| 00:00.15 | nebajoth | hmm |
| 00:00.16 | nebajoth | ok |
| 00:00.18 | nebajoth | cool |
| 00:00.27 | nebajoth | thanks D-Fender |
| 00:01.11 | [TK]D-Fender | nebajoth: Can you describe your planned needs? |
| 00:01.18 | nebajoth | sure thing! |
| 00:01.29 | nebajoth | I have 4 employees |
| 00:01.39 | nebajoth | who spend most of their day on the phone, taking and making sales calls |
| 00:01.51 | nebajoth | I want one number on my business cards |
| 00:02.07 | nebajoth | and when they call, they get through on that line, and if its busy, they get punted over to one of the empty lines |
| 00:02.13 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 00:02.27 | nebajoth | does that make sense? |
| 00:02.28 | martha | nebajoth: I think that has to be done on the telco side |
| 00:02.44 | nebajoth | martha: ok... so I should call them and ask about it |
| 00:02.52 | nebajoth | its probably 'call forwarding' or something |
| 00:03.00 | outtolunc | huntgroup |
| 00:03.10 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:03.19 | nebajoth | huntgroup? |
| 00:03.22 | [TK]D-Fender | nebajoth: Ok, all very basic |
| 00:03.37 | outtolunc | google:telecom glossary |
| 00:03.51 | [TK]D-Fender | nebajoth: the feature you'll ask of your telco is usually "hunt group". |
| 00:04.15 | nebajoth | awesome |
| 00:04.19 | nebajoth | thanks guys |
| 00:04.27 | nebajoth | that's the information I needed |
| 00:04.30 | nebajoth | just didn't have the lingo |
| 00:04.40 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:04.44 | [TK]D-Fender | nebajoth: A warning on embedded devices like that though is that it has a weak CPU so you won't be able to do much in terms of transcoding, erecording, etc |
| 00:05.32 | nebajoth | hmm |
| 00:05.44 | JT | also, POTS doesn't scale, keep that in mind |
| 00:06.38 | nebajoth | [TK]D-Fender: Can you recommend an inexpensive, non-embedded system that works out of the box? Or should I consider just installing it on a computer myself? |
| 00:06.48 | nebajoth | JT: You mean, other than by adding more lines? |
| 00:07.14 | [TK]D-Fender | nebajoth: Regular PC... |
| 00:07.19 | legis | * outside, sip phones inside(nat), what are my options for the sip phones, stun and port forwarding only right? |
| 00:07.46 | JT | nebajoth: for any real amount of lines you should consider T1/E1 (depending on country) PRI circuits |
| 00:07.59 | JT | they are digital and are much better call handling wise |
| 00:08.08 | JT | and you can have practically unlimited DID numbers |
| 00:08.19 | nebajoth | JT: I'm a small company -- would that be cost effective? |
| 00:08.35 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 00:08.41 | JT | i don't know, maybe not for 4 lines |
| 00:08.47 | JT | depends where you are |
| 00:08.50 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 00:09.25 | JT | in metro areas in the US it usually makes sense around 8-10 lines |
| 00:09.38 | JT | here in australia PRI makes sense if your requirement is as little as 4-6 lines |
| 00:09.52 | nebajoth | cool |
| 00:14.50 | [TK]D-Fender | legis: NONE of the above. Go read the guide : |
| 00:14.51 | [TK]D-Fender | ~sipnat |
| 00:14.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:14.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 00:15.15 | [TK]D-Fender | nebajoth: Where are you located? |
| 00:15.40 | [TK]D-Fender | Canada, good. |
| 00:15.56 | nebajoth | yeah |
| 00:16.00 | [TK]D-Fender | nebajoth: You MIGHT want to consider going VoIP for your "lines" |
| 00:16.03 | nebajoth | Why is that good? |
| 00:16.20 | [TK]D-Fender | nebajoth: North America has a lot of options. |
| 00:16.24 | legis | [TK]D-Fender: thx |
| 00:16.28 | nebajoth | hmm |
| 00:16.32 | nebajoth | I've got VOIP at home |
| 00:16.40 | nebajoth | I like it well enough, but it gets dodgy from time to time |
| 00:16.42 | [TK]D-Fender | nebajoth: Got a fax for your business? |
| 00:16.45 | nebajoth | I don't want that to happen at work |
| 00:16.48 | nebajoth | yeah |
| 00:16.53 | nebajoth | we need to have a fax line |
| 00:17.24 | [TK]D-Fender | nebajoth: You could run 1 fax line for a physical fax mahcine, DSL over it, and voip over that. les.net & unlimitel are both decent optios |
| 00:17.46 | [TK]D-Fender | nebajoth: And your ISP is a good one to work with |
| 00:18.00 | nebajoth | [TK]D-Fender: Thanks, I'll check both those links now |
| 00:18.20 | [TK]D-Fender | nebajoth: Would reduce your wiring requirements and telco costs an remove any special hardware requirements |
| 00:19.21 | nebajoth | hmm |
| 00:19.26 | nebajoth | what are the arguments AGAINST it, then? |
| 00:21.54 | JT | reliability |
| 00:21.55 | *** part/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 00:21.57 | *** join/#asterisk yardB (n=yardie@c-68-44-45-241.hsd1.nj.comcast.net) |
| 00:21.58 | [TK]D-Fender | nebajoth: internet issues = phone issues. Quality could be spotty based on things "in the wild". You don have a lot of simultaneous calls though. Single DSl would do fine. |
| 00:22.17 | [TK]D-Fender | nebajoth: Many benifits of course. |
| 00:22.27 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:23.14 | yardB | an asterisk question ..is it possible to answer a call after a specified number of rings? |
| 00:23.41 | JT | no, but after a specified amount of time, yes |
| 00:23.49 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:24.46 | yardB | JT how? i tied wait(20) before answer but it didn't work |
| 00:25.23 | nebajoth | hmm |
| 00:26.07 | yardB | s,1,Wait,20 |
| 00:26.17 | yardB | s,2,Answer() |
| 00:28.28 | yardB | any help on how to specify a time delay before answering? |
| 00:29.02 | nebajoth | I think I'd rather stick to PSTN for now, and use * to use IP phones at the site |
| 00:29.20 | nebajoth | for reliability |
| 00:29.28 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 00:29.41 | LemensTS | s,1,Wait(20) |
| 00:30.38 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:30.43 | JT | yardB: what sort of lines? |
| 00:30.47 | JT | and yeah, what LemensTS said |
| 00:30.48 | LemensTS | nebajoth: That is not a bad idea, you always have the option of VOIP later |
| 00:30.53 | yardB | ok, thanks Lemens |
| 00:31.04 | yardB | analog JT |
| 00:31.10 | JT | ah yep |
| 00:31.21 | yardB | ok, thanks |
| 00:31.24 | JT | they're the only channels calls come in on the "s" priority :) |
| 00:31.53 | legis | Are there any billing apps for asterisk? |
| 00:32.00 | nebajoth | I might even install a VOIP in parallel for a while, and see how it goes |
| 00:32.10 | nebajoth | but I need the reliability |
| 00:32.15 | nebajoth | so much of our work is done over the phone |
| 00:32.47 | nebajoth | has anyone here actually played with an IP04? |
| 00:32.50 | JT | and you still need your incoming lines, if you move them to voip, you'd have to port your numbers to the voip carrier |
| 00:33.05 | LemensTS | nebajoth: Thats what i do, i run pstn, vonage, and ITSP in asterisk. Im constantly testing different ITSP's. 3 years ago vonage was crappy, but I swear we have no problems out of them in the past 1.5 years |
| 00:33.21 | nebajoth | cool |
| 00:33.34 | nebajoth | +1 for vonage |
| 00:33.56 | LemensTS | I remember spending days on the phone with vonage 3 years ago..havent called them in 2 years |
| 00:34.05 | LemensTS | but $$$ |
| 00:35.31 | nebajoth | LemensTS: What are you using for your asterisk box? |
| 00:36.03 | legis | vonage = features |
| 00:36.05 | LemensTS | Ive messed with all the distros, and I find it easiest in Debian |
| 00:37.07 | LemensTS | Probably more documentation in Centos for asterisk tho, since that is what Digium recommends. |
| 00:38.01 | angryuser | debian or centos are both nice, question of preference, yum or apt-get ;) |
| 00:38.10 | nebajoth | I prefer apt-get myself |
| 00:38.38 | nebajoth | what cards do you have in your box? |
| 00:38.58 | LemensTS | legis: you dont need features at the ITSP if you got a feature box like asterisk :D They do have fail-over tho, it will go to pstn that you specify. Im not sure if you can hook vonage strait into asterisk sip/iax..? |
| 00:39.55 | LemensTS | Ive had to work with gentoo alot and all i can say is emerge is intense and confusing. Apt-get is so wonderful |
| 00:40.18 | nebajoth | I ran gentoo for years |
| 00:40.22 | nebajoth | its flexible as all hell |
| 00:40.26 | nebajoth | but I'm running ubuntu now |
| 00:40.30 | nebajoth | so that should tell you someting |
| 00:40.37 | legis | LemensTS: probably not but you are right can't have more features than a * box :) |
| 00:41.08 | legis | LemensTS: aptitude! |
| 00:41.40 | angryuser | i wonder how much in % people are able to economise? , after asterisk install and voip use , i am able to drop 60% of bill |
| 00:41.47 | LemensTS | legis: ive always done apt-get and im used to it. i read you dont want to use aptitude and apt-get on the same machine. |
| 00:41.58 | Qwell | both just use dpkg |
| 00:42.11 | Qwell | nebajoth: and, no longer running gentoo means you got lazy :p |
| 00:42.20 | LemensTS | angruser: our vonage bill is $400-450 a month. Using asterisk and voipstreet or another ITSP, i figured up $50-$150 |
| 00:42.49 | nebajoth | Qwell: too true :D |
| 00:43.01 | LemensTS | .017/minute is alot cheaper then like a few lines at $20-$45 a pop |
| 00:43.07 | Qwell | genmoo ~ # cat /etc/gentoo-release |
| 00:43.07 | Qwell | Gentoo Base System release 1.12.11.1 |
| 00:43.07 | Qwell | :D |
| 00:43.11 | LemensTS | we do alot of calling to |
| 00:43.47 | LemensTS | (and im sure we confuse the hell out of customers calling them on different vonage lines at different numbers...with asterisk i can pass 1 singel number out to customers) |
| 00:44.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:44.26 | angryuser | with the same voice quality ? it took me a month to find a good voip provider, most of them were just a bunch of resellers |
| 00:44.53 | nebajoth | I can't decide if I want to buy these IP04s or build my own box |
| 00:45.06 | nebajoth | embedded systems appeal to me for their low power consumption |
| 00:45.11 | nebajoth | not to mention small size |
| 00:45.14 | Qwell | nebajoth: get something supported |
| 00:45.19 | Qwell | (or build your own box) |
| 00:45.26 | angryuser | nebajoth if you dont need to expand, buy ip04 |
| 00:45.32 | LemensTS | im currently trying a new itsp. We were using voipstreet but ive had some problems lately using them...nothing has compared to vonage yet, and i havent spent alot on something like bandwidth.com yet |
| 00:45.39 | nebajoth | couldn't I just get two ip04s and trunk them together? |
| 00:45.41 | Qwell | personally, I use an AA50 - I love it :D |
| 00:45.47 | nebajoth | AA50? |
| 00:45.55 | Qwell | Digium Appliance |
| 00:46.02 | nebajoth | oh yes |
| 00:46.05 | nebajoth | how much are those? |
| 00:46.36 | Qwell | not sure.. disclaimer: I got mine for free, because I helped create it |
| 00:46.48 | LemensTS | Qwell: i thought about getting an AA50 to test, am i going to be stuck using the gui |
| 00:46.59 | [TK]D-Fender | </notpossiblybiasedopinion> |
| 00:47.00 | file | wishes he had an AA50 |
| 00:47.01 | Qwell | LemensTS: that's sort of the point of buying an appliance |
| 00:47.09 | Qwell | BUT...no. You can edit the configs, it's just...yeah |
| 00:47.13 | Qwell | file: :( |
| 00:47.38 | file | I'd rather an AA60 actually |
| 00:47.41 | nebajoth | Qwell: 1GB enough for voicemail for 4 salesmen? |
| 00:47.47 | Qwell | nebajoth: more than enough |
| 00:47.57 | Qwell | 10mb would probably be more than enough :) |
| 00:48.04 | Qwell | 1GB is...a lot of audio |
| 00:48.06 | nebajoth | hmmm |
| 00:48.34 | Qwell | file: mmm, that too |
| 00:48.47 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 00:48.54 | Qwell | file: Do you have a rack in your apt yet? |
| 00:49.01 | file | Qwell: no |
| 00:49.13 | file | I priced out one for my house though |
| 00:49.16 | Qwell | :D |
| 00:49.33 | Qwell | You could get like a quarter or half rack |
| 00:50.07 | file | yup |
| 00:50.20 | LemensTS | lol i got a 72 inch rack that is like 3.5ft deep..its a monster |
| 00:50.27 | LemensTS | 4 post |
| 00:50.48 | Qwell | isn't that like...standard dimensions? |
| 00:51.57 | nebajoth | Qwell: where can I get numbers on how much voicemail can be stored per megabyte? |
| 00:52.31 | Qwell | umm, well, how big is 1 minute of ulaw? :) |
| 00:53.14 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:53.40 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:53.51 | nebajoth | peers around expectantly |
| 00:54.01 | Qwell | tip: I have no idea. |
| 00:54.07 | Qwell | it's not much though |
| 00:54.50 | *** join/#asterisk rudal123 (n=rudal@216-176-188-22.reverse.wowrack.com) |
| 00:54.55 | nebajoth | hmmm |
| 00:55.13 | nebajoth | anybody know how much voicemail you can store in a megabyte? |
| 00:55.25 | nebajoth | :D |
| 00:55.58 | nebajoth | :( |
| 00:56.16 | Qwell | ulaw is 1 bit per sample, yeah? |
| 00:56.31 | Qwell | * 8000 samples per second = 1000 bytes |
| 00:56.58 | rudal123 | anyone knows anything about 7941G with asterisk based phone system? |
| 00:57.38 | Qwell | so, 1mbyte would be...1000 seconds? |
| 00:57.40 | JT | 8000 bytes per second |
| 00:57.46 | Qwell | oh |
| 00:57.52 | Qwell | so, / 8 |
| 00:57.56 | JT | 64kbps |
| 00:58.05 | rudal123 | im having problem having the 7941G to register to the phone system |
| 00:58.10 | Qwell | so 1mbyte of audio == x length? |
| 00:58.26 | Qwell | 16 seconds? why does that seem wrong? |
| 00:58.45 | Qwell | because it is |
| 00:58.50 | nebajoth | haha |
| 00:58.55 | Qwell | 128 seconds? |
| 00:59.04 | Qwell | I'm tired :p |
| 00:59.18 | nebajoth | you're still moving about five times faster than my brain can |
| 00:59.23 | nebajoth | and I'm NOT tired |
| 00:59.55 | Qwell | if it's 128 seconds, that means 1mbyte = 2 minutes, * 1024 = 2048 minutes = 34 hours? |
| 01:00.31 | [TK]D-Fender | nebajoth: You'll win out on endurance. Get'em kiddo! |
| 01:00.33 | *** join/#asterisk rukus (n=daniel@dsl-245-72-99.telkomadsl.co.za) |
| 01:00.49 | [TK]D-Fender | pushes nebajoth back into the ring |
| 01:01.19 | rukus | been at this now for a few hours, got basic landline out, tow intenernal phones etc, got gtalk into a voip phone, as well as FWD .. any more nice little tricks to try ? |
| 01:01.26 | rukus | it's 3am here :/ |
| 01:01.36 | [TK]D-Fender | rukus: Try SLEEPING |
| 01:01.40 | rukus | haha |
| 01:01.44 | nebajoth | :D |
| 01:01.46 | Qwell | nebajoth: SO, 1gb is either 34 hours, or 4.25 hours. It depends on how wrong my math is. |
| 01:01.52 | rukus | yea, I just saw it's actaully 3am. |
| 01:01.53 | nebajoth | haha |
| 01:01.55 | Qwell | either way...it's a lot :p |
| 01:01.56 | nebajoth | that's a huge difference |
| 01:01.57 | nebajoth | but you're right |
| 01:01.59 | nebajoth | both are enough |
| 01:03.14 | *** join/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 01:03.46 | *** join/#asterisk rukus (n=daniel@dsl-245-77-13.telkomadsl.co.za) |
| 01:04.02 | rukus | not the best of connections out here on adsl |
| 01:06.11 | *** join/#asterisk Quentusrex (n=nobody@c-71-227-241-183.hsd1.or.comcast.net) |
| 01:07.24 | Quentusrex | How do I diagnose one way audio for one of my extensions? |
| 01:09.06 | Quentusrex | When someone calls in they can't hear the local person on the extension talking... |
| 01:09.16 | *** join/#asterisk voxter (n=hardcore@vpn.voxter.com) |
| 01:09.18 | Qwell | ~sipnat |
| 01:09.18 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:09.20 | Quentusrex | but from a different extension the outgoing audio works fine. |
| 01:09.25 | Qwell | Quentusrex: start there |
| 01:09.54 | Quentusrex | asterisk itself works fine. |
| 01:10.01 | voxter | Is there any way to have asterisk do codec selection from the dial plan before calling a phone? For example if the phone has allow=ulaw and allow=g729, can I say that if user XYZ calls, force ulaw, or if user ABC calls force g729? |
| 01:10.11 | Quentusrex | because if I use extension 202 then audio both ways works great. |
| 01:18.51 | [TK]D-Fender | Quentusrex: Your description lacks proper details. Please be very specific in exactly where the call is coming from, and going to with a breakdown of the hardware involved and the networking inbetween * and both ends. |
| 01:21.09 | *** join/#asterisk ZX81 (n=matt@120.89.80.110) |
| 01:25.47 | *** join/#asterisk ZX81 (n=matt@120.89.80.110) |
| 01:25.49 | yardB | gentlemen, the wait command before answer doesn't work ie |
| 01:25.55 | ZX81 | hi all, is there any way to avoid a reload trashing asterisk queue stats? |
| 01:26.21 | yardB | s,1,wait(20) |
| 01:26.27 | ZX81 | or |
| 01:26.31 | yardB | a,2,Answer() |
| 01:26.31 | ZX81 | answer(20000) |
| 01:26.32 | LemensTS | yardB: oh yea? it works on asterisk 1.4 |
| 01:26.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:26.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:26.42 | ZX81 | type show application answer |
| 01:26.55 | yardB | i am using 1.4 |
| 01:27.00 | ZX81 | [Description] |
| 01:27.01 | ZX81 | <PROTECTED> |
| 01:27.01 | ZX81 | answer it. Otherwise, it has no effect on the call. If a delay is specified, |
| 01:27.01 | ZX81 | Asterisk will wait this number of milliseconds before returning to |
| 01:27.01 | ZX81 | the dialplan after answering the call. |
| 01:27.04 | [TK]D-Fender | yardB: PASTEBIN your call at verbose 10 |
| 01:27.07 | [TK]D-Fender | ~pb |
| 01:27.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:27.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 01:27.44 | ZX81 | is parsing show queue x to get sla |
| 01:27.50 | ZX81 | but data disappears on a reload |
| 01:27.58 | ZX81 | [sp] |
| 01:28.28 | *** part/#asterisk Quentusrex (n=nobody@c-71-227-241-183.hsd1.or.comcast.net) |
| 01:31.23 | yardB | ZX81: yea but I don't want the called to be answered until after 4 ring , 20 sec |
| 01:31.49 | [TK]D-Fender | yardB: .... PASTEBIN IT |
| 01:32.00 | yardB | ok |
| 01:35.08 | angryuser | do you allways have the spare parts for * in case of hardware fail? (fxo fxo pri bri T1) |
| 01:35.49 | LemensTS | If you dont want to be down you do. |
| 01:36.08 | LemensTS | A day or 2 of no sales call can be more then a spare card |
| 01:37.07 | angryuser | is there any stat's of hardware life for * ? with normal operation of servers |
| 01:37.27 | russellb | forrrreeevvverrrr |
| 01:37.43 | JT | yeah, if you're serious about uptime, carry spares. |
| 01:37.52 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 01:38.09 | JT | warranties will never give you the same response time |
| 01:39.36 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-41-171.dhcp.mrqt.mi.charter.com) |
| 01:39.53 | angryuser | yes i understand that, it's easy to calculate a price of week of downtime of bri/pri for example, but what is the life cycle ~ of digium products ? |
| 01:40.13 | angryuser | 5 years ? more ? |
| 01:41.05 | JT | what products? |
| 01:41.18 | angryuser | interface pci cards |
| 01:41.30 | JT | oh i dunno |
| 01:41.37 | JT | i think they upped the warranties on them |
| 01:42.28 | angryuser | when you come to alcatel&whatever the will tall you, phone life, pbx life |
| 01:42.45 | JT | heh |
| 01:42.49 | russellb | angryuser: digium cards have a 5 year warranty |
| 01:45.03 | Qwell | actually, Alcatel-Lucents stated warranty is 1 year on voice hardware |
| 01:45.05 | LemensTS | Thier isnt really a way to tell the life of a pci card. Thier are plenty of early 90 pci modems in use today still. |
| 01:45.41 | russellb | Qwell: heard a story on the radio this morning about how that company hasn't been doing well at all |
| 01:45.50 | Qwell | oh? |
| 01:46.16 | LemensTS | alcatel-lucents may be getting 500million from microsoft, microsoft is going to try to appeal tho of course |
| 01:46.24 | russellb | yep ... they were talking about how the merger of a french company and american company turned out to be a lot more difficult than they thought, heh |
| 01:46.29 | russellb | culturally, i think |
| 01:46.46 | angryuser | have they addes that the director went away with 5milE? |
| 01:46.51 | angryuser | added* |
| 01:47.13 | angryuser | and anootehr person who worked a year for them with another 6milE |
| 01:47.58 | angryuser | 1.1 bil lost in total |
| 01:48.09 | angryuser | losses* |
| 01:50.06 | [TK]D-Fender | yardB: So... where's that broken call for us to look at? |
| 01:55.01 | yardB | TK : i got on the site but i wasn't sure what were the steps necessary .. never used it before |
| 01:55.29 | yardB | just put the code on? |
| 01:55.36 | LemensTS | yardB: just paste your code in thier and put your name in and hit submit |
| 01:55.43 | LemensTS | then paste the link it gives you in here |
| 01:55.52 | yardB | ok |
| 01:56.14 | irisht | brb |
| 01:56.22 | LemensTS | btw the name does not have to be your mirc name or anything...i dont even know if it is required |
| 01:56.30 | [TK]D-Fender | it isn't |
| 02:02.13 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 02:02.13 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- Related channels: #asterisknow, #asterisk-gui, #switchvox, #freepbx, #asterisk-commits, #asterisk-bugs, #asterisk-dev |
| 02:02.13 | voxter | [TK]D-Fender: I'd like to be able to do something where when a receptionist (for example) initiates a page, it forces the use of ulaw on the main registration, instead of g729 |
| 02:02.24 | voxter | I think SIPPrefCodec might do the trick, if that ever made it in to the tree |
| 02:02.38 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:02.58 | voxter | Ah, requires patching 1.2 |
| 02:03.17 | [TK]D-Fender | voxter: 1.2 ... lol |
| 02:03.27 | voxter | [TK]D-Fender: if it aint broke... |
| 02:03.41 | [TK]D-Fender | voxter: If it needs patching... ;) |
| 02:03.47 | [TK]D-Fender | (it is) |
| 02:04.12 | [TK]D-Fender | yardB: So, half an HOUR later, where is it? |
| 02:04.50 | yardB | i did put it up |
| 02:05.01 | *** join/#asterisk chendy (n=chatzill@58.251.231.216) |
| 02:05.08 | LemensTS | where is the link, u need to post it in here for us |
| 02:05.13 | [TK]D-Fender | yardB: How nice of you to provide the LINK |
| 02:05.35 | [TK]D-Fender | yardB: Only a dozen popular sites whose posts MUST be obvious right? |
| 02:06.52 | yardB | hold strain .. i am new to this stuff |
| 02:07.06 | voxter | [TK]D-Fender: 1.4 would need to be patched too, there's nothing in 1.4 for it either |
| 02:07.15 | yardB | http://pastebin.com/m7d8056b3 |
| 02:07.48 | LemensTS | thats not a call in the cli at verbose 10, thats your dial plan of extensions.conf |
| 02:08.02 | [TK]D-Fender | yard, And after all of that, you show me some random snippet ot dialplan. I asked for you to show me a broken CALL |
| 02:08.39 | LemensTS | go into your cli 'asterisk -vvvvvvvvvvr' then try to dial that extension that waits 20 seconds. then copy and paste it to pastebin |
| 02:10.12 | yardB | ok, it wasn't clear what u needed .. ok, i understand now however it is not easy to copy .. two independent machines |
| 02:10.38 | LemensTS | are you on windows using putty? |
| 02:10.41 | yardB | i will copy and paste |
| 02:10.41 | [TK]D-Fender | yardB: yes, it really is. SSH : It ain't Raw-Cat Science |
| 02:11.18 | [TK]D-Fender | [21:27]<[TK]D-Fender>yardB: PASTEBIN your call at verbose 10 |
| 02:11.32 | [TK]D-Fender | [21:50]<[TK]D-Fender>yardB: So... where's that broken call for us to look at? |
| 02:12.08 | LemensTS | heh |
| 02:12.10 | [TK]D-Fender | Looks pretty darn explicit to me. |
| 02:12.29 | [TK]D-Fender | CALL <- with like DETAILS even! |
| 02:14.46 | yardB | i will have to do this another time .. thanks for your help ..i have to travel tomorrow and i need to go pack |
| 02:14.58 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:15.00 | LemensTS | ruh row |
| 02:16.37 | [TK]D-Fender | Pathetic. Just plain pathetic. I could code up a small office config from scratch in the time wasted on this last volley |
| 02:17.38 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 02:17.57 | [TK]D-Fender | Go to * CLI. "set verbose 10". ***PLACE CALL**. Leave * CLI. Scroll back & copy. Go to pastebin.com. Paste. Hit "submit". Copy Link. Come back to IRC client. PASTE LINK. |
| 02:18.21 | [TK]D-Fender | Holy McFuck is this too much to ask? |
| 02:18.35 | LemensTS | how do i set the verbose to 10? |
| 02:18.46 | angryuser | vvvvvvvvvvvv |
| 02:18.53 | LemensTS | lol im joking |
| 02:18.53 | [TK]D-Fender | LemensTS: Sorry, no pass on that one, I quoted it exactly :p |
| 02:19.28 | *** join/#asterisk hfb (n=hfb@cpe-76-87-161-213.socal.res.rr.com) |
| 02:19.32 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 02:19.34 | JT | copying and pasting can be a pain in the arse if it's a lot of text |
| 02:19.39 | JT | but then there's full log |
| 02:20.10 | [TK]D-Fender | JT: Yeah, it takes 5 second longer to scroll to the top |
| 02:20.42 | [TK]D-Fender | JT: PuTTY has a nice "clear buffer" option and a single "copy all to clipboard" option. NO &^#%$ING EXCUSE |
| 02:22.17 | bbryant | you can also type !printf "\033c" to clear the terminal buffer from asterisk |
| 02:22.24 | voxter | Is the channel variable SIP_CODEC still supposed to work? it does not seem to be doing anything. |
| 02:23.33 | [TK]D-Fender | voxter: is ANY audio played on the call before your attempt to dial out to something else? |
| 02:24.54 | voxter | [TK]D-Fender: In this test case, it is, but should it not change the codec for the second leg of the call? I am calling into an IVR via a SIP connection (g729) and dialing an extension off that menu, before the Dial() to the new extension, I am setting SIP_CODEC, intending on having the second leg of the call be ulaw. |
| 02:25.16 | voxter | I'm not looking to change the codec of the existing call mid-stream, of course. |
| 02:25.39 | hardwire | when you tap a pot's line.. all you get is mono audio.. right? |
| 02:25.46 | hardwire | you can't differentiate between the caller and the callee |
| 02:26.27 | hardwire | but if that pots line goes to a digital channel bank, it uses echo cancellation of some kind to remove the remote parties voice from the transmitted audio.. |
| 02:26.30 | hardwire | or am I just silly? |
| 02:26.39 | hardwire | is ring and tip equiv to rx and tx? |
| 02:27.20 | JT | no |
| 02:27.30 | JT | one is positive, one is negative |
| 02:27.36 | JT | with respect to telco battery voltage |
| 02:27.43 | hardwire | and it's a monolithic audio channel eh? |
| 02:28.01 | hardwire | I've never really understood POTS all that well |
| 02:28.06 | hardwire | digital is soooo much easier to grok |
| 02:28.19 | JT | i think they're seperate |
| 02:28.25 | JT | different frequency AC signals |
| 02:28.32 | hardwire | really? |
| 02:28.41 | JT | the hybrid in the telephone seperates/combines them |
| 02:29.05 | hardwire | how do party lines work then? |
| 02:29.25 | [TK]D-Fender | hardwire: You vs everybody else |
| 02:29.37 | [TK]D-Fender | hardwire: Same as multiple people in a hoem |
| 02:29.42 | [TK]D-Fender | home* |
| 02:29.47 | hardwire | do you normally hear yourself in the speaker? |
| 02:29.58 | voxter | a little bit, its called sidetone |
| 02:29.59 | hardwire | how does it know to send the different channels to the speaker? |
| 02:30.09 | hardwire | wants to leaarn |
| 02:30.19 | Qwell | hardwire: the phone does that |
| 02:30.25 | Qwell | (or if you get echo) |
| 02:30.32 | voxter | [TK]D-Fender: so am i mistaken in my assumption that setting SIP_CODEC should change the codec for the second leg of the call? |
| 02:31.22 | [TK]D-Fender | voxter: perhaps. Never used it personally. |
| 02:31.55 | Qwell | hardwire: when you hear yourself in the speaker, I believe that's side-tone |
| 02:32.09 | Qwell | http://en.wikipedia.org/wiki/Sidetone |
| 02:32.20 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 02:33.13 | hardwire | Qwell: so if I pick up Party Phone A on POTS 1 I should hear exactly what Party Phone B on POTS 1 is hearing.. Myself. |
| 02:33.33 | Qwell | you would hear both sides |
| 02:33.40 | hardwire | on both phones |
| 02:33.49 | Qwell | less so on your own phone |
| 02:33.54 | hardwire | due to? |
| 02:34.01 | Qwell | a possible lack of sidetone |
| 02:34.03 | [TK]D-Fender | hardwire: Go read a telecom book. Seriously |
| 02:34.11 | hardwire | [TK]D-Fender: aren't those big? |
| 02:34.27 | hardwire | [TK]D-Fender: I can stop asking questions if this is really bugging you. |
| 02:34.32 | hardwire | it's all good. |
| 02:34.46 | [TK]D-Fender | hardwire: You need to get this from a nice consistent write-up |
| 02:34.55 | voxter | hardwire: i think the correct answer is we dont know for sure. |
| 02:35.02 | [TK]D-Fender | ~101 |
| 02:35.03 | jbot | 101 is probably Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:35.16 | hardwire | [TK]D-Fender: :) |
| 02:35.19 | [TK]D-Fender | might cover it in there |
| 02:35.43 | hardwire | I'm doing a quite for digital tapping.. and I have no idea how analog tapping really works.. so I got to thinking |
| 02:35.50 | hardwire | I'll read that when I'm not in a pseudo rush |
| 02:36.12 | [TK]D-Fender | hardwire: here : http://en.wikipedia.org/wiki/Telephone_hybrid |
| 02:37.10 | [TK]D-Fender | http://en.wikipedia.org/wiki/Hybrid_coil |
| 02:38.26 | [TK]D-Fender | thats a rather nice small nugget of info on how a hybrid separates your voice from theirs on a single-pair mono circuit. |
| 02:39.10 | Qwell | that VQR sounds rather cool |
| 02:40.03 | voxter | OK, so this seems to be the verdict. If you have two phones, both allow g729+ulaw, and you set SIP_CODEC=ulaw and call from one to the other, it will make both force to ulaw. If one is g729 only, and one is g729+ulaw and you set SIP_CODEC to ulaw, it wont make one g729 and one ulaw (and transcode), it will just fall back to g729. so presumably SIP_CODEC is used for the calling party, not the party about to be called. |
| 02:40.19 | Qwell | wow, that doesn't sound very good at all O.o |
| 02:43.20 | JT | hardwire: sidetone is audio that the caller hears of the caller's own voice, reintroduced by their handset to reassure them that their telephone is operating correctly and allowing them to properly carry on a conversation |
| 02:43.23 | hardwire | I love asciidoc |
| 02:44.33 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 02:48.09 | bbryant | voxter: I think I found the problem, want to try a patch? |
| 02:48.21 | voxter | bbryant: with regards to SIP_CODEC? |
| 02:48.26 | bbryant | yep |
| 02:48.34 | voxter | bbryant: sure, this box is 1.2.19 though.. |
| 02:48.39 | bbryant | ah |
| 02:49.15 | voxter | bbryant: it just so happens that the behavior i am seeing will work fine for me, but it does seem appropriate that if i set SIP_CODEC in the middle of the call, it will force that for the second leg, is that what you just patched? |
| 02:49.26 | [TK]D-Fender | bbyHe's ALREADY trying another patch |
| 02:49.28 | bbryant | voxter: i'll see if the same patch would apply |
| 02:49.42 | voxter | bbryant: i can always try it on a 1.4 box too, its no big deal |
| 02:49.49 | voxter | [TK]D-Fender: no i'm not., |
| 02:51.05 | bbryant | voxter: it applies to both, http://pastebin.com/f79ee89a3 |
| 02:51.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:51.19 | [TK]D-Fender | voxter: Oh wait, you mentioned it but didn't actually follow through? |
| 02:51.30 | drako | hi |
| 02:51.43 | voxter | [TK]D-Fender: i mentioned a patch that implemented SIPPrefCodec as a function, but found the SIP_CODEC channel variable -should- do the same thing anyways. |
| 02:52.01 | voxter | bbryant's patch may make the SIPPrefCodec function irrelevant |
| 02:52.06 | voxter | bbryant: I'll give it a go right now. |
| 02:53.10 | voxter | Ah, I see what your patch did. |
| 02:53.19 | voxter | makes the preferred codec selection absolute |
| 02:53.48 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 02:54.21 | LemensTS | Im not a c programmer, im assuming that file is c, and all he has to do is edit it and it takes effect? |
| 02:54.21 | bbryant | yep, otherwise it just adds it to the list of supported codecs on both sides |
| 02:54.38 | bbryant | LemensTS: it has to be recompiled too |
| 02:55.03 | voxter | bbryant: so far it didnt work how i expect it would, but let me try a sip debug and see if i notice whats going on. |
| 02:55.24 | JT | c is not generally an interpreted language, you can't just edit the .c and expect it to work, LemensTS |
| 02:55.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:55.52 | LemensTS | bbryant: So he would edit it wherever he installed the asterisk source from, then recompile asterisk? |
| 02:56.12 | bbryant | right |
| 02:56.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:58.16 | LemensTS | JT: yea ive had classes on languages that needed compiled a long time ago. I only do interpreter languages now. That is cool i will play around sometime and see if i can do something. |
| 02:58.41 | voxter | bbryant: ok so I may be doing this the wrong way to test.. What I'm doing is calling in to my * using a SIP DID (which is forced to g729). From the IVR i hit, I am dialing 106 (SIP phone) which has ulaw and g729 enabled. the dial plan sets SIP_CODEC=ulaw then dials 106. They both still merge with g729 |
| 02:59.10 | voxter | bbryant: What i would expect is that the first leg of the call remain g729 (of course) and the second would force to ulaw, but that is not happening. Now, from what I can tell, this may have something to do with the way SIP_CODEC works altogether. |
| 02:59.25 | voxter | bbryant: it seems as though SIP_CODEC tries to set the preference order for the calling party, not the called party |
| 02:59.48 | bbryant | voxter: it probably has to do with some codec optimization that gives first priority to keeping both ends on the same codec |
| 02:59.57 | bbryant | when they do that there's no transcoding, which is much less cpu intensive |
| 03:01.04 | voxter | bbryant: hmm yeah you're probably right. let me re test between two sip devices and see |
| 03:01.27 | voxter | bbryant: but i think the same thing will happen, I dont think you can force one leg of the call like that. |
| 03:02.30 | bbryant | voxter: maybe not, test it against trunk or 1.4 and if it still happens file a bug report on bugs.digium.com |
| 03:03.29 | voxter | bbryant: for sure. Thanks for the patch, I'll keep messing with it and see what i end up with. |
| 03:06.49 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 03:11.29 | rbd | when issuing a "RECORD FILE" command to agi on asterisk 1.4.x, I get back "510 Invalid or unknown command" ...is there something I'm missing here (perhaps the wrong args to it?) |
| 03:13.55 | rbd | the full command is "RECORD FILE /var/lib/recs_processing/lab1-int-011-ac/159422_1217473764+924572.wav.temp wav # 18000000 BEEP s=10" ...perhaps it doesn't like that + in the file name |
| 03:14.15 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:17.08 | rbd | nope...that wasn't it |
| 03:19.37 | LemensTS | how do i view information about a command at the cli? i forget |
| 03:20.09 | LemensTS | like the 'dial' cmd |
| 03:22.50 | LemensTS | or i should say application not cmd |
| 03:22.52 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 03:23.23 | LemensTS | core show application dial |
| 03:23.38 | LemensTS | i figured it out...quit talking to yourself |
| 03:24.12 | LemensTS | bbyrant: i just did my first asterisk .c file change and recompiled and it worked...neeet i did not know it was that simple. |
| 03:24.36 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:27.46 | bbryant | LemensTS: yep, and if you got the source from subversion, you can type "svn diff" to see a diff file of what you've changed |
| 03:31.08 | LemensTS | ~svn |
| 03:31.08 | jbot | Subversion (aka SVN) is version control software that aims to be a better CVS than CVS. See http://subversion.tigris.org/. |
| 03:36.09 | LemensTS | hrrmm..so i should have had subversion installed before i edited the .c file. Subversion keeps a .svn file logged of changes. If i had it done before and did 'svn diff fileIchanged' it would have worked? |
| 03:36.53 | LemensTS | (never used subversion before, just now learning FRS on sourceforg...i wrote an asterisk timeclock i put up) |
| 03:43.05 | gones | How to set music on hold in a diaplan ? |
| 03:44.28 | gones | I use Dial ( channel,timeout, m) , the option of 'm |
| 03:45.03 | gones | explain : Plays music-on-hold to the caller until the call is answered. Cannot be used with r. |
| 03:45.05 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:45.42 | [TK]D-Fender | gones: Whats to explain? You can't RING and hear MOH at the same time. |
| 03:46.31 | LemensTS | I use Dial ( channel,timeout, m) , the option of 'm |
| 03:46.53 | gones | The option of 'm' can plays music-on-hold to the caller until the call is answered. Cannot be used with r. |
| 03:47.27 | [TK]D-Fender | gones: Yes, you've pasted this twice. Whats your point? |
| 03:47.44 | LemensTS | i hear a verbose 10 comment ringin in my head again |
| 03:48.06 | gones | i can't hear music when i use the option 'm' , why ? |
| 03:48.08 | [TK]D-Fender | unload chan_brokenrecord.so |
| 03:48.23 | [TK]D-Fender | gones: maybe you didn't set up any music. |
| 03:48.32 | *** join/#asterisk pigpen2 (n=pigpen@faxserver.emtl.com) |
| 03:48.44 | LemensTS | Gones: yea check the cli as you call. it should say whats going on with the moh. |
| 03:48.47 | gones | may be , I don't how to set music ! |
| 03:48.50 | [TK]D-Fender | gone PASTEBINI a call attempt that you feel should be playing music. Make sure to do "set verbose 10" first. |
| 03:48.51 | [TK]D-Fender | ~pb |
| 03:48.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:48.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 03:49.31 | LemensTS | <[TK]D-Fender> [21:27]<[TK]D-Fender>yardB: PASTEBIN your call at verbose 10 |
| 03:49.31 | LemensTS | <[TK]D-Fender> [21:50]<[TK]D-Fender>yardB: So... where's that broken call for us to look at? |
| 03:50.02 | [TK]D-Fender | LemensTS: Come on, don't drudge that up, lets see if this one has the ability to follow simple direct instructions. |
| 03:50.16 | gones | The display echo "Started music on hold, class 'default' " ,but I can't hear anything . |
| 03:50.41 | [TK]D-Fender | gones: PASTEBIN the ENTIRE call attempt from start to finish |
| 03:51.47 | gones | Executing [99123@default:1] Dial("SIP/33441-081e1958", "Sip/33442|15|m") in new stack |
| 03:51.48 | gones | <PROTECTED> |
| 03:51.48 | gones | <PROTECTED> |
| 03:51.48 | gones | [Jul 31 11:46:32] WARNING[21177]: rtp.c:1147 ast_rtp_read: RTP Read too short |
| 03:51.48 | gones | <PROTECTED> |
| 03:51.48 | gones | <PROTECTED> |
| 03:51.50 | gones | <PROTECTED> |
| 03:51.52 | gones | <PROTECTED> |
| 03:51.54 | gones | <PROTECTED> |
| 03:51.56 | gones | <PROTECTED> |
| 03:51.58 | gones | [Jul 31 11:46:50] NOTICE[21177]: pbx.c:5511 pbx_builtin_hangup: Invalid cause given to Hangup(): " " |
| 03:52.00 | gones | <PROTECTED> |
| 03:52.07 | LemensTS | :D |
| 03:52.08 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 03:52.44 | WhiteWolf | gones: typically, when the word pastebin is in caps, they mean for you to do it that way <.< |
| 03:52.48 | [TK]D-Fender | gones: PASTEBIN. Spam the channel like that again and you'll be leaving skid-marks on your way out |
| 03:53.12 | [TK]D-Fender | and when I have the bot-message explain it in detail and like to 5 of them you'd better pay attention |
| 03:53.57 | [TK]D-Fender | gones: Do you now comprehend what a PASTEBIN is? |
| 03:54.18 | gones | no. |
| 03:54.20 | LemensTS | dumb question - ChanServ set u to moderator or whatever, did it do that because you have been in here without logging out over a certain amount of time or something? |
| 03:54.21 | WhiteWolf | i typically find it better just to embed the urls to where i want something in the message that's directing them to use it so there's no confusion that they could've "lost" the message in channel chatter |
| 03:54.34 | gones | what a PASTEBIN is ? |
| 03:54.39 | LemensTS | ~pb |
| 03:54.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:54.49 | WhiteWolf | LemensTS: he's got rights, and he asked Chanserv to op him so he could do channel management |
| 03:55.08 | WhiteWolf | LemensTS: one of the many advantages of a services based irc network, so that people don't have to worry about losing control over a channel |
| 03:55.37 | [TK]D-Fender | ~pb |
| 03:55.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:55.38 | LemensTS | its like moderators of a forum then? |
| 03:55.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 03:55.41 | gones | Thanks |
| 03:55.48 | WhiteWolf | LemensTS: indeed |
| 03:55.48 | JT | haha since when did you get chanserv access, [TK]D-Fender ? |
| 03:56.05 | file | throws [TK]D-Fender a gin and tonic |
| 03:56.19 | WhiteWolf | LemensTS: +o = operator -- topic/kick/ban etc, +v voice is typically just a "look at me" title |
| 03:56.21 | LemensTS | what if everyone left, that means ChanServ woould take over after ther were no more moderators correct, so the room doesnt close |
| 03:56.28 | [TK]D-Fender | LemensTS: ChanServ manages giving out Op to channel moderators |
| 03:56.40 | [TK]D-Fender | JT: Several months now. |
| 03:56.44 | JT | ah ok |
| 03:56.50 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 03:56.59 | WhiteWolf | LemensTS: on freenode, chanserv doesn't hold channels open but it does keep a list of people and the access they have to the channel |
| 03:57.26 | WhiteWolf | simplistically, anyway |
| 03:57.28 | [TK]D-Fender | JT : As I'm here a rather prodigious amount of time daily it helps to ensure that the peace is kept |
| 03:57.45 | [TK]D-Fender | WhiteWolf: Decent explanation. |
| 03:57.48 | JT | WhiteWolf: actually you can enable ChanServ Guard |
| 03:57.54 | LemensTS | WhiteWolf: i got ya thanks! |
| 03:57.56 | JT | which keeps it in channel |
| 03:58.13 | WhiteWolf | JT: /can/, but there's not much reason to do so |
| 03:58.14 | LemensTS | Yea i see it trollin in #asterisk-cli |
| 03:58.30 | JT | people don't like to lose topics |
| 03:58.49 | WhiteWolf | in a channel with appreciable traffic it's not that big of a deal |
| 03:59.03 | WhiteWolf | since it probably won't ever be totally empty |
| 03:59.07 | WhiteWolf | but for smaller channels, perhaps |
| 03:59.08 | JT | LemensTS: chanserv can op even without being an op or in channel because it's a service, and services generally have at least as much access as an irc operator |
| 03:59.18 | WhiteWolf | or greater |
| 03:59.59 | WhiteWolf | all these newage networks like to restrict opers in every way that pleases them |
| 04:00.09 | *** join/#asterisk timex789 (n=Tommy@65.167.130.36) |
| 04:00.40 | voxter | How would you guys deal with a situation where you've got a person with two line appearances on a phone, and each of those appearances log in as agents (so that calls for queue1 go to line1, queue2 to line2, etc) - how would you make it so that if the user is on a call from queue1, that calls to queue2 dont come in? (causing a call waiting beep) |
| 04:01.44 | LemensTS | can you make an agent a member of 2 queues? |
| 04:01.49 | WhiteWolf | could disable the client side call waiting tone? *shrug* |
| 04:02.00 | voxter | sure you can, but then you cant have calls come to associated line keys for each queue |
| 04:03.31 | voxter | I thought about using chanisavail() somehow to check the other line, but im not sure how i could then exclude the agent from that Queue() call |
| 04:03.48 | LemensTS | Why do you need it on 2 lines |
| 04:03.50 | [TK]D-Fender | gones: now PASTEBIN your "musiconhold.conf" |
| 04:04.12 | voxter | LemensTS: the client has 3 queues, and they wanted calls for each queue to come to a unique line key to identify which queue was being called. |
| 04:04.20 | gones | wait please . |
| 04:04.23 | LemensTS | ok |
| 04:04.47 | bbryant | voxter: in trunk and 1.6.0 there is a new option to AddQueueMember that lets you specify a custom state interface |
| 04:05.08 | voxter | bbryant: interesting. I'll have to wait for 1.6.0 for that i guess! |
| 04:05.22 | [TK]D-Fender | bbySo you can specify a call channel, and a completely separate state-check tech+device? |
| 04:05.35 | [TK]D-Fender | bbryant: ^ |
| 04:05.53 | bbryant | [TK]D-Fender: correct, it can be another tech/device, or something custom you do from the dialpan |
| 04:05.58 | [TK]D-Fender | bbryant: BRILLIANT |
| 04:05.59 | *** join/#asterisk BeeBuu (n=beebuu@219.130.245.145) |
| 04:06.16 | bbryant | you can thank mmichelson for that |
| 04:06.17 | voxter | That is exactly what i need for this. |
| 04:06.23 | BeeBuu | what's the CLI dial command with asterisk 1.2? |
| 04:06.34 | voxter | BeeBuu: there isnt one. |
| 04:06.34 | [TK]D-Fender | BeeBuu: "dial" |
| 04:06.48 | [TK]D-Fender | BeeBuu: perhaps "console dial" |
| 04:07.02 | [TK]D-Fender | BeeBuu: But its a shitty way to do even a test call |
| 04:07.12 | BeeBuu | how? |
| 04:07.16 | voxter | Actually, I think it only exists if you have the alsa module installed. |
| 04:08.25 | BeeBuu | [TK]D-Fender: what's the not shitty way? |
| 04:08.32 | voxter | bbryant: damn, im bummed that the feature has been written for 1.6 now :P I cant think of a way to do this without it. |
| 04:08.53 | [TK]D-Fender | BeeBuu: a call from any outside device. SIP,IAX,ZAP,ETC |
| 04:09.32 | [TK]D-Fender | voxter: plenty of "non-clean" ways to do this now. |
| 04:09.41 | voxter | [TK]D-Fender: suggestion? |
| 04:09.58 | gones | [TK]D-Fender, Thanks your help , I have fixed it . |
| 04:10.12 | BeeBuu | [TK]D-Fender: thanks for you tips. |
| 04:10.34 | [TK]D-Fender | voxter: check if the device is on a call first before actually dialing it. |
| 04:11.24 | voxter | [TK]D-Fender: Well of course. You can do a chanisavail(), the problem is if it returns that the device is on a call, you would have to presumably remove them from the queue and re-add them again afterwards |
| 04:11.58 | voxter | [TK]D-Fender: the challenge here is omitting dialing that queue agent if their other registration was actively on a call. |
| 04:12.12 | [TK]D-Fender | voxter: Who said anything about removing them? I sure didn't |
| 04:12.44 | voxter | [TK]D-Fender: so you check to see if the other registration is busy. And if it is, how do you omit dialing them when using Queue()? |
| 04:12.49 | [TK]D-Fender | voxter: You can' check BOTh before placing either call. |
| 04:13.33 | [TK]D-Fender | voxter: not before calling "queue". That makes no sense. Dial your members via the dialplan and check in there before dialing out the bridging call |
| 04:14.45 | voxter | [TK]D-Fender: Huh? That doesnt make any sense. |
| 04:15.20 | voxter | [TK]D-Fender: What I'm talking about is a user with two line registrations that has independently joined two separate queues as a unique agent id |
| 04:15.35 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:15.43 | [TK]D-Fender | voxter: dial your "agents via Chan_local. Thus when the call goes out for the guy in queue, it uses the dialplan In your exten that will make your "agent"'s phone ring you can check their status before even BOTHERING to call "Dial" |
| 04:16.07 | voxter | [TK]D-Fender: Ah. Local. Right. |
| 04:16.21 | voxter | [TK]D-Fender: that makes sense, you can then put logic in the dial plan before calling them. OK. |
| 04:16.28 | [TK]D-Fender | voxter: Good. |
| 04:16.33 | voxter | [TK]D-Fender: Good! |
| 04:16.35 | voxter | [TK]D-Fender: :) |
| 04:17.05 | voxter | Chan local always slips my mind in situations like this because of the hackiness, i think. |
| 04:17.58 | [TK]D-Fender | voxter: chan_local is the solution to SO many things people should be shot for not knowing it better :) |
| 04:18.09 | [TK]D-Fender | (and the cause of the rest :p) |
| 04:18.25 | voxter | [TK]D-Fender: oh trust me, I use it all over the damn place. It just takes some time to remember to use it when thinking of solutions, sometimes. |
| 04:20.06 | voxter | I'm surprised that there isnt a return variable from ChanIsAvail() that lets you know if *any* of the requested channels came back unavailable. I guess i'll have to hack that into the dialplan |
| 04:21.29 | [TK]D-Fender | voxter: pass a string, use "s" and compare the length. that oughtta do |
| 04:21.48 | voxter | [TK]D-Fender: thats what I'm doing. I know how long the string should be, so that'll work |
| 04:22.07 | *** part/#asterisk timex789 (n=Tommy@65.167.130.36) |
| 04:23.19 | gones | <PROTECTED> |
| 04:24.07 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 04:26.50 | [TK]D-Fender | gones: Unstable, avoid |
| 04:28.06 | gones | But I must use this channel . |
| 04:28.24 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 04:29.15 | *** join/#asterisk selvam (n=selvam__@59.92.110.72) |
| 04:31.57 | [TK]D-Fender | gones: Go use YATE or FreeSWITCH in front to translate calls for you then. |
| 04:32.39 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133) |
| 04:32.41 | JT | yate, freeswitch doesn't really have h.323 |
| 04:33.10 | [TK]D-Fender | yate should |
| 04:33.28 | JT | yeah it does |
| 04:34.15 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 04:37.40 | voxter | discovers SendText works with polycoms |
| 04:39.40 | *** join/#asterisk admin0 (n=admin@bb121-7-229-228.singnet.com.sg) |
| 04:40.37 | voxter | Hmm, guess ill have to do two chanisavail()'s here, since if you pass it two tech/device's to check it will return only up to one device's status |
| 04:41.46 | DigitalIrony | Im getting and error while compiling zaptel on CentOS.....can i post it? |
| 04:42.11 | WhiteWolf | DigitalIrony: /topic |
| 04:42.12 | [TK]D-Fender | PASTEBIN |
| 04:42.14 | [TK]D-Fender | ~pb |
| 04:42.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:42.16 | [TK]D-Fender | ^^^^^^^^^^ |
| 04:42.23 | DigitalIrony | I know |
| 04:42.24 | [TK]D-Fender | ~centos52bug |
| 04:42.25 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 04:42.27 | [TK]D-Fender | ^^ jsut in case |
| 04:42.27 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 04:42.32 | DigitalIrony | I was asking if I could get help |
| 04:42.43 | [TK]D-Fender | DigitalIrony: is it the one listed above? |
| 04:42.45 | DigitalIrony | thats the problem.... thanks |
| 04:42.53 | [TK]D-Fender | DigitalIrony: Then there's the cure :) |
| 04:43.07 | DigitalIrony | yep |
| 04:43.16 | DigitalIrony | thanks |
| 04:43.48 | mchou | ok, asterisk newb here. I have a few DIDs. How do I set up asterisk so that each DID rings only a specific handset? |
| 04:44.12 | DigitalIrony | well I can't get menuselect to work....says ncurses is not installed, but it is |
| 04:44.15 | mchou | I mean how does asterisk differentiate each DID? |
| 04:44.33 | mchou | I'm missing something basic |
| 04:45.24 | DigitalIrony | mchou, well depends on how its comming in, analog or over a digital line? |
| 04:46.03 | mchou | DigitalIrony: digital, thru different ITSPs |
| 04:46.58 | mchou | DigitalIrony: and I only have SIP handsets :) |
| 04:47.07 | DigitalIrony | mchou: are they comming in over sip? over a T1, how do they come in? |
| 04:47.08 | LemensTS | is thier a more efficient way then dropping them into contexts of the users? |
| 04:47.21 | [TK]D-Fender | mchou: either by sending the calls to separate contexts by matching a peer entry on incoming, or by the EXTEN they dial into your system based on your REGISTER |
| 04:47.26 | mchou | DigitalIrony: a few over IAX, a few over sip |
| 04:47.42 | bbryant | DigitalIrony: try make cmenuselect |
| 04:47.49 | DigitalIrony | ^^^^^ thats the right answer |
| 04:48.05 | bbryant | i had that problem a while ago |
| 04:48.20 | DigitalIrony | bbryant thats not on here |
| 04:50.33 | bbryant | DigitalIrony: do you have the libncurses dev installed? |
| 04:50.46 | DigitalIrony | yes |
| 04:52.23 | mchou | [TK]D-Fender: can you give an example of matching a peer entry on incoming? suppose one of my DIDs is 1234567. How Do I programmatically know that's the DID that "rang?" |
| 04:52.32 | [TK]D-Fender | DigitalIrony: that's "make menuselect" (no c) |
| 04:52.56 | [TK]D-Fender | mchou: Look at the call come in. do you see it say "found peer'someone'". |
| 04:53.38 | mchou | [TK]D-Fender: havent set it up yet, just reading the docs and got all confused :) |
| 04:53.41 | [TK]D-Fender | mchou: Calls SHOULD come in auth'd. Next what EXTEN are they dialing? YOU are normally responsible for TELLING them what they should dial based on your REGISTER statement |
| 04:54.18 | [TK]D-Fender | mchou: Normally a call coming from your ITSP should auth the same as any SIP device, etc. |
| 04:54.56 | *** join/#asterisk Jamaal (i=433121ef@gateway/web/ajax/mibbit.com/x-13e5517802f11ffc) |
| 04:55.16 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 04:55.21 | *** join/#asterisk puck (n=puck@leibniz.catalyst.net.nz) |
| 04:55.25 | DigitalIrony | I fixed it... for some reason I had to delete the dir and untar it again and it worked....I tried doing make clean and make distclean and make dirclean but none of those worked until I untared it again |
| 04:56.10 | mchou | [TK]D-Fender: well, I know it's not efficient, but I just want "virtual" extensions (they wont be visible to callers) |
| 04:56.33 | puck | Hey, does anyone know of a script that'll tell you the maximum call load for an Asterisk box based on the CDR data? (If not, I'll go write it, but I'd rather not...) |
| 04:56.43 | mchou | the idea is each DID is mapped to a specific handset...... |
| 04:56.44 | [TK]D-Fender | mchou: There is no magic term for what you want to do. |
| 04:57.03 | [TK]D-Fender | mchou: nothing is mapped. Everything is DIALPLAN. Call comes in, your dialplan says "do this". |
| 04:57.30 | [TK]D-Fender | mchou: Separation is by context & exten. End of story |
| 04:58.22 | mchou | [TK]D-Fender: I'm confused. If I run asterist and I have multiple handsets, I must force callers to dial extensions? |
| 04:58.32 | mchou | asterisk* |
| 04:58.47 | Jamaal | question: I get the error in log " WARNING[2616] chan_sip.c: username mismatch, have <1111>, digest has <19999991111>" .... how do I edit the digest value on a polycom 550? |
| 04:59.34 | [TK]D-Fender | mchou: You don't seem to understand * much. EVERYTHINIG YOU DIAL IS AN EXTENSION. |
| 04:59.35 | mchou | I can't somehow route a DID to a specific handset w/o requiring the caller to dial an extensin? |
| 04:59.54 | [TK]D-Fender | mchou: the call comes IN targeting an EXTENSION <- |
| 05:00.12 | WhiteWolf | mchou: everything in asterisk is a dialplan, pretty much |
| 05:00.16 | bbryant | Jamaal: match_auth_username=yes in sip.conf |
| 05:00.31 | [TK]D-Fender | mchou: And never, for even the slightest second mistake a DEVICE for an EXTENSION. |
| 05:00.32 | mchou | I'm using extension in the sense "dial 1 for steve, dial 2 for ed, etc" |
| 05:00.53 | mchou | that's exposed to the caller. |
| 05:00.59 | mchou | I dont want that |
| 05:01.06 | WhiteWolf | sigh |
| 05:01.14 | [TK]D-Fender | mchou: a "device" is something like a SIP account used by a phone for eaxmple. An "extension" is a NUMBEr that is dialed by some device and gets matched in the dialplan |
| 05:01.39 | [TK]D-Fender | mchou: Did I say "give the guy an IVR menu" anywhere? No. |
| 05:01.44 | mchou | well, that's the proper * terminology |
| 05:02.12 | [TK]D-Fender | mchou: What I've just explained is TELECOM terminology and it applies to * as much as any other tech. |
| 05:02.43 | mchou | maybe. but you gotta admit "extension" is an overloaded term |
| 05:03.04 | [TK]D-Fender | mchou: Just one abused by too many people. |
| 05:04.21 | [TK]D-Fender | mchou: You dial an "extension". The fact that some schmucks PHONE rings is besides the point. the system just happens to have associated that # with "hey I should perform action X like making this particular piece of hardware emit an obnoxious sound" |
| 05:04.38 | [TK]D-Fender | mchou: (as a sample). |
| 05:04.41 | mchou | so lets go back to my example. In my sip.conf there is a [1000] stanza corresponding to a device |
| 05:05.00 | [TK]D-Fender | mchou: With * YOU say exactly what actions * takes for any given number dailed. |
| 05:05.03 | mchou | I have DID 1234567 |
| 05:05.09 | WhiteWolf | ugh |
| 05:05.12 | mchou | 1234567 comes in |
| 05:05.22 | WhiteWolf | it will match a 1234567 ext |
| 05:05.32 | WhiteWolf | you tell that ext in the dialplan to dial your sip endpoint |
| 05:05.33 | WhiteWolf | tada |
| 05:05.56 | mchou | in extension.conf how exactly does 1234567 => Dial(1000,....)? |
| 05:06.00 | pputman | exten => 1234567,1,Dial(sip/1000) |
| 05:06.04 | WhiteWolf | exactly like that |
| 05:06.05 | WhiteWolf | heh |
| 05:06.12 | WhiteWolf | (with a bit more oil on top) |
| 05:06.20 | mchou | ok. That's all I need to know. Thx |
| 05:06.31 | [TK]D-Fender | mchou: mchEvery call is JUST A CALL.the call comes in either A |
| 05:06.35 | WhiteWolf | that convo lasted about 4000% longer than it needed to |
| 05:06.39 | mchou | lol |
| 05:06.39 | [TK]D-Fender | sdarn mashup |
| 05:06.48 | pputman | WhiteWolf, yup |
| 05:06.56 | mchou | WhiteWolf: thanks for the clarification :) |
| 05:07.12 | mchou | WhiteWolf: some ppl get stuck on telecom terminology :) |
| 05:07.26 | WhiteWolf | you should brush up |
| 05:07.31 | WhiteWolf | and get with the game |
| 05:07.45 | pputman | That's definitely true with the word "extension", most people, like you said, do associate the term extension with a line or a phone instead of an action by the system. |
| 05:07.48 | mchou | yeah yeah. reading the docs, that wasnt too clear |
| 05:08.38 | mchou | but now that's it's all cleared up I understand how it's all supposed to work |
| 05:08.56 | [TK]D-Fender | mchou: You should have realixed this the second you made EXTENSIONS in "extensions.conf" (z0mg, obvious enough?) that somehow DIDN'T involve ringing some device |
| 05:09.21 | mchou | [TK]D-Fender: you STILL beating on that dead horse? |
| 05:09.24 | mchou | lol |
| 05:09.33 | WhiteWolf | there's not too much reason to continue badgering him for his ignorance |
| 05:09.41 | [TK]D-Fender | mchou: http://theroosterstrikes.files.wordpress.com/2008/01/beating-a-dead-horse.gif |
| 05:09.56 | [TK]D-Fender | WhiteWolf: I'm quite done now :) |
| 05:10.19 | WhiteWolf | thanks |
| 05:10.55 | Jamaal | anyone using asteriskNow here ? |
| 05:11.19 | [TK]D-Fender | Jamaal: Some but this is not its support channel |
| 05:11.20 | WhiteWolf | those systems are a bit involved |
| 05:12.44 | WhiteWolf | hm, let me say that another way |
| 05:12.46 | WhiteWolf | dialplan hell? |
| 05:13.05 | Jamaal | AsteriskNow is dialplan hell ? |
| 05:14.13 | WhiteWolf | i believe any of the prepackaged dialplan-in-a-box things are overly complicated for what they acomplish |
| 05:15.11 | C4colo | why doesn't the horse's body move at all when hit? |
| 05:17.24 | Jamaal | bbryant: match_auth_username=yes in sip.conf >> what does this setting do ? |
| 05:17.42 | [TK]D-Fender | C4colo: mass vs kinetic energy transfer vs intertia+gravity |
| 05:18.29 | bbryant | Jamaal: instead of matching the user based on the contact header, it matches the username from the Authorization header |
| 05:19.26 | CunningPike | ~seen dlynes |
| 05:19.29 | jbot | dlynes <n=daniel@S01060016b68219f1.vs.shawcable.net> was last seen on IRC in channel #asterisk, 2d 7h 33m 30s ago, saying: 'ZX81: and then voicemail should work just fine, assuming asterisk is running as root user'. |
| 05:19.39 | Jamaal | when you change it in the sip.conf is that a global setting ? |
| 05:19.51 | bbryant | Jamaal: I mean not checking the username based on from, not contact |
| 05:20.32 | Jamaal | so this will override any announcement the polycom is making ? |
| 05:20.40 | bbryant | it's a global option, it should be in the [general] section |
| 05:20.48 | bbryant | what do you mean, announcement? |
| 05:20.59 | Jamaal | but in the PolyCom how would you disable that field |
| 05:21.09 | Jamaal | or change it to mimic the auth user id |
| 05:21.20 | [TK]D-Fender | ok, checkout time. Later all. |
| 05:21.42 | bbryant | the polycom is trying to authenticate as 19999991111, that's what your phone is setup for |
| 05:22.52 | Jamaal | yeah I guess it is cached in the config somewhere |
| 05:22.58 | Jamaal | cant figure out how to get rid of it |
| 05:24.59 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 05:25.03 | C4colo | it is called provisioning the device |
| 05:27.43 | gones | [TK]D-Fender: gones: Go use YATE or FreeSWITCH in front to translate calls for you then. But the YATE use Openh323 too . |
| 05:28.08 | WhiteWolf | their implementation differs |
| 05:29.00 | JT | very different |
| 05:29.10 | JT | yate's h.323 actually seems to work |
| 05:31.18 | rbd | anyone know if there's ubuntu/debian debs for the most recent versions of asterisk? I'm not finding much... |
| 05:32.01 | JT | if you want recent, just compile |
| 05:32.42 | Jamaal | bbryant: I tried the setting "match_auth_username=yes in sip.conf" it phone still doesnt register same error message |
| 05:33.16 | Jamaal | " WARNING[2616] chan_sip.c: username mismatch, have <1111>, digest has <19999991111>" |
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| 05:34.05 | gones | WhiteWolf: Can you show me the different form YATE and Asterisk ? Thank you ! |
| 05:36.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:36.53 | JT | gones: ask a proper question |
| 05:36.59 | JT | they're different projects |
| 05:37.15 | Great_Anta_Baka | if i cant dial local extensions where can the problem lie |
| 05:37.24 | Great_Anta_Baka | i can dial voicemail fine |
| 05:38.37 | Great_Anta_Baka | even dialing extensions in the same context doesnt work |
| 05:39.01 | Great_Anta_Baka | i am extension 304 and when i dial 303 on my snom it says Declined:303 |
| 05:39.42 | *** join/#asterisk s0lid (n=s0lid@122.53.110.155) |
| 05:40.17 | gones | I want to know how the sip channel transition to h323 ? |
| 05:41.34 | JT | gones: look at the yate web site |
| 05:45.12 | gones | Thank you , JT |
| 05:47.55 | Jamaal | setting "match_auth_username=yes in sip.conf" is that the only field responsible for the digest setting ? |
| 05:49.11 | bbryant | Jamaal: that's responsible for how the digest is matched to a sip user |
| 05:49.29 | bbryant | if your phone is sending the wrong digest, you need to provision your phone, not mess with asterisk settings |
| 05:49.40 | *** join/#asterisk rukus (n=daniel@41.247.184.160) |
| 05:49.54 | Jamaal | Polycom IP 550 phones aren't registering ... when I check the /var/log/asterisk/messages ... I get the following error "" WARNING[2616] chan_sip.c: username mismatch, have <1111>, digest has <19999991111>" " Just looking for way to resolve this |
| 05:50.28 | Great_Anta_Baka | is there some setting that i need to set in order to dial local extension? |
| 05:50.51 | *** join/#asterisk cplx (n=cplx@203.206.204.213) |
| 05:53.09 | Jamaal | any good sites for assistance with provisioning the Polycom 550s ? |
| 05:54.12 | Great_Anta_Baka | the user manuals explain everything jamaal |
| 05:54.21 | Great_Anta_Baka | http://www.voip-info.org/ |
| 05:54.28 | Great_Anta_Baka | they have a pretty good tutorial as well |
| 05:55.21 | Jamaal | k thx |
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| 06:15.48 | Great_Anta_Baka | how do i make it so that users in different contexts can call each other? |
| 06:16.00 | Great_Anta_Baka | as in i have an office with two sections |
| 06:16.12 | Great_Anta_Baka | but both sections neeed to be able to call each other |
| 06:16.27 | creativx | include contexts |
| 06:16.40 | Great_Anta_Baka | where do i include the context |
| 06:16.41 | Great_Anta_Baka | ? |
| 06:16.48 | Great_Anta_Baka | thanks for the response :) |
| 06:16.57 | Great_Anta_Baka | been getting nothing these days |
| 06:17.47 | creativx | [default] context, include => office1, include => office2 |
| 06:18.01 | Great_Anta_Baka | ah nice thanks |
| 06:18.11 | creativx | is one way to do it |
| 06:21.43 | Great_Anta_Baka | creativx: would i be able to include the other context in the other eg [office1] context, include => office2 |
| 06:21.56 | Great_Anta_Baka | one in the other* |
| 06:21.59 | creativx | you could do it that way i suppose |
| 06:22.09 | creativx | experiment :-) |
| 06:22.16 | Great_Anta_Baka | well asking cos they way you suggested didnt work but will try :P |
| 06:24.16 | Great_Anta_Baka | ah i see why it didnt work |
| 06:24.24 | Great_Anta_Baka | the extensions arent registered anymore |
| 06:25.58 | mchou | so I have a personal list of blacklisted telemarketer #s I'd like to filter with * |
| 06:26.31 | mchou | I'm not sure if asterisk backticks would work for me |
| 06:27.10 | mchou | basically I want to grep the blacklist to see if # is blacklisted |
| 06:28.00 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 06:29.08 | mchou | and if grep returns 0 (# found) then use something like ex-girlfriend logic |
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| 07:33.44 | bird_of_Luck | hi, I've got 2 asterisks (1.4.21 & 1.2.27) connected via IAX2. link speed is 100mbps, link is not busy. At the moment I'm getting many 'Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 ..' on every call through this IAX. ping -f .. -s 1400 shows no packetlost and delay is ~ 2ms. Where shold I dig ? |
| 07:34.59 | *** join/#asterisk selvam (n=selvam__@59.92.110.72) |
| 07:35.19 | selvam | how to transfer calls in asterisknow |
| 07:36.47 | *** part/#asterisk selvam (n=selvam__@59.92.110.72) |
| 07:38.28 | Great_Anta_Baka | i have the same problem right now selvam |
| 07:39.55 | *** join/#asterisk tapic (n=tap@88.255.77.200) |
| 07:46.19 | Great_Anta_Baka | the cli isnt even noting that i am trying to transfer the call |
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| 08:10.03 | Great_Anta_Baka | can anyone tell me why i cant transfer calls between my internal phones |
| 08:16.18 | tapic | is there a way to bridge to active channels? |
| 08:16.59 | tapic | without meetme |
| 08:19.21 | Great_Anta_Baka | please help me i cant transfer calls at all need help badly |
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| 08:22.19 | xzcvczx | is it possible to use asterisk as a CSD gateway? |
| 08:22.55 | JT | depends what CSD is |
| 08:23.17 | xzcvczx | dial-up from a cellphone |
| 08:23.27 | tapic | great anta baka : what do you mean by transfer |
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| 08:32.13 | raqamy | hi, where can i found asterisk video tutorials? |
| 08:35.43 | dominic1 | youtube is your friend |
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| 08:41.03 | Great_Anta_Baka | i dont suppose i will get any help setting up an attended transfer |
| 08:41.05 | Great_Anta_Baka | tapic |
| 08:41.08 | Great_Anta_Baka | got it sorted\ |
| 08:41.22 | Great_Anta_Baka | was the old trixbox sip.conf file that was still laying around |
| 08:41.36 | Great_Anta_Baka | but i still cant do an attended transfeer |
| 08:42.50 | Great_Anta_Baka | snom phones support attended transfer right? |
| 08:44.56 | tapic | i dont know |
| 08:57.02 | Great_Anta_Baka | mmm trying to set it up in features.conf |
| 08:57.22 | Great_Anta_Baka | but when i dial the numbers i have setup (*2 for atxfer ) nothing happens |
| 08:59.15 | tapic | I also have problems, I cant find a way to bridge two active channels. Now I try to park one of the calls and then redirect the other to the parket extension |
| 08:59.23 | tapic | parked |
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| 09:06.13 | Great_Anta_Baka | i havent touched parked extensions sorry man |
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| 09:16.25 | dickyjoe | Hello all |
| 09:17.38 | dickyjoe | Im an asterisk noob and what like to know how to make asterisk load on startup in Centos 5.2 |
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| 09:20.48 | dickyjoe | anyone home? it seems a bit quiet |
| 09:20.52 | xuser | dickyjoe: If you did 'make config' during install, the /etc/init.d scripts should take care of that. |
| 09:21.12 | dickyjoe | ok |
| 09:21.18 | dickyjoe | i will look |
| 09:21.45 | dickyjoe | i had a set of preconfigured files to copy into the /etc/asterisk folder |
| 09:21.55 | dickyjoe | maybe i didn't run make config |
| 09:25.23 | dickyjoe | yeah i have no asterisk file in init.d |
| 09:26.57 | *** join/#asterisk stoven (n=staven@port-92-196-68-183.dynamic.qsc.de) |
| 09:28.36 | stoven | Hello! I want to install bristuff-0.4.0-RC3b (* 1.4.21.1). Everything worked fine, but if I want to compile zaphfc, there is an error with the zaptel.h |
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| 09:38.34 | dickyjoe | xuser: ok i have checked my init.d directory and i have no file called asterisk |
| 09:38.53 | dickyjoe | xuser: maybe i can manually create one? |
| 09:39.05 | stoven | what does this error mean? ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 09:39.13 | tzafrir_laptop | stoven, could you please pastebin that error? |
| 09:39.30 | tzafrir_laptop | (the compile error) |
| 09:40.44 | stoven | http://rafb.net/p/aTxRtQ96.html |
| 09:41.15 | stoven | tzafrir_laptop: I think there is no more a compile error |
| 09:41.35 | stoven | I just copied the .h-files in the same directory |
| 09:42.26 | tzafrir_laptop | stoven: cat /proc/zaptel/* |
| 09:43.22 | stoven | tzafrir_laptop: Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" (MASTER) |
| 09:43.46 | tzafrir_laptop | rmmod ztdummy; modprobe zaphfc |
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| 09:46.19 | stoven | tzafrir_laptop: seems to work. How to get information about the zap-channels in the asterisk-cli? |
| 09:46.26 | stoven | zap show channeltypes? |
| 09:46.50 | tzafrir_laptop | ztcfg |
| 09:47.07 | tzafrir_laptop | and then you'll probably need to restart asterisk |
| 09:47.11 | tzafrir_laptop | zap show channels |
| 09:47.51 | stoven | http://rafb.net/p/4CGSaG66.html |
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| 09:52.01 | tzafrir_laptop | stoven, you're missing some content in zapata.conf |
| 09:52.30 | tzafrir_laptop | see the sample zapata.conf in the zaphfc directory, or use zapconf / genzaptelconf |
| 09:54.09 | stoven | tzafrir_laptop: thank you. Works fine right now :) |
| 09:55.03 | tzafrir_laptop | stoven, I think that the sample dialplan does not have pridialplan=unknown; but chances are you'll need it |
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| 09:57.35 | DarKnesS_WolF | tzafrir_laptop: question |
| 09:57.45 | tzafrir_laptop | DaPrivateer, shoot |
| 09:58.17 | DarKnesS_WolF | with NT/ET i should use anormal UTP cable ? strate through ? |
| 09:59.38 | *** join/#asterisk |stefan|_ (i=stefan@mjew.eu) |
| 10:00.07 | |stefan|_ | when creating a menu with responsetimeout etc. how come i get autofallthrough status unknown when it's supposed to wait for input ? |
| 10:00.26 | DarKnesS_WolF | tzafrir_laptop: ^ |
| 10:00.33 | DarKnesS_WolF | and also LS did the same problem |
| 10:00.59 | tzafrir_laptop | DarKnesS_WolF, for what device? |
| 10:01.58 | tzafrir_laptop | BRI? E1? |
| 10:04.15 | DarKnesS_WolF | PRI tzafrir_laptop |
| 10:05.45 | tzafrir_laptop | If this happens to be an Astribank: you can use a straight cable. Otherwise you need a loopback cable |
| 10:09.21 | gones | I don't understand the macro in the extensions, can you tell me how to use it ? |
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| 10:48.44 | dominic1 | short question. If I read out a variable from database with following content dial(SIP/123) and do a ${ODBC_mystatement} in the dialplan. Will I be able to execute the stored variable that the system will dial 123? |
| 10:49.17 | tapic | I park a user and get Segmentation fault (core dumped) |
| 10:49.35 | tapic | how crazy is that |
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| 10:57.31 | mort_gib | chan_sip.c: Remote host can't match request BYE to call -What does this mean?? |
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| 11:49.45 | nix8n82 | Anyone know the proper way to read AGI results from perl I found an example but it keeps hanging on me and exiting the agi non-zere |
| 11:50.16 | nix8n82 | with asterisk 1.3 |
| 11:50.20 | nix8n82 | er 1.4 |
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| 12:33.07 | mort_gib | Again. Could anyone help me translate this: chan_sip.c: Remote host can't match request BYE to call |
| 12:35.10 | angryuser | mort_gib it happens at the end of the call i suppose ? |
| 12:36.22 | mort_gib | Yeah, that could be! It's a small systems that is causing some trouble, and I need to rule out different possibilities... |
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| 12:37.04 | angryuser | mort_gib maybe your provider need a different hangup cause ? |
| 12:37.37 | JHilgeman | hi all - quick question. we normally use a T1 line for phone service but it's down and I'm looking for a quick way to set up a VoIP gateway so our sales can at least call outbound until the T1 is fixed. any recommendations? |
| 12:37.55 | mort_gib | Possibly, but I wouldn't be able to control that. -Would I?? |
| 12:40.00 | mort_gib | I can live with normal termination of calls, but people are complaining about dropped calls from time to time |
| 12:40.19 | angryuser | mort_gib i am not relly sure, try tu use soft hangup |
| 12:40.37 | angryuser | dropping call ? |
| 12:41.07 | angryuser | that is not the same thing, yu need to sip trace when call is dropping |
| 12:41.56 | mort_gib | Yes, but it happens maybe 2-3 times a week... |
| 12:42.11 | mort_gib | So up verbose in asterisk.conf ?? |
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| 12:42.41 | creativx | JHilgeman: what kind of phones? |
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| 12:43.01 | angryuser | it hapenns for me sometimes, and i have no idea why ;( i think some upd packet get dropped and call get dropped |
| 12:43.25 | JHilgeman | the phones are Polycom 501, the backend is AsteriskNOW (Asterisk 1.4.19). |
| 12:44.06 | mort_gib | Yeah, I have some 10 systems, but I ONLY get this crap on one of the installations. |
| 12:44.11 | [TK]D-Fender | JHilgeman: Pick an ITSP, sign up, set up your SIP creds and get busy |
| 12:44.29 | mort_gib | I even redid the entire system to no avail |
| 12:44.43 | JHilgeman | right - just looking for recommendations on ITSP |
| 12:44.57 | [TK]D-Fender | ~itsplist-us |
| 12:44.57 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 12:45.00 | angryuser | ask tk who knows ;) |
| 12:45.47 | mort_gib | Tried that, although very helpful (despite his short temper) he only pointed me to network issues |
| 12:46.06 | mort_gib | Which I'm not convinced is the problem |
| 12:46.53 | mort_gib | I have two remote offices connecting via ADSL/VPN and the dropped calls are evenly distributed over "local" and remote users |
| 12:47.21 | mort_gib | I'm beginning to suspect the Dell server might have something to do with it... |
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| 12:47.47 | JHilgeman | thanks |
| 12:48.04 | mort_gib | First system was based on CentOS 4.4 second attempt is on Debian |
| 12:49.50 | [TK]D-Fender | mort_gib: if you're talking VoIP, then hardware has nothing to do with it. |
| 12:50.54 | mort_gib | Well, it seems to be mostly via the ISDN BRI connections |
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| 13:02.37 | mort_gib | TK any ideas?? |
| 13:04.18 | [TK]D-Fender | mort_gib: You've got no real info. Sorry |
| 13:05.51 | LemensTS | TKD-Fender: How many hours do you sleep a night? |
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| 13:07.14 | mort_gib | TK -What do you need?? |
| 13:08.00 | [TK]D-Fender | mort_gib: SOMETHING. |
| 13:09.32 | [TK]D-Fender | LemensTS: about 6 |
| 13:09.34 | creativx | [TK]D-Fender does not require sleep. |
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| 13:12.20 | Dramokl | hellow |
| 13:12.49 | CrashHD | <PROTECTED> |
| 13:12.50 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 13:12.53 | mort_gib | On 29.07.2009 10:44 (User PC is not synchronized with time server) a call is dropped |
| 13:13.01 | Dramokl | I have some trouble with asterisk(with Sangoma hardware) |
| 13:13.28 | Dramokl | in dmesg:wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8be4410 |
| 13:13.28 | mort_gib | Dramokl: What Sangoma hardware?? |
| 13:13.45 | Dramokl | it is hardware troublу? |
| 13:14.22 | Dramokl | Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card |
| 13:14.52 | pbrown985 | I am using the single t1 card and it works nicely. |
| 13:15.21 | mort_gib | TK: I get one line in /var/log/asterisk/messages saying chan_sip.c: Remote host can't match request BYE to call |
| 13:15.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.28 | [TK]D-Fender | Dramokl: read this : http://www.trixbox.org/forums/vendor-moderated-forums/sangoma/problems-setting-multiple-cards and then call up Sangoma support |
| 13:16.52 | [TK]D-Fender | mort_gib: Sounds like the call is gone from * by the time it receives an ACK from the other end |
| 13:17.32 | mort_gib | TK I get NO errors at all no in /var/log/sangoma_mgd.log, /var/log/messages or dmesg |
| 13:17.48 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:17.53 | mort_gib | TK Yeah, so do I need to fix that?? |
| 13:18.14 | Dramokl | [TK]D-Fender, i have one Sangoma card.... |
| 13:18.32 | Dramokl | and it's work before |
| 13:18.57 | [TK]D-Fender | mort_gib: that message doesn't contribute anything much. |
| 13:19.10 | [TK]D-Fender | Dramokl: Call up Sangoma support |
| 13:19.16 | Dramokl | =)))) |
| 13:19.43 | mort_gib | It's not telling mee too much either, because I get this on a regular basis without any problem with the calls |
| 13:20.01 | mort_gib | It's like a normal hangup |
| 13:20.45 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 13:22.32 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
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| 13:27.49 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 13:28.14 | tzanger | coppice: are you aware of any fixed-point echo cans? doesn't necessarily have to be stellar, as it's for FXS only, but floating point sucks on blackfin ;-) |
| 13:29.03 | coppice | I don't know of any floating point echo cans. |
| 13:29.33 | tzanger | ? isn't OSLEC floating point? |
| 13:29.43 | coppice | nope |
| 13:30.16 | tzanger | hmm, I have to take a closer look at it, it doesn't work so hot on blackfin |
| 13:32.06 | coppice | blackfins are very very sensitive to where your buffers are. OSLEC can work pretty well or pretty slowly on one |
| 13:32.18 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:32.20 | tzanger | I'd have put the buffers in L1 |
| 13:32.30 | tzanger | now admittedly I'm not a guru when it comes to this |
| 13:32.50 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:33.56 | coppice | the blackfin floating point emulation library seems very leisurely :-) |
| 13:35.13 | anonymouz666 | coppice: do you know if the reference software (china stuff) they are offering speex codec? |
| 13:35.21 | tzanger | coppice: hahaha |
| 13:35.25 | tzanger | it's... relaxed |
| 13:41.41 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:43.36 | *** join/#asterisk ToTo (n=ToTo@209.8.233.71) |
| 13:44.15 | *** join/#asterisk puzzled (n=patrick@62-177-241-95.dsl.bbeyond.nl) |
| 13:46.23 | *** join/#asterisk UD (n=steve@unaffiliated/underdawg) |
| 13:46.27 | UD | hi |
| 13:46.44 | UD | Where could I get the best deal on SIP service? |
| 13:47.04 | coppice | anonymouz666: what china stuff? |
| 13:47.14 | UD | broadvoice jacked up their prices |
| 13:47.50 | [TK]D-Fender | ~itsplist-us |
| 13:47.51 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:47.58 | [TK]D-Fender | UD: Go shop around |
| 13:48.26 | coppice | but try not to end up at walmart :-) |
| 13:48.49 | UD | what should i look for exactly? |
| 13:49.05 | UD | do some have limits on simultaneous sessions? |
| 13:49.17 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 13:49.17 | *** mode/#asterisk [+o denon] by ChanServ |
| 13:50.02 | [TK]D-Fender | UD: Every company has their own offerings. GO LOOK. |
| 13:52.22 | creativx | anyone here happen to know of anyone selling usb-controllable/programmable visual indicators (led/lights/etc) that can be used for indicating when a person is using the phone? |
| 13:52.43 | creativx | im tired of people disturbing me when im on the phone.. i want a bigass red light! |
| 13:54.17 | [TK]D-Fender | creativx: There are plenty of cheap ones that plug in-line with your handset |
| 13:54.26 | creativx | [TK]D-Fender: problem is i use the handset for mp3 aswell |
| 13:54.42 | [TK]D-Fender | creativx: ...? |
| 13:54.50 | creativx | softphone :-) |
| 13:54.57 | [TK]D-Fender | creativx: BLEH |
| 13:55.04 | creativx | i was thinking of finding something that i can control with our cms and hook up to each users usb port |
| 13:55.07 | creativx | hehe yes bleh indeed |
| 13:55.19 | [TK]D-Fender | creativx: X-10 CM11A + lamp module <- |
| 13:56.02 | creativx | interesting! |
| 13:56.09 | [TK]D-Fender | creativx: If you're looking to announce a whole pile of people you could always run an HDTV as a panel |
| 13:56.34 | creativx | hehe |
| 13:56.37 | UD | [TK]D-Fender: http://www.ee.washington.edu/circuit_archive/circuits/F_ASCII_Schem_Tel.html#ASCIISCHEMTEL_0081 |
| 13:56.38 | creativx | the hdtv is showing other graphs =) |
| 13:56.48 | creativx | i just need some visual indication.. this X-10 CM11A looks brilliant |
| 13:57.18 | [TK]D-Fender | UD: unfortunately he's not using a TELEPHONE. |
| 13:57.39 | [TK]D-Fender | creativx: Run a "ticker" underneath, you'll look like CNN :p |
| 13:57.57 | UD | oh |
| 13:58.14 | [TK]D-Fender | load chan_bigprint.so |
| 13:58.15 | *** join/#asterisk moy (n=moy@nat/ibm/x-9a845ff210a1dab7) |
| 13:58.17 | [TK]D-Fender | :D |
| 14:00.22 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 14:01.08 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 14:02.26 | cy3o3 | sup asterisk gurus |
| 14:03.37 | creativx | [TK]D-Fender: heheh |
| 14:04.09 | cy3o3 | Question.. when using originate from AMI .. channel is an external number, exten is an external number. CallerID: <number> is dandy on the channel it's calling, but cid is not correct for the exten. Aware of any way to fix this? |
| 14:05.07 | UD | i also directed that at the wrong nick whoops |
| 14:05.17 | cy3o3 | From within AMI that is |
| 14:05.40 | [TK]D-Fender | UD: S'ok, gets you busted faster ;) |
| 14:05.44 | creativx | [TK]D-Fender: im gonna look this up tomorrow, it might be a problem with the different power sockets etc. but thanks for the tip :) |
| 14:06.42 | [TK]D-Fender | cy3o3: pastebin a sample attempt & NoOp it from both ends |
| 14:08.34 | *** join/#asterisk c4t3l (n=rcallico@mail.questia.com) |
| 14:08.47 | c4t3l | hello world |
| 14:09.23 | n3hxs | C ? |
| 14:09.52 | c4t3l | hehe |
| 14:10.13 | n3hxs | nothing like a double play on words |
| 14:13.27 | LemensTS | Where would I get information on all the settings in sip.conf for 1.4? voip-info.org is 1.2 blah |
| 14:13.53 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 14:15.45 | [TK]D-Fender | LemensTS: the BOOK, upgrade.txt, the SAMPLE configs, etc |
| 14:18.08 | tapic | is there a wat to bridge two actie channels? |
| 14:18.10 | tapic | way |
| 14:18.19 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 14:18.24 | tapic | is there a way to bridge two active channels? |
| 14:18.37 | [TK]D-Fender | tapic: go read the AMI function list. |
| 14:19.00 | tapic | I have read, you can only redirect to an extension |
| 14:20.54 | c4t3l | tapic: you can also bridge channels |
| 14:21.05 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:21.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:21.15 | tapic | how do you bridge two active channels? |
| 14:22.12 | tapic | I want agi control (interaction) for two not related channels and bridge them at some point. |
| 14:22.31 | [TK]D-Fender | tapic: redirect one to a parking lot and the other to an exten that will retrieve it. |
| 14:22.48 | c4t3l | thats one way to do it |
| 14:22.50 | tapic | I tried that. and asterisk crashes with segmentation fault |
| 14:23.02 | c4t3l | 1.4? |
| 14:23.20 | tapic | yes 1.4.21.2 |
| 14:24.22 | c4t3l | well, i wrote a simple python agi that does what you are asking, but its a little gimpy |
| 14:24.56 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 14:24.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:25.03 | tapic | I originate a call and park it for the other channel to retrieve it. While the parked call is waiting (I am not dialled its park extension yet) after 1-2 seconds asterisk crashes. |
| 14:26.35 | The-Bat | Hie |
| 14:26.37 | The-Bat | it is possible to hear live ongoing calls in queue(or any ongoing call) by a 3rd person ?? |
| 14:26.40 | c4t3l | hmm, sounds like there'a alot more to you issue... what kind of hardware r u using etc? |
| 14:26.56 | The-Bat | Any url related to this is appreciable |
| 14:27.07 | tapic | I only use SIP with ubuntu server 2.6 |
| 14:27.24 | tapic | x86 machine |
| 14:29.27 | *** join/#asterisk webman (n=adamg@200.179.233.220.exetel.com.au) |
| 14:29.41 | tapic | I want the bridge operation to happen, once the originated call is successfully answered,played a sound and read dtmf from callee |
| 14:29.44 | c4t3l | The-Bat: are you Will? |
| 14:30.32 | c4t3l | tapic: ubuntu server, so no X right? |
| 14:30.37 | tapic | yep |
| 14:31.18 | c4t3l | tapic is * compiled natively or is it a downloaded binary? |
| 14:31.55 | tapic | [TK]D-Fender : how do you tell the caller channel that it should connect to the parking lot with extension XXX? |
| 14:32.28 | tapic | I compiled it |
| 14:35.09 | c4t3l | tapic: sounds like strace time |
| 14:36.01 | tapic | what do you mean? |
| 14:39.11 | The-Bat | c4t3l, nope |
| 14:40.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:43.11 | tzafrir_laptop | tapic, what's ubuntu server 2.6? |
| 14:43.46 | UD | is there a certain mechanism to ban certain numbers from being dial()ed? |
| 14:44.23 | [TK]D-Fender | UD: GotoIF <- |
| 14:44.31 | UD | ok |
| 14:44.41 | [TK]D-Fender | UD: Or any other logic you put in the way. |
| 14:46.29 | UD | so say i'd like to keep a dialing service sorta like a phone card for me and a few friends. my friend leaves a piece of paper out for his little brother to find with phone # and extension and uses it to call the sherriffs office and ask if their refridgerator is running |
| 14:46.29 | *** join/#asterisk neisley (n=neisley@www.nwafternoon.org) |
| 14:47.02 | UD | can this be setup in a way to have THEIR # show up on the caller ID and not mine? |
| 14:47.17 | [TK]D-Fender | UD: Depends on what you're using to dial out. |
| 14:47.24 | UD | just using SIP no hardware |
| 14:47.35 | neisley | hello, |
| 14:47.37 | [TK]D-Fender | UD: Depends on your provider. Some allow you to set CID, others don't |
| 14:48.07 | neisley | I'm having an occasional issue where calls get |
| 14:48.10 | [TK]D-Fender | UD: And They should likely be sending ANI, so the cops should be able to see that its "you" regardless of the CID |
| 14:48.30 | neisley | "stuck" between a specific extension and another extension or an inbound or outbound trunk |
| 14:49.36 | neisley | when I say "stuck" I mean asterisk -rx 'show channels' shows a call up between the affected extension and another destination. |
| 14:49.58 | neisley | however, the affected extension isn't in use or is otherwise on hook. |
| 14:50.13 | UD | no way to cause ANI not to be sent? |
| 14:50.34 | *** join/#asterisk spokra (n=spokra@host093-179-177.sea0.speakeasy.net) |
| 14:51.10 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:51.16 | neisley | I've discovered that when this occurs, asterisk is getting a SIP response 400 "Bad Request" back from the affected phone, and any calls are met with circuit-busy and exit with Dialstatus = Congestion. |
| 14:51.33 | [TK]D-Fender | neisley: Please be specific about exactly what devices are involved with each end of the call. |
| 14:51.42 | neisley | ok |
| 14:52.06 | [TK]D-Fender | neisley: And for SIP issues always PASTEBIN CLI output with sip debug enabled. |
| 14:52.07 | [TK]D-Fender | ~pb |
| 14:52.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:52.09 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 14:52.20 | neisley | thanks |
| 14:52.20 | [TK]D-Fender | UD: Typically no. |
| 14:52.36 | [TK]D-Fender | UD: Sorry, you don't get the right to #&^%$ with the phone network. |
| 14:53.09 | CrashHD | Hello everyone |
| 14:53.11 | neisley | this is the situation. I have Extension 120, which is the "receptionist phone", an Aastra 57i CT. the other end of the call may vary from another extension on the phone system to an inbound call |
| 14:53.26 | neisley | which is coming from a PRI |
| 14:54.44 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 14:54.45 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 14:54.49 | CrashHD | any thoughts on what asterisk would do without a timing source? dropped calls? one way audio drops? |
| 14:55.14 | clyrrad | CrashHD: Conference rooms wouldnt work |
| 14:55.18 | neisley | what happens is, that a call doesn't clear properly from the asterisk server when hung up, |
| 14:55.48 | CrashHD | clyrrad: interesting. we actually don't use any conference rooms. |
| 14:55.50 | neisley | such that the output from asterisk -rx 'show channels' still lists a call up between extension 120 and the destination, |
| 14:56.49 | clyrrad | Any of you folks using PostgreSQL + SSL Connections with Asterisk Realtime? |
| 14:57.19 | neisley | since there's still a call up on the server end, and CW is disabled, the calls just sit in the queue and back up |
| 14:57.32 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 14:57.38 | neisley | I'm trying to find out why calls occasionally get stuck in the system like this |
| 14:57.38 | clyrrad | better question would be does Asterisk Realtime support SSL connections to a PostgreSQL Database? |
| 14:57.42 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:57.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.53 | neisley | I think that it may be related to a network issue or a phone issue. |
| 14:58.40 | neisley | when this occurs, the server gets a SIP/400 "bad response" back from the phone, and the call ends with Dialstatus=Congestion. |
| 14:58.56 | neisley | I'm really looking for any insight on this issue, |
| 14:59.20 | [TK]D-Fender | clyrrad: uses ODBC. If your connecteor allows it, then yes |
| 14:59.25 | neisley | this doesn't happen every day, but just enough to be annoying, and alas, I haven't had the chance to do a SIP Debug, or packet capture. |
| 14:59.37 | [TK]D-Fender | neisley: And I asked you for SIP DEBUG to backup your problem |
| 15:00.14 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.28 | neisley | hang on |
| 15:00.33 | neisley | let me see what I can find here. |
| 15:00.38 | clyrrad | [TK]D-Fender: yup using ODBC, let me check the res_odbc.conf file again and see if I can see an option in there for SSL, if there is I must have missed it.... |
| 15:01.10 | [TK]D-Fender | clyrrad: this is your DSN's job, not *'s |
| 15:02.27 | clyrrad | [TK]D-Fender: right that makes sense |
| 15:04.58 | CrashHD | anyone know why sip qualify's are so inconsistent? |
| 15:05.20 | CrashHD | servers on a local switch show lagged and varried responses |
| 15:05.26 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
| 15:06.53 | [TK]D-Fender | CrashHD: Qualify is a SIP timed event, not a "ping" |
| 15:07.17 | [TK]D-Fender | CrashHD: Polycoms respond pretty slow because of thier prioritization, etc |
| 15:07.37 | CrashHD | seems varied though |
| 15:07.44 | CrashHD | between asterisk servers |
| 15:08.06 | [TK]D-Fender | CrashHD: Can be a lot of things |
| 15:08.21 | CrashHD | anyway to troubleshoot |
| 15:08.37 | CrashHD | it's enough where at times a server will report unavailable |
| 15:08.42 | CrashHD | or extremely lagge |
| 15:08.43 | CrashHD | d |
| 15:10.00 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 15:10.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:11.52 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 15:12.30 | gnorbert | Hi, does somebody have an idea, what can be the problem, if I can connect perfectly to asterisk with linphone, jut linphone writes continously "ALSA lib pcm_dsnoop.c:565:(snd_pcm_dsnoop_open) unable to open slave"? Everything can be heard, so it's not a big problem, just it would be more beautiful without it. :) |
| 15:12.45 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:13.07 | gnorbert | Asterisk writes no failure message. |
| 15:13.40 | [TK]D-Fender | gnorbert: "noload => chan_alsa.so" in modules.conf |
| 15:14.01 | gnorbert | [TK]D-Fender: Thank you very much |
| 15:16.20 | thansen|laptop | after changing a sip password how can for it to re-authenticate |
| 15:16.25 | gnorbert | [TK]D-Fender: It still writes the same, no change... |
| 15:16.31 | LemensTS | ~alsa |
| 15:16.32 | jbot | alsa is probably the Advanced Linux Sound Architecture, or at http://www.alsa-project.org/, or available only for Linux (if you intend your code to be portable, ask me about OSS) |
| 15:16.35 | thansen|laptop | force |
| 15:16.40 | [TK]D-Fender | gnorbert: Did you RESTART *? |
| 15:16.47 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 15:17.05 | gnorbert | Yes, twice, because first I thought it doesn't work well because I forgot it. :) |
| 15:17.14 | tzafrir_laptop | ~oss |
| 15:17.14 | jbot | methinks oss is a portable sound interface available in 11 different Un*x systems. The propietary package is available at http://www.opensound.com/. However, free systems like GNU/Linux or *BSD include free GPL/BSD re-implementations. an acronym that stands for "Open Source Software" |
| 15:17.15 | [TK]D-Fender | thansen|laptop: Force what to re-authenticate to what? |
| 15:17.34 | thansen|laptop | [TK]D-Fender: well, the phone is still connected and making calls |
| 15:18.03 | [TK]D-Fender | thansen|laptop: Whats the problem? |
| 15:18.08 | thansen|laptop | via sip show peers and I was able to dial it |
| 15:18.12 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:18.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:18.19 | thansen|laptop | I changed the password on the server |
| 15:18.34 | thansen|laptop | I want the device deactivated until someone changes the password on it |
| 15:19.21 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 15:19.30 | tzafrir_laptop | ~oss |
| 15:19.31 | jbot | oss is, like, a portable sound interface available in 11 different Un*x systems. The propietary package is available at http://www.opensound.com/. However, free systems like GNU/Linux or *BSD include free GPL/BSD re-implementations. The support in Linux is being gradually removed in favor of ALSA. an acronym that stands for "Open Source Software" |
| 15:19.45 | [TK]D-Fender | thansen|laptop: "sip reload" from CLI will apply the changes |
| 15:19.47 | EmleyMoor | Is there a list anywhere of which SIP phones support 2 or more independent accounts? |
| 15:19.59 | [TK]D-Fender | ~phones |
| 15:20.00 | jbot | well, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 15:20.03 | thansen|laptop | [TK]D-Fender: I already did that |
| 15:20.09 | [TK]D-Fender | EmleyMoor: Almost all do. |
| 15:20.23 | [TK]D-Fender | thansen|laptop: then your new cred are in effect |
| 15:20.44 | [TK]D-Fender | thansen|laptop: As long as the config is COMING from sip.conf and something stupid like users.conf |
| 15:21.08 | thansen|laptop | yes, from sip.conf |
| 15:21.58 | thansen|laptop | it's not kicking the device by simply doing a sip reload |
| 15:22.14 | CrashHD | so is sip over tcp a big benifit? |
| 15:22.17 | [TK]D-Fender | thansen|laptop: It won't kick calls already in progress |
| 15:22.26 | thansen|laptop | I'm calling the extension and it's and it's still ringing |
| 15:22.29 | CrashHD | is it just rtp over tcp? or signalling as well? |
| 15:22.52 | thansen|laptop | I'm debuging it and I see the headers coming from it and everything |
| 15:22.55 | [TK]D-Fender | thansen|laptop: becasue it is still registered and the phone doesn't challenge the incoming call |
| 15:23.14 | [TK]D-Fender | thansen|laptop: To fix that you'll have to blow away the registration info from astdb |
| 15:23.20 | thansen|laptop | so how do I unregister the thing since sip reload isn't working |
| 15:23.22 | [TK]D-Fender | thansen|laptop: do "help database" |
| 15:23.46 | [TK]D-Fender | CrashHD: SIP over TCP, not RTP |
| 15:24.01 | CrashHD | so the call audio is still udp? |
| 15:24.05 | [TK]D-Fender | CrashHD: And has limited benifits |
| 15:24.15 | [TK]D-Fender | CrashHD: Yes, and it nearly must be |
| 15:24.29 | thansen|laptop | [TK]D-Fender: I need to make these db calls to get the thing unregistered |
| 15:24.31 | CrashHD | ya I would figure |
| 15:24.35 | CrashHD | realtime and all |
| 15:24.43 | [TK]D-Fender | CrashHD: audio is real-time and PL, jitter, etc can all happen. You can't compensate with TCP |
| 15:24.50 | gnorbert | Does somebody have an other idea? |
| 15:25.08 | [TK]D-Fender | gnorbert: Go look at whatever else might lock up sound on your system. |
| 15:26.12 | gnorbert | [TK]D-Fender: Ok, I try. Thanks for help anyway, whether it works or not. :) |
| 15:27.13 | *** part/#asterisk neisley (n=neisley@www.nwafternoon.org) |
| 15:28.00 | EmleyMoor | Am I right in thinking all Cisco phones need a licence to work with Asterisk? |
| 15:28.52 | [TK]D-Fender | EmleyMoor: You SHOULD have one just to use them period. Cisco should be avoided. |
| 15:29.23 | EmleyMoor | Indeed... that's what I thought |
| 15:29.30 | angryuser | mahahaha welcome to a world of cisco;) |
| 15:29.44 | EmleyMoor | Only reason to go for them is if someone is insisting you do (and paying!) |
| 15:31.21 | angryuser | EmleyMoor not sure, cisco is a known brand, and no one deny the in network they are best, so people continue to buy same brand |
| 15:32.00 | Qwell | angryuser: They aren't the best. |
| 15:33.22 | coppice | if we still had governments who cared about consumers, cisco would be spanked for trying to assert that you need a software licence to use a phone |
| 15:33.34 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 15:34.26 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:34.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.144) |
| 15:34.42 | angryuser | well i am repeating what others say, like a simple consumer, i dont know any trusted organisation who can claim 'this product is the best' , all the rest is marketing as my words ;) |
| 15:36.13 | angryuser | coppice you have a licence to use windows, why not phone firmware ? |
| 15:36.47 | [TK]D-Fender | angryuser: Because windows is not the COMPUTER you're running it on. |
| 15:36.53 | coppice | one is a general purpose machine onto which you load all sorts of stuff, and the other is a bloody phone. |
| 15:38.08 | angryuser | [TK]D-Fender dont see a difference, on COMPUTERS we have a chance to have gpl linux&co without them, same case |
| 15:38.47 | tapic | tzafrir_laptop : server distro of ubuntu ends with -server word |
| 15:39.15 | [TK]D-Fender | angryuser: And you have a choice what you put on your computer. Not with a Cisco phone. So you have to CISCO's hardware, and have to LICENSE it for their own bloody software? |
| 15:39.51 | angryuser | coppice yes, bloody phone, look at snoms, they have linux inside, but snom dont make you pay the licence, cisco is |
| 15:39.55 | EmleyMoor | The firmware on your PC motherboard doesn't need a license as an extra requirement before you can use it |
| 15:40.45 | angryuser | [TK]D-Fender actually you dont have much choice, of course there is some limited versions wiht & linux & freedos, but most of the time oem windows |
| 15:41.05 | coppice | cisco need to decide if they are selling products of kits of parts |
| 15:41.07 | EmleyMoor | It's dishonest to advertise them at some price which is not a "functional" one |
| 15:41.09 | angryuser | [TK]D-Fender in general consumer world |
| 15:41.43 | coppice | EmleyMoor definitely |
| 15:41.43 | [TK]D-Fender | angryuser: General consumer world I live in has hundreds of computer stores in town that beg to differ. Dell is not the world. |
| 15:43.11 | angryuser | i dont like cisco strategy either, all i want to say , they are in position of force like microsoft, and able to dictate their rules |
| 15:43.15 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-5ce1ccda7c64f74c) |
| 15:44.31 | coppice | angryuser: polycom would beg to differ. |
| 15:44.47 | angryuser | [TK]D-Fender yes we got acer hp sony samsung .. whatever, if you want some partucalr model and ask linux in there, they wont sell you and tht is a major par of market there |
| 15:44.57 | angryuser | part |
| 15:45.03 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 15:45.09 | EmleyMoor | Are Grandstream still reckoned to be poor? |
| 15:45.19 | coppice | i think cisco will stop this crap, now they are becoming an also ran in VoIP |
| 15:45.24 | EmleyMoor | angryuser: Yes, but you can refuse the Windows license |
| 15:45.24 | angryuser | ~phones |
| 15:45.25 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 15:46.38 | angryuser | EmleyMoor mhahah yes for example , one guy refused it got his 72$ from vista licence 3 month later and tonns of paper, of course you can |
| 15:46.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:46.47 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 15:47.29 | angryuser | coppice i hope so |
| 15:47.41 | EmleyMoor | ~ |
| 15:50.43 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 15:51.06 | dominic1 | hello guys, what do you think about sangoma cards? |
| 15:51.28 | gr0mit | dominic1, i only use sangoma cards - they just work. |
| 15:51.39 | Qwell | dominic1: They don't contibute to Asterisk. |
| 15:51.46 | gr0mit | (speaking of their PRI cards - cant comment on analogue cards) |
| 15:52.12 | dominic1 | I am looking for a pricard like A104D |
| 15:52.20 | coppice | gr0mit: their analogue cards are fine, apart from the stupid plugs |
| 15:52.22 | Qwell | 1 port? |
| 15:52.31 | dominic1 | 4port |
| 15:52.34 | dominic1 | pri |
| 15:52.42 | Qwell | dominic1: Digium TE4xx |
| 15:53.17 | spokra | are there any sip clients that can be loaded on an iphone yet that you can get for un-jail broke phones |
| 15:53.55 | dominic1 | digiums still the best cards to use with isdn pri's ? |
| 15:54.07 | Qwell | dominic1: I'm biased, but yes. |
| 15:54.14 | [TK]D-Fender | spokra: you'd see it in the app store |
| 15:54.14 | EmleyMoor | Shame there isn't anywhere that specialises in selling these things |
| 15:54.26 | EmleyMoor | (OTC rather than OTW) |
| 15:54.58 | dominic1 | qwell.... okay :-) |
| 15:55.02 | spokra | fender: I've been searching everyday for new apps.. still waiting for one that is not tied to a service. |
| 15:55.30 | angryuser | EmleyMoor a lot of site sel pri interfaces.. |
| 15:55.57 | angryuser | but i have no exp about the quality ;) |
| 15:56.44 | coppice | I wish there were places on the web that sold voip stuff without the mail costing more than the product :-\ |
| 15:57.13 | Qwell | coppice: that's probably pretty difficult from HK.. |
| 15:57.17 | angryuser | coppice that depends where you live |
| 15:57.37 | coppice | it seems like shipping has doubled in the last year |
| 15:57.58 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 15:58.08 | hardwire | voip-info made it sound like app_managerredirect would be included in 1.4 or 1.6 |
| 15:58.18 | hardwire | anybody using app_managerredirect ? |
| 15:58.34 | *** join/#asterisk minaguib (n=mina@modemcable118.145-203-24.mc.videotron.ca) |
| 15:58.50 | Qwell | hardwire: never heard of it |
| 15:59.06 | Qwell | does it maybe mean channelredirect? |
| 15:59.06 | hardwire | Qwell: apparently it's identical to app_channelredirect |
| 15:59.08 | hardwire | but not. |
| 15:59.12 | outtolunc | hardwire: there is a AMI action: Redirect |
| 15:59.38 | minaguib | What do you guys think of this: Voip packet sent with IP header TOS 0x10 from client A on ISP I to client B (same ISP) arrives with TOS 0x0 |
| 15:59.41 | hardwire | outtolunc: that redirects bridged channels? |
| 15:59.58 | outtolunc | redirects upto 2 channels |
| 16:00.29 | dominic1 | anybody which is not biased can tell me something about the quality of the sangoma products? |
| 16:00.35 | *** join/#asterisk Superbartt (n=bart@84-245-35-223.dsl.cambrium.nl) |
| 16:00.47 | angryuser | coppice is is simple to know why, look at the petrol price |
| 16:00.55 | hardwire | I'll need call managers to redirect a bridged channel to a meetme for an externally forces n-way call |
| 16:00.58 | hardwire | forced |
| 16:01.23 | snapple42 | we're using sangoma on a bunch of servers... seem pretty good so far.. been running a couple years now. |
| 16:02.07 | coppice | angryuser: I think when they now want as much to fly 0.5kg from the US to me as they charged last year for 18kg, its a little more than the gas prices |
| 16:02.12 | [TK]D-Fender | minaguib: QoS has no gaurantees outside of your network regardless of if you're on the sam ISP |
| 16:02.25 | dominic1 | will I need to install special drivers for sangomacards |
| 16:02.26 | dominic1 | ??? |
| 16:02.38 | [TK]D-Fender | dominic1: yes |
| 16:03.44 | minaguib | [TK]D-Fender: So you don't think videotron's necessarily being malicious by stripping those fields off to enhance their own VOIP offering ? |
| 16:03.44 | dominic1 | are there any cards on the qualitylevel of digium and is it possible to install cards without a excessive orgy of installing stuff I don't want? |
| 16:04.06 | [TK]D-Fender | minaguib: Oh no, I FULLY believe them capable of screwing its customers over every which way they can get away with. |
| 16:04.25 | [TK]D-Fender | minaguib: Videotron = suck |
| 16:04.37 | Qwell | dominic1: all hardware is going to need some type of driver |
| 16:05.09 | Qwell | most Asterisk hardware will use Zaptel, and Sangoma requires an extra layer |
| 16:05.15 | [TK]D-Fender | dominic1: Every non-clone of Digium requires a driver of some sort |
| 16:05.51 | [TK]D-Fender | dominic1: If thats going to be the deciding factor for you then just stop and buy a Digium branded card right now and save us all the discussion. |
| 16:11.36 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 16:12.20 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:12.40 | adr3nalin3 | where do you change the smtp server for vm notifications? |
| 16:13.43 | [TK]D-Fender | adr3nalin3: To do what exactly? |
| 16:14.03 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 16:14.34 | dominic1 | which manufacturer is using zaptel? |
| 16:14.39 | adr3nalin3 | Just to send out the notifications of a voicemail received. |
| 16:14.48 | Qwell | dominic1: pretty much all of them |
| 16:15.51 | [TK]D-Fender | dominic1: Almost all of those with their own driver use it as a FRONT END to Zaptel. |
| 16:16.02 | [TK]D-Fender | dominic1: Its just one layer in front of it |
| 16:16.08 | dominic1 | I was using beronet for some time, they used misdn.... |
| 16:16.23 | dominic1 | ah okay, that's interesting |
| 16:16.24 | exothermc | is there any SIP clients for the blackberry? |
| 16:16.45 | [TK]D-Fender | adr3nalin3: * calls the "sendmail" binary. Yuo can have this replaced by any other MTA's front end that you want thats compatible |
| 16:17.07 | dominic1 | my problem is, that I am not able to indicat the differences between a 4port digium and a 4port sangoma etc... They all have hardware ec ..... |
| 16:17.08 | [TK]D-Fender | dominic1: Do not mix BRI with PRI. |
| 16:17.17 | adr3nalin3 | ok thanks [TK]D-Fender |
| 16:17.21 | *** join/#asterisk robos (n=robos@70.89.39.222) |
| 16:18.02 | robos | Hi. I would like to send and receive SMS text messages. I'm in the starting phases of figuring out how to do it and I think Asterisk could handle this... ? |
| 16:18.05 | dominic1 | [TK]D-Fender: do you mean bri cards and pri in one system? |
| 16:18.34 | *** join/#asterisk kensukeido (n=kensuke@unaffiliated/kensukeido) |
| 16:18.46 | [TK]D-Fender | dominic1: No, I mean conceptually. Basically jsut about every BRI card is jsut plain trouble with *. |
| 16:19.11 | [TK]D-Fender | dominic1: * PRI support for notmal cards is pretty simple regarless of needing 1 timy layer in front and all that I've seen go through Zaptel |
| 16:20.24 | dominic1 | okay, I have one installation with beronet misdn BRI and a few others with beronet misdn pri. I had no trouble before, but heard there are solutions which better fit to a asterisk system. |
| 16:21.21 | [TK]D-Fender | dominic1: Yeah. NON-BRI with common cards better supported :) |
| 16:22.24 | *** join/#asterisk rue_mohr (n=root@h24-207-89-169.cst.dccnet.com) |
| 16:23.25 | rue_mohr | so, todays question is how to set up asterisk to identify distinctive ring on incomming loopstart lines |
| 16:23.48 | [TK]D-Fender | rue_mohr: "dring" <- WIKI it |
| 16:24.55 | *** join/#asterisk draygon-w (n=draygon@gateway5-pnap.exigo.com) |
| 16:26.39 | adeel | does * support rtp multicast? |
| 16:26.55 | adeel | i saw something on the bug tracker about it, but it didn't include any source |
| 16:27.14 | dominic1 | okay thank you very much fender |
| 16:27.21 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 16:28.00 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:28.56 | rue_mohr | http://en.wikipedia.org/wiki/Asterisk |
| 16:29.06 | rue_mohr | could you remind me the * wiki url? |
| 16:29.28 | [TK]D-Fender | ~wikis |
| 16:29.28 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:29.59 | rue_mohr | thnka ;) |
| 16:31.14 | dominic1 | again a short question.... Can I store a variable with the content var=Dial(SIP/myaccount) and execute this variable? |
| 16:31.28 | Qwell | sure |
| 16:31.33 | Qwell | Exec(${var}) |
| 16:31.40 | Qwell | or something |
| 16:31.50 | Qwell | exten => s,1,Exec(${var}) |
| 16:31.55 | Qwell | ...maybe |
| 16:32.02 | rue_mohr | but I'm seking dring on an fxo |
| 16:32.38 | dominic1 | Qwell: It works like a charm, thank you very much |
| 16:33.31 | rue_mohr | TDM800P fxo some lines have distinctive ring |
| 16:34.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:34.49 | rue_mohr | "Note that specifying a distinctive ring makes sense only when you are ringing an internal extension (e.g. the channel is connected by an FXS interface). " |
| 16:34.53 | c4t3l | TK: have you seen any good iax load balancers out there? |
| 16:35.11 | *** join/#asterisk hoonteke (n=kevin@user-0c2hj2e.cable.mindspring.com) |
| 16:35.26 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 16:38.03 | hoonteke | I have a coworker running xmeeting. He's been on our trixbox all morning, but after a phone call to a local number, he is now unable to get back on. my regular sysadmin is on vacation, and I'm currently at a loss at how to debug this. Can anyone offer me some pointers? |
| 16:38.59 | hoonteke | I'm able to connect just fine to the asterisk instance, and after a reboot. I do note an ERROR[1500] when he connects, but so far GINMF. |
| 16:39.09 | Qwell | ~trixbox |
| 16:39.10 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:39.13 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 16:39.59 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 16:40.02 | hoonteke | uh ... I think it's actually debian etch with the asterisk package |
| 16:40.05 | hoonteke | I just said trix box |
| 16:40.08 | Qwell | ... |
| 16:40.11 | hoonteke | but ... how would I check to be sure? |
| 16:40.32 | Qwell | [TK]D-Fender: listcheck |
| 16:40.50 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 16:41.45 | hoonteke | Qwell: I'm sorry, I don't know what that means? |
| 16:41.46 | [TK]D-Fender | Qwell: ? |
| 16:41.49 | hoonteke | oh, haha |
| 16:41.52 | hoonteke | it's someone, my bad |
| 16:42.11 | Qwell | [TK]D-Fender: ^^ |
| 16:43.13 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 16:43.47 | *** join/#asterisk [tasty]freeze (n=derek@204-181-48-126.skybest.com) |
| 16:43.59 | [tasty]freeze | does anyone in here use the cisco cp-7940? |
| 16:44.09 | clyrrad | i've looked at the sample config file for res_pgsql.conf, there seems to be no ssl parameter in there, I am finding mixed reviews by Googling. Does anyone know definitavly if res_pgsql supports SSL in asterisk 1.4? |
| 16:44.33 | [TK]D-Fender | Qwell: Sorry, don't follow you... |
| 16:44.39 | Qwell | nm |
| 16:44.49 | [TK]D-Fender | clyrrad: its your DSN's job, not *'s |
| 16:45.14 | hoonteke | [TK]D-Fender: I think he's asking if you might have a pointer for my problem? |
| 16:45.16 | hoonteke |  and if he's not, I am. :-) |
| 16:45.18 | clyrrad | [TK]D-Fender: is res_pgsql pulling from odbc.ini though? |
| 16:46.03 | clyrrad | I though if using res_pgsql.conf for realtime would be seperate from using DSN would it not? Or I am mistaken? |
| 16:46.20 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 16:46.53 | outtolunc | hoonteke: i think the part where qwell already did the ~trixbox output (which you ignored) was more of the point |
| 16:47.25 | rue_mohr | [TK]D-Fender so, can asterisk recognice dring on an fxo line? (assume tdm800P) |
| 16:47.41 | Qwell | rue_mohr: there is dring support, yes |
| 16:47.46 | [TK]D-Fender | hoonteke: You have NO useful details for your problem and clearly don't even know what you have. |
| 16:47.52 | rue_mohr | but it says for fxs lines |
| 16:48.07 | Qwell | why would an fxs detect dring? |
| 16:48.16 | Qwell | (hint: fxs never gets ringing) |
| 16:48.17 | [TK]D-Fender | hoonteke: Go sacrifice a goat or something. The voodoo you'll require necessitates it. |
| 16:48.27 | [TK]D-Fender | rue_mohr: You read it wrong |
| 16:48.31 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:48.34 | rue_mohr | my understanding from teh docs I can find are that for an fxs line it can DO dring |
| 16:48.37 | rue_mohr | ok |
| 16:48.46 | rue_mohr | I just wanted to be sure :) |
| 16:48.55 | Qwell | rue_mohr: I believe both are supported - dring detection on fxo is definitely supported though. |
| 16:49.02 | hoonteke | ok, ... |
| 16:49.15 | Qwell | (I don't think I've ever tried to dring out an FXS..) |
| 16:49.32 | clyrrad | [TK]D-Fender: so just want to be clear, were you indicating the res_pgsql is going to use the data I have in odbc.ini? |
| 16:50.36 | [TK]D-Fender | clyrrad: Perhaps I'm mistaken, though my approach woul let you pick the driver that DOES support it |
| 16:51.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 16:52.03 | clyrrad | [TK]D-Fender: indeed, but before going on a driver hunt that supports SSL, just trying to get a definitive answer if res_pgsql has some hidden / undocumented parameter that would enable SSL support, from Googling this it seems there is a paramter 'sslmode', but I can not find documentation for it |
| 16:52.50 | [TK]D-Fender | clyrrad: The source would list it |
| 16:54.30 | clyrrad | hrm, guess i'll have a look in the C code |
| 16:58.51 | clyrrad | well that answers that, checking the code there is definatly no 'sslmode' parameter |
| 17:00.24 | *** join/#asterisk aliver (n=aliver@ip-216-17-160-99.rev.frii.com) |
| 17:01.16 | *** join/#asterisk LND (n=Lee@89.192.9.65) |
| 17:03.37 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 17:04.50 | *** join/#asterisk madison (n=madison@208.89.241.9) |
| 17:09.56 | mandd | where does asterisk store/read sound files |
| 17:10.03 | mandd | on freebsd |
| 17:10.08 | *** join/#asterisk rdgr_ (n=rich@87-194-164-154.bethere.co.uk) |
| 17:10.26 | mandd | configs are in /usr/local/etc/asterisk/ |
| 17:10.37 | [TK]D-Fender | mandd: Fo look in asterisk.conf for the lib dir |
| 17:10.41 | [TK]D-Fender | Go* |
| 17:10.50 | mandd | ok, thanks! |
| 17:11.15 | mandd | astmoddir => /usr/local/lib/asterisk/modules |
| 17:11.25 | mandd | astvarlibdir that? |
| 17:12.08 | mandd | got it! |
| 17:12.12 | mandd | thanks [TK]D-Fender |
| 17:14.50 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:15.10 | *** join/#asterisk pigpen2 (n=pigpen@66.139.29.29) |
| 17:17.02 | madison | how do i get music on hold working? I have a folder called /var/lib/asterisk/mohmp3, I am using the default class in musiconhold.conf and I setup an extension |
| 17:17.27 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 17:18.07 | [TK]D-Fender | madison: Make sure to have completely restarted *. Try using it. Pastebin the CLI output of your attempt. |
| 17:18.09 | [TK]D-Fender | ~pb |
| 17:18.09 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:18.11 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:19.46 | madison | http://pastebin.ca/1088633 |
| 17:20.00 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:20.42 | [TK]D-Fender | madison: Ok you have to do MusicOnHold(default). You need to specifify the class |
| 17:21.13 | madison | checking |
| 17:24.37 | madison | i have specified the class |
| 17:24.50 | madison | still same error |
| 17:25.25 | [TK]D-Fender | madison: pastebin your musiconhold.conf and do "ls -la /var/lib/asterisk/mohmp3" |
| 17:26.58 | *** join/#asterisk [Akemi] (n=akemi@206-248-164-136.dsl.teksavvy.com) |
| 17:27.27 | madison | http://pastebin.ca/1088637 |
| 17:29.24 | [TK]D-Fender | madison: MP3's must be non-VBR, no ID3 tags, and your FOLDER is wrong : directory=/var/lib/asterisk/moh |
| 17:30.11 | mandd | <PROTECTED> |
| 17:30.11 | mandd | sox: Command not found. |
| 17:30.20 | mandd | 1.4 is different? |
| 17:30.25 | madison | i was using class [manual] |
| 17:30.51 | [TK]D-Fender | mandd: sox is NOT part of * |
| 17:31.20 | [TK]D-Fender | madison: then you need to specify that class. |
| 17:31.37 | madison | ok, something seems to be happening now |
| 17:31.45 | [TK]D-Fender | madison: You said you were "I am using the default class in musiconhold.conf" |
| 17:31.52 | [TK]D-Fender | madison: Please be consistent |
| 17:33.13 | madison | sorry, first cup of coffee |
| 17:34.19 | [TK]D-Fender | madison: make it a tripple-short espresso and load the sugar to you max out solubility. |
| 17:35.25 | madison | http://pastebin.ca/1088643 |
| 17:35.43 | madison | no more fast busy but no audio |
| 17:36.40 | [TK]D-Fender | madison: Go verify what version of mpg123 you are running |
| 17:37.32 | madison | well dang, it is missing :p |
| 17:38.42 | [TK]D-Fender | SMRT |
| 17:38.44 | madison | i wish that my boss was not so gungho on using opensuse :p |
| 17:38.56 | madison | I prefer gentoo or centos |
| 17:38.59 | [TK]D-Fender | madison: just use ANtive MoH and stop Fing around |
| 17:39.02 | [TK]D-Fender | Native* |
| 17:39.16 | madison | ok, how do I do that |
| 17:39.32 | *** join/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 17:39.40 | madison | move the mp3's into /var/lib/moh? |
| 17:39.47 | madison | and use class default? |
| 17:41.06 | [TK]D-Fender | madison: mode=files <- |
| 17:41.12 | jsolis | hey people, how many connection support asterisk with AMI ? -- sorry for my english |
| 17:41.18 | [TK]D-Fender | madison: And make sure you have compiled asterisk-addons |
| 17:41.39 | [TK]D-Fender | jsolis: a bunch, and there is astmanproxy if you are looking to do something insane |
| 17:41.48 | [TK]D-Fender | jsolis: What are you thinking of doing? |
| 17:42.15 | jsolis | i make sockets with java to connect with asterisk |
| 17:42.28 | jsolis | via AMI |
| 17:43.04 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
| 17:43.11 | [TK]D-Fender | jsolis: HOW MANY? |
| 17:43.18 | jsolis | 4 |
| 17:44.09 | jsolis | is a good idea have a 4 connections? |
| 17:44.56 | [TK]D-Fender | jsolis: Shouldn't have throught twice about it. |
| 17:45.05 | [TK]D-Fender | jsolis: Don't worry, just do it |
| 17:45.41 | [TK]D-Fender | madison: And remove desktop.ini etc from there. |
| 17:46.14 | jsolis | thx |
| 17:48.01 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 17:48.32 | *** join/#asterisk stoffell (n=kristof@d51A4DC78.access.telenet.be) |
| 17:49.14 | The-Bat | Hie |
| 17:49.15 | The-Bat | is it possible to hear live ongoing calls in queue(or any ongoing call) by a 3rd person ?? |
| 17:49.35 | stoffell | The-Bat: spy on channels.. chanspy |
| 17:49.50 | The-Bat | stoffell, Thanks |
| 17:50.18 | stoffell | yw |
| 17:51.53 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 17:52.06 | madison | ok |
| 17:54.20 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 17:55.51 | rickross | anyone know of a variant of MixMonitor (or similar) that will allow recording of caller on one stereo channel and the call recipient on the other (useful for independent audio processing for podcast recording) |
| 17:56.38 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 17:57.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:57.17 | rickross | afaik, MixMonitor will allow volume adjustement to caller or receiver, but not recording on separate stereo channels |
| 17:58.01 | rickross | which is curious, since * seems to keep temporary files for each channel discretely until the end of the call, when they get combined |
| 17:58.54 | file | MixMonitor combines them as the call goes on |
| 17:58.59 | madison | still nothing |
| 17:59.12 | file | well, each frame of audio is combined in the audiohooks stuff and then provided to MixMonitor as a single frame |
| 17:59.24 | file | Monitor records them separately |
| 18:00.36 | mchou | when I use something like "exten => s,n,Congestion" must the extension be answered first? (like "exten => s,n,Answer") |
| 18:01.13 | file | mchou: if it is not answered the Congestion goes back to the channel driver which usually sends it out of band, for example SIP will translate it into a SIP response instead of playing it as audio |
| 18:01.31 | mchou | file: ok, cool |
| 18:01.41 | [TK]D-Fender | rickross: call your own script to do the combin your way |
| 18:02.34 | rickross | yeah, that looks like it'll be the way to go |
| 18:03.04 | rickross | I bet there are others who would like to use their * systems for podcast recordings - it should be well-suited |
| 18:03.11 | rickross | there are some google hits on the topic |
| 18:04.19 | *** join/#asterisk Corazu (n=justin@bas3-toronto12-1177560563.dsl.bell.ca) |
| 18:04.29 | Corazu | Hello all |
| 18:04.30 | EmleyMoor | On 1.2, is there a way to see all the headers of an incoming SIP call? |
| 18:05.49 | carrar | ngrep |
| 18:06.07 | carrar | tcpdump |
| 18:07.30 | EmleyMoor | Nothing a bit more "native" |
| 18:07.37 | EmleyMoor | ? |
| 18:07.39 | mchou | file: I'd like to set up an asterisk server whose sole purpose is to swat down telemarketing calls by peeking at callerID and comparing it to a precompiled callerID blacklist. |
| 18:07.51 | *** join/#asterisk Nkoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 18:08.01 | file | I use the astdb and some dialplan logic |
| 18:08.06 | file | for that specific purpose |
| 18:08.11 | EmleyMoor | mchou: I do that too |
| 18:08.17 | mchou | file: astdb? |
| 18:08.25 | EmleyMoor | Asterisk database - built in |
| 18:08.33 | file | mchou: berkeley database stored in a local file manipulated via CLI commands, persists through restarts |
| 18:08.34 | carrar | ngrep -d eth0 -s 1500 host X.X.X.X |
| 18:08.42 | mchou | is that some sort of real time blacklist? :) |
| 18:08.47 | Nkoji | Hello i am very new to asterisk and would just like to double check that this will work. If i have i make a call out with my asterisk system can i recieve the information of about the number i am calling |
| 18:08.53 | EmleyMoor | mchou: What do you mean by real time? |
| 18:08.55 | carrar | +q for good measure :) |
| 18:09.04 | Nkoji | such as caller id |
| 18:09.14 | mchou | EmleyMoor: kind of like RBL for email spammers :) |
| 18:09.21 | mchou | community compiled |
| 18:09.38 | EmleyMoor | mchou: No, I know of no realtime version - all entry would be manual |
| 18:09.43 | EmleyMoor | Lists do exist out there |
| 18:09.52 | mchou | EmleyMoor: cool |
| 18:09.53 | carrar | ngrep -q -d eth0 -s 1500 host x.x.x.x and port 5060 |
| 18:09.56 | EmleyMoor | (I have a "marketer" class in mine) |
| 18:10.38 | EmleyMoor | carrar: OK - nothing native then - will have to install ngrep |
| 18:10.53 | The-Bat | stoffell, wot do u think abt ExtenSpy ..?? |
| 18:12.02 | mandd | becides sox, anytihng else that it already build into asterisk, that can convert .wav to .gsm? |
| 18:12.12 | mchou | EmleyMoor: so what exactly do you send a call that determined to be in the marketer class? hang up, congestion, SIT or ? |
| 18:12.27 | mchou | that's* |
| 18:12.50 | hardwire | anybody used all 192 channels at once on an 8 port PCI-Express TDM card? |
| 18:12.54 | hardwire | did you smell smoke? |
| 18:12.59 | EmleyMoor | mchou: A recorded message wanting them to leave voicemail stating who they are calling from |
| 18:13.10 | stoffell | The-Bat: probably what i mean :-) |
| 18:13.19 | EmleyMoor | If I'm not happy, I then Blacklist them - they get SIT but still get a chance to leave voicemail |
| 18:13.32 | mchou | EmleyMoor: hmmm.... |
| 18:13.40 | *** join/#asterisk moy (n=moy@nat/ibm/x-29cab13d78769218) |
| 18:13.58 | mchou | EmleyMoor: I'm thinking just reporting congestion, any downside to that? |
| 18:14.20 | EmleyMoor | It could block companies you do genuinely have dealings with... |
| 18:14.25 | Nkoji | Project Question : Can you receive information on the phone number you are dialing? |
| 18:14.31 | mchou | EmleyMoor: I'm a bit leery that will encourage the autodialers to dial more often though |
| 18:15.23 | Corazu | I've got a bit of a problem with the music on hold; I'm using 16bit [pcm] sampled at 8000. I loaded them into audacity to reduce the volume but asterisk seems to keep playing them at a higher volume. Is this a possibility..or is the problem somewhere else? |
| 18:17.35 | EmleyMoor | Hmmm... not seeing any headers with the caller's ID in with that... |
| 18:18.35 | madison | [TK]D-Fender, http://pastebin.ca/1088688 |
| 18:18.51 | mchou | file: so how does pattern matching work on astdb? Does it use sql or? |
| 18:19.11 | file | nope |
| 18:19.20 | madison | yay it works now |
| 18:19.42 | file | mcab: there is a family, key, and value... you look up via family and key |
| 18:19.45 | *** join/#asterisk stoffell (n=stoffell@d51A4DC78.access.telenet.be) |
| 18:20.26 | mchou | file: hmm....I thinking just using grep and backticks |
| 18:20.50 | mchou | be a lil simpler to set up |
| 18:20.54 | hoonteke | how can I tell asterisk to debug sip stuffs to a file? |
| 18:24.04 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 18:24.09 | LemensTS | http://www.voip-info.org/wiki-Asterisk+debugging |
| 18:33.27 | EmleyMoor | Well, I don't think my provider gives me any "interesting" SIP headers... |
| 18:34.17 | EmleyMoor | Just an offchance anyway |
| 18:37.08 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 18:38.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:38.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:40.55 | Corazu | Anyone have any ideas about my problem with MoH? |
| 18:41.21 | madison | is there a way to have a menu system to choose different music on hold choices? ie, press 1 for country, press 2 for jazz |
| 18:44.19 | [TK]D-Fender | madison: that still has the WRONG FOLDER that I warned you about over an hour ago |
| 18:45.18 | madison | [TK]D-Fender, it is working now. i fixed it |
| 18:45.48 | *** join/#asterisk Vec (n=Vec@nsabfw1.nsab.se) |
| 18:47.27 | Vec | Hi, when I user is on an incoming zap call and then does a transfer to do a Dial(SIP/xxx&SIP/xxx) |
| 18:47.34 | Vec | oops bumped enter :p |
| 18:47.56 | Vec | then the user can't transfer the zap call to the answered party. |
| 18:48.05 | Vec | Any idea how I could get around this ? |
| 18:49.30 | EmleyMoor | How does he do the transfer? |
| 18:50.10 | madison | what is a quick and easy way to remove id3 tags from mp3 |
| 18:50.17 | madison | files in linux? |
| 18:50.46 | Vec | EmleyMoor : he puts the caller that is on zap on hold, then dials 500 which does the Dial(SIP&SIP) and then pushes transfer. |
| 18:52.19 | Vec | When the transfer button is pushed I see music on hold turning off and then immidately back on on the asterisk CLI output but nothing happens. |
| 18:52.25 | lmadsen | madison: I think you're in the wrong channel for that question |
| 18:52.37 | tzafrir_laptop | madison, "apt-cache search mp3 | grep -i 'id.*tag'" gives, e.g. id3tool |
| 18:52.56 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:53.31 | madison | tzafrir_laptop, thank you |
| 18:55.08 | Vec | How do I know if my zap channels are ulaw or alaw ? |
| 18:56.21 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:58.37 | [TK]D-Fender | Vec: What are those SIP devices? |
| 18:59.05 | Vec | [TK]D-Fender : Counterpath Bria soft phones. |
| 18:59.38 | Vec | [TK]D-Fender : does the tT options have to be set in the Dial cmd ? |
| 18:59.57 | [TK]D-Fender | Vec: No, your PHONE should have a transfer button of their own |
| 19:00.42 | Vec | [TK]D-Fender : it does and transfers work normally, but if I transfer to a established call that was dialed using multiple devices, then the transfer does not go through. |
| 19:00.48 | EmleyMoor | Vec: Is the call being handled on a SIP phone?# |
| 19:01.02 | EmleyMoor | How do you dial a call using multiple devices? |
| 19:01.44 | Vec | EmleyMoor : yeh Bria soft phone, only supports SIP |
| 19:02.06 | [TK]D-Fender | Vec: Sorry your wording is very hard to follow |
| 19:02.13 | EmleyMoor | Not one I've used... what happens if you press "Transfer" while on the call? |
| 19:02.15 | Vec | EmleyMoor : dial a call to multiple devices Dial(SIP/num&Sip/num2etcetc) |
| 19:02.19 | [TK]D-Fender | \vecYou Bria's should always be capable of transferring a call |
| 19:02.35 | EmleyMoor | Vec: It can't be on multiple devices... it's "first to answer" |
| 19:02.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.11 | Vec | [TK]D-Fender : r those tT options in the Dial cmd for the asterisk transfer application, is call transfer independent of asterisk ? |
| 19:03.11 | Vec | EmleyMoor : yeh sorry thats what I mean |
| 19:03.26 | [TK]D-Fender | Vec: SIP Transfers have NOTHING to do with "Tt" |
| 19:03.28 | EmleyMoor | The tT options are primarily for Zap phones |
| 19:03.42 | [TK]D-Fender | and SHITTY softphones |
| 19:03.43 | Vec | cool thanks |
| 19:03.57 | Vec | Would you sau that Bria is a shity osftphone ? |
| 19:04.26 | madison | spa941 ftw |
| 19:04.35 | EmleyMoor | What happens if you "Transfer" without putting the call on "Hold"? |
| 19:05.09 | [TK]D-Fender | Vec: Never used it, but because its a paid version last I heard I'm sure it has the basic features. |
| 19:06.43 | Vec | [TK]D-Fender : it seems to have too many featuers, I think it may actually be running out of memmory and that why the call transfer doesn't go through, although that is a long shot, the comps have 256MB memory with XP and lots of other apps open. |
| 19:07.43 | [TK]D-Fender | Vec: BS Excuse. Either it has a transfer button or it doesn't. If you are attempting a transfer go look at the SIP debug of your attempt from the moment the get the call to the end |
| 19:08.45 | Vec | [TK]D-Fender : yeh good point, it certainly has the transfer button but after pushing the button nothing happens. |
| 19:09.05 | [TK]D-Fender | Vec: NEVER, or a select few times? |
| 19:09.40 | Vec | [TK]D-Fender : never when using Dial(with multiple devices) |
| 19:09.56 | EmleyMoor | ... and always when not??? |
| 19:10.00 | *** part/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 19:10.18 | [TK]D-Fender | Vec: only time I'm going to say this : Dial doesn't have ANYTHING to do with a "Transfer" button on your soft-phone |
| 19:10.34 | [TK]D-Fender | Vec: Doesn't matter if 1 device was called, or a HUNDRED |
| 19:10.42 | EmleyMoor | Vec: You alluded to putting the call on hold - why? |
| 19:10.47 | Vec | [TK]D-Fender : yeh I understand that, thats why its odd |
| 19:11.07 | [TK]D-Fender | Vec: I think you need to actually look at sip debug, and read the manual for it |
| 19:11.26 | Vec | the phone automatically puts the first call on hold when doing the transfer, ps: its an attended transfer not a blind transfer |
| 19:11.44 | EmleyMoor | Vec... OK... so then what do you do? |
| 19:12.11 | madison | tzafrir_laptop, how does that program strip off the id3 tag? |
| 19:12.18 | madison | i see no flag for it |
| 19:12.58 | tzafrir_laptop | madison, never used it |
| 19:13.04 | Vec | I definately don't think its something I am doing wrong in terms of using the soft phone, let me investigate a little more. |
| 19:13.13 | EmleyMoor | Vec: What is the process by which you are trying to transfer the call? |
| 19:13.37 | *** join/#asterisk mcargile (n=mcargile@mail.in-florida.net) |
| 19:13.47 | CrashHD | so I'm having this issue where at certain times all connections seem to die from our asterisk servers |
| 19:15.11 | CrashHD | asterisk server to asterisk server on same lan and asterisk to phones over t1 |
| 19:15.27 | CrashHD | and two different asterisk servers to gateway on same lan drop |
| 19:15.59 | Vec | EmleyMoor : I push the "Call then transfer button" it puts the user on hold and prompts me to dial, I dial the new number say 555, it rings multiple devices, when the first person picks up, I say I have a call for you, I then push the transfer button, this works normally. |
| 19:16.08 | Vec | i.e when just dialing another extension directly |
| 19:16.18 | CrashHD | I see a debug message saying DEBUG[15785] sched.c: Request to schedule in the past?!?! |
| 19:16.39 | EmleyMoor | Vec: Hmmm... not quite how I expected... what's different when it fails? |
| 19:17.48 | Vec | in asterisk there is Dial(multiple devices) rather than Dial(single device) |
| 19:18.12 | EmleyMoor | Yes - and you've now said it works with multiple devices... |
| 19:19.44 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:21.17 | CrashHD | would a timer issue affect sip messages or just rtp? |
| 19:21.50 | EmleyMoor | Vec: What are you doing differently when it fails? (never mind what Asterisk is doing) |
| 19:22.22 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 19:22.27 | hsv-al | hello fellow internet addicts |
| 19:22.39 | hsv-al | are we all looking forward to another long & glorious Thursday of irc ? :) |
| 19:22.57 | Vec | EmleyMoor : nothing!, I will investigate more |
| 19:24.11 | Vec | EmleyMoor : thanks |
| 19:24.12 | [TK]D-Fender | Vec: Something is jsut flatly BS about your scenario. To each device called, it is jsut another SIP call. There is no such concept as "multi" vs "single". |
| 19:24.43 | Vec | [TK]D-Fender : I believe you guys, thats why I need to do some more experimenting. |
| 19:24.49 | Vec | thanks |
| 19:25.50 | Vec | Does it matter if the Dial(local/number) is used to make the call, or will asterisk replace local with Tech ? |
| 19:26.10 | [TK]D-Fender | Vec: the CALL to your SIP device is never "special". |
| 19:26.32 | Vec | [TK]D-Fender : yeh makes sence. |
| 19:27.42 | CrashHD | what would this debug message indicate? sched.c: Request to schedule in the past?!?! |
| 19:27.56 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 19:28.05 | EmleyMoor | CrashHD: That you are trying to do something before "now" |
| 19:28.25 | CrashHD | about 3 seconds after that messsage sip qualifies to various points became unreachable |
| 19:28.39 | [TK]D-Fender | "When will then be now? SOON!" - Spaceballs |
| 19:28.46 | CrashHD | lol classic movie |
| 19:28.56 | [TK]D-Fender | CrashHD: usually indicative of a timing issue. |
| 19:29.13 | CrashHD | I have ztdummy running |
| 19:29.21 | CrashHD | 99.98's + |
| 19:29.22 | [TK]D-Fender | CrashHD: zttest it |
| 19:29.33 | *** join/#asterisk LND (n=Lee@89.192.133.131) |
| 19:29.42 | [TK]D-Fender | CrashHD: using mpg123 by any chance? |
| 19:29.51 | CrashHD | nope |
| 19:29.54 | CrashHD | and what is weird |
| 19:30.00 | CrashHD | is the problem is across a couple of boxes |
| 19:30.00 | [TK]D-Fender | CrashHD: Ok, then I'm fresh out of ideas |
| 19:30.00 | bijit | ~buybook |
| 19:30.01 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 19:30.09 | CrashHD | all similar setups |
| 19:30.16 | CrashHD | been having ongoing call drop issues |
| 19:30.19 | CrashHD | and call quality issues |
| 19:30.26 | CrashHD | I found we had no timing installed |
| 19:30.35 | CrashHD | so I installed ztdummy last night |
| 19:30.43 | CrashHD | and enabled rtc support |
| 19:30.54 | CrashHD | and cranked the sample rate to 2048 with module option |
| 19:30.59 | CrashHD | to get my zttest where it should be |
| 19:31.12 | [TK]D-Fender | CrashHD: Check your kernel timing |
| 19:31.21 | CrashHD | default for centos is 1000hz |
| 19:31.26 | [TK]D-Fender | crash you can get this is it was done with something other than 1000hz |
| 19:31.27 | CrashHD | dunno where to actually see the current |
| 19:31.36 | [TK]D-Fender | CrashHD: Ok, you SHOULd be fine then.. |
| 19:31.40 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 19:31.44 | [TK]D-Fender | CrashHD: dmesg shows it IIRC |
| 19:32.36 | CrashHD | let me see if I can find it |
| 19:32.46 | CrashHD | any grep keyword |
| 19:32.46 | CrashHD | ? |
| 19:36.31 | [TK]D-Fender | CrashHD: "hz" |
| 19:36.32 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 19:37.00 | CrashHD | dmesg | grep hz has no output |
| 19:37.02 | CrashHD | lol |
| 19:37.04 | CrashHD | weird |
| 19:39.46 | CrashHD | Hz |
| 19:40.18 | CrashHD | I don't see anything indicating 1000 though? |
| 19:40.28 | CrashHD | hpet0: 3 64-bit timers, 14318180 Hz |
| 19:40.39 | CrashHD | I see some cpu MHz things |
| 19:40.54 | CrashHD | an assumption about a 33MHz system bus |
| 19:41.01 | CrashHD | and something about nic cards |
| 19:41.29 | friendly_fire | network interface card cards, my favorite |
| 19:43.31 | *** join/#asterisk [Akemi] (n=akemi@69-196-154-216.dsl.teksavvy.com) |
| 19:43.32 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:44.52 | *** part/#asterisk mcargile (n=mcargile@mail.in-florida.net) |
| 19:47.30 | CrashHD | and get this |
| 19:47.35 | CrashHD | I just got a report of |
| 19:47.46 | CrashHD | a caller dialed the office |
| 19:47.58 | CrashHD | it rang 4 times then dropped from the perspective of the caller |
| 19:48.07 | CrashHD | 2 minutes later that phone call rang in on the system |
| 19:48.12 | CrashHD | and made all the phones ring haywire |
| 19:48.47 | *** join/#asterisk n9urk (n=IceChat7@rrcs-70-62-74-122.midsouth.biz.rr.com) |
| 19:50.05 | *** join/#asterisk LoOoD (n=gman@64.201.247.2) |
| 19:50.31 | LoOoD | How can I tell which codec a sip connection is using? |
| 19:50.43 | CrashHD | sip show channel .... |
| 19:51.11 | n9urk | Hi, would someone please paste a link to where I can find the latest information for using MP3s for music on hold? I know things have changed with mpg123 and what not, but I can't find any current info. Thanks |
| 19:51.27 | CrashHD | voip-info has some pretty good info I believe |
| 19:51.54 | n9urk | CrashHD -> voip-info was where looked. Do you have an exact page there in mind? |
| 19:52.19 | [TK]D-Fender | ~book |
| 19:52.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:52.21 | [TK]D-Fender | ^^^ |
| 19:53.09 | CrashHD | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf -> Title: Using native Asterisk format_mp3 for Music on Hold * |
| 19:53.10 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 19:53.32 | n9urk | [TK]D-Fender looking at the book now. |
| 19:53.51 | *** join/#asterisk AMUG (n=junky@modemcable156.137-20-96.mc.videotron.ca) |
| 19:54.15 | [TK]D-Fender | AMUG: Salut mon ostie! :p |
| 19:54.42 | hsv-al | Tha KillaZ |
| 19:54.45 | n9urk | CrashHD -> Thanks. It kept returning mpg123+faking+it for me. I appreciate the help |
| 19:55.01 | CrashHD | how would a tickless system or dynamic ticks affect asterisk? |
| 19:55.08 | CrashHD | in a kernel build? |
| 19:57.04 | LND | would've thought that not being able to clock 20ms or 30ms samples would be a bad thing ? |
| 19:58.46 | n9urk | [TK]D-Fender -> You told me to look at the book about MOHMP3, which advice appears to be depricated |
| 20:00.57 | [TK]D-Fender | n9urk: P.301 begs to differ <- |
| 20:01.24 | n9urk | [TK]D-Fender: looking now |
| 20:03.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:03.35 | n9urk | [TK]D-Fender: isnt the SOX stuff depricated as well? |
| 20:03.59 | *** join/#asterisk FuturePrimitive (n=stephen@c-24-7-186-196.hsd1.ca.comcast.net) |
| 20:04.19 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 20:04.27 | FuturePrimitive | Hey Everyone - Using FreePBX 2.4 with Asterisk 1.4.21 on CentOS 5.2. We have created several queues, and currently to log on to the queues my staff have to type queuenumber* and then it asks additional questions such as the extension. Is there an easy way to make it so the user just dials the queue number they want to login to and it either logs in or logs out? |
| 20:04.57 | grandpapadot | Has anyone ever seen an issue where asterisk appears to process two inbound channels (same channel identifier). The console will show two occurances of an call being processed such as outlined here: http://pastebin.com/m5ed9b341 |
| 20:05.21 | grandpapadot | Could this just be duplicate console chatter? |
| 20:05.46 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 20:06.06 | [TK]D-Fender | n9urk: How can SOX be deprecated... its not PART of asterisk. |
| 20:06.26 | [TK]D-Fender | FuturePrimitive: FreePBX is NOT supported here |
| 20:06.30 | [TK]D-Fender | ~freepbx |
| 20:06.31 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:06.51 | FuturePrimitive | This is more of an asterisk question. I was just laying out what I had. |
| 20:07.46 | FuturePrimitive | I was hoping someone might have a sample dialplan they use. I can figure out how to make it work in my setup. Just need something to get me kickstarted. |
| 20:08.04 | FuturePrimitive | I am in the FreePBX room as well, they are just a bit too quiet. :) |
| 20:08.18 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279395000.dsl.bell.ca) |
| 20:08.23 | [TK]D-Fender | FuturePrimitive: Read the instructions for the apps used to log them in/out |
| 20:08.34 | Corydon76-dig | FuturePrimitive: that should tell you something about supportability |
| 20:08.42 | [TK]D-Fender | grandpapadot: pastebin your dialplan. |
| 20:08.52 | mandd | any info on how to get asterisk to forward calls onto another phone number |
| 20:09.05 | [TK]D-Fender | mandd: * doesn't "forward" anything |
| 20:09.09 | n9urk | [TK]D-Fender: ok, so now I guess I am confused. How then does p.301 play mp3s natively in Asterisk? I need them to play MP3 just as we had in asterisk1.2.x and not convert them each time |
| 20:09.21 | mandd | redirect ? |
| 20:09.24 | [TK]D-Fender | mandd: and a "number" is not a self-encompassing "thing" either. |
| 20:09.51 | [TK]D-Fender | mandd: Asterisk DIALS. Don't go giving DIAL (app_dial.so" any more magical meaning. |
| 20:10.17 | CrashHD | hey tk would timing affect sip messages as well as rtp? such as qualify options sent? |
| 20:10.57 | grandpapadot | [TK]D-Fender: Hi TK -- It's hundreds of lines but for troubleshooting purposes I've simplified it to just the call coming in to the inbound context and the did as 2565551234,1,GoTo(someaccount|888|1) as you see in the pastbin (http://pastebin.com/m5ed9b341). There's only two lines, 888,1,Answer and 888,n,Wait(10). |
| 20:11.19 | grandpapadot | But both are processed twice as the console output in the pastebin implies... |
| 20:11.29 | grandpapadot | Not consistently, about 3/5 calls. |
| 20:11.29 | [TK]D-Fender | grandpapadot: please pastebin the DIALPLAN as requested |
| 20:11.39 | mandd | okay, a call comes in, to asterisk, it waits for a user to answer, and if there is no answer, asterisk then transfers a call to a cell phone |
| 20:11.55 | n9urk | mandd |
| 20:11.57 | [TK]D-Fender | mandd: then DIAL IT like you would any other call. |
| 20:12.11 | mandd | oh, and just put in the number? |
| 20:12.36 | [TK]D-Fender | mandd: Every call is jsut a call. Stop thinking its special because you want to reacht he PSTN. It isn't. |
| 20:12.42 | [TK]D-Fender | mandd: A call is a call is a call |
| 20:12.43 | grandpapadot | [TK]D-Fender: http://pastebin.com/m231f98c6 (thanks for the help) |
| 20:12.54 | mandd | ok |
| 20:13.02 | mandd | I understand, thanks [TK]D-Fender |
| 20:13.09 | n9urk | add a line saying something like this exten => 500,n,Dial(IAX2/user@iaxorsip/19999999999,30,r) |
| 20:13.39 | [TK]D-Fender | grandpapadot: Show me real dialplan with real CLI at max debug & verbose. |
| 20:14.04 | LemensTS | TK: whats the max debug at the cli |
| 20:14.10 | n9urk | [TK]D-Fender: did you see my last question to you? |
| 20:14.30 | tzafrir_laptop | Should there be any problem if some of the lines in an asterisk config file end with \r\n instead of \n ? |
| 20:14.32 | [TK]D-Fender | LemensTS: I think 4 is the functional max, but 10 sounds "important". Just ask George Carlin |
| 20:14.46 | [TK]D-Fender | tzafrir_laptop: dos2unix :) |
| 20:15.34 | tzafrir_laptop | [TK]D-Fender, I've edited a text file, removed some ^M-s, added a config directive, and problem solved |
| 20:15.35 | [TK]D-Fender | n9urk: * can play moh from any kind of file it has a FORMAT to support. for mp3 you need to have format_mp3 installed which comes from asterisk-addons |
| 20:15.53 | tzafrir_laptop | I know that the config directive did it |
| 20:16.08 | tzafrir_laptop | But I also wonder if the ^M-s had any effect |
| 20:16.13 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 20:17.57 | n9urk | [TK]D-Fender: You told me to read the book, and was kind enough to send the link (which I actually have read a couple times) and then when I said the book was depricated you said it was not. Am I missing something here? |
| 20:18.39 | [TK]D-Fender | n9urk: I've got it open here and everything in that chapter was accurate. |
| 20:19.27 | tzafrir_laptop | n9urk, but why would you actually want to play mp3 files? |
| 20:19.33 | outtolunc | all knowledge gained from reading documents for older versions of asterisk are not automatically shitcan material, the basic functions and most for the design and flow are still viable. |
| 20:20.11 | [TK]D-Fender | n9urk: Anyways, go install asterisk-addons like I've jsut told you. |
| 20:20.41 | n9urk | [TK]D-Fender: thanks for all of your help, it is always appreciated. You spend a lot of time in here, and i am always grateful for it. |
| 20:23.17 | *** join/#asterisk stoffell (n=stoffell@d51A4DC78.access.telenet.be) |
| 20:26.43 | *** join/#asterisk stoffell (n=stoffell@d51A4DC78.access.telenet.be) |
| 20:27.03 | [TK]D-Fender | heading home, BBIAB |
| 20:34.43 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 20:34.51 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:43.57 | mandd | is there a way to get voicemail.conf nofity SIP phone that there is a voicemail ? |
| 20:44.15 | mandd | or only email? |
| 20:44.33 | Strom_C | mandd: you have to set mailbox= in the sip phone's entry in sip.conf |
| 20:44.37 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:44.42 | mandd | aha, thanks! |
| 20:45.02 | Yourname`` | Ok, this is insane. 400 and 401 are new IP330 phones. They are not able to call each other. They're able to make outbound calls and it's fine, no deadair, etc. When they call each other, there's a 20 second dead air (according to the dialplan, ring for 20 min and then go to the next pririoy) http://pastebin.ca/1088851 Please help! |
| 20:46.18 | scooby2 | whats up with asterisk-1.4.21.2.tar.gz? gzip: asterisk-1.4.21.2.tar.gz: invalid compressed data--format violated |
| 20:46.58 | [TK]D-Fender | Yourname``: passtebin [general] from sip.conf and describe these phone's location |
| 20:48.27 | Yourname`` | one sec [TK]D-Fender , and what do you mean about the phone's location? They're behind a 3com NBX switch.. |
| 20:48.31 | fogo | Does voicemail use a dialplan like everything else? Where is that dialplan located? |
| 20:49.19 | Yourname`` | [TK]D-Fender: http://pastebin.ca/1088872 |
| 20:49.57 | [TK]D-Fender | fogo: No, Voicemail is an APPLICATION and its functioniing is hard coded in C |
| 20:50.27 | [TK]D-Fender | Yourname``: pastebin a failed call with SIP debug |
| 20:50.50 | fogo | [TK]D-Fender: I assumed as much, and that actually makes me happy ... the boss wanted me to make some changes to it; which I'm not sure I wanted to do :) |
| 20:52.38 | Yourname`` | oh man sip debugs, lol.. with no way of filtering :( |
| 20:52.58 | Yourname`` | [TK]D-Fender: I'm gonna have to say no to that for now... :( Remember the last time we tried to do one? lol |
| 20:53.28 | [TK]D-Fender | Yourname``: Then never ask why something fails in here again |
| 20:54.08 | [TK]D-Fender | Yourname``: Are these phones on the same LAN segment as *? |
| 20:56.10 | Yourname`` | [TK]D-Fender: no |
| 20:56.27 | Yourname`` | [TK]D-Fender: The * is offsite on a direct IP with no fw or iptables. |
| 20:56.52 | Yourname`` | [TK]D-Fender: and these phones are behind this 3com NBX thing. |
| 20:59.15 | [TK]D-Fender | Yourname``: ok, well I'm not touching this without backup |
| 21:02.43 | Yourname`` | [TK]D-Fender: You mean the sip debug? yeah.. I can give it to you like you ask for it, but I know what you're going to say |
| 21:02.45 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 21:03.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.09 | [TK]D-Fender | Yourname``: debug 1 IP |
| 21:05.09 | Yourname`` | [TK]D-Fender: What do you mean goodsir? |
| 21:05.31 | Yourname`` | Actually. |
| 21:05.43 | Yourname`` | I think I can do that, since there's only two-4 of there phones on that IP. |
| 21:07.37 | madison | anyone know of a good tutorial for followme recipes? |
| 21:09.36 | *** part/#asterisk FuturePrimitive (n=stephen@c-24-7-186-196.hsd1.ca.comcast.net) |
| 21:10.35 | [TK]D-Fender | madison: "core show applications" |
| 21:10.48 | [TK]D-Fender | madison: "core show functions" |
| 21:11.20 | tzafrir_laptop | core show application <tab><tab> |
| 21:12.03 | Yourname`` | [TK]D-Fender: Got some, formatting it 4u. |
| 21:15.50 | madison | ok |
| 21:16.01 | *** part/#asterisk puck (n=puck@leibniz.catalyst.net.nz) |
| 21:16.01 | Yourname`` | [TK]D-Fender: pastebin'd |
| 21:16.15 | madison | i am looking for a recipe for extensions.conf |
| 21:16.57 | angryuser | madison ha youchecked wiki ? |
| 21:17.12 | angryuser | ~wiki |
| 21:17.19 | angryuser | ~wikis |
| 21:17.20 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:17.57 | madison | basically what i want to do is have a call ring my extension, if i am not there, it plays a message and then transfers to my cell phone and allows me the option to accept or toss the call. if i choose to toss the calls, i want to send it to my voicemail |
| 21:19.57 | [TK]D-Fender | madison: "core show application dial" |
| 21:20.38 | madison | i did that |
| 21:21.00 | angryuser | madison it is simple and can be done with different ways, "core show application follow me" ;) |
| 21:21.43 | [TK]D-Fender | madison: M() option. |
| 21:36.05 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
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| 21:40.09 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:43.57 | *** part/#asterisk hoonteke (n=kevin@user-0c2hj2e.cable.mindspring.com) |
| 21:48.21 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584032.dsl.bell.ca) |
| 21:55.15 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 21:55.26 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
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| 21:56.46 | CrashHD | how do you setup a symmetric precache model? |
| 21:57.16 | AlexTO | Hi, i need a little help, i'm making a dialplan that catch like 14 digits and remove fisrt 3, but all the time I get error 603 (rejected) and looks like fine. http://pastebin.ca/1088942 |
| 21:58.14 | *** join/#asterisk AJayMN (i=4360bbfe@gateway/web/ajax/mibbit.com/x-346b1e28e867020c) |
| 21:58.32 | [TK]D-Fender | AlexTO: Jul 31 17:51:53 WARNING[28713]: pbx.c:1697 pbx_extension_helper: No application 'NoOp ' for extension (from- <----------------- |
| 21:58.37 | AJayMN | My /var/spool/asterisk/voicemail directory got messed up, so trying to fix it, i deleted it and trying to recreate the ext directories inside and getting an error.. |
| 21:58.45 | AJayMN | mkdir 203 |
| 21:58.57 | AJayMN | mkdir: cannot create directory `203': No such file or directory |
| 21:59.01 | [TK]D-Fender | AJayMN: don't create them by hand. Just leave a VM and * will recreate them itself |
| 21:59.03 | Qwell | AJayMN: You don't need to create them |
| 21:59.17 | Qwell | and it's because you deleted the directory underneath where you are. `cd .` |
| 22:00.42 | AlexTO | Ok let me check |
| 22:01.12 | AJayMN | The VM isnt creating them.. i screwed up the file table or somthing haha |
| 22:01.22 | CrashHD | I'm getting a no auth response from dundi peer in which I'm trying to precache with |
| 22:01.26 | AJayMN | I cant create any directory under /var/spool/asterisk/voicemail..... |
| 22:01.28 | CrashHD | any thoughts? |
| 22:01.47 | AJayMN | no wait.. i lied.. if its not an extension it makes the directory... |
| 22:02.18 | AJayMN | under /var/spool/asterisk/voicemail/ mkdir trial makes a trial directory... mkdir 203 Errors out with previous error |
| 22:03.18 | Qwell | again - `cd .` |
| 22:03.50 | *** join/#asterisk NickKoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 22:04.11 | AJayMN | Qwell type cd. then mkdir? |
| 22:04.13 | AJayMN | same issue |
| 22:04.48 | NickKoji | I have a simple pretty simple question. Do you receive information about the phone number you are dialing just like the phone that is receiving the call |
| 22:04.55 | NickKoji | Ei: Caller Id |
| 22:05.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:08.47 | AJayMN | QWell am i miss understanding you? |
| 22:08.52 | Qwell | yes |
| 22:10.04 | AJayMN | please explain hah |
| 22:10.56 | AJayMN | is it possible to change where the voicemail stores its files? Is there a conf file i could change to point it to a new directory? |
| 22:14.27 | *** part/#asterisk AJayMN (i=4360bbfe@gateway/web/ajax/mibbit.com/x-346b1e28e867020c) |
| 22:14.33 | *** join/#asterisk AJayMN (i=4360bbfe@gateway/web/ajax/mibbit.com/x-346b1e28e867020c) |
| 22:17.12 | AJayMN | [TK]D-Fender is there a place i can change where voicemail is stored? |
| 22:17.18 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 22:17.58 | AlexTO | [TK]D-fender, i remove some lines but i'm still getting disconnect http://pastebin.ca/1088962 |
| 22:18.15 | scooby2 | any way to switch a voice t1 between two machines remotely? |
| 22:18.53 | *** join/#asterisk anthm (n=anthm@99-205-196-249.area4.spcsdns.net) |
| 22:23.57 | AlexTO | http://pastebin.ca/1088962 |
| 22:34.23 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 22:42.57 | AJayMN | is it possible to move where asterisk saves the voicemail files? |
| 22:43.54 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) |
| 22:45.45 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 22:48.04 | mattwj2002 | hi guys |
| 22:48.12 | mattwj2002 | I have an Asterisk question |
| 22:48.25 | mattwj2002 | I am thinking about buying a Cisco IP phone |
| 22:48.47 | mattwj2002 | I was wondering how hard is it to hook it up to Asterisk? |
| 22:49.02 | LemensTS | Matt: as hard as any other phone |
| 22:49.12 | LemensTS | Most of us recommend polycom. |
| 22:49.52 | mattwj2002 | I don't need a conference phone |
| 22:49.53 | mattwj2002 | :) |
| 22:50.00 | mattwj2002 | but thanks for the idea |
| 22:50.50 | LemensTS | They have all types of phones |
| 22:51.42 | mattwj2002 | aren't polycoms now made by Cisco |
| 22:52.40 | JT | lol no |
| 22:52.55 | JT | it's the other way around |
| 22:53.01 | JT | polycom make some of cisco's phones |
| 22:53.02 | *** join/#asterisk [AMB] (n=amb@r190-135-5-11.dialup.adsl.anteldata.net.uy) |
| 22:53.15 | JT | the conference bridges from cisco are rebadged polycoms |
| 22:53.27 | LemensTS | So get a cisco conference phone :p |
| 22:53.39 | JT | mattwj2002: polycom is still the recommendation for handsets generally |
| 22:53.48 | JT | don't buy a cisco ip phone |
| 22:53.57 | JT | they're a pain |
| 22:54.17 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 22:54.20 | mattwj2002 | why do you say that JT? |
| 22:54.51 | JT | because cisco up until recently has been focusing on the SCCP firmware to work with their CCM instead of SIP |
| 22:55.03 | JT | and you need to pay for firmware licenses on ciscos |
| 22:55.12 | JT | the polycom sip firmware has more features |
| 22:56.03 | mattwj2002 | anyone know how much a Cisco firmware license cost? |
| 22:56.05 | JT | they're also heaps cheaper |
| 22:56.53 | scooby2 | $25 per phone per year when I looked |
| 22:57.48 | mattwj2002 | hang |
| 22:57.52 | mattwj2002 | *dang |
| 22:58.37 | JT | why are you so preoccupied over cisco phones if i may ask? |
| 22:58.53 | mattwj2002 | just because that is what I work with at work |
| 22:59.42 | scooby2 | their older phones appear to be cheaper. around $8 or $9 per year |
| 22:59.47 | plik | a cisco system at work will be different to a cisco phone on asterisk anyway |
| 22:59.51 | scooby2 | per phone |
| 23:00.04 | JT | is this for a business system or personal mattwj2002 ? |
| 23:00.14 | mattwj2002 | personal |
| 23:00.45 | JT | you could always use dodgy firmware yourself... but i think polycoms are nicer anyway |
| 23:00.47 | mattwj2002 | personal asterisk system |
| 23:01.02 | mattwj2002 | these polycoms have me interested |
| 23:01.07 | mattwj2002 | you have a favor model? |
| 23:01.44 | JT | the cheapest models (around USD$90) need PoE or a PSU, so you have to work out if it's more economical to get a model that comes with a PSU |
| 23:01.51 | JT | the IP320 is the cheapest |
| 23:05.56 | hardwire | does 1.4.x stable support Monitor and ChanSpy on the same channel yet? |
| 23:06.13 | *** join/#asterisk iamthelostboy (n=chatzill@203-110-29-111.nzwireless.co.nz) |
| 23:06.20 | hardwire | doesn't mind tailing a monitor file to get the playback, if needed. |
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| 23:31.51 | AJayMN | Can you configure asterisk to write the voicemail files to a different directory? |
| 23:32.18 | *** part/#asterisk Mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:35.21 | *** part/#asterisk c4t3l (n=rcallico@mail.questia.com) |
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| 23:39.50 | xacatecas | hi guys, is there any way to given an audio clip, originate a call with asterisk and then when the call gets answered play back that clip? |
| 23:40.22 | scooby2 | AJayMN: of course |
| 23:40.36 | *** join/#asterisk Guest9877 (n=root@189.141.98.158) |
| 23:42.58 | Guest9877 | Hello guys, do you know if asterisk can send customized SIP INFO messages? |
| 23:45.26 | xacatecas | Guest9877, knowing asterisk there probably is. however, i'd have no idea where to start looking. good luck :). |
| 23:45.30 | xacatecas | i'm off to bed. |
| 23:47.01 | AJayMN | scooby2 what conf file is it? im having issues with the /var/spool/asterisk/voicemail directory.. cant delete or make any new directories inside it |
| 23:49.07 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 23:54.33 | Guest9877 | do you know if asterisk can send customized SIP INFO messages? |
| 23:54.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 23:57.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
