| 00:57.15 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 00:57.24 | dandate2 | does agent queue weight penalties work in conjunction with auto-fill? |
| 01:18.06 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 01:18.38 | dijib | hello all |
| 01:37.28 | dijib | j'y'all not around ou what? |
| 01:38.12 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 01:44.44 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:26.29 | SeRi|afk | dijib, you got the reseller stuff sort it out? |
| 02:38.16 | *** join/#asterisk f2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 02:39.39 | f2Knight | Q: Just wondering how others have configured users... I am wondering how others assign usernames... do you assign them as there phone number? something else? |
| 02:40.29 | f2Knight | How about your extension paterns? Do you prefix the number dialed with the users ID? or some other method to help prevent toll fraud for example. |
| 02:44.15 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 02:46.11 | f2Knight | Q: Another question... I already know that you can do a Dial... followed by another Dial... to supply a failover effect. But how does anyone here impliment this for say a more dynamic way. |
| 02:46.41 | f2Knight | e.g. Lets say you have 20 trunks, that you can all out over.. do you list them all one after another? |
| 02:47.17 | f2Knight | better yet.. lets say you keep a database of call costs, how do you perform your LCR logic? |
| 02:48.42 | f2Knight | I would think it best to run a query against your database to get the records then try each in the result. but the ael and .conf makes that a little hard to accomplish, and running your Dial from an AEL seems a resource waste |
| 02:57.33 | p3nguin | User names? I don't have user names. |
| 02:57.44 | f2Knight | p3nguin, when you have a sip phone... |
| 02:57.56 | p3nguin | ~devicenames |
| 02:57.57 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 02:58.28 | p3nguin | My phones are given peer names equal to the MAC address. |
| 02:58.29 | f2Knight | p3nguin, okay devicenames... when assigning devicenames. |
| 02:58.53 | p3nguin | If it's a PC with a soft phone, I use the MAC address of the primary adapter. |
| 02:59.08 | p3nguin | Most devices only have one network device, though, so it's easy. |
| 02:59.29 | f2Knight | okay .. so ... when an outside caller calls in.. you map them to there mac address... |
| 02:59.35 | p3nguin | No. |
| 02:59.40 | p3nguin | Calls go to extensions. |
| 03:00.08 | p3nguin | If they dial 8005551212, the call goes to extension 8005551212. |
| 03:00.14 | f2Knight | p3nguin, okay call comes in it goes to extensions.conf or .ael, then there is a pattern that gets matched to do something with the call |
| 03:01.04 | f2Knight | so when the incoming call 8005551212 in your example needs to reach the 'device' 00:00:00:12:12:12 |
| 03:01.12 | p3nguin | If extension 8005551212 is a DID for a single person, then extension 8005551212 will eventually Dial() the device used by the person who uses that phone number. |
| 03:01.33 | f2Knight | p3nguin, what I am getting at is this |
| 03:01.47 | p3nguin | If it's a SIP device, exten => 8005551212,1,Dial(SIP/000000121212,36) |
| 03:01.57 | f2Knight | I have noticed some sip providers require you to 'prefix' your number dialed with your 'account code' |
| 03:02.29 | f2Knight | so what you dial might actually be 0123456_5555551212 |
| 03:02.30 | p3nguin | I've never heard such a thing. |
| 03:02.42 | p3nguin | But even if that's the case, it's trivial to add it. |
| 03:02.49 | f2Knight | Oh I know it is.. |
| 03:02.59 | f2Knight | I am just trying to understand the logic behind it is all |
| 03:03.12 | f2Knight | and if it serves a good reason to do so. |
| 03:03.37 | p3nguin | I've never heard of any provider requiring that. |
| 03:03.52 | f2Knight | the only reason I can see is to help prevent toll fraud, as if a remote attacker did gain access there would never be a patter match to make the call anyways |
| 03:04.26 | p3nguin | My providers simply require me to be registered first, plus require every call to auth when it is made. |
| 03:05.31 | f2Knight | Flowroute will require a prefix if you use ser or have your account right on the phone.. it works normal with a register statement on asterisk |
| 03:05.44 | f2Knight | just was wondering what the reason would be for was all. |
| 03:10.46 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 03:25.13 | phix | p3nguin: Should I ring that number? |
| 03:25.53 | f2Knight | phix, sure ring that number its toll free directory assistance :) |
| 03:33.47 | phix | :D |
| 03:33.52 | phix | What name should i ask for? |
| 03:34.40 | phix | Probably wont be toll free for me :( |
| 04:07.47 | *** join/#asterisk dandate2 (~dan@122.3.171.41) |
| 04:07.55 | dandate2 | http://forums.digium.com/viewtopic.php?t=74878 does anyone know if this issue was ever resolved? |
| 04:22.39 | phix | No idea, my Internets isn't working :( |
| 04:22.51 | phix | Only SSH and IRC seem to operate correctly |
| 04:22.58 | ChannelZ | I think the forums are a little hosed |
| 04:29.19 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:30.42 | dijib | summers over kids |
| 04:30.58 | dijib | p3nguin, f2knite isnt talking asterisk i guess |
| 04:31.58 | dijib | SeRi, no i didnt get it sorted. |
| 04:34.43 | phix | ah |
| 04:35.06 | phix | It is Spring now, Summer is comming up :D |
| 04:35.08 | phix | Another month |
| 04:35.18 | dijib | damn you |
| 04:35.31 | phix | 40C days! wooo! |
| 04:35.39 | dijib | in a month i will br driving in snow and its your fault |
| 04:35.39 | phix | Something to look forward to |
| 04:35.40 | phix | haha |
| 04:40.11 | dijib | australia? |
| 04:40.16 | dijib | melbourne? |
| 04:40.28 | dijib | nantuckit? |
| 04:40.40 | phix | Sydney |
| 04:40.58 | dijib | oh i just opened a frothy guiness stout. and my livers starting to hurt |
| 04:41.03 | phix | Melbourne doesn't get to 40C, it rains too much for that to happen |
| 04:41.09 | dijib | i need some asterisk-tainment |
| 04:41.10 | phix | hehe |
| 04:41.19 | dijib | ahk |
| 04:41.22 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-138-172.chyn.qwest.net) |
| 04:41.40 | dijib | ive got a buddy in sydney, and biotch or three in toowoomba. |
| 04:41.52 | phix | dijib: 100,1,Dial(SIP/phix@phix.net) |
| 04:42.00 | phix | nice |
| 04:42.18 | dijib | anybody else in there? |
| 04:42.31 | phix | nope, not a valid address either :P |
| 04:42.57 | dijib | oh lol technically its valid.. just no dns resolution |
| 04:43.11 | phix | yup :) |
| 04:43.18 | dijib | how do i do that with my asterisk. ive got dyndns |
| 04:43.29 | phix | and does meet your asterisk-tainment guideline :) |
| 04:43.52 | dijib | why can i have a dial(SIP/user@my.dydnsdomain.com); ? |
| 04:44.18 | phix | you can if you want to |
| 04:44.22 | dijib | i want to. |
| 04:44.47 | dijib | makes it easy to talk, better then typing... and i dont give that much of a eff about my bandwidth using SIP |
| 04:46.37 | *** join/#asterisk kerx_ (~kerx@li254-60.members.linode.com) |
| 04:46.45 | dijib | phix, can i PM yuou? |
| 04:47.04 | dijib | and do you have an asterisk server available to dial me |
| 04:47.38 | phix | sure |
| 04:47.47 | phix | umm not atm |
| 04:48.02 | phix | pm is fine :) |
| 04:49.24 | kerx_ | Hi all. On an outbound SIP call I set the caller ID name and number, however only the number is set. The name shows up as 'Unknown'. Am I missing something that is required to do on outbound SIP calls for Caller ID name? |
| 04:50.06 | phix | support for caller id from your SIP provider? |
| 04:51.43 | kerx_ | phix: Oh, I wasn't aware that a SIP provider needs support for outbound caller ID. |
| 04:52.04 | phix | They can choose to support it or not |
| 04:52.15 | phix | they dont have to forward on your callerid, they can get rid of it if they choose |
| 04:52.21 | kerx_ | Well, they set the caller ID number, but the name comes back as Unknown |
| 04:52.25 | dijib | kerx_, the answer to this is a little complex. but the long and short of it is. The CallerID(num) is the only thing sent. Telco's use a database to whcih i cant recall the name to get the name info in reference to the number |
| 04:53.03 | kerx_ | I see. So, there is a whole process beyond my Asterisk's boxes control |
| 04:53.07 | dijib | between SIP providers you have a chance of the CallerID(name) attribute sent aswell but dont expect it against Ma'Bell |
| 04:53.12 | phix | kerx_: correct |
| 04:53.38 | kerx_ | Ok, roger that. I'm sending a call from my provider to AT&T, so I guess that's considered Ma'Bell. Unfortunately. |
| 04:53.43 | dijib | not attribute, variable. |
| 04:53.47 | dijib | right? ? |
| 04:54.32 | kerx_ | I think that terminology depends on the context. |
| 04:54.39 | dijib | yes. now me and p3nguin had a conversation about this and he told me the DB name and i think there is a process you can follow to register with them.. possibly paid. |
| 04:54.43 | kerx_ | Who cares though. Attribute or Variable... Whatever |
| 04:54.54 | dijib | ill look for it |
| 04:58.17 | *** join/#asterisk radic (~radic@dslb-178-002-237-202.pools.arcor-ip.net) |
| 04:58.40 | dijib | not sure if this is it. http://ciddb.com/index.php |
| 04:59.32 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 04:59.51 | kerx_ | yea, that looks right |
| 05:00.03 | kerx_ | just ran a check on a number, and it comes back correct |
| 05:00.33 | p3nguin | Caller ID name is not something that is sent over the PSTN. |
| 05:00.50 | p3nguin | Number is sent, name it looked up. |
| 05:00.56 | p3nguin | s/it/is/ |
| 05:01.16 | kerx_ | is there a single registry that maintains this database? |
| 05:01.26 | kerx_ | i remember hearing a lot of things about DIPs |
| 05:01.58 | p3nguin | There isn't one single LiDB, no. |
| 05:02.10 | p3nguin | Many telcos have their own LiDB. |
| 05:02.31 | p3nguin | There are some general ones that companies do use, though. |
| 05:02.56 | kerx_ | So, for example, a CLEC gets a call from a number owned by AT&T. How do they know they need to query AT&T's LiDB, and how do they query AT&T's LiDB? |
| 05:03.16 | kerx_ | Also, is there a way to tap into this database in both a Read/Write method? |
| 05:03.20 | p3nguin | They query their own LiDB, or whatever one they are subscribed to. |
| 05:03.55 | p3nguin | It is possible to update the CNAM databases of some carriers. |
| 05:04.08 | p3nguin | And you can subscribe to some. |
| 05:04.31 | p3nguin | I'm not active in that level of operations, so I can't say just exactly how you'd update one. |
| 05:05.21 | p3nguin | Many ITSPs, for example, will offer the updating of CNAM information to the customer... if the carrier of the DID supports it. |
| 05:05.39 | p3nguin | I figured it's something that a CLEC or ILEC can do. |
| 05:06.50 | dijib | LiDB i was busy looking unsucecsfilly |
| 05:08.15 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 05:08.37 | p3nguin | You may also be interested in SS7. |
| 05:10.59 | p3nguin | I guess the major databases may be networked together. That would sure be useful if they are. |
| 05:17.35 | p3nguin | You can get access to a CNAM database and pay per dip. It isn't all that expensive unless you're doing a ton of lookups. |
| 05:18.09 | p3nguin | I think you'll pay like 0.8 cents per dip or something. |
| 05:27.10 | *** join/#asterisk CaptWho (Capt@unaffiliated/captwho) |
| 05:28.26 | CaptWho | i'm configuring a private phone system and i'd like to start all the numbers with a 2#nnnnnn. does anyone have any suggestions for that? |
| 05:30.56 | SeRi | I think I pay 88 cents last time for a month worth of cnam look ups. its cheap. |
| 05:31.40 | SeRi | p3nguin, I am hoping to get everything in place next week. |
| 05:32.41 | *** join/#asterisk serafie (~erin@nat/digium/x-wriptwiwzxfudjbo) |
| 05:33.22 | WIMPy | CaptWho: The # may get you in trouble on many sip devices. Otherwise: Just do it. |
| 05:33.49 | [TK]D-Fender | CaptWho: Start them with 2#nnnnnn |
| 05:53.16 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-203.telkomadsl.co.za) |
| 06:08.56 | dijib | im hoping to sober upp |
| 06:09.40 | dijib | how do i make my own SIP/user@domain.com using dyndns |
| 06:13.04 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:44.21 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 06:44.34 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 06:44.35 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:44.43 | dandate2 | where can i adjust the length of time a member is paused for by autopause=yes ? what is the default anyway or how do they become unpaused? |
| 07:27.14 | *** join/#asterisk SparFux (~raoul@rl2-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 07:40.08 | *** join/#asterisk irroot (~irroot@41-132-57-229.dsl.mweb.co.za) |
| 07:41.31 | SparFux | hi irroot |
| 07:42.07 | irroot | SparFux hi there how the work on HFC bits coming ? |
| 07:42.26 | SparFux | irroot: great. I renamed the sf.net project :-) |
| 07:42.34 | SparFux | It's dahdi-hfcs now. |
| 07:42.36 | irroot | cool you nailed it |
| 07:42.48 | irroot | i need to get the usb module in there ASAP |
| 07:42.54 | SparFux | I just created a new project and put the git on. |
| 07:43.13 | irroot | i keep a mISDN v1 tree alive now running on linux 3.1 |
| 07:43.14 | SparFux | What does usb do with it? |
| 07:43.34 | irroot | there is a USB hfc chipset we use extensivly |
| 07:44.26 | irroot | mISDN v2 does not seem to work with the b410 card so cant use that even though i ported lcr to v10 i dont use it |
| 07:44.57 | irroot | so if dhahdi_zap supports usb then it makes it easier |
| 07:44.58 | SparFux | which cards does misdn support which dahdi doesn't? |
| 07:45.12 | irroot | just the USB AFAIK |
| 07:45.20 | SparFux | there is a xpp_usb module for some digium stuff. |
| 07:45.30 | irroot | yeah that is xorcom |
| 07:45.37 | irroot | its a channel bank |
| 07:46.14 | irroot | been usb it should not be too complicated to port |
| 07:46.26 | irroot | but have not got there yet |
| 07:46.36 | SparFux | nope. not yet. |
| 07:46.56 | SparFux | you say it would be best to use xpp_usb to supppport hfc-s too? |
| 07:48.19 | SparFux | perhaps better just put the usb stuff into dahdi_hfcs |
| 07:52.18 | SparFux | cool: http://www.colognechip.com/hfc-s-usb.pdf |
| 07:54.24 | irroot | yip :P in the dahdi hfs project big diff to xorcom |
| 07:54.53 | irroot | the misdn driver should have all the good stuff |
| 07:55.22 | SparFux | you mean dahdi hfcs? |
| 07:55.32 | irroot | of course if you do it from the spec sheet you can contribute it to main dhahdi project it |
| 07:55.55 | irroot | is there a a hfcs driver ?? |
| 07:56.12 | irroot | mmm i must double check last i looked there was not |
| 07:57.19 | *** join/#asterisk KNERD (~KNERD@99.41.13.82) |
| 08:00.30 | SparFux | in the dahdi project there is not. that's the whole point of the dahdi_hfcs git tree I set up :-P |
| 08:04.43 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:38.56 | *** join/#asterisk nafg (~quassel@ool-4355e4a2.dyn.optonline.net) |
| 08:39.31 | nafg | Hi, I just attempted to install app_swift. When I do module load app_swift.so I get: |
| 08:39.44 | nafg | Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close |
| 08:39.48 | nafg | Any ideas? |
| 08:56.43 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:01.04 | irroot | swift libraries ?? |
| 09:13.54 | nafg | irroot: What? |
| 09:14.10 | irroot | libswift ?? only a guess |
| 09:15.50 | irroot | http://nerdvittles.com/index.php?p=202 |
| 09:16.29 | irroot | nafg ^^ may help |
| 09:16.50 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 09:18.27 | nafg | That's what I followed in the first place. |
| 09:42.03 | *** join/#asterisk jacobkiers (~jacobkier@82-168-168-85.ip.telfort.nl) |
| 09:57.07 | *** join/#asterisk aglenday (~Impatient@59.167.161.74) |
| 10:44.01 | *** part/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 11:16.27 | *** join/#asterisk WWGD (~WWGD@109.70.64.45) |
| 11:18.47 | *** part/#asterisk WWGD (~WWGD@109.70.64.45) |
| 11:20.57 | *** join/#asterisk WWGD (~WWGD@109.70.64.45) |
| 12:05.37 | *** part/#asterisk WWGD (~WWGD@109.70.64.45) |
| 12:14.10 | *** join/#asterisk rethus (~suther@p50879721.dip.t-dialin.net) |
| 12:14.32 | rethus | can i change actual moh song live in cli? |
| 12:18.38 | rethus | how can i increment channal volume in cli ? |
| 12:22.40 | *** join/#asterisk wonderworld (~ww@port-92-201-57-158.dynamic.qsc.de) |
| 12:23.26 | wonderworld | hi, http://www.digium.com/en/mediacenter/viewpress/digium-and-open-source-community-release-asterisk-10-at-astricon states, that asterisk 10 is released, but i can only find beta2 on the ftps? |
| 12:32.47 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 12:39.32 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-094-216-160-070.pools.arcor-ip.net) |
| 12:42.47 | irroot | wonderworld the svn ?? |
| 12:46.29 | irroot | wonderworld nope not packaged yet |
| 12:46.53 | wonderworld | i see, thanks |
| 12:53.50 | *** part/#asterisk rethus (~suther@p50879721.dip.t-dialin.net) |
| 12:54.46 | irroot | wonderworld i have been using it for a while now from branches/10 that will be it when packaged |
| 13:03.06 | *** join/#asterisk krotos (~d3v1l@b2b-client.as48500.net) |
| 13:03.17 | krotos | hi channel :) |
| 13:07.29 | *** join/#asterisk fipu (~fipu@fipu.usr.sh) |
| 13:11.43 | *** join/#asterisk TimeRider (steve@5ad51264.bb.sky.com) |
| 13:12.39 | wonderworld | irroot: i will pull it now. did you have any unexpected problems? |
| 13:13.26 | irroot | nope not at this point i have not used it in heavy duty enviroments but the smaller sites < 20 extensions i have put in at are happy |
| 13:13.44 | irroot | some of the features i have backported to 1.8 |
| 13:13.54 | irroot | so have tested them well |
| 13:14.36 | wonderworld | nice, thanks. i will give it a try for a smaller site as well. really looking forward to test out confbridge |
| 13:16.34 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 13:17.10 | irroot | wonderworld hehe i have not played with all the toys yet |
| 13:17.54 | irroot | wonderworld if you have a ATA with T38 and a TDM line you will find faxing works better |
| 13:18.24 | wonderworld | i gave up on faxing a long time ago |
| 13:18.34 | irroot | lol dont blame you |
| 13:18.56 | wonderworld | i even stopped using pci hardware |
| 13:19.00 | irroot | we have big demand for it still |
| 13:19.01 | wonderworld | i go for media gateways all the time |
| 13:19.04 | wonderworld | hassle free |
| 13:19.16 | irroot | if you have decent bandwidth |
| 13:19.35 | irroot | not so here |
| 13:19.55 | wonderworld | nope. ISDN -> Meda Gateway -> SIP -> Asterisk |
| 13:20.04 | wonderworld | no need for decent bandwidth |
| 13:20.05 | irroot | ah ok |
| 13:20.14 | irroot | yeah |
| 13:20.18 | wonderworld | just a blackbox to get an always working SIP-channel |
| 13:20.31 | irroot | paton/mediacodes |
| 13:20.35 | irroot | know em |
| 13:20.38 | wonderworld | yeah |
| 13:20.51 | wonderworld | euro-isdn drivers for most pci-hardware are horrible |
| 13:21.39 | irroot | i use mISDN v1 still keep it going for new kernels currently on 3.1 |
| 13:22.08 | irroot | big user of USB ISDN interfaces for small sites with 1/2 BRI |
| 13:22.30 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 13:24.19 | wonderworld | maybe i'll retry misdn. pattons are a real cost raiser. but they provide peace of mind |
| 13:26.08 | wonderworld | nice -> Connected to Asterisk SVN-branch-10-r342715 |
| 13:26.16 | wonderworld | hope it won't explode :) |
| 13:27.54 | WIMPy | The old mISDN has been ok for me if used in TE mode only. |
| 13:29.01 | WIMPy | With the new one the only issue I see are the module usage counts, but they don't matter in use. |
| 13:29.52 | wonderworld | i remember an old setup of mine with mISDN which gave me downtimes at random once a month. i never really found out why. |
| 13:29.56 | wonderworld | guess i am just too lazy :) |
| 13:43.39 | irroot | wonderworld if it blows up shout we need to iorn out things ASAP |
| 13:43.47 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 13:44.28 | irroot | the problem with any new release its mostly only develeopers running it and it sometimes needs to get some legs that said the testing and test "suite" has improved |
| 13:53.10 | irroot | WIMpy yeah noticed that moduse problem its a problem if you want to reload |
| 13:57.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 13:57.29 | *** join/#asterisk Vince-0 (~AndChat@41-132-156-89.dsl.mweb.co.za) |
| 14:02.35 | WIMPy | Yes, replacing the modules will usually require a rmmod -f. |
| 14:06.02 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:09.55 | *** join/#asterisk master_of_master (~master_of@p57B52BA8.dip.t-dialin.net) |
| 14:25.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:30.37 | *** join/#asterisk Foxi352 (~Foxi352@ppp-104-254.adsl.restena.lu) |
| 14:55.01 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 15:01.32 | *** join/#asterisk Heishiro (~mitsurugi@190.77.88.124) |
| 15:04.02 | Heishiro | Hello everyone. One quick question. If i have a TDM400P which one should i use? Zaptel or DAHDI? I'm very confused on this point. According to elastix without tears i have to modify zapata.conf, and other files, but when i look for those files in my asterisk now installation, everything is in dahdi. My problem is that i cannot dial from or receive calls in my FXO's. So not sure if the problem |
| 15:04.02 | Heishiro | is the zaptel/dahdi thingy... Help anyone? |
| 15:09.36 | WIMPy | Dahdi is the new name for zaptel. |
| 15:09.39 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 15:14.08 | Heishiro | So it's the same at this point... Then i have a problem here configuring the pbx. I could create a sip trunk, and i can dial from it using a software sip phone, but there's no way to dial using my FXO.. Keep reading, i guess.. Thanks for your answer WIMPy |
| 15:15.09 | WIMPy | If you're using Elastix, you should check #elastix. |
| 15:15.51 | Heishiro | No, i'm not on Elastix. I'm on Asterisk Now with freepbx. |
| 15:16.09 | WIMPy | Ok, then #freepbx. |
| 15:16.31 | Heishiro | ok. I'll try that one. Thanks a lot. |
| 15:41.11 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 15:49.45 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-203.telkomadsl.co.za) |
| 15:51.13 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 16:03.00 | depressed | Hello all |
| 16:10.01 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 16:13.24 | depressed | I have an account with pingdom.com and was wondering what options should I select to check the uptime through the SIP port? |
| 16:20.54 | [TK]D-Fender | ? |
| 16:21.38 | [TK]D-Fender | There is no "uptime" * SIP is UDP by default and is stateless |
| 16:21.56 | [TK]D-Fender | What you've got is qualify time on a peer and that's about it |
| 16:26.38 | depressed | I see. There are options for string to send and string to expect - is there nothing I can specify for this? |
| 16:28.24 | WIMPy | If it's long enough, you might be able to put a whole SIP oprions message there. |
| 16:28.47 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 16:28.50 | depressed | WIMPy ok I will try that, thank you |
| 16:29.12 | WIMPy | But unless you know exactely what you're doing, you should use something that it up to your task. |
| 16:40.50 | WIMPy | ebay is fascinating. 30,50 for an octobri and a minute later 36,09 for a quadbri. |
| 16:52.55 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:55.45 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 16:57.23 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:05.23 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 17:09.22 | *** join/#asterisk joat (~joat@ip70-160-216-251.hr.hr.cox.net) |
| 17:12.16 | *** join/#asterisk MiserySoft (~LeeD@host81-148-65-181.in-addr.btopenworld.com) |
| 17:16.19 | p3nguin | Is there any way to increase the amplitude of voice in a phone call (a SIP device, calling outbound). |
| 17:16.22 | p3nguin | s/./?/ |
| 17:19.15 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:20.19 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 17:20.45 | dijib | hey all... how do i make my own Dial(SIP/user@mydomain.com) ??? |
| 17:20.52 | dijib | i want to make an IP Confrence line |
| 17:22.20 | p3nguin | You want to accept calls via SIP URI. |
| 17:22.29 | p3nguin | And the call will go to a conference. |
| 17:22.37 | WIMPy | p3nguin: The phones settings? Or 'core show function VOLUME'. |
| 17:22.41 | dijib | yes |
| 17:23.02 | p3nguin | I'll tell you how, and even write the dial plan... but you have to copy what I write and use it rather than fucking with it and then asking why it doesn't work. |
| 17:23.25 | dijib | hahahahaha |
| 17:23.27 | dijib | your too funny. |
| 17:23.31 | p3nguin | you're |
| 17:23.34 | p3nguin | not your |
| 17:23.38 | dijib | you are |
| 17:23.41 | WIMPy | Ne. He's got experience. |
| 17:23.41 | dijib | j'esus |
| 17:23.49 | p3nguin | Right, you are... you're. |
| 17:23.57 | dijib | p3nguin, only if you join the confrence. |
| 17:24.02 | p3nguin | I might do that. |
| 17:24.18 | dijib | ok then... see if we cant get something #asterisk -confrence happening. |
| 17:24.27 | dijib | how much bandwidth could it potentially use? |
| 17:24.33 | dijib | if say 10ppl are in it? |
| 17:24.39 | p3nguin | Potentionally? All of it. |
| 17:24.48 | p3nguin | err, potentially |
| 17:24.49 | dijib | would it kill my 700kbps upstream |
| 17:25.10 | dijib | could i limit users to 6? |
| 17:25.13 | p3nguin | There is always potential to destroy 700 kbps upstream. |
| 17:25.26 | dijib | well lets see if we can mess it up |
| 17:25.29 | p3nguin | I think 6 would be good. |
| 17:26.06 | p3nguin | I need to gather a couple pieces of information from you. |
| 17:26.11 | dijib | shoot |
| 17:26.14 | dijib | dyndns? |
| 17:26.22 | WIMPy | I think 8 should fit IF 1. it is not used by anything else or 2. has working traffic control. |
| 17:26.25 | dijib | i might make a second that i can disconnect easily |
| 17:26.25 | p3nguin | In sip.conf, you have a context in the general section. What context is it? |
| 17:26.39 | WIMPy | Or better both. |
| 17:26.50 | dijib | yes has a working traffic controller, but is also used by other users on the home network |
| 17:27.32 | WIMPy | Not any more then. |
| 17:27.47 | p3nguin | I'm going to write the dial plan to incorporate a max count of 6. |
| 17:27.48 | dijib | what context? i have general voipms |
| 17:28.06 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 17:28.28 | p3nguin | I'll start over. There is a section entitled [general], and there is a context in that section... before you ever get to your voipms peer entry. What context is it? |
| 17:28.46 | p3nguin | Often 'default' or in my case, 'misc_calls'. |
| 17:29.37 | dijib | none defined |
| 17:29.59 | p3nguin | Somebody still hasn't read the book. |
| 17:30.12 | dijib | dude... do u want ssh access? |
| 17:31.11 | dijib | http://pastebin.com/3fEgC5Ji |
| 17:31.28 | dijib | under this context its the user accounts then voipms context. |
| 17:31.40 | p3nguin | You're missing the context. |
| 17:31.46 | p3nguin | You should fix that soon. |
| 17:31.56 | p3nguin | I prefer to not use 'default' for the context. |
| 17:32.23 | p3nguin | I like something else that reflects what the calls are. In my case, they are miscellaneous calls, so the context I use is misc_calls. |
| 17:32.24 | dijib | so what? |
| 17:33.05 | dijib | ok now |
| 17:33.05 | dijib | context=misc_calls |
| 17:33.08 | dijib | in general |
| 17:33.19 | p3nguin | Okay, now we go to extensions.conf. |
| 17:33.32 | WIMPy | Yes, you might need "default" to contain something else. |
| 17:33.35 | dijib | k find the mic_calls context.. i think i have you wrote. |
| 17:33.58 | p3nguin | Create a context [default] if it does not exists, and create [misc_calls] which we will use. |
| 17:34.09 | dijib | http://pastebin.com/cDvCrb76 |
| 17:34.29 | p3nguin | You crashed the pastebin. |
| 17:34.44 | dijib | there is default context... add it? |
| 17:34.53 | p3nguin | If there IS one, you don't need to ADD one. |
| 17:35.16 | dijib | ok nevermind i do have both default and misc_calls context |
| 17:35.19 | dijib | s |
| 17:35.19 | p3nguin | If there isn't one already, create it. |
| 17:35.20 | p3nguin | Good. |
| 17:36.07 | p3nguin | Now, in the misc_calls context, you'll create the extension to accept calls to the conf. You want your conference SIP URI to be 'conf@yourhost.dyndns.com'? |
| 17:36.14 | dijib | so then do i just define like asterisk extension so that asterisk@mydyndns.com |
| 17:36.30 | dijib | lets make it asterisk |
| 17:36.42 | p3nguin | I don't want to actually process the conference in that context. We'll use a Goto(). |
| 17:36.53 | dijib | ok |
| 17:36.57 | dijib | ? |
| 17:37.17 | WIMPy | I'ts worth noting, that "asterisk" is the default voicemail extension. |
| 17:37.22 | p3nguin | I personally don't recommend using asterisk. |
| 17:37.36 | p3nguin | I would use 2663 or conf, or both. |
| 17:37.50 | p3nguin | or something completely arbitrary. |
| 17:38.03 | p3nguin | or something you change periodically. |
| 17:39.56 | dijib | go with 2663 then |
| 17:40.07 | dijib | im thinking the dyndns is going to be asterisk.serveirc.com |
| 17:40.29 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 17:40.56 | p3nguin | In the misc_calls context, create a new extension. exten => 2663,1,Goto(conference,${EXTEN},1); |
| 17:41.10 | p3nguin | Then create a new context: [conference] |
| 17:42.24 | dijib | then define 2663 with a confbridge in the confrence context? |
| 17:42.41 | dijib | and also include confrence context in the misc_calls context? |
| 17:43.19 | p3nguin | http://pastebin.com/ChjEJ8Ei |
| 17:43.35 | p3nguin | Only do what I said. Don't add things like includes. |
| 17:45.09 | p3nguin | Once you have the Goto in the misc_calls context, and your conference context looks like mine, save, exit. Then sip reload and dialplan reload. |
| 17:45.53 | p3nguin | Now your conference SIP URI is the extension you created in misc_calls @ your host name. |
| 17:45.58 | dijib | http://pastebin.com/NpR15wTS |
| 17:46.09 | p3nguin | such as 2663@dijib.dyndns.com |
| 17:47.41 | p3nguin | Give me the real host name and I'll call it to test. |
| 17:48.05 | dijib | asterisk.serveirc.com should work |
| 17:48.19 | p3nguin | You already pointed it at your system? |
| 17:48.52 | p3nguin | I guess so. It resolves. |
| 17:49.12 | dijib | yeah its working |
| 17:49.36 | dijib | yeah i have that dyndns pointing to my other dyndns. |
| 17:49.44 | dijib | both url's work from my tests |
| 17:50.08 | dijib | how do i directly dial 2663 from my phones context? include? |
| 17:50.59 | p3nguin | You could either "include => conference" in your phones context, or make another goto in your phones context exactly the same as the one you put in misc_calls. |
| 17:51.27 | dijib | ok |
| 17:52.20 | dijib | k so SIP/2663@asterisk.serveirc.com |
| 17:52.28 | dijib | should get you in/ |
| 17:52.36 | p3nguin | Your SIP URI is 2663@asterisk.serveirc.com |
| 17:52.44 | dijib | you going to test? |
| 17:52.52 | p3nguin | It can also be expressed as sip:2663@asterisk.serveirc.com |
| 17:53.00 | p3nguin | In a minute I will. |
| 17:53.26 | dijib | so at 6 users this will show as congested to new users eh? |
| 17:53.31 | p3nguin | Yes. |
| 17:53.39 | p3nguin | Adjust as needed. |
| 17:53.40 | dijib | sweet |
| 17:53.53 | dijib | then #asterisk... use this as a conf |
| 17:54.09 | dijib | incomming |
| 17:54.11 | dijib | i see you in |
| 17:54.59 | dijib | im stepping out for a smoke |
| 17:55.04 | dijib | crap you dropped eh |
| 17:55.50 | p3nguin | It said I was the only person in the conference, then played music to me. I pressed * to get the menu, but it didn't tell me my options. |
| 17:56.16 | dijib | no? |
| 17:56.27 | dijib | i joined after i saw you in it... then i hear a dtmf tone |
| 17:56.38 | dijib | i just changed the default MOH class |
| 17:56.39 | dijib | also |
| 17:57.09 | p3nguin | I pressed *, and then I guess you pressed something too. |
| 17:57.14 | dijib | how would i dial that sip uri from x-lite |
| 17:57.32 | dijib | # is for options, not * |
| 17:57.44 | dijib | i pressed 1. |
| 17:57.47 | dijib | but yes i did |
| 17:58.11 | p3nguin | <PROTECTED> |
| 17:58.19 | p3nguin | hmm |
| 17:58.24 | p3nguin | <PROTECTED> |
| 17:58.27 | p3nguin | <PROTECTED> |
| 17:58.29 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 17:58.29 | dijib | nope # |
| 17:58.32 | dijib | i just tried |
| 17:58.49 | p3nguin | So you're saying the author is wrong? |
| 17:58.58 | dijib | or it was updated or something yes |
| 17:59.08 | dijib | meet me in there and try it... . |
| 17:59.12 | dijib | talk to me while i smoke |
| 17:59.13 | dijib | ;) |
| 17:59.35 | dijib | see. # |
| 17:59.51 | dijib | k im joining |
| 17:59.55 | *** join/#asterisk irroot (~irroot@41-135-187-42.dsl.mweb.co.za) |
| 18:00.15 | *** part/#asterisk irroot (~irroot@41-135-187-42.dsl.mweb.co.za) |
| 18:43.17 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 18:44.03 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 19:02.17 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 19:02.19 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 19:02.27 | p3nguin | FLOOD! |
| 19:02.38 | p3nguin | I got config reload ... to actually show something. |
| 19:03.07 | dijib | how? |
| 19:03.20 | dijib | my liver hurts |
| 19:03.28 | p3nguin | I changed moh conf and reloaded it using config reload, and it spewed all sorts of stuff about sending SIGHUP to MOH processes. |
| 19:03.30 | *** join/#asterisk irroot (~irroot@197.108.239.38) |
| 19:03.53 | p3nguin | irroot will join the conf. |
| 19:04.16 | irroot | ?? |
| 19:04.22 | irroot | what where when |
| 19:04.24 | dijib | :D |
| 19:04.28 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 19:04.40 | dijib | now, |
| 19:04.55 | irroot | id love too but bw sux |
| 19:05.04 | irroot | cant call over this shitty 3G |
| 19:05.05 | dijib | bandwidth? |
| 19:05.10 | dijib | ahh |
| 19:05.13 | irroot | yeah |
| 19:05.36 | dijib | shouldnt 3g have plenty of bandwidth? |
| 19:05.44 | dijib | and how to dial that through X-Lite? |
| 19:06.25 | irroot | yeah got 7.2Mbs to tower then i may have 9k6 to out the country |
| 19:06.43 | irroot | also having a thunderstorm ATM could be 300bps :P |
| 19:07.03 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:07.17 | irroot | VOIP is actually shaped by the providers so its pissing into a hurricaine |
| 19:07.58 | p3nguin | sip:2663@asterisk.serveirc.com |
| 19:08.20 | p3nguin | If 2663@asterisk.serveirc.com doesn't work, prefix it with sip: |
| 19:08.45 | dijib | cant even do the @asterisk.serveirc.com |
| 19:08.54 | dijib | part... only numbers in X-Lite |
| 19:10.11 | p3nguin | If you enable the setting that allows you to call without being registered to anything, then will it work? |
| 19:10.34 | irroot | have changed the whole way locking works in app_queue |
| 19:10.41 | dijib | i dont even know of the setting but i would assume no, as there would be nothing to handle the call? |
| 19:10.45 | irroot | split queues / members |
| 19:11.35 | irroot | should improve performance substantially the queues were been locked while the members were traversed no longer the case |
| 19:12.21 | p3nguin | The user agent is all that is needed to handle the call. |
| 19:12.27 | p3nguin | That's what user agents (phones) are for. |
| 19:12.38 | p3nguin | And that's the nice thing about a SIP URI. |
| 19:13.49 | p3nguin | I don't know where the setting is, but there is one that allows making calls without being registered. |
| 19:14.12 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 19:15.15 | p3nguin | There is also a work-around using the phone book. |
| 19:16.21 | p3nguin | But if you're registered to your own asterisk, you don't need to change the setting for dialing while not registered. Just enter the SIP URI -- toggle the alpha-numeric keypad instead of only numeric keypad. |
| 19:16.39 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-138-172.chyn.qwest.net) |
| 19:16.41 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:17.11 | p3nguin | Dialing while not registered it normally used for someone like me who want to just dial some random SIP URI from any computer I can get my hands on without entering my own asterisk credentials. |
| 19:17.26 | dijib | i dont even see an alphanumeric dialpad |
| 19:17.41 | p3nguin | I think you just have to press the space bar. |
| 19:18.07 | p3nguin | (when your cursor is in the main input box) |
| 19:20.35 | p3nguin | By the way, the method I used to call your conf was: originate SCCP/001562FFFFFF-a application Dial SIP/2663@asterisk.serveirc.com |
| 19:22.21 | p3nguin | When my phone rang, I picked it up and it immediately called your SIP URI. |
| 19:28.58 | *** join/#asterisk covici (~user@pool-173-72-192-149.clppva.fios.verizon.net) |
| 19:31.07 | dijib | im going to be joining the confrence line, and going to have a smoke and talk to my neighbor guy |
| 19:31.33 | p3nguin | Ask about number portability? |
| 19:31.37 | covici | Hi. I have a strange problem where sip has suddenly stopped working for asterisk. I have tried fs and that works so my network is working, but I would like |
| 19:32.45 | covici | to get asterisk working again -- and as far as I can tell, no changes were made in the configs. I am using freepbx also. |
| 19:33.15 | dijib | covici, join SIP/2663@asterisk.serveirc.com for assistance |
| 19:33.24 | dijib | lol |
| 19:33.29 | dijib | nevermind your sip doesnt work |
| 19:33.31 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 19:33.34 | p3nguin | haha |
| 19:33.36 | covici | all internal phones say they are unreachable |
| 19:34.03 | p3nguin | What does "module show like sip" say? |
| 19:35.30 | covici | hang on and let me see -- I had some problems loading chan_sip, but it is loaded, but not doing much. |
| 19:35.57 | p3nguin | If it is loaded, unload it. Then load it and copy the entire output, and put it in the pastebin. |
| 19:36.26 | covici | pastebin.com? |
| 19:36.30 | p3nguin | yes, please |
| 19:36.38 | covici | OK, hang on. |
| 19:37.36 | covici | I am restarting asterisk. |
| 19:38.15 | p3nguin | *shrug* |
| 19:38.27 | p3nguin | I really just asked for you to unload and load chan_sip. |
| 19:38.43 | covici | Its only two lines so here iit is. |
| 19:38.46 | covici | Module Description Use Count |
| 19:38.47 | covici | app_adsiprog.so Asterisk ADSI Programming Application 0 |
| 19:38.56 | p3nguin | Okay, chan sip is not loaded. |
| 19:38.57 | covici | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 19:39.02 | p3nguin | now it is. |
| 19:39.24 | p3nguin | Does "sip show peers" show all your peers? |
| 19:39.39 | covici | but sip show peers says no such command |
| 19:40.21 | p3nguin | Now do what I asked you to do, and pastebin it. |
| 19:41.58 | covici | You mean unload and reload? |
| 19:42.04 | p3nguin | unload, then load. |
| 19:42.11 | p3nguin | module unload chan_sip.so |
| 19:42.15 | p3nguin | module load chan_sip.so |
| 19:42.21 | covici | OK. |
| 19:42.23 | p3nguin | Pastebin whatever it outputs. |
| 19:42.33 | p3nguin | If nothing, pastebin nothing. |
| 19:43.29 | p3nguin | If it only says loading, that also doesn't need pastebinned. |
| 19:43.33 | covici | No output from those. |
| 19:43.47 | p3nguin | core set verbose 3 |
| 19:43.50 | p3nguin | Do it again. |
| 19:43.53 | covici | Its 4 now. |
| 19:43.57 | p3nguin | okay |
| 19:44.00 | p3nguin | That's sufficient. |
| 19:44.08 | p3nguin | When you load it, it should show that it parsed sip.conf. |
| 19:44.30 | covici | I think its in the logs, but not seeing in the console -- this is 1.8. |
| 19:45.54 | covici | After a long time it said unloaded chan_sip.so -- must have taken 3 minutes. |
| 19:49.37 | covici | Now its actualling registering peers, so I am not sure what is happening. |
| 19:49.49 | covici | But it is slow as heck . |
| 19:50.39 | covici | And sorry, I could only pastebin from the logs the load is off the screen. |
| 19:52.52 | wdoekes2 | dns issues? |
| 19:53.21 | covici | Unlikely, but I can check. |
| 19:53.30 | covici | Hang on -- phone call. |
| 19:57.05 | p3nguin | Sorry, temporary network outage. |
| 19:59.11 | p3nguin | What do you mean by "slow" in this context? |
| 20:07.28 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:09.50 | covici | I mean that the load took 2 or 3 minutes. and when I execute a command it takes quite a while. |
| 20:10.13 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 20:10.32 | p3nguin | Maybe the system is heavily loaded. |
| 20:10.45 | p3nguin | Can you check top/htop as well as iotop? |
| 20:11.55 | covici | Nothing else on the system -- load average is 0. |
| 20:12.21 | p3nguin | iotop -oPa |
| 20:12.22 | p3nguin | Let it run for a few minutes. |
| 20:13.16 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 20:13.40 | covici | don't have iotop -- will try to install now. |
| 20:15.15 | covici | OK its going now, but its getting 0's . |
| 20:22.02 | p3nguin | It should have collected something by now. |
| 20:22.13 | covici | hang on. |
| 20:22.39 | covici | just 31k/sec. |
| 20:23.05 | p3nguin | At least one thing would have either written to disk or read from disk by now. |
| 20:23.12 | *** join/#asterisk doug (doug@breakout.telerama.com) |
| 20:23.39 | p3nguin | 17879 be/4 asterisk 0.00 B 8.00 K 0.00 % 0.01 % asterisk |
| 20:23.41 | p3nguin | for example. |
| 20:23.50 | covici | hang on. |
| 20:25.58 | doug | is there a variable that'll tell me which iax channel i'm coming in on? |
| 20:26.18 | doug | i'm using an iax client, loudhush, to connect up to asterisk. |
| 20:26.22 | p3nguin | Not a variable, no. But you can use the CHANNEL() function. |
| 20:26.54 | p3nguin | If you just need to see it now, you can use "core show channels verbose" on the CLI. |
| 20:27.17 | doug | ah, channel(peername) seems like it'd do the trick... |
| 20:27.43 | p3nguin | core show channels seems a lot easier if you just need to see it now rather than every time. |
| 20:27.58 | covici | asterisk gets 16022 be/4 asterisk 0.00 B 380.00 K 0.00 % 0.00 % asterisk -f -U asterisk -G asterisk -vvvvg -c |
| 20:28.36 | p3nguin | Okay, so you've got at least that much disk activity. Now I'm really more interested in the read/write at the top of the list. |
| 20:28.56 | covici | another phone call. |
| 20:31.24 | doug | > Function channel not registered |
| 20:31.32 | doug | it's not case sensitive, is it? |
| 20:31.34 | p3nguin | It's "CHANNEL" |
| 20:31.43 | p3nguin | core show function CHANNEL |
| 20:31.55 | doug | wow, it is. |
| 20:32.01 | doug | rock it like it's 1972 |
| 20:32.07 | p3nguin | All functions are in caps. |
| 20:32.17 | *** join/#asterisk MDesade (~chatzilla@ip70-162-84-98.ph.ph.cox.net) |
| 20:32.30 | p3nguin | But you can type core show function cha and hit Tab and it will complete and change case for you. |
| 20:32.41 | *** join/#asterisk cerberus_za (~coert@8ta-151-134-105.telkomadsl.co.za) |
| 20:34.20 | *** join/#asterisk ruied (~ruied@po-217-129-154-119.netvisao.pt) |
| 20:36.13 | covici | OK, I am back -- I will try sip show peers again and see if it registered things. |
| 20:36.29 | doug | lemme crank out a replacement punchcard...then it'll work. |
| 20:37.12 | covici | Most of them say unspecified so they are not registering. Its faster now, however. |
| 20:37.28 | covici | And these are on the internal network! |
| 20:37.44 | doug | p3nguin++ |
| 20:38.15 | p3nguin | Give them a few minutes and then check again. |
| 20:38.25 | covici | OK. |
| 20:38.55 | covici | But the funny thing is another sip server is working. |
| 20:39.05 | covici | I tried freeswitch just to see. |
| 20:41.53 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 20:44.09 | p3nguin | lsof -i udp:5060 |
| 20:44.15 | p3nguin | Show me what that says. |
| 20:45.20 | covici | Here is the output of sip show peers http://pastebin.com/xHViq9N7 |
| 20:53.34 | *** join/#asterisk francisvgarcia (~networker@186.1.90.193) |
| 20:53.53 | covici | The output is: asterisk 16022 asterisk 12u IPv4 271943 0t0 UDP *:sip |
| 20:54.27 | francisvgarcia | Hi Folks |
| 20:54.38 | francisvgarcia | I got a question for you |
| 20:54.55 | [TK]D-Fender | covici: Actually connect to CLI and stay there, don't do this with -rx |
| 20:55.14 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:55.16 | covici | OK, but I get a lot of dnsmgr output. |
| 20:55.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:55.41 | WIMPy | Patch that away :-) |
| 20:56.47 | francisvgarcia | As the 2.6.28 kernel has the support for the OSLEC, If I have a CentOS version with this kernel, do I have to activate the OSLEC echo cancel algorithm in system.conf |
| 20:57.57 | scubes13 | our phone system requires that we always dial a 1 before any outgoing phone call with the remaining 10 digits. what could I add to the dial plan (guessing that is where it goes) so that it would add the 1 if someone did not dial the full 11 digits? |
| 20:58.12 | WIMPy | francisvgarcia: How do those two things fit together? |
| 20:59.32 | [TK]D-Fender | scubes13: put the 1 in your Dial |
| 21:00.02 | francisvgarcia | WIMPy: It is just a question |
| 21:00.10 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 21:00.18 | scubes13 | [TK]D-Fender - such as 1+NXXNXXXXXX ? |
| 21:00.39 | [TK]D-Fender | scubes13: No, in the DIAL() |
| 21:00.42 | covici | You would need to test the first digit and if not 1 then add the one before the extension variable. |
| 21:01.44 | [TK]D-Fender | covici: He didn't say anything about 1 not being a valid first digits. He said if they dial 10 digits, put a 1 in front |
| 21:01.59 | francisvgarcia | something like this _XXXXXXXXXX,1, Dial(CHANNEL/1+${EXTEN},120,tT) |
| 21:02.06 | covici | But probably people will do both. |
| 21:02.13 | [TK]D-Fender | No "+" |
| 21:02.29 | francisvgarcia | Exactly not with the + |
| 21:02.41 | WIMPy | francisvgarcia: system.conf sounds like dahdi, but dahdi is not in Linux. |
| 21:02.42 | p3nguin | covici: core set verbose 3 |
| 21:02.53 | p3nguin | covici: That will get rid of the dnsmgr crap. |
| 21:02.57 | covici | Verbose is 4 now. |
| 21:03.46 | p3nguin | You should not be using tT in that dial command. That allows both the caller and the callee to transfer the call. |
| 21:03.52 | francisvgarcia | WIMPy: Yes it is dahdi, I wonder if have to activate it in dahdi before asterisk can use it |
| 21:04.04 | p3nguin | covici: I'll tell you for a third time: core set verbose 3 |
| 21:04.08 | p3nguin | I won't say it again. |
| 21:04.57 | francisvgarcia | exten => _XXXXXXXXXX,1,DIAL(CHANNEL/1${EXTEN,120}) |
| 21:05.08 | [TK]D-Fender | francisvgarcia: Yes you have to specify teh EC to use in system.conf |
| 21:05.08 | p3nguin | fail |
| 21:05.15 | WIMPy | francisvgarcia: Yes, but why did you mention the kernel? |
| 21:05.40 | covici | output of that is "verbosity was 4 and is now 3" |
| 21:05.42 | p3nguin | scubes13: http://pastebin.com/Piqv4Egj Scroll down and look at lines 118-124. |
| 21:05.55 | p3nguin | covici: Good. Now you won't have dnsmgr filling the screen. |
| 21:06.05 | p3nguin | Now you can do whatever [tk]d-fender wanted you to do. |
| 21:06.57 | francisvgarcia | WIMPy: Because this kernel has the support for OSLEC and I don't have to compile it in dahdi apparently |
| 21:07.21 | francisvgarcia | all the starting from the 2.6.28 |
| 21:08.14 | covici | tk]d-fender: OK, now what do you want me to do in my cli? |
| 21:08.29 | [TK]D-Fender | cosip show peers |
| 21:08.37 | [TK]D-Fender | covici: sip show peers |
| 21:09.28 | covici | OK, now some of the peers are coming up as registered. |
| 21:09.50 | covici | last line is 59 sip peers [Monitored: 28 online, 12 offline Unmonitored: 13 online, 6 offline] |
| 21:10.01 | p3nguin | If you keep waiting, the others may come online, too. |
| 21:10.17 | [TK]D-Fender | -rx doesn't wait for a full response <- |
| 21:10.34 | [TK]D-Fender | If you hit a delay, you get a partial response regardless of the proper output |
| 21:10.55 | covici | But its been up for 45 minutes why is it so slow? |
| 21:11.23 | p3nguin | The user agents aren't trying to register every second of their existence. |
| 21:11.55 | covici | But I have them set to 5 minutes or less -- I can check, but its something like that. |
| 21:12.25 | covici | And the module seemed to take a long time even to load. |
| 21:22.38 | covici | Well, thank you guys for all your help -- I wil monitor and see what happens. |
| 21:23.01 | *** part/#asterisk covici (~user@pool-173-72-192-149.clppva.fios.verizon.net) |
| 21:24.50 | *** join/#asterisk thebitguru (~thebitgur@75-134-26-190.dhcp.mdsn.wi.charter.com) |
| 21:29.47 | thebitguru | Hi |
| 21:30.34 | thebitguru | if I am using SIP/VOIP only to call normal phone lines can I still use dahdi for echo cancellation? I am a litte confused about my echo cancellation options |
| 21:30.50 | p3nguin | There's no echo on SIP. |
| 21:31.14 | thebitguru | p3nguin: SIP to a normal cell phone line results in echo on the other side |
| 21:31.45 | thebitguru | I am not sure what I am not doing right |
| 21:32.01 | francisvgarcia | thebitguru: are you using a SIP Trunk? |
| 21:32.10 | p3nguin | If you hear echo, it's either the handset having an issue with the volume, or the echo is introduced in the analog part of the call (which you don't have control over). |
| 21:32.12 | p3nguin | ~siptrunk |
| 21:32.12 | infobot | siptrunk is probably something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 21:32.30 | p3nguin | The SIP trunk is a lie. |
| 21:32.34 | thebitguru | I see |
| 21:32.50 | thebitguru | the thing is the echo is heard on the other side, sound is good on my side |
| 21:32.51 | WIMPy | Where on earth would you find analog cellphones? |
| 21:33.06 | thebitguru | WIMPy: I am not saying they are analog, just that they hear the echo |
| 21:33.23 | p3nguin | I also didn't say the cell phone are analog. |
| 21:33.38 | francisvgarcia | Could be an issue with the TELCO |
| 21:33.41 | WIMPy | Now fit the 3 together. |
| 21:33.51 | WIMPy | thebitguru: What kind of phone are you using? |
| 21:33.53 | francisvgarcia | that can be using TDM for routing the Cell Phone Calls |
| 21:34.21 | thebitguru | WIMPy: softphone for now. I have tried x-lite on windows, on mac and 3cx on android, all with similar results |
| 21:34.34 | thebitguru | I have tried two different SIP providers (sipstation and vitelity) with similar results |
| 21:34.52 | [TK]D-Fender | You could be gettingacoustic echo from your headset speakers feeding into the mic... or your ITSP could suck |
| 21:34.56 | francisvgarcia | thebitguru: Does it happen to regular phone numbers? |
| 21:35.13 | WIMPy | Maybe your internet connection has too much delay? |
| 21:35.18 | thebitguru | francisvgarcia: for sipstation it did, but not with vitelity |
| 21:35.42 | thebitguru | WIMPy: 52ms ping to vitelity's server, what else can I check to validate that connection isn't the problem? |
| 21:35.42 | p3nguin | Call some other numbers and see if they also hear echo. |
| 21:36.32 | francisvgarcia | thebitguru: The Echo is only hear by the other end ? |
| 21:36.43 | thebitguru | francisvgarcia: correct |
| 21:36.59 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 21:37.40 | francisvgarcia | thebitguru: and does it happen when anybody else call to the same cell phone or only when you call? |
| 21:37.49 | thebitguru | I have tried laptop speakers, a logitech headset with mic on the desktop, and my cellphone's mic/speakers with similar results so I don't think it echo is introduced by my equipment. |
| 21:37.58 | thebitguru | francisvgarcia: only me. when I call from my cell then they hear OK |
| 21:38.20 | thebitguru | two different cell phone numbers behaved similarly |
| 21:38.53 | francisvgarcia | echo is an issue only on TDM lines when using dahdi |
| 21:39.17 | thebitguru | weird |
| 21:39.24 | thebitguru | what else could be the cause? |
| 21:39.32 | francisvgarcia | try using headseats and try again |
| 21:39.56 | thebitguru | I have tried asterisk on base ubuntu with freepbx, and with pbx in a flash with similar results |
| 21:40.03 | WIMPy | Listen to p3nguin. Echo is always analog, unless you use a softphone and have set up a loop in your mixer. |
| 21:40.27 | thebitguru | WIMPy: so if they are hearing echo then it is probably something with my provider? |
| 21:40.37 | francisvgarcia | can be the speake feeding the mic |
| 21:40.46 | francisvgarcia | a feedback |
| 21:40.53 | thebitguru | francisvgarcia: even with headset? |
| 21:40.57 | p3nguin | Your digital equipment can have echo, but it's produced from the analog parts of it -- usually speaker-to-mic issues like [tk]d-fender mentioned. |
| 21:40.58 | WIMPy | I wouldn't know how they could produce echo. |
| 21:41.14 | thebitguru | maybe I should confirm that my network isn't causing this |
| 21:41.27 | thebitguru | other than the ping time what else can I check? |
| 21:41.29 | [TK]D-Fender | [17:40]thebitgurufrancisvgarcia: even with headset? <- yes |
| 21:41.44 | WIMPy | Yes, even headsets can produce echo. But if you've got one with the mic on the boom instead of just a tube, you should be safe. |
| 21:41.45 | francisvgarcia | anyway dahdi echo cancel is only applicable to TDM hardware |
| 21:41.54 | thebitguru | francisvgarcia: how about oslec? |
| 21:41.55 | francisvgarcia | and Digital Circuits |
| 21:42.15 | WIMPy | You only get EC for hardware interfaces. |
| 21:42.21 | thebitguru | I see |
| 21:42.25 | francisvgarcia | oslec is only applicable to hardware |
| 21:42.30 | p3nguin | oslec is a type of echo cancellation for analog hardware. |
| 21:42.31 | francisvgarcia | such as TDM |
| 21:42.43 | thebitguru | the weird thing is that the other side hears their own voice, everything is OK on my side |
| 21:42.45 | p3nguin | As I mentioned before, SIP does not echo. |
| 21:43.11 | *** join/#asterisk pietro (~pietro@88-149-227-165.dynamic.ngi.it) |
| 21:43.13 | WIMPy | SIP doesn't even transfer audio. |
| 21:43.26 | [TK]D-Fender | thebitguru: Mute your mic. Do they still hear themselves? |
| 21:43.30 | p3nguin | If you want to get down to it, that's true. |
| 21:43.35 | p3nguin | Not what I meant, but true. |
| 21:43.41 | thebitguru | [TK]D-Fender: that's a good idea. let me try that |
| 21:43.58 | WIMPy | But IP introduces delays and they regularly make EC fail. |
| 21:44.02 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:47.02 | p3nguin | I have a headset with a voice tube, and no one ever told me they hear any echo from it. |
| 21:47.29 | p3nguin | I thought about getting the other style (with the noise canceling boom mic), but never did it. |
| 21:47.32 | *** join/#asterisk sparrW (~kvirc@pdpc/supporter/active/sparr) |
| 21:47.35 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 21:47.47 | sparrW | what linux distro is best suited for asterisk? packaged library versions, etc |
| 21:47.49 | WIMPy | Maybe your volume isn't high enoug or it has better quality. |
| 21:48.02 | p3nguin | sparrw: AsteriskNOW |
| 21:48.23 | p3nguin | I just use a Cisco phone with a Plantronics headset. |
| 21:48.31 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:48.36 | thebitguru | I think it might just be the softphone/headset issue. I will try a few other options |
| 21:48.38 | thebitguru | brb |
| 21:50.14 | francisvgarcia | thebitguru: Try using a hardphone |
| 21:50.30 | francisvgarcia | and let's see if the issue carries on |
| 21:50.35 | WIMPy | But a digital one. |
| 21:50.51 | p3nguin | a regular IP phone |
| 21:50.56 | sparrW | p3nguin: sorry, a mainstream distro. my virtual server provider doesn't allow custom images |
| 21:51.06 | p3nguin | sparrw: CentOS 5.5 |
| 21:51.11 | sparrW | thanks |
| 21:51.45 | p3nguin | Don't go with C6 because there haven't been any packages made as of the last time I looked. |
| 21:52.53 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:52.59 | francisvgarcia | and IP Phone |
| 21:58.01 | francisvgarcia | I got a question for something that I haven't done yet, or if you know that it'll be included in future asterisk releases |
| 21:58.27 | francisvgarcia | Extension Mobility !!! |
| 21:58.38 | p3nguin | What does that phrase mean to you? |
| 21:58.49 | francisvgarcia | without queues |
| 22:00.03 | francisvgarcia | p3nguin: I mean, that an user can dial something like a code in the dialpad and his extension come to him wherever he logs in |
| 22:00.27 | p3nguin | Your knowledge of asterisk terminology could use some work. |
| 22:00.46 | p3nguin | And the mechanism you're looking for is known as "hot desking." |
| 22:00.58 | francisvgarcia | Hot Desking |
| 22:01.05 | francisvgarcia | that is |
| 22:01.14 | p3nguin | Extensions don't change; only the device changes. |
| 22:01.35 | sparrW | p3nguin: thanks. i have a server set up, the asterisk packages installed via yum, and the asterisk service running... and i'm failing horribly at figuring out the next step, there aren't any links from the installation parts of the wiki to the usage parts. do i have to just dive in or can you recommend a getting started tutorial? |
| 22:01.47 | p3nguin | So if I go to SIP phone 000011112222, my extension (762) will Dial(SIP/000011112222)... |
| 22:02.07 | p3nguin | or If I go to SIP phone 222233334444, now my extension (762) will Dial(SIP/222233334444). |
| 22:02.11 | p3nguin | hot desking |
| 22:02.29 | p3nguin | ~book |
| 22:02.29 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 22:02.32 | francisvgarcia | yeap, that is |
| 22:02.37 | p3nguin | sparrw: Read the book. |
| 22:03.12 | sparrW | I'm not the person who is going to be administering the asterisk installation |
| 22:03.15 | sparrW | that seems like overkill |
| 22:03.28 | p3nguin | What, specifically, do you want to know about? |
| 22:03.33 | [TK]D-Fender | sparrW: Then hand your system over to that other person |
| 22:04.36 | francisvgarcia | p3nguin: do you know if a Hot Desking application will be included in future asterisk releases? |
| 22:04.44 | [TK]D-Fender | ... |
| 22:04.48 | [TK]D-Fender | appllication? |
| 22:04.59 | sparrW | i think what i was looking for is that freepbx is the software i probably want to be dealing with |
| 22:04.59 | [TK]D-Fender | francisvgarcia: Hot-desking is a concept. |
| 22:05.00 | sparrW | will try that |
| 22:05.04 | [TK]D-Fender | francisvgarcia: Not an "application" |
| 22:05.53 | p3nguin | It's not an application, it's a concept. |
| 22:05.58 | p3nguin | You just build it. |
| 22:06.00 | francisvgarcia | thanks for the correction, but I mean If there is an easy way to do it |
| 22:06.02 | p3nguin | The tools exist. |
| 22:06.07 | sparrW | [TK]D-Fender: i think he is using application in the engineering sense (ala "Application Note" as a form of documentation), not in the software package sense |
| 22:06.28 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:07.02 | p3nguin | Any admin worth a shit wouldn't rely on FreePBX. |
| 22:07.51 | p3nguin | FreePBX is for that manager who thinks he needs to run a phone system instead of pay a real admin. |
| 22:08.25 | ChannelZ | And why would any manager want to run a phone system, unless he's an ubernerd - in which case he wouldn't care about FreePBX |
| 22:08.37 | p3nguin | That's what they do. |
| 22:08.42 | sparrW | i have a server to which i am not giving root access to the person who wants to manage the pbx |
| 22:08.49 | p3nguin | "I'm in charge; I can run the phone system." |
| 22:09.03 | p3nguin | He wouldn't need root access to run asterisk. |
| 22:09.42 | p3nguin | He needs only a regular user account and enough sudo rights to admin asterisk. |
| 22:09.44 | ChannelZ | Ooooh, just got a big spat of anon SIP call attempts. I don't get what these people think they are accomplishing if the replies don't even make it back to them. |
| 22:09.53 | sparrW | and I don't know that, because I'm too lazy to read an entire book explaining how asterisk works |
| 22:09.55 | sparrW | hence, freepbx |
| 22:10.09 | p3nguin | That has nothing to do with Asterisk nor the book. |
| 22:10.17 | p3nguin | That's SysAdmin 101. |
| 22:10.18 | ChannelZ | If you're too lazy to do that, you probably shouldn't be running a phone system |
| 22:10.26 | sparrW | ChannelZ: I'm not. He is. |
| 22:10.31 | sparrW | that's the problem. my server. his PBX. |
| 22:10.37 | ChannelZ | He, you, they, whatever |
| 22:11.32 | sparrW | p3nguin: do you not understand that I don't know how asterisk works? I don't know which, or how many, services it needs or wants to run. i don't know what binaries are used to admin it. i don't know what files the user adminning it needs access to. |
| 22:11.46 | sparrW | and, moreso, I don't WANT to know those things. i'm not the one adminning it. |
| 22:11.49 | p3nguin | So you need to hire a sys admin, too. |
| 22:12.05 | p3nguin | Or just ask your questions already. |
| 22:12.45 | [TK]D-Fender | sparrW: Go ask the person who you're doing it for what they want |
| 22:12.53 | sparrW | right now my question is... how do i find out the admin login and password for my freepbx installation, installed from the YUM repositories documented in the asterisk wiki? |
| 22:13.05 | p3nguin | ~freepbx |
| 22:13.05 | infobot | [~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, or fpbx |
| 22:13.13 | sparrW | yep, already asking there |
| 22:13.49 | sparrW | the default setup in the rpm packages for asterisk seem to want root to do the administration |
| 22:14.07 | francisvgarcia | does anyone know how to correct the TDM400p issue that It rings two times before answering the call |
| 22:14.17 | p3nguin | Add that to my list of reasons to not use FreePBX and instead learn how to admin Asterisk. |
| 22:15.12 | p3nguin | If you use FreePBX to admin it, you have little choice of non-root controlling things. |
| 22:16.06 | p3nguin | If you deal with it from a normal systems administrator perspective, you have all the control you want or all the control you want to delegate to someone else. |
| 22:16.19 | sparrW | i'd prefer the packages just install as a new user |
| 22:16.22 | sparrW | like apache does |
| 22:16.38 | [TK]D-Fender | francisvgarcia: It isn't an issue. It's waiting for CALLERID |
| 22:16.45 | sparrW | instead i'm left trying to figure out what my asterisk-admin user needs access to |
| 22:16.54 | [TK]D-Fender | francisvgarcia: Feel free to lose callerid by disabling it. |
| 22:17.17 | francisvgarcia | I did it already |
| 22:17.21 | francisvgarcia | and It does not work |
| 22:17.46 | [TK]D-Fender | francisvgarcia: Wouldn't bet on it being done right or applied... |
| 22:17.55 | [TK]D-Fender | francisvgarcia: Would help if you showed us |
| 22:17.58 | p3nguin | Caller ID is received between the first and second rings. If you don't wait for it, you won't get it. |
| 22:18.16 | francisvgarcia | The customer don't care |
| 22:18.21 | francisvgarcia | about caller id |
| 22:18.24 | p3nguin | They don't want CID? |
| 22:18.28 | francisvgarcia | no |
| 22:18.38 | francisvgarcia | he wants to |
| 22:18.39 | p3nguin | Reconfigure the thing to answer right away. |
| 22:18.57 | francisvgarcia | he wants the phone to ring automatically |
| 22:19.09 | francisvgarcia | no matter if he loose the caller id |
| 22:19.31 | francisvgarcia | some of the strange things that we see over the time |
| 22:19.44 | p3nguin | Seems odd, but give him what he wants. |
| 22:21.08 | thebitguru | yup, the equipment seems to be the issue in my case. |
| 22:21.36 | thebitguru | francisvgarcia, [TK]D-Fender, p3nguin, WIMPy: thanks for guiding me in the right direction. I appreciate your help! |
| 22:26.00 | francisvgarcia | thebitguru: you welcome |
| 22:39.24 | francisvgarcia | It still rings two times even if I place callerid=no |
| 22:39.33 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:39.34 | p3nguin | Check your dial plan. |
| 22:39.35 | dijib | there you fo |
| 22:39.38 | dijib | go |
| 22:39.45 | dijib | oh man ive quite a few beers in now |
| 22:39.50 | p3nguin | You probably have a Wait() in it where you're creating that pause. |
| 22:39.57 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:40.58 | [TK]D-Fender | [18:39]francisvgarciaIt still rings two times even if I place callerid=no <- this isn't the right parameter at all |
| 22:42.07 | francisvgarcia | sorry |
| 22:42.13 | francisvgarcia | it's usecallerid=no |
| 22:42.26 | francisvgarcia | thinking sip |
| 22:43.09 | [TK]D-Fender | francisvgarcia: And as I said before it'd help if you showed us. |
| 22:43.24 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:43.43 | dijib | im going to spam as if spam hasent been invented |
| 22:45.20 | [TK]D-Fender | Ops? |
| 22:45.41 | p3nguin | Special Ops, to be specific. |
| 22:45.42 | francisvgarcia | Here we go |
| 22:45.42 | francisvgarcia | [channels] |
| 22:45.42 | francisvgarcia | usecallerid = no |
| 22:45.42 | francisvgarcia | hidecallerid = no |
| 22:45.42 | francisvgarcia | ;cidsignalling=v23 |
| 22:45.43 | francisvgarcia | ;cidstart=polarity |
| 22:45.43 | francisvgarcia | loadzone = uk |
| 22:45.44 | francisvgarcia | busydetect=yes |
| 22:45.44 | francisvgarcia | busycount=3 |
| 22:45.45 | francisvgarcia | inmediate=yes |
| 22:45.45 | francisvgarcia | ;hanguponpolarityswitch=yes |
| 22:45.45 | p3nguin | Don't do that. |
| 22:45.46 | francisvgarcia | sendcalleridafter=1 |
| 22:45.46 | francisvgarcia | defaultzone = uk |
| 22:45.47 | francisvgarcia | language = es |
| 22:45.49 | francisvgarcia | signalling=fxs_ks |
| 22:45.51 | p3nguin | WTF |
| 22:45.51 | francisvgarcia | faxdetect=no |
| 22:45.53 | francisvgarcia | callwaiting = yes |
| 22:45.55 | francisvgarcia | threewaycalling = yes |
| 22:45.57 | francisvgarcia | transfer = yes |
| 22:45.59 | francisvgarcia | echocancel = 64 |
| 22:46.01 | francisvgarcia | echotraining= no |
| 22:46.02 | p3nguin | No one ever taught you not to flood a channel? |
| 22:46.03 | francisvgarcia | echocancelwhenbridged = no |
| 22:46.05 | francisvgarcia | rxgain= 0.0 |
| 22:46.06 | [TK]D-Fender | facepalms |
| 22:46.07 | francisvgarcia | txgain = -4.5 |
| 22:46.12 | dijib | stop flooding francisvgarcia Pastebin.com for that shit |
| 22:46.17 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:46.18 | p3nguin | Are people really this silly? |
| 22:46.19 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:46.26 | dijib | yes we are |
| 22:46.26 | p3nguin | Or do they just play this silly on IRC? |
| 22:46.52 | [TK]D-Fender | p3nguin: Welcome to "Full-Retard" Weekend |
| 22:46.53 | francisvgarcia | Sorry |
| 22:46.56 | francisvgarcia | I didn't know |
| 22:47.04 | p3nguin | It's not just today. |
| 22:47.33 | p3nguin | You didn't KNOW we didn't want that flood? |
| 22:47.51 | p3nguin | You thought we wanted our screen to be filled with stuff we don't care about? |
| 22:49.18 | dijib | well irregardless Dial(SIP/2663@asterisk.serveirc.com); |
| 22:49.59 | p3nguin | ir = without. regardless = lack of regard. irregardless = without the lack of regard = having much regard. |
| 22:50.20 | dijib | i know, dont worry its proper american |
| 22:50.25 | [TK]D-Fender | irregardless = Not an actual word because of how redundent it is. |
| 22:50.27 | p3nguin | huh? |
| 22:50.49 | p3nguin | <dijib> i know, dont worry its proper american <--- does not compute |
| 22:50.51 | dijib | its a double negative |
| 22:51.06 | p3nguin | double negative is affirmative |
| 22:51.09 | dijib | but americans use it because they thing their always right |
| 22:51.31 | [TK]D-Fender | [18:50]dijibi know, dont worry its proper american <- "I" , "don't", "it's", "American", and lack of a period at the end. Almost every single word in that sentence was wrong. |
| 22:51.53 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:51.55 | p3nguin | Not all Americans are that uneducated. I'm an American, for instance. |
| 22:51.59 | francisvgarcia | dijib I am in |
| 22:52.31 | dijib | really |
| 22:52.36 | dijib | give me one second |
| 22:53.15 | francisvgarcia | It is a conference bridge |
| 22:53.24 | dijib | yes |
| 22:53.27 | dijib | you hear me? |
| 22:53.31 | francisvgarcia | not |
| 22:53.34 | p3nguin | And if I've ever used the term "irregardless," it was due to negative influence from places where people say "irregardless." |
| 22:53.36 | francisvgarcia | no I don't hear you |
| 22:53.50 | p3nguin | Don't forget to unmute. |
| 22:53.57 | dijib | you should be able to hear me francis |
| 22:53.58 | francisvgarcia | I was hearing the music on hold |
| 22:54.03 | dijib | now nothing? |
| 22:54.14 | francisvgarcia | yes, nothing |
| 22:54.28 | dijib | now music on hold? |
| 22:54.29 | francisvgarcia | but you hear me? |
| 22:54.32 | francisvgarcia | yes |
| 22:54.33 | dijib | no not at all |
| 22:54.39 | francisvgarcia | check ur mic |
| 22:54.43 | dijib | i did |
| 22:54.52 | francisvgarcia | but do u hear me? |
| 22:55.01 | dijib | no i didnt |
| 22:55.06 | dijib | im rejoining |
| 22:55.20 | dijib | im in again |
| 22:55.26 | francisvgarcia | It stopped |
| 22:55.28 | dijib | i cant hear you at all. |
| 22:55.30 | [TK]D-Fender | That's what SHE said |
| 22:55.38 | dijib | lol |
| 22:55.40 | francisvgarcia | the music stoped |
| 22:55.46 | dijib | [TK]D-Fender, join the conference |
| 22:55.54 | dijib | EVERYBODY put this is your dialplan and join US! Dial(SIP/2663@asterisk.serveirc.com); |
| 22:56.02 | dijib | im in here |
| 22:56.02 | [TK]D-Fender | dijib: You're annoying enough here... |
| 22:56.06 | dijib | :D |
| 22:56.07 | p3nguin | ;) |
| 22:56.09 | dijib | you know it |
| 22:56.17 | [TK]D-Fender | dijib: Speaking of which stop spamming that junk around. |
| 22:56.21 | dijib | p3nguin, why is this broken would you summize? |
| 22:56.26 | dijib | lol |
| 22:56.29 | p3nguin | What's broken? |
| 22:56.41 | francisvgarcia | <PROTECTED> |
| 22:56.50 | francisvgarcia | to avoid nat issues |
| 22:56.55 | dijib | i cant hear francisvgarcia |
| 22:57.08 | dijib | i didnt know IAX2 was the answer to that |
| 22:57.24 | p3nguin | IAX2 does not get affected by NAT like SIP/RTP does. |
| 22:57.33 | dijib | interdasting |
| 22:57.42 | dijib | any why dont i rebuild my server to use IAX |
| 22:57.47 | [TK]D-Fender | Which is not to say "completely unaffected" |
| 22:57.59 | [TK]D-Fender | However far friendlier at face value |
| 22:58.06 | dijib | francisvgarcia, you dropped? |
| 22:58.06 | p3nguin | There is no "rebuilding" to use IAX2. You just turn on IAX2. |
| 22:58.11 | francisvgarcia | yes |
| 22:58.14 | francisvgarcia | I did it |
| 22:58.14 | dijib | i would rebuild |
| 22:58.21 | dijib | i have MOH in my ear now |
| 22:58.22 | francisvgarcia | I'll be back in a while |
| 22:58.23 | [TK]D-Fender | You don't turn it on.. you configure devices for it. |
| 22:58.24 | p3nguin | You're an idiot, then. |
| 22:58.40 | dijib | p3nguin, why so rude? is it because ive been drinking |
| 22:58.56 | p3nguin | Configure iax.conf, module load chan_iax2.so = turn on IAX2 |
| 22:59.40 | dijib | whoever is in the confrence bridge dial # for options not * |
| 22:59.49 | p3nguin | That's so weird. |
| 22:59.58 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:00.01 | p3nguin | #1 to unmute. |
| 23:00.01 | dijib | is that you in there p3nguin ? |
| 23:00.04 | p3nguin | yes |
| 23:00.15 | francisvgarcia | to be somethink like Dial(IAX2/266@asterisk.serveirc.com) |
| 23:00.18 | dijib | oh shit yeh i keep forgetting whats the option to not have the mutted option |
| 23:00.25 | p3nguin | I have to figure out why it's # instead of *. |
| 23:00.32 | dijib | why? |
| 23:00.35 | dijib | oh ok |
| 23:00.51 | p3nguin | Do you really want people to come in unmuted? |
| 23:01.52 | p3nguin | If yes, remove the m from ConfBridge(). |
| 23:01.55 | *** join/#asterisk wooster (~footlocke@ns1.sextube.ro) |
| 23:02.13 | wooster | i upgraded to 10.0.0, now all of my SIP clients are getting 401 unauthorized |
| 23:02.47 | p3nguin | I'm connected, but not talking right now. |
| 23:05.57 | p3nguin | What was the question? |
| 23:06.00 | dijib | p3nguin, did you hear his question? |
| 23:06.06 | p3nguin | Negative. |
| 23:06.28 | wooster | did something change with SIP friend authentication in 10.0? |
| 23:07.35 | p3nguin | I'm still saying it's dial plan. There is probably a Wait(). |
| 23:08.04 | p3nguin | If there is Wait before Answer, it will ring before it answers. |
| 23:09.11 | p3nguin | PASTEBIN IT |
| 23:09.13 | p3nguin | ~pb |
| 23:09.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 23:12.41 | dijib | <PROTECTED> |
| 23:13.51 | p3nguin | I'll have to see the dial plan to be convinced it isn't the dial plan. |
| 23:16.02 | dijib | i think i want to keep spamming |
| 23:16.14 | dijib | this is slowely starting to get fun |
| 23:20.00 | francisvgarcia | p3nguin: http://pastebin.com/NuLdXxK8 |
| 23:21.18 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 23:22.48 | [TK]D-Fender | francisvgarcia: According to your /etc/asterisk/chan_dahdi.conf you have NO channels whatsoever |
| 23:23.05 | dijib | Goto(context,extension,priority) |
| 23:23.09 | [TK]D-Fender | inmediate=yes <- Misspelled and wrong to have as yes once corrected |
| 23:23.10 | francisvgarcia | I have just paste |
| 23:23.22 | francisvgarcia | only the relevant information |
| 23:23.41 | p3nguin | You pasted some of the relevant information. |
| 23:24.29 | [TK]D-Fender | francisvgarcia: Why on Earth would we believe that? |
| 23:24.55 | [TK]D-Fender | francisvgarcia: Things aren't working and you're deliberately not showing us important bits. |
| 23:32.43 | francisvgarcia | Ok here we go |
| 23:32.44 | francisvgarcia | http://pastebin.com/vtThanNc |
| 23:33.48 | [TK]D-Fender | francisvgarcia: Show us the call |
| 23:38.55 | *** part/#asterisk ChannelZ (channelz@burner.com) |
| 23:38.58 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 23:39.00 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 23:39.01 | ChannelZ | whoops |
| 23:39.07 | p3nguin | Special Price Analysis and Marketing = SPAM |
| 23:45.45 | [TK]D-Fender | And there goes the last of the evidence... |
| 23:57.37 | carrar | adios amigo |
| 23:59.13 | p3nguin | So is there a setting for the card that causes it to wait for two rings? I would have thought that is all dial plan. |