| 00:08.05 | *** join/#asterisk globetrotterz (globgfgetr@pool-72-80-171-83.nycmny.fios.verizon.net) |
| 00:32.31 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 00:38.16 | *** join/#asterisk nueral8 (~nueral8@pool-108-12-102-200.hstntx.dsl-w.verizon.net) |
| 00:42.07 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 00:48.05 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 00:57.20 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 01:00.00 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 01:00.28 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 01:00.30 | *** join/#asterisk rene- (~rene@189.214.91.84.cable.dyn.cableonline.com.mx) |
| 01:00.46 | rene- | hey guys, anybody using adaptive_cdr features of asterisk 1.6+ ? |
| 01:02.07 | rene- | my question is, the user added fields, do they get populated with an SQL update after asterisk has created the cdr record using the insert with just the basic fields? or is everything deferred until we hit the h extension and everything is inserted at that point? |
| 01:06.20 | rene- | ohh i am looking at endbeforehexten=no setting of cdr.conf |
| 01:07.40 | rene- | this is interesting, now the question is if endbeforehexten is set to yes, will the adaptive cdr engine be able to write to cdr records, with an update? |
| 01:15.49 | *** join/#asterisk neurosys (~neurosys@c-98-219-76-3.hsd1.fl.comcast.net) |
| 01:17.26 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 01:19.59 | *** part/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 01:29.26 | *** join/#asterisk f2Knight (~ben@c-24-21-102-161.hsd1.or.comcast.net) |
| 01:31.23 | f2Knight | Q: How do I make the DIAL command stay in the current context? |
| 01:35.01 | *** join/#asterisk nightrid3r (~kvirc@91.176.0.167) |
| 01:37.22 | *** join/#asterisk fyrfaktry (~dread@adsl-070-145-120-039.sip.pns.bellsouth.net) |
| 01:40.58 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 01:48.15 | *** join/#asterisk nightrid3r (~kvirc@91.176.53.237) |
| 02:12.58 | The_REV | hey... has anyone attempted or have a good write up on howto control conf file with GIT or SVN? |
| 02:38.26 | ChannelZ | f2Knight: It does |
| 02:39.52 | f2Knight | ChannelZ, it does?? |
| 02:40.26 | ChannelZ | There's nothing inherent in Dial that would make it switch contexts. You might be doing some other weird things that cause it to |
| 02:40.40 | f2Knight | ChannelZ, Okay , yes I am got a moment? |
| 02:41.56 | ChannelZ | Local channels for instance, |
| 02:42.57 | f2Knight | ChannelZ, This is what I am doing... The queue is calling dynamic agents at local/s@agents |
| 02:43.22 | f2Knight | ChannelZ, the agents context then does a DIAL() to the external agent. |
| 02:44.04 | f2Knight | ChannelZ, I need to continue running logic in the agents context after the call is disconnected by EITHER party, |
| 02:46.03 | f2Knight | ChannelZ, the main reason for this is that I need to record to a database the inbound callerID with the number of minuets they spoke with an agent. Capturing time before and after the actual dial to the agent will let me keep track how long that call was in a realtime database, |
| 02:46.43 | f2Knight | ChannelZ, I am open to other ways of gathering this information just I have been hitting walls all week and I need to come up with some type of solution to this issue fast. |
| 02:47.36 | ChannelZ | Are you using the F and g flags in the dial? |
| 02:50.16 | f2Knight | ChannelZ, I have tried the g But that doesnt seem to work all the time., and I don't fully understand how to use the F, g only seems to work in, some of the times. |
| 02:51.01 | f2Knight | ChannelZ, but 'g' is in my options yes |
| 02:51.15 | ChannelZ | well g is if the destination channel hangs up (which would be your agent I guess) and F is if the source channel hangs up (your caller) |
| 02:55.59 | ChannelZ | I'm not actually sure if doing that on a local channel will return back to the original context/priority when done... |
| 02:57.15 | ChannelZ | You're probably better off coming up with different logic than using Local channels for your queue agents and instead call their devices directly so you can do all your logic in one place |
| 03:00.08 | ChannelZ | (although I just tried it here and it's working fine) |
| 03:01.11 | f2Knight | ChannelZ, I am open to suggestions, my issue is the agents are not "in-house" and will not have SIP phones, they will instead be using a callback, That was my reason for using local/ with the queues... |
| 03:02.00 | f2Knight | ChannelZ, can you paste bin me what you tried? maybe its just a small change from what I have been stumped against. |
| 03:02.12 | ChannelZ | well as I said, it seems to be working here (although I am not testing from a queue, I'm not sure that would matter) - Dial(Local/whatever@wherever,,Fg) followed by a NoOp(And then this happened) |
| 03:02.48 | ChannelZ | If I dial the exten which dials the Local channel (which is an exten dials a SIP channel in another part of the dialplan/context) when either side hangs up I see my NoOp execute |
| 03:03.59 | *** join/#asterisk file (~file@neutrino-114-86.joshua-colp.com) |
| 03:05.29 | f2Knight | ChannelZ, what did you use for your F context? |
| 03:05.57 | f2Knight | ChannelZ, from my understanding when using F you have to provide a context to put the call in to. |
| 03:06.05 | ChannelZ | http://pastebin.com/izWaTexK |
| 03:06.17 | ChannelZ | There are two syntaxes for F |
| 03:09.11 | f2Knight | ChannelZ, I am going to paste bin you in a min what I got, maybe you can see what I am missing. (cause your example did work) Maybe its the Queue use I don't know but its driving me crazy |
| 03:12.51 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:24.30 | ChannelZ | well I'm shutting down for a bit to vacuum out my server, fans keep revving up |
| 03:45.52 | *** join/#asterisk adolfomaltez (~taro@190.86.151.235) |
| 03:49.10 | f2Knight | ChanServ, helps if i pasted it. http://pastebin.com/Ft8g0Css |
| 03:57.03 | f2Knight | http://pastebin.com/Ft8g0Css |
| 04:15.41 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
| 04:37.22 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:38.59 | *** join/#asterisk wizzr (~bingr@50-57-115-214.static.cloud-ips.com) |
| 04:39.34 | wizzr | what's one of the better sip providers to use with asterisk for a business that needs unlimited intl calling? |
| 04:46.30 | wizzr | ?!?!?!?! |
| 04:55.29 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 04:56.59 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:03.18 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 05:09.24 | *** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net) |
| 05:23.51 | ectospasm | wizzr: that's usually a bad question to ask. Not many providers will quote a low price for unlimited international calling, IIRC. If you have a really high volume, you might be able to lock in rates by buying 1M-minute chunks |
| 06:07.08 | *** join/#asterisk xNinja (~xNinja@74.115.1.48) |
| 06:08.13 | xNinja | hello...is there a simple all in one box which is sip server or trunk server whatever i am new to this but just wants to connect a box to my dsl to make free calls while i am outside home using my home line |
| 06:08.44 | xNinja | through mobile which is connected to the net |
| 06:10.04 | xNinja | i did install asterisk/asterisknow before but still seems to have much work to make it done i dont need complicated stuff just normal dial thats it |
| 06:14.05 | ChannelZ | Not really that I'm aware of |
| 06:14.58 | xNinja | why it has to be so complicated and configured from scratch its just a simple need there must be something simple as that! |
| 06:15.47 | ChannelZ | What is your mobile? Android? |
| 06:16.16 | xNinja | iphone |
| 06:16.28 | xNinja | but noproblem if thereis a solution i will change to android |
| 06:16.30 | ChannelZ | Is there a SIP phone app? |
| 06:16.50 | xNinja | yeah a lot |
| 06:17.01 | ChannelZ | ok.. well you're probably better off signing up with an ITSP and just using them rather than trying to run it yourself off your home line. |
| 06:18.52 | xNinja | why? plus i want to run it in my home so that it will be totally free! when i use another sip service that means it will be international call! |
| 06:20.13 | ChannelZ | No, you get an ITSP in whatever country it is you're wanting to make calls in. The link between your iPhone and the ITSP is over the net. |
| 06:20.29 | ChannelZ | To interface with your home line, you need an ATA. Not "totally free." |
| 06:20.58 | ChannelZ | Then you need to configure Asterisk or something else to interface the two, which you don't want to learn how to do. |
| 06:21.35 | xNinja | i heard some ppls using their phone line + dsl with asterisk box and thats it! |
| 06:21.44 | xNinja | what i am asking is same but simple |
| 06:22.06 | ChannelZ | You heard wrong if you're talking about an analog phone line |
| 06:22.23 | ChannelZ | You need a hardware interface |
| 06:22.58 | xNinja | yeah like digium ? |
| 06:25.00 | xNinja | i just want a call transfer or voip gateway... |
| 06:25.30 | xNinja | from mobile through net i connect to my home server and using the land line to call any number |
| 06:25.37 | xNinja | what you suggest for that ? |
| 06:27.29 | ChannelZ | Linksys makes a couple of boxes, ~$60 US |
| 06:27.31 | xNinja | what do i need for that like...fxo fxs......too many stuff lol cant sort them |
| 06:29.09 | ChannelZ | You need an FXO |
| 06:30.31 | xNinja | i did get spa3000 or 3200 something like that but so many configuration and didnt find something clear to how to do it |
| 06:32.44 | ChannelZ | well yes this stuff is tricky to get running |
| 06:33.40 | xNinja | is there somewhere where i can get an embedded system for my needs and ready configured ? just changing my ip/open port on my router/set passwords ? |
| 06:34.01 | xNinja | i dont mind pay little more for that depend on the price |
| 06:39.36 | ChannelZ | I don't know of any to do what you want that aren't hundreds and hundreds of dollars |
| 06:43.15 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 06:43.58 | f2Knight | ChannelZ, how did the server vacumining go? |
| 06:44.41 | f2Knight | ChannelZ, not sure if you got the link or not before you signed out http://pastebin.com/Ft8g0Css |
| 06:45.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 06:46.01 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 07:23.05 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 07:23.57 | sawgood | [Jul 24 00:22:03] WARNING[27520]: chan_skype.c:1071 request: No active Skype user 'myaccount-admin' |
| 07:24.09 | sawgood | Any tips on what file I need to edit to make this error go away? |
| 07:24.31 | sawgood | I tried /etc/asterisk/chan_skype.conf, but I do not think that is correct |
| 07:46.46 | ChannelZ | f2Knight: lovely I hope, I moved a fan too to hopefully help |
| 07:47.32 | ChannelZ | sawgood: I'm sure that is correct |
| 07:48.11 | f2Knight | ChannelZ, I hope it works for you . I got a Dell Precession workstation that has like 4 fans IN it, and a a box fan outside to keep it cool (Dual quad core xeons make a good heater) |
| 07:48.15 | ChannelZ | Did you make a Skype user to use with SFA through the business control panel? |
| 07:48.52 | ChannelZ | My case is not very good. There was a fan in the front but there's almost no ventilation through the case in front so it was kind of pointless |
| 07:49.13 | f2Knight | ChannelZ, hate to bug you but I am still smashing my head against a wall here, I cant figure out anyway to get the infor I am looking for :( |
| 07:51.40 | ChannelZ | I'd have to setup a queue and test it to see if I'm getting the same thing |
| 07:52.11 | ChannelZ | Give me a bit I need to run off for a min |
| 07:53.44 | f2Knight | ChannelZ, no problem. I can get in to more detail when you get back I am going to be up all night working on this again other wise. |
| 07:53.56 | *** part/#asterisk godmachine-x6 (~godmachin@h18.171.140.67.dynamic.ip.windstream.net) |
| 07:53.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 07:57.39 | *** join/#asterisk godmachine-x6 (~godmachin@h18.171.140.67.dynamic.ip.windstream.net) |
| 08:00.17 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 08:12.25 | ChannelZ | f2Knight: try adding /n to the end of your Local queue members, like "Local/s@test/n" |
| 08:13.27 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 08:13.42 | f2Knight | ChannelZ, okay let me give that a try.. whats that supposed to do btw? |
| 08:14.53 | ChannelZ | well it's a little hard to explain, internally asterisk tries to remove the local channel from the chain of what it's doing.. and this causes dial options to get lost |
| 08:15.41 | ChannelZ | Why this works in a straightforward dial (like that little sample I posted hours ago) but not from the queue I'm not sure |
| 08:17.34 | sawgood | I am now able to make an outbound call from Asterisk to a Skype account |
| 08:18.10 | ChannelZ | praise jesus! |
| 08:18.32 | sawgood | The concern now is ... how to seutp a dialplan which will allow 'name' dialing from a SIP phone ... |
| 08:18.48 | sawgood | 400 million Skype usernames you know! |
| 08:18.55 | f2Knight | ChannelZ, That almost looks like it is working now... A odd way that it handles something though is it throughts the default 'h' before returning back to the test context when hanging up... but my epoch times seem to be showing up now.. |
| 08:19.16 | ChannelZ | sawgood: setting up the dialplan is easy, typing usernames into your phone is the part you have to figure out |
| 08:19.23 | f2Knight | ChannelZ, I am going to do some testing but by George that might have been what I have been fighting with all week. |
| 08:19.55 | ChannelZ | well yes the 'h' extension will add another step to the process.. |
| 08:21.00 | f2Knight | ChannelZ, do you know how this might work if the Queue has the option for the user to exit the queue? I will try it myself but was wondering if you knew off the top of your head. |
| 08:21.04 | sawgood | ChannelZ: Do you think others have put stuff in place like this though? |
| 08:21.43 | *** join/#asterisk irroot (~irroot@41.55.68.171) |
| 08:21.49 | ChannelZ | sawgood: no idea. Most people use SFA to making phone calls, not Skype-to-Skype calls |
| 08:22.44 | sawgood | ChannelZ: Well how about incoming Skype calls ... (at least this would allow any Skype user to call your Asterisk box) ... incoming calls would be valueable! |
| 08:23.04 | ChannelZ | You can make numeric extensions to dial set usernames, or maybe your phones will let you enter letters into speed dials through their web interface.... |
| 08:23.10 | ChannelZ | Incoming calls work fine |
| 08:23.41 | sawgood | How does "Skype" know where to send the call to (the IP address of your Asterisk box) ... where is this configured at? |
| 08:23.44 | f2Knight | sawgood, I have put a box up using googleVoice as outbound trunks, inbound works too though sometimes there is a connection delay on inbound |
| 08:24.09 | ChannelZ | It's in your chan_skype - whatever Skype accounts you configure for it to be playing client for |
| 08:24.48 | ChannelZ | You have to use the Skype Business Control Panel to make the user(s) though, you can't just use existing normal Skype accounts |
| 08:24.55 | sawgood | oh excellent .... so if I have 15-20 Skype accounts inside /etc/asterisk/chan_skype.conf (they are 'online' waiting for Skype calls) |
| 08:25.08 | sawgood | ChannelZ: understood ... thanks for that tip! |
| 08:25.19 | ChannelZ | yes, SFA actually acts as the Skype client for those users, logging into the network as them |
| 08:25.43 | ChannelZ | When someone calls them (whether via a DID you have through Skype or just Skype-to-Skype to their username) it rings into Asterisk per your config |
| 08:26.02 | f2Knight | I heard that skype for asterisk was being dropped |
| 08:26.14 | ChannelZ | Yes |
| 08:26.14 | sawgood | so, say an existing company has a Skype account called something like (mybusiness) ... if we create a new Skype account for them called something like (mybusiness-asterisk) (a new totally different account) ... we are ok then ... |
| 08:26.17 | *** join/#asterisk sourcode (~code@ppp-115-87-217-89.revip4.asianet.co.th) |
| 08:26.31 | f2Knight | I think that has to do with Microsoft buying skype and not wanting to let asterisk be in the game |
| 08:26.37 | ChannelZ | Skype didn't renew the licensing with Digium and will only support it for another year at which point it may or may not continue to work |
| 08:26.47 | sawgood | of course 'creating' this new account via the Skype Manager 'portal' |
| 08:26.56 | ChannelZ | sawgood: correct |
| 08:27.09 | sawgood | It might be 2 years of support (which is what I read when I bought the license) |
| 08:27.15 | ChannelZ | I can't remember |
| 08:27.43 | f2Knight | I just heard about the drop the other day on an rss feed. |
| 08:27.55 | sawgood | Is there anyway to 'merge' and existing Skype account into the Skype Manager process (which has replaced the Skype Business Control Panel) ... |
| 08:27.57 | ChannelZ | I guess it is 2 years |
| 08:28.17 | sawgood | This way, you can keep your 'cool' and existing' Skype account without having to make a new one? |
| 08:28.17 | ChannelZ | f2Knight: They will stop selling channel licenses on Tuesday |
| 08:28.59 | sawgood | Problem is ... you probably cannot be logged in 'two places' ... so it would be best to have a separate stand alone SFA type Skype account in place |
| 08:29.03 | ChannelZ | sawgood: I don't think so.. I seem to recall the panel lets you 'invite' existing users but internally they aren't the same and don't work |
| 08:29.11 | f2Knight | ChannelZ, Ahh |
| 08:29.19 | sawgood | ChannelZ: I believe you are 110% correct! |
| 08:29.52 | ChannelZ | I think it mentions that in the docs somewhere or maybe a note on the forum |
| 08:29.59 | sawgood | only new accounts which do not exist yet AND are added as 'members' or 'clients' from your Skype Manager account will work |
| 08:30.21 | f2Knight | ChannelZ, I am going to go put this through some tests, but I think your suggestion for /n hit it on the head. (how you knew or found that I don't know - been looking for help for a week on it.) |
| 08:31.02 | ChannelZ | Search the Wiki for Local channels, I think it's documented and explained better than I did |
| 08:31.19 | f2Knight | wiki being voip-info? |
| 08:31.35 | ChannelZ | no the new asterisk.org one |
| 08:31.36 | ChannelZ | ~wiki |
| 08:31.37 | sawgood | So for 'inbound' SFA to 'work' ... do I need a context in sip.conf for the SFA account and/or just something in extensions.conf for the SFA name? |
| 08:31.44 | ChannelZ | bahh |
| 08:31.51 | sawgood | ChannelZ: Have you ever met Russell Brand? |
| 08:32.00 | ChannelZ | wiki.asterisk.org |
| 08:32.08 | ChannelZ | sawgood: yeah |
| 08:32.11 | ChannelZ | points at russellb |
| 08:32.17 | sawgood | oh ... really? |
| 08:32.38 | ChannelZ | really.. though he's leaving Digium |
| 08:33.00 | f2Knight | ChannelZ, ahh! will do. Thank you for taking the time to help! |
| 08:33.00 | sawgood | I enjoyed both the 3rd edition of Asterisk Guide and the Asterisk Cookbook |
| 08:33.03 | ChannelZ | if you mean met in person, no |
| 08:33.38 | *** part/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 08:33.52 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 08:34.00 | ChannelZ | f2Knight: sure good luck |
| 08:34.56 | f2Knight | Thank you ChannelZ, |
| 08:35.37 | ChannelZ | dang I need to clean up my own dialplan, this thing is a train wreck |
| 08:35.51 | f2Knight | sawgood, I meet Mark Spenser a few times in person. |
| 08:36.00 | sawgood | I had a girlfriend once who was a train wreck |
| 08:36.58 | f2Knight | ChannelZ, My production server is that way I have a new one I am moving to with realtime accounts, but I told my self all incoming DID's will call macros to do the job and be clean.... well see how long that lasts lol |
| 08:37.20 | f2Knight | I am switching everything to AEL though cause i think its easier to read and do stuff with |
| 08:37.54 | ChannelZ | I just have a ton of test extensions and junk in here I need to clean out |
| 08:38.09 | ChannelZ | my home one does relatively little |
| 08:38.24 | irroot | channelZ f2knight have a dialplan from hell of sorts mostly for the gui we supply has overkill options |
| 08:38.50 | ChannelZ | one size fits all! |
| 08:38.58 | f2Knight | irroot, what gui? |
| 08:39.19 | irroot | yeah 1 size fits all couple func_odbc calls too decide |
| 08:39.36 | irroot | 150 systems deployed |
| 08:39.43 | f2Knight | I used to use Druid by voiceroute, (looks like they abandoned what was a good project) because it didn't mutilate your configs |
| 08:39.45 | sawgood | Would something like this work for an extensions.conf context (to 'receive' incoming SFA calls): exten => skypeaccount,1,Dial(SIP/1111/${EXTEN}) |
| 08:39.54 | ChannelZ | sawgood: yes |
| 08:40.28 | irroot | we built a linux distro for the local market here with a PHP frontend |
| 08:40.35 | ChannelZ | though ${EXTEN} has little meaning |
| 08:41.02 | sawgood | [Jul 24 01:36:45] WARNING[28211]: pbx.c:4852 __ast_pbx_run: Channel 'Skype/accountname-b795ebb8' sent into invalid extension 's' in context 'default', but no invalid handler |
| 08:41.19 | sawgood | This is the error I get from an SBC trying to send the call to an Asterisk server ... |
| 08:41.33 | sawgood | I have the context definied in extensions.conf |
| 08:41.46 | ChannelZ | default? |
| 08:41.57 | sawgood | I do not know why it says 'default'? |
| 08:42.00 | irroot | have a "s" exten in default ?? |
| 08:42.19 | ChannelZ | Skype calls go into whatever context you tell them to in chan_skype.conf |
| 08:42.29 | sawgood | oh ... inocming calls ... |
| 08:42.30 | ChannelZ | I generally put them in their own 'skype' context and then handle them from there |
| 08:42.30 | irroot | the skype conf needs a context option added |
| 08:42.31 | sawgood | my bad ... |
| 08:42.35 | sawgood | sorry! |
| 08:43.01 | f2Knight | ChannelZ, just read up on the /n in the wiki.. curious though /b ? where would you use something like that? |
| 08:43.56 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 08:44.01 | ChannelZ | hmm I don't know of b |
| 08:44.09 | ChannelZ | what's the link |
| 08:44.21 | f2Knight | https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers |
| 08:44.49 | sawgood | how to 'reload' chan_skype.conf without having to restart Asterisk? |
| 08:45.12 | f2Knight | it says something about connecting the orginal channel.. I don't think that would work for my needs as my original channel is a Queue, well .. the out side call really but not sure thought I would ask. |
| 08:45.33 | ChannelZ | try module reload chan_skype |
| 08:45.47 | f2Knight | sawgood, some modules have to restart or do a module reload chan_skype.so |
| 08:45.54 | sawgood | ty! |
| 08:46.04 | ChannelZ | f2Knight: I dunno, by the explanation I think it means that it causes the channel to look like the original channel, rather than Local/whatever |
| 08:46.13 | f2Knight | sawgood, but that does not work on some ...like gtalk.. you have to restart, :( |
| 08:46.36 | ChannelZ | you can reload gtalk |
| 08:46.58 | ChannelZ | manually.. just unload it and then load it again - don't have to restart asterisk as a whole |
| 08:47.34 | f2Knight | when i was setting up my google voice it wouldn't allow it too.. or maybe it was the jingle/jabber |
| 08:47.59 | ChannelZ | it won't let you module reload, but you can module unload chan_gtalk and the module load chan_gtalk |
| 08:48.02 | ChannelZ | same with res_jabber |
| 08:48.24 | f2Knight | yea thats what i did |
| 08:48.42 | f2Knight | later.. much latter lol |
| 08:49.19 | f2Knight | it would be nice if they supported realtime interfaces for it so I could add them on the fly with out dropping them on a reload |
| 08:49.55 | ChannelZ | it'll probably get there eventually as the modules mature |
| 08:51.03 | irroot | once some one is frustrated enough it will be done :)\ |
| 08:51.39 | irroot | think with G+ taking off it will be sooner than latter |
| 08:52.23 | irroot | for example i added DTMF support to chan_gtalk just to see if i could use it for IVR / interbranch calls |
| 08:52.41 | irroot | thats how progress happens |
| 08:57.59 | f2Knight | it didn't support DTMF? |
| 08:58.20 | irroot | nah that was long time back |
| 08:58.32 | f2Knight | btw, if you guys want to add me to your G+ gplus.to/tuxpowered |
| 08:58.58 | ChannelZ | I'm not on that mess |
| 08:59.09 | irroot | https://issues.asterisk.org/view.php?id=9401 |
| 09:00.14 | f2Knight | ChannelZ, lol I must confess It is more technical then FB, now thats a mess. but I have meet more tech people on G+ then I have other places. (just no asterisk guys yet other then leifmadsen |
| 09:00.16 | irroot | hehe 2007 |
| 09:00.34 | irroot | FB is like a uni frat house |
| 09:00.50 | ChannelZ | Like anything it's what you make of it |
| 09:02.06 | irroot | added f2knight |
| 09:02.28 | f2Knight | ChannelZ, this is true |
| 09:03.04 | sawgood | In order to receive incoming calls from Skype, will I need an entry in sip.conf? |
| 09:03.44 | irroot | no sawgood unless |
| 09:04.07 | irroot | unless you using the sip proxy |
| 09:04.22 | irroot | but you using chan_skype ?? |
| 09:04.33 | sawgood | yes I am using chan_skype |
| 09:04.41 | ChannelZ | No sip has nothing to do with Skype |
| 09:04.55 | sawgood | the context= (should this be the place in extensions.conf) where the calls should look at? |
| 09:05.44 | ChannelZ | the context, yes |
| 09:06.02 | sawgood | so if it = from-trunk (right now) that could be a concern |
| 09:06.21 | ChannelZ | I don't know, I know nothing about the rest of your config |
| 09:07.15 | sawgood | in chan_skype.conf, I put the context=Skype-incoming |
| 09:07.30 | ChannelZ | you can direct the calls exactly though by setting context=whatever and exten=whatever for each of your skype 'peers' |
| 09:07.32 | sawgood | in extensions.conf, I created a context called [Skype-incoming] |
| 09:07.59 | ChannelZ | sounds fine |
| 09:08.02 | irroot | cool now sawgood put context=Skype-incoming into the config |
| 09:08.24 | irroot | to make the calls go there |
| 09:08.48 | sawgood | I did ... I am trying now ... thank you! |
| 09:13.14 | sawgood | Here is a copy/paste of the 4 lines I get when I 'receive' a call from Skype (sorry for the long paste) |
| 09:13.24 | sawgood | == Starting Skype/skypeme100-b6d009f8 at from-trunk,skypeme100,1 failed so falling back to exten 's' |
| 09:13.24 | sawgood | == Starting Skype/skypme100-b6d009f8 at from-trunk,s,1 still failed so falling back to context 'default' |
| 09:13.24 | sawgood | [Jul 24 02:11:14] WARNING[28647]: pbx.c:4852 __ast_pbx_run: Channel 'Skype/skypeme100-b6d009f8' sent into invalid extension 's' in context 'default', but no invalid handler |
| 09:13.24 | sawgood | [Jul 24 02:11:14] NOTICE[28647]: core.cpp:2150 sfa_call_hangup: ending call |
| 09:14.07 | sawgood | in chan_skype.conf, I have context=Skype-incoming |
| 09:14.24 | sawgood | in extensions.conf, I have [Sype-incoming] |
| 09:14.44 | sawgood | I meant [Skype-incoming] |
| 09:14.53 | ChannelZ | first don't paste big chunks here |
| 09:15.14 | sawgood | I do know better than that ... sorry again! |
| 09:15.21 | ChannelZ | second did you reload because it looks like nothing has changed |
| 09:15.58 | irroot | its going to from-trunk |
| 09:16.16 | irroot | skypeme100, |
| 09:16.18 | sawgood | I see that ... and I do not kno why? |
| 09:16.26 | ChannelZ | because you didn't reload |
| 09:16.48 | sawgood | thank you (reload was needed) |
| 09:16.53 | sawgood | things are working 100% now ... |
| 09:16.54 | sawgood | sorry! |
| 09:20.06 | sawgood | I think Skype 'must' use G729 only |
| 09:21.19 | ChannelZ | no it'll do ulaw too |
| 09:21.36 | irroot | i always thought skype prefered ilbc |
| 09:21.42 | ChannelZ | It supports a few |
| 09:22.05 | ChannelZ | Asterisk 10 I think does SILK now (but dunno if they've even built SFA for the beta) |
| 09:22.35 | ChannelZ | Oh, they do. |
| 09:24.09 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 09:27.52 | ChannelZ | okey bed time |
| 09:29.11 | sawgood | ChannelZ: Thank you ... inbound/outbound SFA working ... thank you! |
| 09:29.16 | sawgood | good night ... |
| 09:29.51 | f2Knight | ChannelZ, good night, thanks again for your help |
| 09:32.17 | irroot | cheers have a good one |
| 09:33.43 | *** part/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 09:43.07 | *** join/#asterisk wonderworld (~ww@port-92-201-75-20.dynamic.qsc.de) |
| 10:06.20 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 10:08.19 | eject_ck | Hi all, I have 10 outgoing sip trunks and want to use round robin trunks selection in my extension. What's best way to achieve this goal ? Only thing what I have in mind at the moment is global variable usage. |
| 10:10.53 | irroot | you could put them into a ACD |
| 10:10.57 | irroot | app_queue |
| 10:11.03 | irroot | and use that |
| 10:11.42 | irroot | have to ask why so many ?? you multihomed |
| 10:12.10 | eject_ck | irroot: that's outbound SIP trunks |
| 10:12.16 | eject_ck | ACD ? |
| 10:12.17 | irroot | yeah |
| 10:12.27 | irroot | Automatic Call Distribution |
| 10:12.30 | eject_ck | it works for outgoing as well ? |
| 10:14.01 | eject_ck | how it works for outbound calls ? |
| 10:14.24 | irroot | outgoing and incoming are all the same |
| 10:14.38 | irroot | its a point of perspective |
| 10:14.56 | irroot | yes you can route calls out via app_queue |
| 10:15.15 | irroot | all you do is load the trunks as q memebers |
| 10:15.29 | irroot | place a call to Queue not Dial |
| 10:15.37 | irroot | and off it goes |
| 10:15.45 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 10:22.41 | eject_ck | irroot: checking for examples |
| 10:24.49 | eject_ck | irroot: can't figure how it should be in dialplan |
| 10:25.28 | irroot | exten => ....,n,Queue(<Qname>) |
| 10:25.38 | irroot | see "core show application queue" |
| 10:25.52 | irroot | set up a queue with members that are the trunks |
| 10:28.13 | eject_ck | ok, I have queue with members SIP/1212XXX and SIP/1212YYY called OUTQUEUE, I want to make outgoing call |
| 10:28.53 | eject_ck | for examle |
| 10:28.53 | eject_ck | exten => _X.,1,Queue(OUTQUEUE) |
| 10:29.15 | eject_ck | where should I paste my ${EXTEN} in this case? |
| 10:29.36 | eject_ck | with Dial it will looks like DIAL(SIP/1212XXX/${EXTEN}) |
| 10:37.20 | *** join/#asterisk remx (~remx@coreapps.us.gnics.net) |
| 10:37.33 | eject_ck | any clue ? |
| 10:37.54 | remx | Hello. Does Asterisk need an MTA for anything? |
| 10:42.08 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 10:50.15 | tzafrir_laptop | remx, for voicemail |
| 10:50.22 | tzafrir_laptop | (optional) |
| 10:51.40 | remx | tzafrir_laptop: Does it connect to local port 25, honour mail.rc or is the MTA specified in a configuration file? |
| 10:53.27 | irroot | eject_ck ah yes |
| 10:53.42 | tzafrir_laptop | It uses sendmail (the command) by default. You can give it an alternative command |
| 10:53.49 | irroot | need to set a variable and use a local chan |
| 10:55.58 | remx | tzafrir_laptop: Thank you very much |
| 10:56.05 | eject_ck | irroot: what var you mean ? |
| 10:56.25 | irroot | its been a while .... |
| 10:56.52 | irroot | set a var in the DB and then use a local channel to proxy the sip side |
| 10:57.15 | irroot | set up a context that the local channels use |
| 10:57.20 | irroot | [outgoing] |
| 10:57.45 | irroot | perhaps |
| 10:57.55 | irroot | then get the exten here and dial out |
| 11:07.45 | *** join/#asterisk nightrid3r (~kvirc@91.176.53.237) |
| 11:10.38 | remx | If I may, I'm constantly seeing this message in the logs, "[2011-07-24 12:02:19] NOTICE[32025] indications.c: Removed default indication country 'uk'" what can I do about this? I know it's minor but I'm getting anal about it |
| 11:41.03 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:597a:fcbb:1dfc:ef3b) |
| 12:03.14 | *** join/#asterisk godmachine-x6 (~godmachin@h239.250.19.98.dynamic.ip.windstream.net) |
| 12:07.04 | *** join/#asterisk box (~box@unaffiliated/box) |
| 12:07.09 | box | can someone please tell me what the hell this means Unable to register extension 's', priority 1 in 'macro-outisbusy', already in use |
| 12:07.53 | box | my extensions work perfectly fine :s |
| 12:12.35 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 12:16.20 | irroot | box you have a duplicate |
| 12:16.26 | irroot | in the config |
| 12:31.17 | box | irroot: I don't |
| 12:31.47 | irroot | ah but you do asterisk says so ... |
| 12:31.57 | irroot | look at any includes too |
| 12:32.20 | irroot | if you include a context with s,1 it will conflict with the s,1 in the context |
| 12:36.51 | box | irroot: sir my sip.conf has only 1 extension and 1 trunk, here is my extension: http://pastebin.com/TmiTFfQG |
| 12:37.15 | irroot | its in extensions.conf you must look |
| 12:37.24 | box | sorry, i meant extensions |
| 12:37.45 | box | but as you probably have far greater knowledge, I can search through my configuration files for a term, if you could tell me what that term would be? |
| 12:40.03 | irroot | post extensions.conf then |
| 12:40.07 | irroot | i can see |
| 12:40.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 12:40.15 | box | irroot: ok sir bare with me |
| 12:46.18 | remx | box are you using freepbx? |
| 12:46.40 | box | remx: yes sir, can i ask why? |
| 12:46.56 | remx | ^^ #freepbx |
| 12:47.38 | box | remx: did not know there was a channel for them! im new to this :$ |
| 12:48.39 | *** part/#asterisk box (~box@unaffiliated/box) |
| 12:49.04 | irroot | ~freepbx |
| 12:49.04 | 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 |
| 13:04.39 | *** join/#asterisk ggd (~ggd@pool-173-72-205-2.clppva.fios.verizon.net) |
| 13:07.24 | *** join/#asterisk [gnubie] (~marvin.pa@cm36.kappa8.maxonline.com.sg) |
| 13:07.32 | [gnubie] | waves |
| 13:08.02 | [gnubie] | i'm running asterisk-1.8.5 from digium's .deb repository on my squeeze box |
| 13:08.36 | [gnubie] | my asterisk box is behind the firewall, hence it's using a single private ip address |
| 13:11.36 | [gnubie] | i noticed that when i set the externip's value to the asterisk box's private ip address, when the sip phones within the same private lan calls to a pots number, the callee can hear the voice from the caller.. in short, the call is ok.. |
| 13:12.50 | [gnubie] | but the sip accounts that are coming from the internet that are registered to my asterisk box do not hear any voice.. |
| 13:13.10 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 13:14.48 | [gnubie] | now, if i set the externip to the public ip address of the firewall/router, those sip accounts that are registered to my asterisk box that are coming from the internet have no problem with the voice.. the problem now is on the sip phone within the private lan.. |
| 13:16.42 | [gnubie] | please advice. what shall i do here? |
| 13:17.23 | *** part/#asterisk remx (~remx@coreapps.us.gnics.net) |
| 13:36.56 | Gugge | [gnubie]: localnet |
| 13:37.21 | [gnubie] | localnet=192.168.1.0/255.255.255.0 |
| 13:37.28 | [gnubie] | it's already set |
| 13:38.32 | Gugge | do your local phones connect to the external or the internal ip? |
| 13:39.44 | Gugge | maybe turning off reinvites, to force the audio through asterisk |
| 13:40.35 | [gnubie] | internal ip |
| 13:40.55 | [gnubie] | where shall i turn off the reinvites? |
| 13:41.56 | Gugge | dont know if canreinvite is still valid in 1.8 |
| 13:42.04 | Gugge | search the sample sip.conf maybe :) |
| 13:43.00 | Gugge | i think it was renamed to directmedia or something like that |
| 13:43.31 | [gnubie] | ok |
| 13:45.04 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 13:46.43 | jasoncarter | hi guys. I have a sip account with a provider that I'd like to add as an extension for the purpose of adding it to a queue. is this possible? |
| 13:48.43 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 13:54.04 | *** join/#asterisk ChUbB (~IceChat7@62.31.213.230) |
| 13:56.45 | [gnubie] | Gugge: by the way, my scenario is presented as a diagram at http://imagebin.org/164662 |
| 13:57.15 | Gugge | jasoncarter: any SIP/ peer is a valid queuemember |
| 13:57.42 | Gugge | or you could even add a LOCAL/ member, and let it go through some dialplan logic |
| 13:59.17 | Gugge | [gnubie]: yep that was how i pictured it too. But im just guessing, i never put asterisk behind NAT |
| 14:01.04 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:04.45 | [gnubie] | Gugge: my asterisk box is using an intel atom based motherboard that comes only with a single on-board ethernet port |
| 14:05.00 | [gnubie] | the updated diagram is at http://imagebin.org/164663 |
| 14:05.21 | Gugge | My firewall with 5 networks has a single NIC |
| 14:05.25 | Gugge | VLANS is nice :P |
| 14:08.03 | [gnubie] | both for your internet and lan connections? |
| 14:08.40 | Gugge | What is your question? |
| 14:08.52 | Gugge | Yes, VLANS is nice both for internet and lan connections. :P |
| 14:10.25 | [gnubie] | this is going to be ot here but i'm interested to know how vlan works on your side.. i don't have a manageable switch here and i do have a single access point/router/firewall |
| 14:19.57 | Gugge | you need network equipment that supports vlans ofcourse |
| 14:30.36 | [gnubie] | Gugge: yeah. that's the problem. |
| 14:38.12 | [gnubie] | gtg now.. thanks.. |
| 14:38.17 | [gnubie] | waves |
| 14:38.20 | *** part/#asterisk [gnubie] (~marvin.pa@cm36.kappa8.maxonline.com.sg) |
| 14:50.37 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 15:02.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 15:07.34 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-131-213.home.otenet.gr) |
| 15:34.09 | *** join/#asterisk DelphiWorld (~Miranda@openvpn/user/DelphiWorld) |
| 15:34.13 | DelphiWorld | slaps pabelanger around a bit with a large trout |
| 15:34.14 | DelphiWorld | :) |
| 15:35.26 | *** join/#asterisk iscario (~div@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 15:35.47 | pabelanger | DelphiWorld: Thank you?!? |
| 15:35.55 | DelphiWorld | pabelanger: :P |
| 15:36.15 | DelphiWorld | pabelanger: just for a fun time |
| 15:37.54 | pabelanger | DelphiWorld: Ah, I see |
| 15:48.03 | DelphiWorld | pabelanger: :) |
| 15:48.22 | DelphiWorld | pabelanger: in US who's your cell carrier ? |
| 15:48.47 | pabelanger | DelphiWorld: I live in Canada, but it was Rogers |
| 15:48.56 | DelphiWorld | pabelanger: and is still roger ? |
| 15:49.12 | pabelanger | No, I don't have one right now |
| 15:49.26 | DelphiWorld | lol pabelanger cell less :P |
| 15:50.01 | pabelanger | Indeed, no reason to have one right now |
| 15:50.14 | DelphiWorld | pabelanger: why ? no voice calls on the go ? no sms ? no data ? :D |
| 15:50.25 | pabelanger | plus, cell phones in Canada are expensive |
| 15:50.45 | pabelanger | DelphiWorld: I can do all that from my Desktop or Laptop |
| 15:51.02 | DelphiWorld | pabelanger: :) |
| 15:51.14 | DelphiWorld | pabelanger: now i want to receyve sms through sip |
| 15:51.16 | DelphiWorld | pabelanger: any clue ? |
| 15:53.09 | DelphiWorld | anyone know a carrier delivering sms through sip ? |
| 15:53.39 | pabelanger | DelphiWorld: Asterisk 10 has SIP MESSAGE support, so you could use that if you found an SMS provider |
| 15:53.50 | DelphiWorld | pabelanger: lol |
| 15:53.53 | DelphiWorld | pabelanger: i need the sms provider |
| 15:54.29 | pabelanger | google? |
| 15:54.35 | pabelanger | I don't know of any |
| 15:54.37 | DelphiWorld | pabelanger: didn't found >:( |
| 15:56.19 | pabelanger | It would be interesting to see if Asterisk sees the message from Google Voice SMS |
| 15:56.28 | pabelanger | http://www.google.com/intl/en_us/mobile/sms/voice/ |
| 15:58.15 | DelphiWorld | pabelanger: but not through sip :( |
| 16:01.09 | pabelanger | true, it would likely be over jingle |
| 16:01.37 | DelphiWorld | pabelanger: lol then forward jingle to sip :P |
| 16:02.44 | pabelanger | DelphiWorld: why do you want SMS over SIP? |
| 16:02.59 | DelphiWorld | pabelanger: to kill my sim card |
| 16:03.20 | pabelanger | DelphiWorld: but what are you using it for? Personal or business |
| 16:03.33 | DelphiWorld | pabelanger: personal right now |
| 16:03.44 | DelphiWorld | pabelanger: i would keep my business one otherwise algeria telecom will fire me out |
| 16:03.49 | *** join/#asterisk Tas01019 (~IceChat77@c-76-106-227-203.hsd1.fl.comcast.net) |
| 16:05.25 | DelphiWorld | anyone a customer of voxbone ? |
| 16:11.17 | Tas01019 | anyone have an idea on how to prefix (or suffex) a caller ID with the extension a customer put into the IVR? |
| 16:14.35 | Gugge | Tas01019: you set CALLERID(num) to the output from CALLERID(num) plus the var from the ivr |
| 16:15.23 | Tas01019 | sorry gugge... where would i set that? extensions.conf? |
| 16:18.34 | Gugge | Tas01019: in whatever file you have your dialplan yes. |
| 16:20.24 | Tas01019 | cool - thank you for your help... one more question, hopefully... how do i tell what the variable is from the IVR? |
| 16:21.40 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 16:23.27 | Gugge | Tas01019: by reading the dialplan :) |
| 16:24.13 | Tas01019 | lol of course it would be that easy... sorry... using freepbx so i haven't manually created these files |
| 16:24.14 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 16:24.47 | Tas01019 | thank you gugge |
| 16:24.57 | Gugge | Tas01019: if you use freepbx, then go ask freepbx people how to do it .. dont edit dialplan yourself |
| 16:25.11 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 16:25.33 | Tas01019 | ok - will continue to ask them then... thanks |
| 16:29.07 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 16:34.32 | iscario | hello, I would like to add video support for my asterisk server. Would you have resources for me ? thx (I would be interested to know solutions compatible with IAX, but if there are others, i ' ll have a look. Thx) |
| 16:34.53 | *** join/#asterisk coppice (~chatzilla@116.92.27.206) |
| 16:36.47 | *** join/#asterisk irroot (~irroot@197.173.167.3) |
| 16:37.02 | *** join/#asterisk nightrid3r (~kvirc@91.176.53.237) |
| 16:41.45 | *** join/#asterisk nightrid3r (~kvirc@91.176.53.237) |
| 16:45.24 | iscario | or just advice maybe ;) |
| 16:53.08 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Video+Telephony |
| 16:53.28 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.48.137) |
| 16:54.56 | iscario | I already read this, it is not that useful^^ thanks anyway |
| 17:00.22 | ChannelZ | well if you're looking for devices, go browse Polycom's site, or Cisco's site... |
| 17:00.36 | DelphiWorld | ChannelZ: :P cisco ? :P :P :P |
| 17:04.29 | ChannelZ | buy what you want, I don't care |
| 17:04.59 | irroot | Yealink .... |
| 17:06.15 | DelphiWorld | vote for irroot suggestion |
| 17:06.24 | iscario | ChannelZ, i would like to know what i need to add to my server. Once I have an IAX server (Asterisk 1.8) working well for VoIP, do I have to add something special on my server ? |
| 17:07.56 | ChannelZ | bandwidth and CPU power possibly... I don't know |
| 17:09.07 | DelphiWorld | ChannelZ: general question |
| 17:09.09 | iscario | ok. In the best case, maybe i have nothing to do then :D |
| 17:09.12 | DelphiWorld | do asterisk support AMR codec ? |
| 17:10.09 | iscario | maybe i'll ask to the asterisk-users mailing list.... |
| 17:10.11 | ChannelZ | Not now, Asterisk 10 might but I think it has patent issues so I'm guessing still no |
| 17:10.42 | DelphiWorld | >:( |
| 17:11.30 | irroot | asterisk 10 does have initial video conf support |
| 17:12.42 | DelphiWorld | irroot: i know ... need AMR |
| 17:13.29 | iscario | amr, i'll google this^^ |
| 17:14.55 | DelphiWorld | iscario: :P |
| 17:15.07 | DelphiWorld | adaptive multirate codec |
| 17:15.39 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 17:16.29 | iscario | ok^^ |
| 17:17.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21c:23ff:fe20:991f) |
| 17:18.39 | ChannelZ | << >> |
| 17:22.44 | *** join/#asterisk nightrid3r (~kvirc@91.176.53.237) |
| 17:24.54 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 17:26.06 | ChannelZ | Where's lunch? |
| 17:26.45 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:28.26 | DelphiWorld | ChannelZ: << |
| 17:31.15 | *** join/#asterisk MiserySoft (~lee@host81-148-35-209.in-addr.btopenworld.com) |
| 17:42.10 | *** join/#asterisk sourcode (~code@ppp-115-87-217-89.revip4.asianet.co.th) |
| 18:00.00 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
| 18:03.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:07.12 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 18:07.24 | *** part/#asterisk MiserySoft (~lee@host81-148-35-209.in-addr.btopenworld.com) |
| 18:15.44 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 18:16.44 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 18:21.14 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:35.20 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:50.07 | ChannelZ | Wow. I drove out to work to pick up a CD, took the long way home and stopped for lunch.. and not a single thing has been said here in all that time! |
| 18:56.59 | singler | good for you :) |
| 18:57.11 | irroot | we waiting for your wisdom |
| 18:58.21 | ChannelZ | DO NOT GET THE FISH |
| 19:00.18 | irroot | oops thought that was chicken ... dont think it will end well |
| 19:11.11 | mickecarlsson | This is a test: |
| 19:11.13 | mickecarlsson | ~trixbox |
| 19:11.13 | infobot | trixbox is probably SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 19:11.17 | mickecarlsson | LOL |
| 19:13.27 | DelphiWorld | ~elastix |
| 19:13.27 | infobot | from memory, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 19:13.44 | *** part/#asterisk dr0ck (~dr0ck@nat/digium/x-xqxiwkfvcnffuape) |
| 19:14.06 | DelphiWorld | ~pabelanger |
| 19:14.07 | infobot | it has been said that pabelanger is a bug marshal and all around good guy! |
| 19:14.27 | DelphiWorld | infobot: pabelanger is someone that you can never see it |
| 19:14.27 | infobot | ...but pabelanger is already something else... |
| 19:14.31 | DelphiWorld | :P |
| 19:15.13 | ChannelZ | infobot ChannelZ is something else |
| 19:15.13 | infobot | okay, ChannelZ |
| 19:15.41 | irroot | ~irroot |
| 19:15.48 | irroot | hehe im safe :P |
| 19:15.52 | DelphiWorld | infobot: DelphiWorld is nothing |
| 19:15.52 | infobot | DelphiWorld: okay |
| 19:16.11 | DelphiWorld | infobot: irroot is ir root ! |
| 19:16.11 | infobot | DelphiWorld: okay |
| 19:17.10 | irroot | ~irroot |
| 19:17.10 | infobot | you are probably ir root ! |
| 19:17.20 | irroot | ir baboon |
| 19:17.34 | irroot | who be my weasel |
| 19:17.53 | irroot | i need to get T printed again with IR Root upside down on it |
| 19:19.51 | ChannelZ | cafepress |
| 19:35.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 19:41.59 | *** join/#asterisk puzzled (~patrick@53533D3D.cm-6-4a.dynamic.ziggo.nl) |
| 19:42.15 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:44.26 | *** join/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
| 19:45.39 | tmrhmdv | Does installing asterisk from packages (debian) will install LibPRI and DAHDI too automatically? |
| 19:49.02 | EmleyMoor | tmrhmdv: Yes to dahdu... |
| 19:49.06 | EmleyMoor | dahdi |
| 19:49.12 | tmrhmdv | Thank you sir |
| 19:51.03 | EmleyMoor | ... and libpri |
| 19:51.17 | tmrhmdv | Thank you again :) |
| 19:51.37 | EmleyMoor | uses asterisk installed from Debian packages |
| 19:54.06 | tzafrir_laptop | tmrhmdv, apt-cache depends asterisk |
| 19:54.52 | tzafrir_laptop | The current packages in Sid / Wheezy have a separate asterisk-dahdi that depends on dahdi and libpri |
| 19:55.17 | irroot | dahdi also linked to a kernel version |
| 19:55.24 | EmleyMoor | makes a note, in case he needs to do that |
| 19:56.14 | EmleyMoor | The dahdi modules will need to be built with module-assistant, whenever the kernel is changed |
| 19:56.30 | *** join/#asterisk radic (~radic@tmo-097-28.customers.d1-online.com) |
| 19:56.35 | EmleyMoor | (don't suppose there's dkms to automate that) |
| 19:59.27 | ChannelZ | packages, feh! |
| 20:01.44 | DelphiWorld | hi tzafrir_laptop |
| 20:03.59 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:14.11 | tmrhmdv | Thank you guys, installing from package went smooth :) |
| 20:29.00 | Bipul | p3nguin, alive ? |
| 20:29.51 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 20:34.03 | *** join/#asterisk wonderworld (~ww@port-92-201-75-20.dynamic.qsc.de) |
| 20:39.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:41.50 | *** join/#asterisk flyman_ (~flyman_@91-113-70-187.adsl.highway.telekom.at) |
| 20:48.39 | *** join/#asterisk tzafrir_laptop (~tzafrir@static-8-143-241-92-p2mp.broadband.blic.net) |
| 21:00.06 | *** join/#asterisk radic (~radic@tmo-096-58.customers.d1-online.com) |
| 21:13.21 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 21:14.13 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 21:26.56 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 21:27.40 | DelphiWorld | nighty night .P |
| 21:27.45 | DelphiWorld | pabelanger: bye / later |
| 21:32.06 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
| 21:37.18 | joker2u | does the manager api use only manager.conf or does it draw from peer authentication as well? error -> "manager.c:1114 authenticate: 127.0.0.1 failed to pass IP ACL as 'test'" |
| 21:38.58 | ChannelZ | just to login to manager it should only be using manager.conf |
| 21:39.59 | nightrid3r | did you setup the ip and acl in manager.conf for test |
| 21:43.16 | joker2u | nightrid3r yes I did. it was a complex password but when I saw the error I switch to simple/simple |
| 21:43.36 | nightrid3r | ok |
| 21:46.31 | joker2u | it was my deny=0.0.0.0/0.0.0.0 |
| 21:47.07 | joker2u | but I am still getting this one -> ERROR[7812]: utils.c:1177 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 21:49.35 | joker2u | when using -> http://pastebin.com/dx11dzyv |
| 21:53.04 | nightrid3r | i have no idea what could be wrong, i don't know how to program stuff |
| 21:57.43 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 21:58.09 | ChannelZ | You probably need to read more often |
| 21:59.00 | ChannelZ | also I've never reloaded through AMI, I wonder if that is breaking the pipe since manager is probably being reloaded.... |
| 22:00.24 | ChannelZ | and your syntax is wrong |
| 22:00.27 | nightrid3r | could very well be the problem |
| 22:00.35 | ChannelZ | Action: Command |
| 22:00.40 | ChannelZ | Command: reload |
| 22:01.18 | ChannelZ | (and I just tried, it doesn't knock you off..) |
| 22:01.29 | joker2u | ChannelZ the syntax comes from http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+PHP |
| 22:02.09 | ChannelZ | yeah, it's right - your paste has it wrong |
| 22:05.03 | joker2u | ChannelZ I appologize, I am not seeing the difference? |
| 22:05.46 | ChannelZ | I told you 3 lines ago - |
| 22:05.49 | joker2u | ChannelZ other than replacing $timeout with the #1, it's a cut and paste? |
| 22:05.51 | ChannelZ | See line 9 of your script and compare it |
| 22:06.00 | ChannelZ | No, it's not |
| 22:06.19 | ChannelZ | you have 'Action: Command" and then "reload" |
| 22:06.25 | ChannelZ | But it's "Command: reload" |
| 22:07.09 | joker2u | ChannelZ how in the? i see now. don't know how it got deleted. |
| 22:07.15 | joker2u | ChannelZ thankx |
| 22:07.32 | ChannelZ | yup |
| 22:09.28 | joker2u | ChannelZ oh, now I remember, this was why -> PHP Parse error: syntax error, unexpected T_STRING in /home/asterisk.php on line 8 |
| 22:10.02 | ChannelZ | eh? |
| 22:10.51 | joker2u | ChannelZ http://pastebin.com/NMb8hL0U |
| 22:11.05 | joker2u | ChannelZ Yeah, it didn't make sense to me either. |
| 22:12.36 | ChannelZ | You have an extra \ at the end of line 6 |
| 22:12.55 | ChannelZ | which turns everything that follows into an open quote |
| 22:14.31 | joker2u | ChannelZ -> back to ->ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 22:15.25 | ChannelZ | Probably your script terminates before you've actually read everything Asterisk has sent back |
| 22:15.42 | ChannelZ | You're only getting 1 line but Asterisk has sent you several from logging in and from the reload |
| 22:17.09 | joker2u | ChannelZ console output http://pastebin.com/Twxhijsj |
| 22:18.03 | joker2u | ChannelZ where can I get info on how asterisk reads the parse? |
| 22:18.20 | ChannelZ | I don't understand the question |
| 22:18.22 | ChannelZ | reads the parse of what |
| 22:21.42 | joker2u | ChannelZ the api interface? because this -> http://www.asterisk.org/astdocs/api/index.html doesn't help much to connect the dots between how the objects are passed from asterisk to php. |
| 22:23.26 | ChannelZ | maybe start here instead https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+%28AMI%29 |
| 22:32.59 | joker2u | ChannelZ I can't find a single place where it mentions how objects are passed |
| 22:34.42 | ChannelZ | I don't know what you're talking about. https://wiki.asterisk.org/wiki/display/AST/AMI+Command+Syntax |
| 22:35.15 | ChannelZ | Thats defines how you send commands. Each command has different arguments and things and will reply differently, which is defined in each command |
| 22:36.16 | ChannelZ | The easiest way is to just telnet to port 5038 and issue commands live to see how they respond |
| 22:36.16 | Nugget | telnet is eeeeeeevil! |
| 22:36.47 | joker2u | ChannelZ that's pretty much what I was looking for. thank you! |
| 22:41.25 | *** join/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
| 22:42.05 | tmrhmdv | is it possible to run "make menuselect" if I installed asterisk from packages? if yes, in which dir I would run it? |
| 22:43.01 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:43.01 | ChannelZ | no |
| 22:43.32 | ChannelZ | unless you download the source package and basically rebuild it yourself |
| 22:43.59 | tmrhmdv | Ok, thanks |
| 22:45.38 | tmrhmdv | How can I install "Core Sound Packages" then? |
| 22:46.15 | ChannelZ | they're probably a separate package in your distro already |
| 22:46.47 | ChannelZ | look for something starting with 'asterisk-sounds' for instance |
| 22:48.23 | tmrhmdv | my asterisk book says it's in /var/lib/asterisk/sounds, but there is only one empty folder named 'custom' |
| 22:48.41 | tmrhmdv | in /var/lib/asterisk/sounds |
| 22:48.59 | ChannelZ | RE: you installed Asterisk from a package, the sounds are in a separate package |
| 22:49.07 | ChannelZ | (they come in many formats, different languages..) |
| 22:49.19 | ChannelZ | Go search in your repository the same way you did for Asterisk in the first place |
| 22:49.47 | tmrhmdv | OK, thank you |
| 22:49.49 | ChannelZ | You can manually go dig around on downloads.digium.com and they're there somewhere |
| 22:57.20 | *** join/#asterisk kaushal (~kaushal@14.99.141.23) |
| 22:57.25 | kaushal | Hi |
| 22:57.52 | ChannelZ | Aho |
| 22:57.54 | ChannelZ | +y |
| 22:59.41 | kaushal | I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ |
| 22:59.44 | kaushal | http://pastebin.ubuntu.com/651451/ |
| 22:59.53 | kaushal | Can someone please suggest |
| 23:01.02 | kaushal | Error code 101 is "Message not compatible with call state". The explanation for this is "The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. |
| 23:01.10 | kaushal | Any clue ? |
| 23:06.38 | kaushal | I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is "Message not compatible with call state". The explanation for this is "The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. |
| 23:10.37 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:17.23 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 23:25.05 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 23:27.32 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
| 23:28.33 | WIMPy | kaushal: Still not working? Or wasn't that you the other day? |
| 23:29.34 | kaushal | WIMPy: please give me a moment |
| 23:30.23 | kaushal | I have Package libpri-1.4.11.5-1_centos5.x86_64 already installed and latest version on CentOS 5.6, is there a rpm version of 1.4.12 for CentOS 5.6 as per http://downloads.asterisk.org/pub/telephony/libpri/ ? |
| 23:31.18 | kaushal | WIMPy: I am faced with Error 101 |
| 23:31.19 | WIMPy | So that wasn't you where it happened after a power failure? |
| 23:31.50 | kaushal | WIMPy: i am afraid no |
| 23:32.04 | WIMPy | Interesting. |
| 23:32.07 | kaushal | WIMPy: Please suggest about my issue ? |
| 23:32.41 | WIMPy | So maybe the other guy upgraded somethning that wans't activated until the forced reboot. |
| 23:33.02 | WIMPy | I couldn;t make any sense to that situation, but no it smells like a bug. |
| 23:33.14 | WIMPy | Can you see incoming calls? |
| 23:34.24 | WIMPy | The other situatipn was like: No alarms on either end. outgoing calls failed. incomming calls were connected but then released again after a few seconds. |
| 23:35.30 | WIMPy | It looked like Q931 was ok, but Q921 failed. |
| 23:35.49 | WIMPy | Anything differen for you? |
| 23:38.30 | kaushal | WIMPy: Any clue to my query ? |
| 23:42.26 | WIMPy | Is it the same as I described? |
| 23:42.31 | WIMPy | Or anything different? |
| 23:45.24 | kaushal | WIMPy: Not sure i understand that |
| 23:45.39 | kaushal | Am i on the same page as you are ? |
| 23:46.55 | WIMPy | That's what I'd trying to find out. |
| 23:47.26 | kaushal | WIMPy: Shall i explain again my query to you ? |
| 23:48.18 | WIMPy | You didn't tell that much so far. Only the cause 101. |
| 23:48.42 | WIMPy | Did you try to make a call TO that line? What happens then? |
| 23:52.46 | kaushal | WIMPy: yes |
| 23:52.57 | kaushal | the call hungs |
| 23:55.16 | WIMPy | Does it arrive? |
| 23:55.55 | WIMPy | Or just paste the output of 'pri debug 2 span ...' of both an outgoing and an incoming call. |
| 23:56.31 | kaushal | sure |
| 23:56.59 | kaushal | WIMPy: at the CLI> mode right ? |
| 23:57.06 | WIMPy | yes |
| 23:57.09 | kaushal | sure |
| 23:58.32 | kaushal | No such command 'pri debug 2 span' (type 'core show help pri debug' for other possible commands) |
| 23:58.49 | kaushal | host0038*CLI> pri debug 2 span |
| 23:58.56 | WIMPy | You need to give the span number to debug. |
| 23:59.01 | WIMPy | hence the ... . |
| 23:59.27 | kaushal | is it pri debug 2 span 2 ? |
| 23:59.33 | kaushal | for example ? |
| 23:59.44 | WIMPy | If it's you span number 2, yes. |
| 23:59.50 | kaushal | ok |