/min or less |
| 20:36.48 | macTijn | hmm |
| 20:36.48 | b00mer_ | I'm the one who should be unhappy... debuging ideas? |
| 20:36.57 | Assid | less would be better tho |
| 20:37.20 | macTijn | anyone who knows how CallerID is done on analog lines in the Netherlands ? |
| 20:37.33 | file | b00mer_: switch to SIP, you'll save yourself a headache |
| 20:37.40 | justinu | probably the same way it's done in other places |
| 20:37.47 | *** join/#asterisk keith80403 (n=keith@24-56-189-80.co.warpdriveonline.com) |
| 20:38.49 | macTijn | justinu: well, is it DTMF based, ADSI, some other form of pre-voice--signalling ? |
| 20:39.02 | justinu | i would guess it works like ADSI does |
| 20:39.06 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 20:39.19 | macTijn | justinu: I need to know for sure :) |
| 20:39.25 | *** part/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 20:39.39 | justinu | yeah, i don't know for sure... |
| 20:39.51 | macTijn | ah, DTMF |
| 20:40.09 | justinu | weird |
| 20:40.30 | OliverX | my routing is so: handy -> isdn -> sip the call is ringing and i can answer to the call with my xlite client, but i have no sound transfer |
| 20:40.40 | OliverX | have anyone here an idea? |
| 20:41.02 | macTijn | hmm |
| 20:41.08 | macTijn | apparently it's FSK aswell |
| 20:41.10 | macTijn | or so |
| 20:41.32 | OliverX | ehm no firewall |
| 20:41.43 | OliverX | it is all in the sambe subnet ;) |
| 20:41.46 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 20:42.34 | OliverX | also: handy -> handy provider -> isdnsystem -> isdn card in a asterisk server -> asterisk transfer the call to an sip user -> x-lite |
| 20:44.55 | dlynes_ | is anyone aware of a bug wrt mode=files for musiconhold, when the remote caller is on iax? |
| 20:45.14 | dlynes_ | the problem i'm encountering is when the remote caller calls in and gets placed on hold |
| 20:45.29 | b00mer_ | file: so sip would really be better than IAX? |
| 20:45.38 | b00mer_ | file: in my case? |
| 20:45.46 | file | when dealing with Asterlink, yes |
| 20:45.47 | dlynes_ | when the callee takes the caller off hold, the callee can hear the caller, but the caller can not hear the callee |
| 20:46.01 | dlynes_ | but when using an external music on hold player, everything is just fine |
| 20:46.08 | b00mer_ | file: I'll give it a try |
| 20:46.12 | b00mer_ | file: thanks! |
| 20:46.23 | lesouvage | TNO a leading research institute is lookingf for someone to make an Asterisk implementation of the MELPe codec. This is a codec that needs very little bandwidth and used by the NATO. (see http://www.tno.nl/informatie-_en_communicatietechnologie/werken_bij_tno/afstudeeropdrachten/mobiele_telecommunicatie_1/een_codec_voor_de_open_so/) |
| 20:46.41 | dlynes_ | when two extensions on the same asterisk box use the music on hold feature in files mode, it works fine, also |
| 20:47.03 | dlynes_ | Anyone have any ideas on what might be causing the problem? |
| 20:47.23 | dlynes_ | Also, the problem is replicable every time |
| 20:49.09 | lesouvage | If there are any people from Holland on the channel. There are still some tickets left for the Meet Asterisk Europe Tour event in Amsterdam on the 26 of april. (see www.meetasterisk.com) |
| 20:50.29 | GerbilWrk | can a ring type be sent with a call? |
| 20:51.14 | GerbilWrk | for example, someone dials a number, it goes to a special queue that rings all of our phones with a different ringer then the normal queue? |
| 20:52.02 | dlynes_ | Are these analog phones, or voip phones? |
| 20:52.10 | GerbilWrk | Voip, Snom360's |
| 20:52.20 | dlynes_ | then no, not afaik |
| 20:52.55 | GerbilWrk | guess i'll have to set them up as different extensions on one of the lines, with different ringers, and tie them to a different queue :( |
| 20:53.58 | TheGreyElf | GerbilWrk: There is alert-info option that can be sent to a Snom to do auto-answer and custom ring. you should be able to create dialplan code to do it. Check out the voip-info wiki "asterisk snom phone" |
| 20:54.11 | GerbilWrk | ok |
| 20:54.12 | GerbilWrk | thanks |
| 20:54.22 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 20:54.42 | Zodiacal | anyone know if the new cisco 7960 sip firmware v8.x supports hints? |
| 21:02.04 | *** join/#asterisk alvaro- (i=jeremiah@31.Red-213-98-123.staticIP.rima-tde.net) |
| 21:03.29 | Greek-Boy | whats the default to check balance with sip providers? |
| 21:03.31 | *** part/#asterisk alvaro- (i=jeremiah@31.Red-213-98-123.staticIP.rima-tde.net) |
| 21:04.51 | tzanger | Greek-Boy: that would be very much sip-provider-specific... |
| 21:05.50 | mmmmmToop | justinu: found problem...how irritating...forgot to move jumper to make it E1!! ...anyway |
| 21:06.45 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 21:07.46 | Z_God | I'm having problems getting asterisk to pick up my ISDN card |
| 21:08.01 | Z_God | it seems it isn't able to initialise the ISDN card properly |
| 21:08.17 | Z_God | with minicom everything seems to work |
| 21:08.44 | Z_God | I'm using asterisk version 1.0.7 (from debian sarge) with the modem channel |
| 21:08.51 | Z_God | my isdn card uses the hisax driver |
| 21:09.05 | *** join/#asterisk ToTo (n=ToTo@host125-166.pool879.interbusiness.it) |
| 21:09.17 | Z_God | any idea how I can check how asterisk initialises the isdn card? |
| 21:10.26 | *** join/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 21:11.32 | *** part/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 21:13.12 | dlynes_ | has anyone encountered problems getting music on hold to return the caller properly after hold is released? |
| 21:13.44 | dlynes_ | I've been having problems getting it to work properly with aastra 9133i's since asterisk 1.2.4 |
| 21:14.24 | justinu | mmmmmToop: :) |
| 21:14.37 | dlynes_ | Z_God, i've never used the isdn cards, but have you tried dmesg? |
| 21:14.52 | mmmmmToop | always the silly things u miss! ...anyway...wont forget to check that again! |
| 21:17.09 | *** join/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 21:18.39 | Assid | hrmm |
| 21:19.56 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-59.apple.com) |
| 21:20.04 | Z_God | yep |
| 21:20.08 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:20.16 | Z_God | dmesg just says the calls are being ignored |
| 21:20.24 | *** part/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 21:21.00 | Z_God | if the ISDN card is initialized properly, dmesg says RING instead |
| 21:21.29 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 21:21.59 | *** join/#asterisk Sedorox (n=penbra67@smartserv/cna/Sedorox) |
| 21:24.24 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 21:25.58 | *** part/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 21:29.10 | marl | cn anyone tell me if there is an easy way to set up asterisk so that a customer phoneing in, can enter an account code, and then be forwarded to an operator, and the account code can be passed to another program as the call is forwarded? same idea ass using CLI, but can be used by customers that dont send CLI when calling in? |
| 21:30.02 | brettnem | marl: show application dial |
| 21:30.08 | brettnem | marl: send URL |
| 21:30.14 | brettnem | activate a popup |
| 21:30.24 | brettnem | speak the account number when the agent answers. |
| 21:30.30 | brettnem | do some digging, the answers are out there. |
| 21:31.01 | marl | what about grabbing dtmf codes prior to transfering a call? |
| 21:31.22 | marl | been looking, but i think im not using the corect terms to get a sensible result :( |
| 21:31.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:32.42 | *** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net) |
| 21:35.20 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 21:41.08 | macTijn | hah |
| 21:41.09 | macTijn | found it |
| 21:41.15 | macTijn | we do DTMF signalling |
| 21:41.21 | macTijn | freaking KPN bastards |
| 21:41.42 | Z_God | ah, handig om te weten ;) |
| 21:42.00 | macTijn | haha :) |
| 21:42.07 | macTijn | keep it english ;) |
| 21:42.29 | Z_God | k :) |
| 21:42.33 | Hmmhesays | na na na don't phunk with my heart |
| 21:43.42 | macTijn | Z_God: I just read KPN PSTN specifications (terrible to read), an ETSI standard (even worse), and then I had to create a summary. |
| 21:43.53 | b00mer_ | file : thanks about the SIP info... working great now |
| 21:44.00 | macTijn | oh |
| 21:44.01 | file | b00mer_: aye |
| 21:44.07 | macTijn | and now my outlook crashes |
| 21:44.17 | macTijn | freaking piece of sh*t |
| 21:44.43 | xachen | oi |
| 21:44.57 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 21:45.11 | xachen | perl can be such a pain in the ass |
| 21:45.18 | macTijn | true that |
| 21:45.37 | xachen | i don't get what I'm doring wrong with my regexes |
| 21:45.59 | Z_God | >macTijn, interesting |
| 21:46.10 | Z_God | can those specs be found online? |
| 21:46.15 | macTijn | Z_God: well, it actually is terribly boring |
| 21:46.18 | macTijn | yes |
| 21:46.20 | Z_God | hehe |
| 21:46.24 | macTijn | hold on |
| 21:46.30 | Z_God | is it similar for ISDN stuff? |
| 21:46.30 | macTijn | oh, right |
| 21:46.33 | macTijn | nono |
| 21:46.40 | macTijn | ISDN is all euroisdn signalling |
| 21:46.58 | macTijn | thank god they didn't mess up that aswell |
| 21:47.05 | Z_God | k, hehe |
| 21:47.12 | Z_God | only need to deal with isdn here :) |
| 21:47.46 | macTijn | I haven't had such good experiences with asterisk + bri |
| 21:48.50 | macTijn | mostly when I used NT mode |
| 21:49.03 | macTijn | together with a piece of shit Siemens DECT station |
| 21:49.13 | macTijn | asterisk and the dect station kept crashing |
| 21:49.20 | Z_God | bri? |
| 21:49.33 | macTijn | BRI, or ISDN-2 |
| 21:49.38 | Z_God | k :) |
| 21:50.41 | Z_God | I'm pretty new to all this stuff, trying to get the phones over here to be able to call SIP clients and vice-versa |
| 21:51.15 | Z_God | had it working before in slackware (now using debian), but I seem to have some new problems on both sides now :S |
| 21:51.22 | *** part/#asterisk lars-ut-away (n=lars-ut@70.103.228.158) |
| 21:51.35 | Z_God | ISDN isn't being initialised correctly and my firewall is too strict for rtp ;) |
| 21:51.44 | macTijn | heh |
| 21:51.58 | Sedorox | 0_o |
| 21:51.59 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:52.48 | macTijn | hmm |
| 21:52.49 | macTijn | rtp |
| 21:52.51 | macTijn | good point |
| 21:52.54 | macTijn | brb :) |
| 21:53.13 | Z_God | k? |
| 21:58.12 | *** part/#asterisk mysfitt (n=joe@cpe-24-165-167-122.midsouth.res.rr.com) |
| 21:59.09 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 22:00.30 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 22:00.38 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:01.18 | *** join/#asterisk saftsack (n=saftsack@p54A7E939.dip.t-dialin.net) |
| 22:02.36 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 22:03.49 | brodiem | If I have a TE210P (span1=chan 1-24, span2=chan 25-49), and I add a TDM400P, is it correct that the first port on the tdm400p would be port 50? |
| 22:04.00 | brodiem | er channel 50 |
| 22:04.32 | ManxPower | brodiem, that would depend on which card driver you load dirst |
| 22:04.35 | ManxPower | first too |
| 22:04.47 | brodiem | right sorry assuming te210 was loaded first |
| 22:05.43 | ManxPower | yes, then the new card would start at channel 50 |
| 22:06.06 | ManxPower | actually at channel 48 |
| 22:06.08 | ManxPower | 49 |
| 22:06.15 | ManxPower | 2x24=48 |
| 22:07.10 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 22:07.17 | *** part/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 22:08.33 | brodiem | er right yeah |
| 22:08.33 | brodiem | thanks |
| 22:09.24 | ManxPower | the power here keep flickering |
| 22:11.15 | *** join/#asterisk bonfire1 (n=bon@bzq-88-154-199-55.red.bezeqint.net) |
| 22:11.44 | bonfire1 | Hey everybody! Anyone here can answer my questions about deploying a VoIP network using asterisk? |
| 22:12.05 | Hmmhesays | nope, we only talk about pizza and beer in here |
| 22:12.13 | bonfire1 | ok I can do that |
| 22:12.20 | bonfire1 | what toppings do you like best? |
| 22:12.34 | Hmmhesays | pepperoni |
| 22:12.40 | Hmmhesays | served by a nekkid wimmins |
| 22:13.23 | bonfire1 | now really- |
| 22:13.37 | Hmmhesays | what do you want to know |
| 22:13.53 | *** join/#asterisk manxpower (n=ewieling@dpc67142183150.direcpc.com) |
| 22:14.00 | bonfire1 | say I have an IP network |
| 22:14.06 | manxpower | Ah. I knew I forgot to plug something into the battery backups |
| 22:14.28 | bonfire1 | And I want to convert the telephony setup to an IP based one. |
| 22:15.02 | bonfire1 | well basically I know how it works with SIP and RTP and stuff, and I know I'm supposed to have a linux server with asterisk |
| 22:15.17 | bonfire1 | But I have two questions (later I'll remember a few more) |
| 22:15.17 | Hmmhesays | asterisk is not the only solution, but go on |
| 22:15.26 | bonfire1 | yeah ok, but I hear it's the best. |
| 22:15.37 | bonfire1 | I mean, if I'm not rich and can't afford CallManager |
| 22:15.38 | Hmmhesays | best is relative to your needs |
| 22:15.43 | Hmmhesays | anyway, go on |
| 22:16.04 | Hmmhesays | this helmet is getting heavy on my head, ask your question |
| 22:16.20 | bonfire1 | How do I convert to PSTN-calls |
| 22:16.30 | bonfire1 | Do I just install a modem on the linux server |
| 22:16.31 | Hmmhesays | 1. an fxo gateway, 2. an fxo card |
| 22:16.37 | bonfire1 | and just plug-it in |
| 22:16.38 | Hmmhesays | pick one |
| 22:16.44 | bonfire1 | what's an fxo card? |
| 22:16.51 | Hmmhesays | pci card with fxo ports on it |
| 22:17.07 | Hmmhesays | personally I use fxo gateways, little more expensive, but worth the trouble |
| 22:17.13 | bonfire1 | what fxo? (sorry I'm probably asking noob questions). |
| 22:17.17 | Hmmhesays | ~docs |
| 22:17.19 | jbot | hmm... 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 |
| 22:17.26 | Hmmhesays | ~fxofxs |
| 22:17.27 | jbot | hmm... fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 22:17.55 | manxpower | ~fxsfxo |
| 22:17.57 | jbot | i heard fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 22:18.21 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 22:18.32 | Egonis | I want to do an include => externalfile.conf... how does this work? |
| 22:18.42 | Hmmhesays | same way you do any includes |
| 22:18.45 | bonfire1 | so what's fxo gateway? |
| 22:18.46 | manxpower | Eggplant, no. |
| 22:18.54 | Hmmhesays | an external gateway |
| 22:18.54 | manxpower | you would #include externalconf.conf |
| 22:19.08 | Hmmhesays | converts ip<->tdm |
| 22:19.16 | bonfire1 | like a server someone runs and I just use it? |
| 22:19.20 | Egonis | manxpower: as literal? i.e. #include externalconf.conf and then include => [context]? |
| 22:19.25 | Hmmhesays | no |
| 22:19.27 | Hmmhesays | ~gateway |
| 22:19.30 | jbot | well, gateway is The computer or device onto which the first hop needs to go to get out of your network. As you know a TCP/IP route uses many hops to get from here to there. |
| 22:19.34 | Hmmhesays | i dunno if jbat knows |
| 22:19.38 | Hmmhesays | *jbot even |
| 22:19.45 | Hmmhesays | go read on the wiki |
| 22:19.49 | manxpower | Eggplant, #include works EXACTLY like C #include, i.e. it's like you inserted that file in at that point |
| 22:19.59 | bonfire1 | ok |
| 22:20.11 | Egonis | manxpower: awesome, ty! |
| 22:20.13 | Hmmhesays | there is plenty of info there |
| 22:20.35 | Hmmhesays | i'm going to go freeze my ass of on this bike ride now |
| 22:20.36 | Hmmhesays | later |
| 22:20.37 | froguz | where can i find info about creating a itsp (voip provider) using Asterisk? i can't find anything on the wiki |
| 22:20.37 | manxpower | great, a tornado warning for the county I'm in, fortunatly it's not near me. |
| 22:20.54 | *** join/#asterisk eipi (n=eipi@OL54-55.fibertel.com.ar) |
| 22:21.00 | Hmmhesays | talk to someone who already knows |
| 22:21.08 | eipi | anyone on freepbx? |
| 22:21.11 | manxpower | froguz, First learn telephony, then learn networking, then learn routing, then learn Linux, then come back here. |
| 22:21.25 | manxpower | eipi, many people, but not here. notice the /topic. |
| 22:22.08 | Egonis | manxpower: do I need to specify the full path? |
| 22:22.21 | manxpower | Eggplant, unless it's in /etc/asterisk yes |
| 22:23.39 | Egonis | manxpower: ty again |
| 22:24.25 | froguz | manxpower, i've been working on asterisk systems since about 6 months ago. i think i got the skills, but i need a guide of alternatives for the architectures, ser + asterisk? asterisk array?... |
| 22:27.39 | *** join/#asterisk santiago (n=santiago@208.195.215.51) |
| 22:28.34 | Egonis | when specifying voicemail contexts, do I use mailbox=100@blah? |
| 22:29.17 | froguz | just need to know the alternatives i have to start working on a little itsp... i've been selling Asterisk PBX's and i don't want to continue using external voip providers on my customers boxes. i want to do mine own |
| 22:29.44 | froguz | my* |
| 22:30.01 | *** join/#asterisk undertalk (n=sonicyou@201.14.63.2) |
| 22:30.02 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:30.20 | froguz | can i avoid using SER? |
| 22:31.34 | riddlebox | can you dial a code while on the phone to park a call? |
| 22:31.50 | froguz | well... i'm going to try later |
| 22:31.53 | froguz | thanks |
| 22:31.54 | froguz | bye |
| 22:34.00 | riddlebox | ha nevermind, google strikes again |
| 22:37.47 | Z_God | hmm, this is weird, calling to ISDN is working fine.... |
| 22:38.17 | Z_God | only from ISDN isn't working |
| 22:38.39 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:39.05 | Z_God | and indeed seems ISDN isn't getting initialized correctly looking at the callerid on my phone .... |
| 22:39.10 | Z_God | it^ |
| 22:42.48 | riddlebox | hrmm this is weird I read press # to initiate a transfer, then dial 700 |
| 22:42.57 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 22:43.09 | riddlebox | it works when a call comes in but when call someone it wont work? |
| 22:43.26 | kink0 | good night , I am in emergency, my asterisk now did not see my PRI , can somebody give me some help ? |
| 22:43.27 | eipi | help, i'm newbie: i have asterisk running under "asterisk" user, and apache under "apache" user, what permissions i have to give to asterisk directories (what command i have to run) to be possible to write under asterisk directories |
| 22:44.10 | manxpower | eipi, run Asterisk as "apache" or run Apache as "asterisk" |
| 22:44.14 | kink0 | T4XXP (PCI) Card 0 Span 1 RED 0 0 0 |
| 22:44.44 | eipi | manxpower, that's the only way? |
| 22:44.52 | manxpower | kink0, RED means "I don't see ANYTHING." Means a bad cable or a bad T-1 |
| 22:45.12 | eipi | i have to run all my processes and virtual sites as one user? |
| 22:45.23 | manxpower | eipi, I'm sure there are at least a dozen ways. The way I suggested is the easiest |
| 22:45.42 | manxpower | eipi, you did not find another ways when you searched the mailing lists? |
| 22:45.44 | manxpower | ~mailinglist. |
| 22:45.46 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 22:45.46 | eipi | manxpower :) thanks for reply, but that alternative its working now |
| 22:46.03 | eipi | ok |
| 22:46.12 | *** join/#asterisk tzafrir_home (n=morphing@bzq-179-75-202.cust.bezeqint.net) |
| 22:46.43 | VeNoMouS_ | http://packetstorm.setnine.com/filedesc/SIPcrack-0.1.tar.html |
| 22:46.44 | VeNoMouS_ | lol |
| 22:46.51 | VeNoMouS_ | : |
| 22:46.53 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:47.00 | kink0 | manxpower, yes, i have try with fews cables, in the event the cable that was running fails, but the same. |
| 22:47.50 | kink0 | manxpower, the problem starts after a reboot of machine |
| 22:48.39 | manxpower | kink0, then you have a SERIOUS problem. |
| 22:48.48 | manxpower | do yu have more than one card? |
| 22:48.54 | kink0 | manxpower, yes... I have, and also is a production machine |
| 22:49.13 | kink0 | manxpower, no, I just have one TE405 card. |
| 22:49.48 | bonfire1 | I have a question about ITSPs: Is there an ITSP that operates gateways around the world which I can register my company's asterisk to make international calls? Something like Skype but one that will allow asterisk server to use it, and not the SkypeClient thingie. |
| 22:49.57 | Hmmhesays | yes |
| 22:49.59 | Hmmhesays | there are plenty |
| 22:50.39 | manxpower | bonfire1, Any time you send your calls over the internet you CANNOT expect it to be as reliable as the phone company. |
| 22:50.43 | bonfire1 | I saw some names in Wikipedia but they have only in Canada USA and UK, and to the rest of the countries it's just normal international calls |
| 22:50.52 | bonfire1 | of course... |
| 22:50.58 | bonfire1 | I know that |
| 22:51.09 | bonfire1 | But isn't this the whole point of VoIP? |
| 22:51.18 | bonfire1 | (well, not the whole) |
| 22:51.32 | b00mer_ | anybody know what this means?? : NOTICE[9305]: channel.c:1904 ast_read: Dropping incompatile voice frame on Local/92125551212@internal-a187,2 of formal ulaw since our native format has changed to slin |
| 22:51.48 | b00mer_ | only happens when I use the fwd call button on the phone |
| 22:51.49 | manxpower | I manage at least 5 asterisk servers and we never send calls over the interne |
| 22:51.49 | manxpower | t |
| 22:51.55 | bonfire1 | (if i'd run my company that's the top reason why I would like to change my telephony infrastructure to VoIP). |
| 22:52.04 | bonfire1 | really? so why do you use asterisk? |
| 22:52.11 | bonfire1 | instead of good 'ol PBX |
| 22:52.23 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-59.apple.com) |
| 22:52.40 | manxpower | bonfire1, because it costs 1/2 as much of traditional PBXs and I can route calls over the corporate WAN, and can write AGIs ans other useful things. |
| 22:53.00 | manxpower | Hell, one of the legacy PBX vendors we use at one of the non-asterisk offices refuses to let us or anyone else manage the PBX. |
| 22:53.35 | b00mer_ | anybody have a clue on that error? |
| 22:53.41 | bonfire1 | can I PM you? |
| 22:53.45 | bonfire1 | manxpowe |
| 22:53.52 | manxpower | bonfire1, not unless you have a lot of cash to hand me. |
| 22:54.02 | bonfire1 | I have a lot of bytes I can give you |
| 22:54.08 | Hmmhesays | i thought i read somewhere the new sounds were in slin? |
| 22:54.46 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 22:55.27 | Druken | mooooooo |
| 22:55.49 | riddlebox | how come call parking only works for inbound calls? |
| 22:56.10 | Druken | why the hell would you park an outgoing call ? |
| 22:56.26 | russellb | it works both ways |
| 22:56.33 | russellb | it's just a regular transfer |
| 22:56.37 | manxpower | riddlebox, Call parking will work with ANY CALL you can do a supervised transfer on. |
| 22:57.22 | riddlebox | manxpower, I can get it to work when I call into my system from another phone, but if I call out on the system it wont work? |
| 22:57.40 | manxpower | riddlebox, you have a problem that is NOT related to call parking. |
| 22:57.42 | *** join/#asterisk wm4k (n=wm4k@dsl213-218-233-192.as15444.net) |
| 22:58.13 | manxpower | riddlebox, perhaps you have a transfer problem. |
| 22:58.18 | riddlebox | manxpower, do I need to add a t, to my outgoing trunk? |
| 22:58.56 | kink0 | manxpower, I get Layer 1 and Layer 2 ok, when cable is connected, but I am unable to get layer 3 ok |
| 22:59.08 | manxpower | riddlebox, I don't know. Do you? I don't have a single "t" in my entire dialplan and my users transfer calls all day. |
| 22:59.13 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:59.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:59.20 | cytrak | hey guys I got a problem .... I have setup a wireless lan on 10.10.10.x and just now I'm testing my idefisk to try to connect to asterisk and registration fails whenever I'm on the 10.10.10.0 |
| 22:59.25 | riddlebox | hrmm weird |
| 22:59.30 | Creathir | kudos to the guys that develop asterisk |
| 22:59.39 | Creathir | you guys have created an awesome product |
| 22:59.45 | cytrak | it works fine if I'm on another wireless that is on 192.168.12x.0 |
| 23:00.05 | Creathir | I have this thing up and running... and it works fantastic.... |
| 23:00.13 | Creathir | just wanted to say good job |
| 23:00.16 | manxpower | I never could understand the fetish people have for DTMF transfers. |
| 23:00.18 | russellb | Creathir: thanks :) |
| 23:00.23 | file | Creathir: thankies |
| 23:00.26 | cytrak | My main router has everything right so that 192 can talk to 10 and vice versa .. I'm using IAX |
| 23:00.34 | cytrak | any ideas ? |
| 23:00.38 | manxpower | If you need DTMF transfers then you either didn't plan well or were too cheap. |
| 23:00.42 | riddlebox | manxpower, it is cause I just have analog phones |
| 23:00.55 | manxpower | riddlebox, why not use flash based transfers? |
| 23:01.37 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 23:02.24 | cytrak | nevermind i got it how many networks can the permit option have ? |
| 23:03.06 | Druken | manxpower: too cheap :) |
| 23:03.08 | manxpower | cytrak, why use permit/deny when you can use usernames and passwords? |
| 23:05.17 | riddlebox | manxpower, that does seem to work easier |
| 23:05.23 | cytrak | manxpower: I'm using username/passwords |
| 23:05.37 | riddlebox | but it is kinda slow either way |
| 23:05.51 | b00mer_ | any body know about this error: unable to find codec translation path from g723 to ulaw? then I get unable to open vm-intro (format g723): no such file or directory |
| 23:05.54 | bonfire1 | I have a general question about jobs in this area. In my country the whole VoIP thing is not very common and I think I can make some money by deploying it. So I thought I'd start a company that deploys asterisk servers over companies WANs. Anyone here already doing it in his country and cantalk to me about it? |
| 23:05.56 | manxpower | riddlebox, did you enable transfer and treewaycalling in zapata.conf? |
| 23:06.03 | *** join/#asterisk gmaruzz (n=maruzz@217-133-80-112.b2b.tiscali.it) |
| 23:06.07 | riddlebox | I am using a sip provider |
| 23:06.10 | manxpower | b00mer_, Asterisk does not support G723.1 |
| 23:06.33 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 23:06.39 | bkw__ | what? |
| 23:06.40 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:06.43 | bkw__ | doesn't support g723.1? |
| 23:06.51 | b00mer_ | aparently my aastra wants to do that.. ok.. I'll disallow it |
| 23:07.28 | manxpower | I wish they would remove allow=all and t/T from all Asterisk related docs. |
| 23:07.34 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:07.50 | riddlebox | dont forget r |
| 23:07.58 | Druken | manxpower: rewrite them for us :) |
| 23:08.01 | manxpower | riddlebox, yeah, that too. |
| 23:08.01 | Hmmhesays | i actually use R |
| 23:08.03 | file | it supports G723.1 passthru |
| 23:08.14 | bkw__ | the truth of the the matter is digium had g723.1 and g723.1b in CVS for ages.. you could just check it out into codecs/ and build a working codec. You can also download the ipp codec or find the lib |
| 23:08.16 | bkw__ | so lets stop the lies |
| 23:08.23 | bkw__ | it does infact support g723.1 |
| 23:08.29 | bkw__ | look at codec_g723_1.c |
| 23:08.39 | b00mer_ | ok |
| 23:08.40 | bkw__ | its just missing the lib that was hidden on digiums CVS for ages |
| 23:08.43 | bkw__ | and now is gone |
| 23:08.56 | b00mer_ | not a big deal for me to find another codec |
| 23:09.01 | bkw__ | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 23:09.19 | bkw__ | as an example |
| 23:09.22 | manxpower | file, saying "asterisk support G723.1 passthru" is like saying "killing yourself keeps you out of jail". Yes, both are true, but neither is very useful for solving the problem. |
| 23:09.42 | b00mer_ | what about this problem? |
| 23:09.46 | manxpower | bkw_, is that a new URL for it? |
| 23:09.48 | b00mer_ | only happens when I use the fwd call button on the phone |
| 23:09.53 | bkw__ | asterisk and codecs |
| 23:09.54 | b00mer_ | anybody know what this means?? : NOTICE[9305]: channel.c:1904 ast_read: Dropping incompatile voice frame on Local/92125551212@internal-a187,2 of formal ulaw since our native format has changed to slin |
| 23:09.54 | bkw__ | joys |
| 23:10.00 | bkw__ | it will always do g723.1 if its offered |
| 23:10.03 | bkw__ | even on pass thru |
| 23:10.07 | bkw__ | its restarded |
| 23:10.07 | VeNoMouS_ | b00mer_ u doing fax? |
| 23:10.08 | cytrak | manxpower: so i can remove the deny/permit strings entirely ? |
| 23:10.23 | b00mer_ | VeNoMouS_: some time I will |
| 23:10.26 | manxpower | cytrak, most people don't use them |
| 23:10.35 | Hmmhesays | i use them and I use R |
| 23:10.40 | VeNoMouS_ | b00mer_ its just the only time ive ever seen that is with faxes |
| 23:10.40 | Hmmhesays | er.. r even |
| 23:10.41 | riddlebox | manxpower, the funny thing, yesterday I saw a post on the forums that said you could do |m(/path/to/file) at the end of a dial string, I tried it and it didnt even work :) |
| 23:11.08 | manxpower | riddlebox, what does "show application dial" say? |
| 23:11.15 | b00mer_ | nope its when I set the phone to forward from the pri to the pri |
| 23:11.28 | manxpower | bkw_, so OpenPBX will come with G723.1 and G729 support? |
| 23:12.28 | manxpower | I suppose I need to write the IP holders about the illegal codec locations again. |
| 23:13.03 | riddlebox | hrmm why would you want the called party to transfer the calling party? |
| 23:13.29 | manxpower | riddlebox, It's a good way for people to get free long distance calls and bill them to you. |
| 23:13.48 | russellb | riddlebox: makes perfect since within an office ... |
| 23:14.14 | anthm | b00mer_ it's a race condition happens when the format of the channel is changed to raw for a temporary operation and a frame of the original format that the channel really should be is read during that window |
| 23:14.27 | bkw__ | manxpower, haha I'm not involved with OpenPBX so go for it.. they don't include it |
| 23:14.48 | manxpower | bkw_, for some reason I thought you were. |
| 23:14.57 | bkw__ | nope I work on FreeSWITCH now |
| 23:15.09 | manxpower | bkw_, another fork or a totally different code base? |
| 23:15.14 | bkw__ | new code |
| 23:15.15 | b00mer_ | anthm : fix? |
| 23:15.27 | anthm | what is the situation when it happens |
| 23:15.29 | anthm | is there 1 |
| 23:15.32 | anthm | or many of them |
| 23:15.42 | b00mer_ | many |
| 23:15.48 | manxpower | bkw_, Ah. I've basically stopped doing new Asterisk installs until some issues are resolved. |
| 23:15.52 | anthm | scrollling like mad? |
| 23:15.55 | b00mer_ | yep |
| 23:15.58 | b00mer_ | then it stops |
| 23:16.04 | b00mer_ | before the call is complete |
| 23:16.05 | anthm | what is the situation that causes it? |
| 23:16.22 | b00mer_ | when I call a phone that is forwarded |
| 23:16.32 | b00mer_ | forwarded at the phone |
| 23:17.32 | anthm | where are you forwarding to |
| 23:17.39 | anthm | channel type |
| 23:17.52 | b00mer_ | pri |
| 23:17.57 | b00mer_ | ? that help |
| 23:18.04 | bkw__ | 302 redirect right? |
| 23:18.20 | b00mer_ | no idea... I still new to this |
| 23:18.30 | b00mer_ | haven't sniffed the sip traffic |
| 23:18.30 | anthm | so an exten that has Dial(Zap) something or other ? |
| 23:19.01 | b00mer_ | Dial(Zap/g1 |
| 23:19.52 | b00mer_ | does that help? |
| 23:19.54 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
| 23:20.01 | bkw__ | its a sip phone doing a 302 it uses chan_local to exec the call do you see that in on the console? |
| 23:20.23 | bkw__ | Do you see Local/ when the redirect happens? |
| 23:20.25 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 23:20.43 | anthm | is the inbound call also zap ? |
| 23:20.52 | b00mer_ | yes |
| 23:20.55 | b00mer_ | on the zap |
| 23:20.57 | b00mer_ | also |
| 23:21.19 | b00mer_ | yes to the local |
| 23:21.43 | b00mer_ | just like the error message I sent |
| 23:22.13 | b00mer_ | <PROTECTED> |
| 23:22.14 | MoutaPT | I've *1.2.5 and i found this: SipUser agent "A" calls PSTN user "B", then puts MOH, and after that exits XLite, some times Asterisk doesn't notice that SIP user Agent has left and keeps MOH to PSTN , PSTN only ends when B hangs the call, and user "A" is unavailable in Asterisk CLI! |
| 23:22.49 | *** join/#asterisk gmaruzz (n=maruzz@217-133-80-112.b2b.tiscali.it) |
| 23:23.12 | MoutaPT | I've tried maxexpirey=120 and didn't solve |
| 23:23.50 | anthm | can you reproduce atm? |
| 23:24.06 | MoutaPT | anthm talknig to me? |
| 23:24.13 | b00mer_ | anthm: yes |
| 23:24.36 | b00mer_ | anthm: atm? |
| 23:24.43 | anthm | at the moment |
| 23:24.46 | b00mer_ | yes |
| 23:24.53 | b00mer_ | just did it |
| 23:25.23 | anthm | k 1 sec |
| 23:27.15 | Druken | uhg,.... who wants to send their yonger good looking sister to me? :) |
| 23:27.31 | Hmmhesays | she's 15 |
| 23:27.38 | luke-jr_ | Will using 'Monitor' prevent reinviting/transferring? |
| 23:27.39 | Druken | too yonge |
| 23:27.50 | anthm | b00mer_ you knoe where astxs is ? |
| 23:27.58 | Druken | Hmmhesays: i do have values... :) |
| 23:27.58 | b00mer_ | no |
| 23:28.11 | Hmmhesays | suuuure |
| 23:28.11 | b00mer_ | anthm: but I could do a find for it |
| 23:28.16 | Hmmhesays | and i'm the queen of england |
| 23:28.21 | Hmmhesays | want to see my boobies? |
| 23:28.32 | Druken | ewhh... man boobies |
| 23:28.43 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 23:28.45 | anthm | contrib/scripts/astxs |
| 23:29.04 | anthm | it cp it to /bin and chmod a+rx it |
| 23:29.32 | rhowe | Has anyone had experience hooking asterisk up to a TDM PBX using (say) a BRI interface? |
| 23:30.08 | rhowe | Well, I'm sure someone has - me, for one - anyone *here*? :) |
| 23:30.34 | b00mer_ | anthm: ok |
| 23:30.37 | b00mer_ | anthm: done |
| 23:31.03 | *** join/#asterisk PBXtech (i=nik@182.sub-70-218-33.myvzw.com) |
| 23:31.15 | anthm | k no |
| 23:31.16 | anthm | now |
| 23:31.42 | anthm | is asterisk running ? |
| 23:31.43 | PBXtech | does a dual core have much performance increase? (with smp) |
| 23:31.50 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@80.178.56.52.adsl.012.net.il) |
| 23:31.52 | b00mer_ | anthm: yes |
| 23:32.02 | anthm | astxs -autoload -install http://www.freeswitch.org/asterisk_stuff/app_changeformat.c |
| 23:32.10 | anthm | i just made this quick for you as a test |
| 23:32.16 | anthm | to prove a theory |
| 23:32.23 | anthm | it's an app that will change a channels format |
| 23:32.31 | anthm | not necessarily a good idea |
| 23:32.36 | anthm | but in your case |
| 23:32.45 | b00mer_ | anthm: eeek |
| 23:32.50 | b00mer_ | anthm: didn't compile |
| 23:33.00 | anthm | add ChangeFormat(ulaw) in the dialplan |
| 23:33.11 | cytrak | echo problem.... I got asterisk connected to a siemens PBX using PRI and I'm able to call the siemens phones and they are able to call my asterisk users.. the only problem is with echoing |
| 23:33.26 | anthm | ah |
| 23:33.28 | PBXtech | yea echoing sucks |
| 23:33.38 | b00mer_ | anthm: http://pastebin.com/670394 |
| 23:34.01 | cytrak | when I call a phone user using my iax account I get very litle echo almost unoticable |
| 23:34.17 | b00mer_ | anthm: any particular place? |
| 23:34.21 | cytrak | but the phone users hear an echo whenever they talk |
| 23:34.30 | anthm | try again |
| 23:34.50 | b00mer_ | anthm: ok |
| 23:34.54 | cytrak | PBXtech: any way around that |
| 23:35.02 | b00mer_ | anthm: done... registered application |
| 23:35.08 | anthm | add ChangeFormat(ulaw) in the dialplan |
| 23:35.16 | anthm | before dial out to the pri |
| 23:35.28 | anthm | on the exten you forward with |
| 23:35.33 | b00mer_ | ok |
| 23:35.58 | *** join/#asterisk gmaruzz (n=maruzz@217-133-80-112.b2b.tiscali.it) |
| 23:36.43 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:36.52 | anthm | my hypothesis is that it will change the read format of the channel to ulaw and not piss off the zap channel |
| 23:37.43 | b00mer_ | didn't help the messages |
| 23:37.43 | Az_au | i've just gone from v4 to v5 firmware on an snom360 and it seems to take a hell of a lot longer to register.. anyone know why this is? |
| 23:37.52 | b00mer_ | anthm: I did mention the call forwards |
| 23:38.07 | b00mer_ | anthm: right... its just the error messages on the console |
| 23:38.35 | anthm | so it comes to a stop after a while? |
| 23:38.42 | anthm | and starts working ? |
| 23:38.54 | b00mer_ | starts working like nothing is wrong |
| 23:39.06 | b00mer_ | spits out the error about 100 times |
| 23:39.08 | anthm | then that is the side effect of the chan_local |
| 23:39.24 | anthm | it does some voodoo where it trades souls |
| 23:39.31 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 23:39.34 | anthm | do you see the local chan exit right when it starts working |
| 23:39.47 | PoWeRKiLL | any idea why I get chan_iax2.c: Maximum trunk data space exceeded to XXX.XXX.XXX.XX:4569 |
| 23:39.56 | PoWeRKiLL | I check MAX_TRUNKDATA it's set to 200 channels and my server was with only about 10 or 20 channels when the message was happening and I didn't have audio anymore on my calls |
| 23:39.57 | b00mer_ | ok... so were chalking it up to beign |
| 23:40.07 | b00mer_ | no .. its working before the errors stop |
| 23:40.12 | robin_sz | meep? |
| 23:40.14 | b00mer_ | and works after they stop |
| 23:40.18 | *** join/#asterisk oej (n=oej@h2.ast.sipit.net) |
| 23:40.24 | anthm | it's the local |
| 23:40.25 | eKo1 | Any reason why I would be getting Got SIP response 488 "Not Acceptable Here" when calling from one extension to another. |
| 23:40.40 | oej | eKo1: Codec problem |
| 23:40.56 | b00mer_ | anthm: ok... cool... I'll move along... thanks for your help |
| 23:40.57 | anthm | i used to have that problem in chan_iax so i turned up that data var to a giant number that was all i ever could do |
| 23:41.08 | anthm | np |
| 23:41.17 | b00mer_ | anthm: do I need to unload this module? |
| 23:41.20 | robin_sz | alephcom, I used to have one .. until I "upgraded" the firmware :( |
| 23:41.28 | anthm | wont hurt anything |
| 23:41.34 | anthm | but you can |
| 23:41.35 | b00mer_ | will it just die off when I reload? |
| 23:41.51 | PoWeRKiLL | anthm I can set it to 20000 * 640 ? |
| 23:41.58 | alephcom | robin_sz: that kind of hurts. |
| 23:42.21 | anthm | that is what mark told me to do long ago |
| 23:42.31 | anthm | he said he would get to the bottom of the real reason some day |
| 23:43.08 | anthm | side effect is you use more ram but not that much really |
| 23:43.31 | rowter | anyone had a similar scenario, 2 Quad E1's 1 for 4 E1, 120 lines, and 2 for 3 Rhinos channel banks.. on a supermicro dual xenon.. |
| 23:43.33 | PoWeRKiLL | #define MAX_TRUNKDATA 640 * 2000 |
| 23:43.39 | anthm | just do unload app_changeformat in the cli and rm the .so file if you wish |
| 23:44.29 | PoWeRKiLL | anthm But the bug will come back some day right ? |
| 23:44.38 | anthm | probably |
| 23:44.48 | anthm | i think it's related to network latency or something |
| 23:44.51 | PoWeRKiLL | Cause when I got it I have to stop asterisk restart Zaptel and restart * |
| 23:45.05 | anthm | or it's deadlocking |
| 23:45.14 | PoWeRKiLL | If I only restart * it's not working back |
| 23:45.18 | anthm | there is only 1 thread for all the iax io of all your calls combined |
| 23:45.38 | anthm | maybe you are deadlocking zap then |
| 23:45.41 | PoWeRKiLL | So it's have to do something with zaptel too right ? |
| 23:45.48 | robin_sz | alephcom, : why hurts? |
| 23:46.04 | anthm | well or blocking moreso |
| 23:46.23 | russellb | anthm: not anymore ... |
| 23:46.27 | anthm | you should try it with no trunk and see if it still breaks to find out |
| 23:46.47 | PoWeRKiLL | You mean remove trunk=yes in the iax.conf ? |
| 23:46.49 | robin_sz | alephcom, trust me, it hurts me more than it does you ... I have now a non-working phoen and no way of going back :( |
| 23:47.00 | anthm | ya just to rule it out |
| 23:47.14 | anthm | cos if something is stuck the trunk data would surly overflow |
| 23:47.16 | russellb | chan_iax2 is now multithreaded. it even supports dynamically increasing the size of the thread pool if the load gets high |
| 23:47.17 | alephcom | robin_sz: There's no way "debrick" it? |
| 23:47.23 | robin_sz | nope. |
| 23:47.42 | PoWeRKiLL | anthm I will try it |
| 23:47.46 | alephcom | :-( |
| 23:47.47 | anthm | oh that is an improvement |
| 23:47.49 | robin_sz | no way to revert to "stable" version from beta |
| 23:48.22 | PoWeRKiLL | russellb your advise will be to upgrade to svn trunk is it ready for production mode ? :) |
| 23:48.36 | russellb | PoWeRKiLL: i would never recommend trunk in production ... |
| 23:48.42 | robin_sz | in effect, what I now have is a plant-pot, with an ethernet port :( |
| 23:48.53 | russellb | just more of a preview of what is to be released in a couple months |
| 23:49.33 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 23:49.39 | tehdely | howdy kiddos |
| 23:50.05 | tehdely | q: i am writing a rather elaborate AGI and there are parts where i want to step back into dialplan logic, do something, and return control to the AGI |
| 23:50.08 | tehdely | how can i accomplish this |
| 23:50.16 | tehdely | with the agi commandset |
| 23:50.32 | tehdely | would a GoSub, when returned, pass control back to the AGI or does that cmd only work from within the dialplan |
| 23:50.49 | robin_sz | only within the dialplan AFAIK |
| 23:51.01 | anthm | save your env into vars that you can pull back out or stash them in a cookie jar in a database |
| 23:51.17 | robin_sz | youd have to do a system call from the AGI script to asterisk |
| 23:51.31 | tehdely | crikey |
| 23:51.32 | tehdely | that's nuts |
| 23:51.33 | rhowe | russellb: So that 8 way dual core opteron box is going to be able to handle a decent number of IAX calls, then? :) |
| 23:51.40 | tehdely | anthm: how do i pass out of the agi |
| 23:51.45 | anthm | exit it |
| 23:51.52 | anthm | think like cgi |
| 23:51.54 | tehdely | ah so the next priority after the agi is called |
| 23:51.55 | asterboy | anyone see front page of digg recently? |
| 23:51.58 | tehdely | should be where i throw the extra logic |
| 23:52.03 | anthm | you save the state and pull it back |
| 23:52.06 | asterboy | mythTV + Asterisk + Home Security |
| 23:52.08 | asterboy | nice |
| 23:52.10 | tehdely | when it exits, performs logic, and then reexecs the agi |
| 23:52.11 | tehdely | ok |
| 23:52.18 | tehdely | that's twisted but it should work :) |
| 23:52.27 | robin_sz | very twisted ... |
| 23:53.01 | asterboy | the site is getting the digg affect |
| 23:53.02 | *** join/#asterisk Shaun2222 (i=Shaun@tina.ndcservers.net) |
| 23:53.21 | robin_sz | id just call asterisk client from within the agi, much like id call any other binary from a cgi |
| 23:53.30 | tehdely | also, with fastagi: if i use the AGI Dial() command and place a call, and the answering caller hangs up |
| 23:53.39 | anthm | what kinds of things do you need to do? |
| 23:53.39 | tehdely | does asterisk stop the agi and drop the channel right there |
| 23:53.48 | tehdely | or can i be dropped back into it and check dial status, etc. |
| 23:53.56 | anthm | dial inside agi is daft |
| 23:54.22 | tehdely | what is the main problem with that approach |
| 23:54.37 | tehdely | should i be creating callfiles or using manager interface instead for dialing? |
| 23:55.31 | PoWeRKiLL | I'm trying to popup a url page on my computer when I got a call any idea ? |
| 23:55.47 | anthm | you tend to end up with zombies cos you are doing fork and exec in those agi |
| 23:55.53 | tehdely | PoWeRKiLL: linux? |
| 23:56.14 | PoWeRKiLL | I've done a AJAX/PHP page that check a url and an AGI that set a flag |
| 23:56.34 | PoWeRKiLL | So when a call come the page got refresh with some info about the caller |
| 23:56.57 | PoWeRKiLL | tehdely no I need it to be client independant |
| 23:57.17 | tehdely | anthm: Dial causes a fork? |
| 23:57.26 | anthm | the agi does |
| 23:57.48 | tehdely | but if it Dials it is just sending control back to the parent process, right? |
| 23:57.53 | tehdely | over named pipe? |
| 23:58.03 | tehdely | why does this cause an additional fork+exec is what i am asking |
| 23:58.26 | anthm | not additional one but the one from the agi |
| 23:58.35 | anthm | ends up zombies a lot of times |
| 23:58.44 | anthm | if the call ends oddly |
| 23:58.45 | tehdely | let us put it this way, this agi is basically reimplementing basic dialplan logic in ruby |
| 23:58.53 | tehdely | so the agi will exist for pretty much the whoel lifetime of any call |
| 23:58.56 | tehdely | made into or through the pbx |
| 23:59.09 | tehdely | and the way im doing it i want it to just die when a channel is hung up, from either end |
| 23:59.11 | anthm | yah most ppl turn to that approach |
| 23:59.57 | tehdely | what would you suggest for placing calls then |
