| 00:03.13 | *** join/#asterisk salz212 (~chatzilla@182.178.225.12) |
| 00:03.36 | salz212 | Hi can someone tell how to get the DIALSTATUS like variables in Call file? |
| 00:05.53 | salz212 | any one? |
| 00:07.31 | WIMPy | You don't get anything in a call file. You set things. |
| 00:09.27 | salz212 | yes, suppose I have a call file where I am setting things to be dialed.. and then I need to have the time or Status of the call in the call file.. how do I get it.. I know how to get it in dialplan ... |
| 00:10.19 | WIMPy | Yes, that's how you do it. |
| 00:12.52 | salz212 | but .. suppose I am dialing an outbound call from call file.. |
| 00:13.14 | salz212 | how do I get the Variables |
| 00:13.31 | WIMPy | From your dialplan. |
| 00:13.44 | WIMPy | Don't call out directly. Use a local channel. |
| 00:14.30 | salz212 | hmm local.. so no choice with . direct IP call |
| 00:14.32 | p3nguin | Dial plan has no concept of where your call is going. It doesn't care of you're calling another phone on your LAN or a phone on the other side of the planet. Variables are used in dial plan the same way as any other call. |
| 00:24.55 | *** join/#asterisk sflemming (~stefan@85.183.53.64) |
| 00:26.45 | sflemming | hi all, i just switched from lua to del and wonder how i can have an individual h extension that is not global for a context but for an extension. can someone give me a hint? |
| 00:27.22 | sflemming | sorry, from extensions.lua to extensions.ael (not del) |
| 00:35.11 | *** part/#asterisk C4colo (~DJpyro@184-96-203-31.hlrn.qwest.net) |
| 00:41.52 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 01:22.15 | SeRi | guys I have callcentric setup to receive calls but I am unable to recive the calls. I get "Sending fake auth rejection for device" |
| 01:22.17 | SeRi | any ideas? |
| 01:22.47 | p3nguin | CallCentric sucks. |
| 01:22.57 | SeRi | yea I know |
| 01:22.58 | p3nguin | I'd drop 'em like a bad habit. |
| 01:23.02 | SeRi | I only use it for fax |
| 01:23.10 | SeRi | my main is voip.ms |
| 01:23.57 | SeRi | any ideas why I am getting that error msg? |
| 01:24.21 | p3nguin | Not really. |
| 01:24.57 | SeRi | mhhhhh..... |
| 01:25.23 | SeRi | voip.ms does not incomming fax in there service :( |
| 01:25.47 | p3nguin | Can you repeat that, but in English this time? |
| 01:27.15 | SeRi | sorry I am typing in so many places... |
| 01:27.52 | SeRi | voip.ms does not support incoming fax calls in there service or outbound |
| 01:28.10 | p3nguin | How do I fax through them, then? |
| 01:28.35 | WIMPy | I think, fake auth rej = username not found. |
| 01:29.15 | SeRi | I was told that from a tech support guy :/ |
| 01:29.28 | p3nguin | Their tech support doesn't know very much. |
| 01:29.44 | SeRi | p3nguin, I can say that much :) |
| 01:38.16 | SeRi | p3nguin, would exten = fax,1,Gosub(fax-rx,s,1) |
| 01:38.16 | SeRi | exten = fax,n,Hangup() go after my extension or before? |
| 01:38.40 | p3nguin | That question does not make any sense. |
| 01:39.50 | SeRi | well didnt meant to say extension but to my incoming context. for ex: |
| 01:39.57 | SeRi | exten = fax,1,Gosub(fax-rx,s,1) |
| 01:39.57 | SeRi | exten = fax,n,Hangup() |
| 01:39.57 | SeRi | exten => s,n,Dial(SIP/1004) |
| 01:40.28 | p3nguin | I still have no idea what you're trying to ask. |
| 01:40.37 | SeRi | I am trying to detect incoming fax. if it is not a fax to pass it to my ext. |
| 01:41.18 | p3nguin | You have to have an extension to process the call. |
| 01:41.30 | p3nguin | It's usually your phone number. |
| 01:41.54 | p3nguin | I don't know how you are doing fax detection, but when a fax is detected it should go to extension fax. |
| 01:48.29 | *** join/#asterisk hippieua (~v.v.gura@82.193.109.199) |
| 01:50.43 | SeRi | well because i am doing a tif convert and emailing it. I am not sending it to a real fax machine |
| 01:50.50 | SeRi | :) |
| 01:50.59 | p3nguin | Yeah, so? |
| 01:52.13 | SeRi | ok so I still need to send it to an ext? The context its in front if my incommign number. |
| 01:52.21 | SeRi | of* |
| 01:52.26 | p3nguin | Calls start at an extension. |
| 01:52.59 | p3nguin | Calls come into a context and match an extension. The extension then runs an app or does something else useful. |
| 01:54.49 | p3nguin | So yes, the call will go to an extension. Otherwise, it's not a call. |
| 01:55.09 | p3nguin | The type of call does not matter; it can be a fax or a voice call. |
| 01:56.28 | SeRi | got it so it should read like this: |
| 01:56.31 | SeRi | [voipms-inbound] |
| 01:56.31 | SeRi | exten = XXXXXXXXX,fax,1,Gosub(fax-rx,s,1) |
| 01:56.31 | SeRi | exten = XXXXXXXXX,fax,n,Hangup() |
| 01:56.31 | SeRi | exten => XXXXXXXXX,n,Dial(SIP/1003,35) ;your DID |
| 01:56.31 | SeRi | exten => XXXXXXXXX,n,Voicemail(4222,u) |
| 01:56.39 | p3nguin | no. |
| 01:56.44 | SeRi | where XXX is my number |
| 01:57.01 | p3nguin | exten = XXXXXXXXX,fax,1,Gosub(fax-rx,s,1) |
| 01:57.06 | p3nguin | ^^ not valid. |
| 01:58.00 | p3nguin | exten => _NXXNXXXXXX,1,Gosub(fax-rx,s,1); <-- valid, matches NANP phone numbers. |
| 01:59.14 | SeRi | ok I see. |
| 02:02.28 | SeRi | testing now :) |
| 02:02.49 | p3nguin | You still didn't tell me how you're doing fax detection. |
| 02:05.24 | SeRi | ok let me do a pastebin for you its pretty extensive. |
| 02:09.10 | SeRi | p3nguin, http://pastebin.com/mU1gi27N |
| 02:10.10 | p3nguin | That's invalid, and there's no fax detection in there. |
| 02:10.22 | SeRi | :( |
| 02:10.34 | p3nguin | What that means is... |
| 02:10.37 | p3nguin | That shit won't work. |
| 02:10.49 | SeRi | rofl |
| 02:11.04 | p3nguin | exten = XXXXXXXXX,fax,1,Gosub(fax-rx,s,1) <--- not valid |
| 02:11.09 | p3nguin | (2057.59) <p3nguin> exten => _NXXNXXXXXX,1,Gosub(fax-rx,s,1); <-- valid, matches NANP phone numbers. |
| 02:12.31 | SeRi | I know. if you looked in my note I left in the top I noted that I have my external number there. I only used XXX so I wont put my real number |
| 02:12.50 | SeRi | 1. Please note that in the field XXXXXXXXX under Dial and Voicemail I have my real number I just put XXX to give you an example. |
| 02:13.06 | p3nguin | Doesn't matter, it's still invalid even if you replaced those Xs with a valid phone number. |
| 02:13.32 | p3nguin | Try it, you'll find out. |
| 02:14.19 | SeRi | thats funny I have been reciving phone calls like that IE: 123456789,n,Dial(SIP/1003,35) |
| 02:14.40 | SeRi | Thats what voip.ms told me to put my number there |
| 02:15.10 | p3nguin | 123456789,n,Dial(SIP/1003,35) is valid, so I would expect it would work. |
| 02:16.30 | SeRi | ok so I am lost. where do I have the XXX? |
| 02:16.48 | p3nguin | I don't understand your question. |
| 02:18.25 | p3nguin | I'll show you a valid extension one more time: exten => _NXXNXXXXXX,1,Gosub(fax-rx,s,1) |
| 02:18.43 | p3nguin | Compare it to your invalid extension: exten = XXXXXXXXX,fax,1,Gosub(fax-rx,s,1) |
| 02:19.03 | pabelanger | SeRi: And exten => only take 3 arguments (EG: exten => 555,1,Answer()) You are trying to pass 4 (exten => 555,fax,1,Answer()), which is not valid |
| 02:19.20 | p3nguin | If you're going to tell me again that you replaced your number with Xs, I'll tell you the same thing again. |
| 02:19.39 | pabelanger | so in your example it would be exten => fax,1,GoSub(fax-rx,s,1) |
| 02:19.42 | p3nguin | exten => EXTEN,PRIORITY,APP |
| 02:23.54 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:24.07 | p3nguin | BUT... |
| 02:24.11 | SeRi | ? |
| 02:24.13 | SeRi | :) |
| 02:24.24 | p3nguin | Even if you define a fax extension, that does not provide fax detection. |
| 02:24.38 | p3nguin | Having a fax extension just allows for processing a fax which has been detected. |
| 02:24.49 | p3nguin | I still want to know how you are detecting the fax. |
| 02:24.52 | pabelanger | right, you need to enable it in the channel driver |
| 02:25.23 | p3nguin | It's automatic with no dialplan tricks required? |
| 02:25.49 | SeRi | exten = s,n,ReceiveFAX(${FAXFILE}.tiff) |
| 02:25.57 | SeRi | I thought that was what that context did? |
| 02:25.58 | p3nguin | That's not fax detection. |
| 02:26.04 | p3nguin | That's not a context. |
| 02:26.05 | pabelanger | if you are using SIP, you set faxdetect = yes in sip.confg |
| 02:26.14 | pabelanger | s/confg/conf |
| 02:26.27 | SeRi | I am using iax2 |
| 02:27.01 | pabelanger | SeRi: fax detect is not support on IAX2 |
| 02:27.05 | pabelanger | DAHDI or SIP |
| 02:27.24 | SeRi | ahhhhh damn. ok so I have to fall back to call centric. |
| 02:27.33 | SeRi | I new there was a reason why i was using them. |
| 02:27.46 | p3nguin | You can still receive a fax over IAX2 as long as you aren't trunking and you are using ulaw. |
| 02:28.04 | p3nguin | There's just no detection of it using a shared phone number. |
| 02:28.26 | pabelanger | exactly |
| 02:29.26 | SeRi | ok I see. |
| 02:29.28 | p3nguin | But in the case of SIP, fax detection is automatic as long as it is enabled in sip.conf? |
| 02:29.38 | pabelanger | yes |
| 02:29.52 | pabelanger | see faxdetect setting |
| 02:29.58 | p3nguin | A call comes to me, my attendant answers and begins playing back a message... |
| 02:30.11 | p3nguin | That incoming call begins playing a fax tone... |
| 02:30.22 | p3nguin | It will jump to exten fax all by itself? |
| 02:30.49 | pabelanger | as long as exten => fax,1,blah() exists in the same context |
| 02:30.49 | SeRi | lol |
| 02:31.27 | pabelanger | It is documented in sip.conf.sample |
| 02:31.37 | p3nguin | That's interesting. I have a dedicated phone number for fax over sip over ip, so I never had to do detection. |
| 02:32.22 | SeRi | pabelanger, thats what I am trying to do. |
| 02:32.39 | pabelanger | SeRi: well if you SIP or DAHDI, it should work |
| 02:33.05 | SeRi | ok so I have the fax detection enabled now. but callcentric craper is giving me errors: Sending fake auth rejection for device |
| 02:33.08 | p3nguin | If you aren't using IAX2 for the trunking, you could easily switch to SIP. |
| 02:33.22 | SeRi | p3nguin, I am using both |
| 02:33.45 | p3nguin | Why would you use both channel technologies for one ITSP? |
| 02:33.46 | SeRi | so I have switched just need to fix the callcentric issue |
| 02:34.01 | SeRi | two providers |
| 02:34.09 | SeRi | and just because :) |
| 02:34.16 | p3nguin | Why would you use both channel technologies for one ITSP? |
| 02:34.23 | p3nguin | key word: one |
| 02:35.05 | SeRi | I would use IAX in callcentric if I could :) |
| 02:35.11 | p3nguin | So if you want fax detection on your number from voipms, change to SIP. If you aren't using IAX2 for the trunking ability, changing to SIP shouldn't be a problem. |
| 02:36.34 | p3nguin | I use IAX2 because of trunking. |
| 02:36.45 | SeRi | I will once I get call centric all settled. The only reason I am using IAX2 its because I have not been able to get sip reliably working behind my satanic firewall :) |
| 02:36.54 | p3nguin | I use SIP because of faxing, because I trunk my IAX2 account. |
| 02:37.53 | p3nguin | CallCentric sucks, and they send calls from unlisted IP addresses, so the calls never match the peer entries you've configured for CallCentric. |
| 02:39.26 | SeRi | well for now is my test account I cant play with voip.ms because I have 10 ext going in it and is the main line right now. |
| 02:39.28 | p3nguin | That's interesting that the fax detect is magical like that. I may have to play with that soon. |
| 02:40.02 | SeRi | p3nguin, so what I posted should be right. |
| 02:40.10 | p3nguin | You say you have 10 ext going in it, but do you mean you have 10 DIDs with them? |
| 02:40.25 | p3nguin | No, what you posted was INFUCKINGVALID. |
| 02:41.15 | SeRi | p3nguin, NO. This is right as per what pabelanger said... fax,1,Gosub(fax-rx,s,1) |
| 02:41.26 | p3nguin | That part is okay. |
| 02:41.33 | SeRi | thats what I meant :) |
| 02:41.41 | p3nguin | That's where the call will go after SIP does fax detection. |
| 02:41.43 | SeRi | not the pastebin. |
| 02:42.13 | SeRi | the pastebin was the modifications I made after the confusion. |
| 02:42.33 | SeRi | awesome. thats the way I wanted to work |
| 02:43.11 | SeRi | ok so now.... I need to figure out wtf to do with callcentric issues... |
| 02:43.31 | p3nguin | dr op |
| 02:43.37 | p3nguin | Bye bye. |
| 02:43.42 | p3nguin | They suck. |
| 02:43.51 | SeRi | p3nguin, I have 3 did's and 10 extensions |
| 02:43.57 | SeRi | with voip.ms |
| 02:43.58 | p3nguin | That doesn't make sense. |
| 02:44.11 | SeRi | sorry the extensions are internal |
| 02:44.18 | SeRi | just 3 did's :) |
| 02:44.19 | p3nguin | also doesn't make sense. |
| 02:44.26 | p3nguin | Maybe you have 10 phones. |
| 02:44.34 | SeRi | fuck nothing makes since. lol |
| 02:44.48 | p3nguin | Most of what you've said doesn't. |
| 02:44.53 | SeRi | well 10 phones that have one extension each. |
| 02:45.01 | p3nguin | I feel like this is your first day working on Asterisk. |
| 02:45.57 | SeRi | p3nguin, some what. the problem here is between my none native language and communicating with you. |
| 02:46.29 | SeRi | You are looking fro exact words I dont know :) |
| 02:46.40 | SeRi | that I dont know* |
| 02:46.50 | SeRi | for* |
| 02:46.53 | SeRi | damn |
| 02:46.55 | SeRi | lol |
| 02:46.58 | p3nguin | Even when I see some inexact words, I can sometimes make a sentence appear from them. |
| 02:47.25 | p3nguin | Like fro... I know if a typo of for. |
| 02:48.05 | SeRi | ok those are every day typos that we are use to. but my sentences do not make since to you because I do not know how to express them correctly |
| 02:48.23 | SeRi | in English |
| 02:48.25 | p3nguin | That's very possible. |
| 02:48.40 | SeRi | :) |
| 02:49.07 | SeRi | Most people would of have told me to f* off by now. Thank you for your patience :) |
| 02:49.46 | p3nguin | I rarely tell people to fuck off. Most people do not deserve to fuck off just because they don't understand certain things. |
| 02:50.10 | p3nguin | They just need to put some more time and effort into the topic and then they might understand those things. |
| 02:50.41 | SeRi | thats awesome. I get rejected in IRC a lot because of my inability to type correct sentences. |
| 02:51.22 | SeRi | p3nguin, I am trying. I bought a book called "Practical Asterisk" |
| 02:51.40 | p3nguin | Did you read The Book? |
| 02:51.42 | p3nguin | ~book |
| 02:51.43 | 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 |
| 02:52.56 | SeRi | p3nguin, I am. Just takes time to process. Its in English :P |
| 02:53.12 | SeRi | I got that link booked marked :) |
| 02:53.12 | p3nguin | Which language would you prefer? |
| 02:53.31 | SeRi | Spanish :) |
| 02:53.58 | SeRi | I have not found anything in Spanish for asterisk locally |
| 02:54.25 | p3nguin | I don't know if there was any discussion about translations of the book. |
| 02:54.41 | p3nguin | I know there are plenty of people that could use them in other langs. |
| 02:55.18 | SeRi | ++++++1111111111 :) |
| 02:56.53 | SeRi | Trying to get sip working now. |
| 02:57.12 | p3nguin | What kind of firewall are you fighting? |
| 02:58.06 | SeRi | pfsense |
| 02:58.25 | SeRi | The problem that pfsense in mangling the packet |
| 02:58.32 | SeRi | the problem is* |
| 02:58.41 | SeRi | sip packets |
| 02:58.56 | SeRi | I think I have found the issue so I am working on it |
| 03:07.24 | SeRi | I think I know what you mean right now about callcentrics IP |
| 03:08.07 | SeRi | That really blows. That deafeats the purpose of firewalls. |
| 03:12.59 | *** join/#asterisk wesphillips (~wphill04@adsl-99-161-159-69.dsl.hstntx.sbcglobal.net) |
| 03:13.25 | *** part/#asterisk slobirdrop (~slobirdro@c-69-181-185-170.hsd1.ca.comcast.net) |
| 03:16.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 03:52.19 | SeRi | p3nguin, I had to disable allowguest = no to get CC working. I am open to the world now? |
| 03:54.01 | p3nguin | That's not going to fix it. |
| 03:54.13 | *** join/#asterisk wesphillips (~wphill04@adsl-99-161-159-69.dsl.hstntx.sbcglobal.net) |
| 03:54.31 | p3nguin | But yes, if allowguest=yes, now you allow everyone to call in. |
| 03:54.40 | p3nguin | That's why CallCentric should not be used. |
| 03:57.54 | SeRi | not the firewall issue I fix that. But I was still unable to get calls in because of allowgesu = no. |
| 03:57.59 | SeRi | guest* |
| 03:58.11 | SeRi | I disabled it and now it works I can get calls in. |
| 03:58.15 | p3nguin | I'm trying to explain it to you, but you don't seem to understand. |
| 03:58.17 | SeRi | but this is bad |
| 03:58.56 | p3nguin | So I guess I'm done. Good luck in your call ventures. |
| 04:12.58 | SeRi | I do understand. Get rid of callcentric :) and so I did. allowguest=yes its a big no no. so that is it for me and CC. |
| 04:13.03 | SeRi | Thanks for the help p3nguin |
| 04:13.38 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 04:37.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:50.02 | p3nguin | I can't figure out why anyone would accept that kind of service from CC. |
| 04:50.20 | dijib | who is CC |
| 04:50.39 | p3nguin | CallCentric |
| 04:50.58 | dijib | whats wrong with their service? |
| 04:51.14 | p3nguin | They have a really jacked up way of doing things that requires you to allow anonymous calls and accept calls from them in your general context. |
| 04:52.17 | dijib | i dont know what you mean by anonymous calls how is this handles normally? |
| 04:53.33 | p3nguin | Normally, you'd create a peer entry in sip.conf which matches calls from hosts. |
| 04:54.03 | p3nguin | Then you can send the call to a context of your choice with extension of either your choice or the provider's choice. |
| 04:55.36 | p3nguin | Anonymous calls are those not matching a defined peer. |
| 04:56.07 | p3nguin | You either allowguest or you don't. If you do, anonymous calls will go to the context defined in the general section. |
| 04:56.24 | p3nguin | If you don't, they will be rejected. |
| 04:56.34 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.143.118.240) |
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| 05:01.16 | SeRi | cc sucks donkey balls! |
| 05:03.16 | SeRi | p3nguin, does voip.ms accept connections in any other port other than 5060? |
| 05:03.24 | p3nguin | yes |
| 05:03.31 | *** join/#asterisk paec (~user@AAmiens-551-1-68-36.w92-142.abo.wanadoo.fr) |
| 05:04.11 | p3nguin | The alternate SIP ports are 5080 and 42872. |
| 05:04.17 | SeRi | sweet! |
| 05:04.23 | p3nguin | They know some ISPs block SIP. |
| 05:04.36 | SeRi | I got it working under 5080 so switching from IAX to sip now :) |
| 05:04.39 | p3nguin | s/SIP/SIP by port/ |
| 05:05.01 | SeRi | hehehehe lol |
| 05:06.04 | SeRi | Thanks |
| 05:06.27 | p3nguin | Now you're going to play with the fax detect in sip.conf? |
| 05:23.31 | SeRi | p3nguin, I got it working and is awesome! |
| 05:23.41 | SeRi | I get an email with a bash script I set |
| 05:23.48 | SeRi | it converts the tiff to a pdf :D |
| 05:23.57 | SeRi | elegant ;) |
| 05:23.59 | p3nguin | You don't need a script for that. |
| 05:24.34 | p3nguin | http://pastebin.com/6RQV9nEx |
| 05:25.28 | SeRi | ooo mhhhh very interesting... let me thinker here for a sec |
| 05:26.03 | SeRi | ooo wow it can all be done in asterisk. very nice |
| 05:26.12 | SeRi | is reading |
| 05:29.04 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 05:31.18 | SeRi | ok I think I am going to tweak asterik now to do all this for me instaed of a secript |
| 05:31.24 | SeRi | brb |
| 05:33.22 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 05:38.48 | dijib | i agree p3nguin thats rediculus\ |
| 05:39.44 | p3nguin | I used to mess around with CallCentric a few years ago and I don't ever remember having any problems like people have reported in the last several months. |
| 05:40.08 | dijib | maybe its user error |
| 05:40.37 | p3nguin | It's possible, but I'm leaning more toward poor design changes. |
| 05:41.45 | dijib | i would gander you would most liely be right |
| 05:41.46 | SeRi | guys I am happy to say I am all SIP now :D |
| 05:41.55 | SeRi | voip.ms is awesome :) |
| 05:42.12 | dijib | IAX2 is any better? |
| 05:42.22 | dijib | for volume? less bandwidth? |
| 05:43.19 | SeRi | no I just wanted fax over sip :) |
| 05:43.29 | p3nguin | With high call volume and trunking enabled, IAX2 can provide significant bandwidth savings. |
| 05:43.30 | SeRi | faxdetect = yes :) |
| 05:43.47 | dijib | ive got fax detect working thanks to P3 |
| 05:44.01 | p3nguin | How are you detecting fax? |
| 05:44.08 | dijib | emails to me and everything. was that your code right ? |
| 05:44.17 | dijib | faxdetect |
| 05:44.30 | dijib | ive disabled it to test the silent calls and thats not it. |
| 05:44.33 | SeRi | I got it working just now with help of p3nguin and pabelanger |
| 05:44.49 | p3nguin | All I did was write some fax extension stuff. |
| 05:44.51 | dijib | i want email with pdf attachment to FAXout |
| 05:45.01 | SeRi | Thats how I have it |
| 05:45.13 | SeRi | it converts tiff to pdf and emailes out |
| 05:45.22 | dijib | no i want the other way around |
| 05:45.39 | dijib | also i want ASR to email. |
| 05:45.40 | SeRi | o fax out |
| 05:46.00 | SeRi | nice. That would be next in my "to do" list |
| 05:46.02 | dijib | automated speech recognition |
| 05:46.13 | dijib | to email. |
| 05:46.20 | dijib | makes my salivate |
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| 06:08.20 | dijib | installing lumenvox now |
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| 08:42.27 | Dovid | hi all. when it says this was applied to trunk: https://issues.asterisk.org/jira/browse/ASTERISK-17293 |
| 08:42.43 | Dovid | wold that be in 1.8.X or 1.8.X is no longer going from trunk ? |
| 08:55.49 | irroot | Dovid patches get applied to branches/x [as applicable oldest version first] then moved to all branches up to trunk or to where its needed ... periodically branches/x get tagged tags/x.y-rc1 then a new cycle starts until x.y is released |
| 08:56.10 | irroot | once tagged/released there will be no changes to that tree further |
| 08:56.58 | irroot | if needed a x.y-rc[next] or x.y.z will be tagged/relleased |
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| 09:19.37 | Bihu | clear |
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| 09:29.43 | Dovid | wow. that was confusing |
| 09:29.59 | Dovid | basically. is: https://issues.asterisk.org/jira/browse/ASTERISK-17293 included in 1.8.X ? |
| 09:37.00 | irroot | if it is it will be in 1.8.x released after the commit date |
| 10:17.27 | Dovid | ok. thanks |
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| 13:35.14 | DND | guys my asterisk machine is on 100% cpu usage |
| 13:35.28 | DND | from the top command it says asterisk is the culprit |
| 13:36.05 | DND | but how can i get the details which part of asterisk is actually using high cpu? |
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| 14:55.12 | cusco | DND: have you looked at the logs? |
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| 15:03.51 | mtbf | When i do exten => someext,n,Noop(/var/spool/asterisk/monitor/${CDR_FNAME}) current context stucks in the infinite loop, |
| 15:04.32 | p3nguin | What does that even mean? |
| 15:04.47 | p3nguin | Contexts don't do anything but contain extensions. |
| 15:04.47 | mtbf | I'm trying to use that path as a parameter for mail sending program, when i use hardcoded path it works, when i use above path it doesn't, just like /var/spool/blabla caused the problem. |
| 15:05.15 | mtbf | Do i have to escape those slashes somehow? |
| 15:05.19 | p3nguin | Contexts don't execute anything or read anything. |
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| 15:05.29 | p3nguin | no. |
| 15:05.31 | p3nguin | What are you trying to do? |
| 15:05.44 | mtbf | Send an e-mail with CDR wave attached. |
| 15:05.51 | p3nguin | CDR wave? |
| 15:05.57 | mtbf | Yup. |
| 15:05.59 | p3nguin | Do you even know what CDR is? |
| 15:06.15 | mtbf | A wave file generated by the Monitor app, ok. |
| 15:06.25 | p3nguin | That's not related to CDR. |
| 15:06.44 | tonsofpcs | cdrwave is apparently a plus-size clothing outlet, per google |
| 15:07.02 | kaldemar | you don't have to escape slashes. show the whole extension. |
| 15:07.06 | p3nguin | How was the recording made? When should it be emailed? |
| 15:09.03 | mtbf | It should be emailed by the h extension, after hangup, but now I'm just trying to check whether the path is correct, and when i noop() that variable context gets looped. |
| 15:09.10 | p3nguin | When I record calls and need to email the file, I use the h extension and the MIXMONITOR_FILENAME variable. |
| 15:09.40 | p3nguin | mutt -a ${MIXMONITOR_FILENAME} |
| 15:09.59 | kaldemar | mtbf: show the extension and a CLI output. |
| 15:10.10 | p3nguin | Saying "context gets looped" does not make any sense. Contexts don't do anything. |
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| 15:13.26 | p3nguin | Just show us the entire extension so we can fix it. |
| 15:14.56 | mtbf | http://pastie.org/2589433 |
| 15:15.02 | mtbf | Here you go. |
| 15:15.21 | mtbf | Thanks for that filename hint, I'll start to use it. |
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| 15:21.20 | p3nguin | http://pastebin.com/dxVBj8Nd |
| 15:21.48 | p3nguin | Although I do not understand the whole Set() and SHELL() thing you're doing, I still recycled it for you. |
| 15:21.55 | mtbf | Mhm, looks like mutt just gives up in that testing case, cause file doesn't exist, but it'll work for hangup ext, now i got it working with some hardcoded filename. |
| 15:21.58 | p3nguin | I would do it with System(). |
| 15:22.42 | root52 | Hi All, So I am trying to record a conversation with the MixMonitor(). I now have a requirement to allow the users to pause and unpause the recording. I have been hacking away and can not figure out how to do that. I tried using the PauseMonitor and UnPauseMonitor as features code with the Monitor() and that work but it just combines the two files so it is not a real conversation just one leg talking and then the other. If I use those two fe |
| 15:22.48 | mtbf | Thanks p3nguin, what are those -- before e-mail address for? |
| 15:23.21 | p3nguin | http://pastebin.com/tw5QDmDR |
| 15:23.28 | p3nguin | man mutt |
| 15:23.32 | mtbf | I just wanted to debug it somehow, cause mutt was not sending e-mail, that's why i switched from System to SHELL. |
| 15:23.36 | mtbf | K. |
| 15:24.51 | p3nguin | You can debug it with System() just as easily. |
| 15:26.38 | kaldemar | with func SHELL you get command output, app System only gives you the SYSTEMSTATUS channel variable. |
| 15:26.38 | mtbf | You mean the SYSTEMSTATUS value or some another way too? |
| 15:26.58 | p3nguin | I meant something else. |
| 15:27.32 | p3nguin | Verbose output shows what is running in asterisk, and you can simply redirect the output of the command into a file and read it. |
| 15:28.14 | p3nguin | Either way, I've given you suggestions using both ways. |
| 15:28.35 | p3nguin | There's no reason I can see that it won't work. |
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| 15:29.40 | mtbf | Thank you p3nguin, by the way, I didn't and still don't see that -- parameter in the mutt man, but it's ok, I'll find it out, thanks again. |
| 15:30.21 | p3nguin | <PROTECTED> |
| 15:30.25 | p3nguin | <PROTECTED> |
| 15:31.37 | p3nguin | Even though my email address does not start with a dash, if I do not use -- before the address, mutt chokes. |
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| 16:11.07 | root52 | Got it. The trick was using the Application Map to point to a Macro so I had access to the recording file name, then just call MixMonitor again with the "a" option and exit said macro. Thanks!! |
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| 17:23.45 | K3rmit | My Message length reported by voicemail always shows incorrect. Like half the actual message length...I dunno why |
| 17:24.02 | p3nguin | Upgrade your Asterisk. |
| 17:24.18 | K3rmit | why does that make a difference? |
| 17:24.30 | pabelanger | it was fixed? |
| 17:24.36 | K3rmit | it was a bug? |
| 17:24.37 | p3nguin | Because the problem was said to be fixed. |
| 17:24.42 | pabelanger | yes |
| 17:24.45 | K3rmit | ah k |
| 17:24.58 | p3nguin | What version are you using? |
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| 17:26.07 | mtbf | p3nguin: Now I'm trying to use h,1,System() in the ivr-3 context, but this one is generated automatically by freepbx, so I added it to ivr-3-custom, which is included by ivr-3, where the hangup occurs, but it doesn't seem to be catched, I mean 1,h,Noop(something) doesn't show. |
| 17:27.03 | p3nguin | *shrug* Don't know anything about FreePBX. |
| 17:27.03 | K3rmit | p3nguin 1.8 |
| 17:27.11 | p3nguin | That's a branch. I asked what version. |
| 17:27.34 | p3nguin | core show version |
| 17:28.07 | mtbf | Ok, so if context a includes context b with h,1,Noop(something) and hangup occurs in the context a, 'something' should occur in the log, right? |
| 17:28.51 | p3nguin | Only if you have verbose turned up enough to see it. |
| 17:29.01 | mtbf | Mhm. |
| 17:29.26 | mtbf | It is set to 12. |
| 17:30.11 | p3nguin | extension h will run in the context where the call begins, unless you Goto() another context. |
| 17:30.36 | pabelanger | K3rmit: I believe 1.8.7.0 has the fix |
| 17:30.57 | K3rmit | p3nguin 1.8.3.3 digium maverick |
| 17:30.57 | pabelanger | otherwise 1.8.8.0-rc1 will |
| 17:35.22 | p3nguin | Is there any way to run one extension all by itself? If I use originate to run an extension, I have to specify a channel to connect with first. |
| 17:48.33 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:48.33 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:48.53 | p3nguin | The current 1.8 version seems to be 1.8.7.0. |
| 17:54.49 | irroot | l1nuxman "svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-svn-1.8" |
| 17:55.12 | irroot | that will have it but it wont be available in a package for at least a weeek |
| 17:55.22 | irroot | it will be 1.8.8.0-rc1 |
| 17:55.28 | p3nguin | But you can make a package out of it. |
| 17:57.30 | irroot | r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines ASTERISK-16981 ASTERISK-2234 |
| 17:57.32 | irroot | Fix for incorrect voicemail duration |
| 17:57.38 | l1nuxman | hmm any clue the directory? |
| 17:58.04 | p3nguin | What do you mean? |
| 17:58.07 | l1nuxman | how do you know tjat version is higher than 1.8.33 |
| 17:58.24 | irroot | its the latest not yet released |
| 17:58.58 | l1nuxman | I just need the one that fixes the voicemail |
| 17:59.06 | p3nguin | (1254.48) <irroot> l1nuxman "svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-svn-1.8" |
| 17:59.13 | p3nguin | Enjoy. |
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| 17:59.16 | l1nuxman | hmm ok |
| 18:00.27 | irroot | l1nuxman its recomended you use a recent version as .3 has many known fixed bugs |
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| 18:03.33 | l1nuxman | alright p3nguin looks like its all 1.8 version but doesn't say which one. I have 1.8.3.3 already |
| 18:03.46 | l1nuxman | irroot, only reason I want to upgrade is to fix voicemail |
| 18:03.58 | p3nguin | Are you having a problem understanding that you need to upgrade? |
| 18:04.16 | l1nuxman | I'm not sure how exactly then |
| 18:05.12 | p3nguin | (1254.48) <irroot> l1nuxman "svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-svn-1.8" |
| 18:05.12 | l1nuxman | I mean usually I just did a package name |
| 18:05.12 | p3nguin | There's no package. |
| 18:05.12 | irroot | you can try check out the commit and merge it but it may not work so far out and you need to be able to mod it |
| 18:05.12 | l1nuxman | hmmm then which of those do I need? |
| 18:05.12 | l1nuxman | there's so many |
| 18:05.12 | p3nguin | (1254.48) <irroot> l1nuxman "svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-svn-1.8" |
| 18:05.12 | l1nuxman | I did that |
| 18:05.24 | irroot | ok so you have a new directory |
| 18:07.00 | irroot | asterisk-svn-1.8 |
| 18:07.00 | p3nguin | then go into the directory, ./configure, make, make menuselect, checkinstall. |
| 18:07.00 | irroot | cd into it |
| 18:07.00 | p3nguin | Then you have your package. |
| 18:07.00 | irroot | then run configure ;make menuconfig;make ;make install |
| 18:07.10 | irroot | when you want to up date it go into the directory and type "svn up" |
| 18:07.31 | p3nguin | used to drink 7up |
| 18:07.32 | irroot | if you want to use a release you can switch to it |
| 18:08.12 | irroot | im not sure i should go into to much detail on svn |
| 18:08.14 | l1nuxman | cool thanks I"ll try |
| 18:11.58 | pabelanger | l1nuxman: if 1.8.8.0-rc1 is released this will, we'll be rolling new packages for it |
| 18:12.16 | pabelanger | s/will/week |
| 18:12.38 | p3nguin | Oh, so close! |
| 18:12.58 | p3nguin | s/close/far away/ |
| 18:13.14 | irroot | how long is a piece of string :P |
| 18:13.37 | coppice | |<------>| |
| 18:13.49 | p3nguin | I have a string several feet long. |
| 18:14.19 | irroot | i only have one foot :P snap |
| 18:27.21 | p3nguin | Can settings like astvarrundir be used in dialplan? If yes, are there variables for the settings? |
| 18:28.52 | pabelanger | p3nguin: yes, but only in Aserisk 10.0+ |
| 18:29.12 | pabelanger | heh, batteries dying in keyboard |
| 18:29.13 | pabelanger | http://svnview.digium.com/svn/asterisk?view=revision&revision=301729 |
| 18:30.52 | pabelanger | You could also try AST_CONFIG |
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| 19:12.26 | SeRi | guys I setup a sub account in voip.ms its up and register. but when I try to call in I get a bust tone. any ideas? |
| 19:12.55 | p3nguin | Did you assign a DID to the account? |
| 19:13.04 | SeRi | p3nguin, yes sr. |
| 19:13.32 | SeRi | Does it need a different port than the main account? |
| 19:13.46 | p3nguin | no |
| 19:13.54 | SeRi | I thought so since is registered |
| 19:14.01 | p3nguin | But you have to define the extension in the context you set for it. |
| 19:14.31 | SeRi | I did. I did the same thing I did for the main account. |
| 19:14.52 | SeRi | it shouldnt be any different except it goes to a different extension. |
| 19:14.56 | p3nguin | Show me the relevant configs. |
| 19:15.01 | SeRi | yes sr |
| 19:15.38 | SeRi | one sec. using pastebin now |
| 19:20.37 | SeRi | p3nguin, http://pastebin.com/LzJNsU6R |
| 19:21.31 | p3nguin | The phone number you are calling is 123456789? |
| 19:21.51 | SeRi | no sir. |
| 19:21.58 | SeRi | thats an example number |
| 19:22.01 | p3nguin | Then how do you expect this to work? |
| 19:22.12 | p3nguin | How many phone numbers do you have with voipms? |
| 19:22.33 | SeRi | I have my real number in the config. I just didnt want to post my number for the whole world to know :) |
| 19:22.41 | SeRi | <PROTECTED> |
| 19:22.58 | p3nguin | Does each number do something different? |
| 19:23.49 | p3nguin | Do you have a purpose for each number, or will all three numbers be used for the same thing? |
| 19:24.19 | SeRi | The main number rings my office. the second number rings my fax. my thrird number is off site at work. |
| 19:24.42 | p3nguin | Which one are you trying to configure now? All three? |
| 19:24.55 | SeRi | in this case is the second number which go to the same place as my office at home. |
| 19:25.19 | SeRi | no. I have my office all ready working just fine. now I want to route my second number to fax to email |
| 19:25.32 | p3nguin | You don't need a new account on VoIP.ms for that. |
| 19:25.44 | p3nguin | One asterisk system, one account. |
| 19:26.09 | SeRi | so I can link the same number to the same account? |
| 19:26.12 | p3nguin | Assign your DIDs to your one account. |
| 19:26.20 | SeRi | ah! ok |
| 19:26.30 | p3nguin | If you have three DIDs that all go to the same place, put all three DIDs on the same account. |
| 19:26.34 | p3nguin | Configure the account in sip.conf. |
| 19:26.41 | p3nguin | Add an extension for each DID. |
| 19:26.47 | p3nguin | Do something with each number. |
| 19:26.54 | p3nguin | I'll show you an example. |
| 19:27.12 | SeRi | hahaha d00d it went right over my head. lol Thanks let me change all this! |
| 19:27.13 | p3nguin | http://pastebin.com/Piqv4Egj |
| 19:27.20 | p3nguin | Line 29... |
| 19:27.59 | p3nguin | Look at line 37 for my fax number. |
| 19:28.12 | SeRi | ah yes nice! ok ok one sec. |
| 19:29.53 | p3nguin | And with a dedicated fax number, you do not need faxdetect. |
| 19:30.14 | p3nguin | Faxdetect is for a shared number, shared fax/voice. |
| 19:30.37 | p3nguin | I have a dedicated fax number, so I don't know too much about faxdetect. |
| 19:30.43 | p3nguin | But I'm going to play with it soon. |
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| 19:31.55 | irroot | me is working on new faxdetect for res_fax |
| 19:32.14 | irroot | make it more flexi |
| 19:33.11 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:38.35 | *** join/#asterisk bipul (~root@unaffiliated/bipul/x-4918593) |
| 19:39.14 | SeRi | p3nguin, I dont think the issue is on my end. I am getting a busy signal right away. and my logs do not show an attempt to reach the number in my system... |
| 19:39.36 | SeRi | is like is not been routed properly. |
| 19:39.51 | p3nguin | It probably isn't. |
| 19:40.01 | p3nguin | You probably don't have it configured right. |
| 19:40.14 | p3nguin | Are you using the main account or a sub account? |
| 19:40.33 | SeRi | the main account. I deleted the sub account since is not need it. |
| 19:40.50 | p3nguin | Did you go to DID management and put your new number on the main account? |
| 19:41.22 | SeRi | yes sr. |
| 19:41.57 | p3nguin | http://pastebin.com/fJgNLGLM |
| 19:44.10 | SeRi | all most the same settings here :) |
| 19:44.18 | SeRi | main account works just fine. |
| 19:44.48 | SeRi | mhhh I think I found my issue one sec |
| 19:44.48 | p3nguin | What is the context assigned to your voipms peer? |
| 19:47.30 | SeRi | one sec |
| 19:48.10 | p3nguin | Do you understand call flow? |
| 19:50.15 | SeRi | yes |
| 19:50.49 | p3nguin | Then my help is not needed. |
| 19:52.02 | p3nguin | Now, I have a question. On 1.4, I could originate say 20 calls from a shell script and they would all be concurrent. Now that I am using 1.8, if I do the same thing using the same script, calls are consecutive. Why? |
| 20:00.34 | bipul | how are you p3nguin |
| 20:00.42 | Micc | p3nguin, are you using .call files? |
| 20:01.22 | p3nguin | No, I am using a shell script that runs originate. |
| 20:01.45 | Micc | asterisk -rx 'originate ...' ? |
| 20:01.49 | p3nguin | right |
| 20:01.58 | Micc | where is it blocking? |
| 20:02.05 | p3nguin | Well, technically it's channel originate now, but still the same thing. |
| 20:02.37 | Micc | are you running multiple asterisk -rx at the same time? |
| 20:02.38 | p3nguin | In 1.4, it would run as many originates as I had numbers in the list. I actually had to limit the numbers in blocks to keep it from calling them ALL at once. |
| 20:02.50 | bipul | p3nguin, can i pm you |
| 20:03.00 | p3nguin | Now that I am using 1.8, I expected the same behavior, but it runs one at a time. |
| 20:03.31 | Micc | can you do multiople channel originate from the cli? |
| 20:03.43 | p3nguin | I don't know. I don't think so. |
| 20:03.53 | p3nguin | I think the CLI "sticks" when I run one. |
| 20:04.04 | p3nguin | I can't run anything else until it clears the originate. |
| 20:04.14 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 20:04.30 | p3nguin | I'll test that in a minute. |
| 20:04.41 | Micc | so that must be where its blocking, so maybe you can get around that by running multiple asterisk -rx at the same time. |
| 20:05.43 | Micc | something probably changed in channel originate cli command |
| 20:05.50 | p3nguin | Well the shell script runs a loop, reading numbers from a list, running originate with each number. Asterisk CANNOT block that from happening. |
| 20:06.03 | p3nguin | So it must be queueing up the originates or something. |
| 20:06.28 | Micc | if the cli is blocking then it would block the asterisk -rx command and you'll only get to execute one at a time. |
| 20:06.34 | Micc | unless your putting it in the background. |
| 20:06.38 | p3nguin | ~book |
| 20:06.38 | 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 |
| 20:06.40 | p3nguin | bipul: ^^^ |
| 20:06.41 | Micc | which is what I would do. |
| 20:07.12 | p3nguin | You think asterisk -rx is sticking? |
| 20:07.20 | p3nguin | I'll try backgrounding it. |
| 20:07.27 | Micc | well it would have to if it is sticking in the cli |
| 20:07.45 | Micc | running asterisk -rx will wait until the command is completed before returning. |
| 20:08.03 | p3nguin | It didn't in 1.4. |
| 20:08.09 | p3nguin | That must be what is different. |
| 20:08.26 | Micc | yeah, they must have changed something in the channel originate cli command. |
| 20:08.29 | *** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:08.57 | Micc | maybe they did it for a good reason and by going around it, you might cause some other problem. |
| 20:09.01 | Micc | but its worth a try. |
| 20:10.46 | p3nguin | It worked! |
| 20:11.09 | p3nguin | I just background the command in the script, and now it calls concurrently rather than consecutively. |
| 20:12.37 | p3nguin | The reason I wanted to make the calls at the same time is because of the possibility of failure being increased over a longer period of time. |
| 20:13.27 | p3nguin | If I have 60 numbers to call, and each message takes approx 30 seconds, it'll take half an hour to make all the times one by one. |
| 20:13.28 | Micc | I would watch it to be sure it doesn't deadlock or something. It seems strange that they would change that functionality on accident. |
| 20:14.09 | p3nguin | If I can make the calls in 20-number blocks, I'll only be making three blocks of calls. Now the time is reduced to just 1.5 minutes. |
| 20:14.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:15.21 | p3nguin | I'm not worried about system resources nor bandwidth. |
| 20:17.52 | Korolev | I am so tempted to sell you the termination for that dialer traffic! :P |
| 20:18.06 | p3nguin | You mean try. |
| 20:18.19 | p3nguin | I already have service. |
| 20:18.27 | Korolev | :D |
| 20:19.10 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 20:19.27 | p3nguin | I've been using asterisk 1.8 for a week, after having used 1.4 for years. Some things are slightly different... this problem with originate was one of those things. |
| 20:19.30 | Micc | Korolev, I'm always interested in looking at pricing for that kind of thing. |
| 20:20.31 | Micc | p3nguin, I used 1.8.5 for a day and had to revert back to 1.6 because it didn't support parking the same way yet. |
| 20:20.53 | Micc | I think 1.8.7 may work correctly. I hacked 1.8.5 but it was still hanging every 30 minutes. |
| 20:21.59 | Korolev | Micc, if its USA or Canada, its very very flexible depending on coverage |
| 20:22.38 | Korolev | goes from way below half a cent to nearly one cent |
| 20:22.48 | p3nguin | What company? |
| 20:22.59 | Korolev | Me Inc. :D |
| 20:28.57 | l1nuxman | does the sip.conf [name] have to be the same for exten => name,1 .....in extensions.conf? |
| 20:29.03 | p3nguin | no |
| 20:29.22 | l1nuxman | where do you use exten => name |
| 20:29.26 | l1nuxman | like the name |
| 20:29.27 | p3nguin | ~devicenames |
| 20:29.27 | 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. |
| 20:29.48 | l1nuxman | oh |
| 20:29.59 | p3nguin | It's extremely hard to dial 'name' from my phone, so I don't use a name for an extension. |
| 20:30.37 | p3nguin | Name the phones accordingly. Then determine an extension for dialing the phone. |
| 20:30.59 | p3nguin | For example, my phone is named 000011112222. My extension is 762. |
| 20:31.31 | p3nguin | 000011112222 is my phone's MAC address. 762 is "rob" on a keypad. |
| 20:32.31 | p3nguin | exten => 762,1,Dial(SCCP/000011112222,30) |
| 20:32.44 | l1nuxman | so my fxo device has a name that logs in to sip.conf listed [name] |
| 20:33.00 | p3nguin | okay |
| 20:33.07 | l1nuxman | yes? |
| 20:33.28 | p3nguin | If 'name' is a reasonable name for the device, I'd do it. |
| 20:34.05 | p3nguin | A distinguishable name or the MAC address would be reasonable names for it. |
| 20:35.19 | *** join/#asterisk ImTheBitch (Apples@ultra30.tptp.cc) |
| 20:35.28 | ImTheBitch | Hi. I have an issue w/ calls ending after 15 minutes. |
| 20:35.35 | ImTheBitch | Is anyone familiar w/ that problem? |
| 20:37.35 | Korolev | exactly 15 minutes? |
| 20:37.54 | p3nguin | Yep. |
| 20:37.59 | p3nguin | Check your session timer. |
| 20:39.07 | ImTheBitch | Exactly 15 minutes, yeh. |
| 20:39.40 | p3nguin | Calls over Dahdi? |
| 20:39.51 | ImTheBitch | Wat? |
| 20:40.04 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 20:40.23 | p3nguin | What channel technology are you using? |
| 20:41.09 | ImTheBitch | I have no idea. I just run an asterisk server on my edge router for my hardware sip phone. |
| 20:41.14 | ImTheBitch | I make calls w/ sipgate+google evoice. |
| 20:41.56 | ImTheBitch | I don't know an awful lot about asterisk, tbh. |
| 20:42.42 | ImTheBitch | Is this what I'd like to change? ;session-expires=600 |
| 20:44.13 | p3nguin | Try session-timers=refuse |
| 20:44.33 | ImTheBitch | https://issues.asterisk.org/view.php?id=16748 |
| 20:44.36 | ImTheBitch | I guess this is the same problem. |
| 20:45.38 | p3nguin | I've made a suggestion. The rest is up to you. |
| 20:47.08 | *** part/#asterisk ImTheBitch (Apples@ultra30.tptp.cc) |
| 20:47.12 | *** join/#asterisk ImTheBitch (Apples@ultra30.tptp.cc) |
| 20:47.24 | ImTheBitch | Alright, thanks. |
| 20:54.43 | *** join/#asterisk paec (~user@AAmiens-551-1-68-36.w92-142.abo.wanadoo.fr) |
| 20:59.29 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:05.27 | *** part/#asterisk paec (~user@AAmiens-551-1-68-36.w92-142.abo.wanadoo.fr) |
| 21:11.36 | *** join/#asterisk gogasca (~Adium@c-71-202-75-108.hsd1.ca.comcast.net) |
| 21:13.12 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:27.18 | *** part/#asterisk catphish_ (~charlie@2001:9d8:2005:2::3) |
| 21:42.52 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:47.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:53.47 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 21:55.08 | Micc | p3nguin, refuse doesn't work with some providers, they will refuse the call without timers. |
| 22:00.06 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:02.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:04.04 | Micc | There is an easy hack to fix the problem. |
| 22:04.29 | Micc | I think it may be a problem with timers and sonus equiopment. |
| 22:04.51 | Micc | I know google voice uses 360, and they have sonus equiopment. |
| 22:04.55 | Micc | not sure about sipgate. |
| 22:06.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 22:21.21 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:52.37 | l1nuxman | I'm confused. If I have an FXO device and it looks like this. Asterisk <->HT503 -> Phone line, & also HT503 -> Phone. Which are the FXO and FXS? |
| 22:55.18 | p3nguin | Do the ATAs have multiple jacks? |
| 22:55.26 | l1nuxman | yes |
| 22:55.33 | p3nguin | Are they labeled? |
| 22:55.49 | l1nuxman | line,phone,lan,WAN |
| 22:56.05 | p3nguin | You don't know what plugs into each of those things? |
| 22:56.33 | l1nuxman | no I have them hooked up physically, but the configuration in extensions and sip conf are confusing me |
| 22:56.53 | p3nguin | Did you create a peer entry in sip.conf for the device already? |
| 22:57.17 | l1nuxman | how does it work when I want to call an extension 101 internally on my phone to ASterisk. And how does it differ when someone calls from outside analog line to Asterisk |
| 22:57.32 | l1nuxman | let me show you |
| 22:57.41 | p3nguin | The Line will have one peer entry, and the Phone will have another. |
| 22:57.56 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 22:58.13 | p3nguin | If a call comes in on the line, it will go to the extension you've configured in the Line section of the device. |
| 22:58.14 | l1nuxman | the labels might be wrong and the names are being changed constantly. http://pastebin.com/1NrLMa19 |
| 22:58.50 | l1nuxman | Line Section the FXO? |
| 22:58.58 | p3nguin | Yes. |
| 22:59.10 | p3nguin | Phone section is FXS. |
| 22:59.22 | l1nuxman | or Unconditional Call Forward to VOIP: |
| 22:59.48 | p3nguin | If you want all calls coming in on the phone line to go to Asterisk, I'd enable that. |
| 23:00.47 | l1nuxman | http://pastebin.com/PfqT4ZKh p3nguin |
| 23:01.55 | p3nguin | That doesn't look to be unreasonable. |
| 23:04.27 | p3nguin | In that style of device, you have to determine which calls will go through asterisk and which ones will go direct to the phone line. |
| 23:04.42 | p3nguin | There is a dial plan in the device to control it, based on the dialed number. |
| 23:11.50 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 23:17.22 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 23:26.52 | l1nuxman | ok I'm getting a username mismatch, authenticate error but everything looks ok... |
| 23:26.59 | l1nuxman | http://pastebin.com/yGVad5zz |
| 23:29.08 | p3nguin | You've configured the device to have a username of 'myhousephones'? |
| 23:30.18 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 23:35.27 | *** join/#asterisk drynish (~drynish@modemcable069.177-58-74.mc.videotron.ca) |
| 23:35.33 | drynish | I need an idea |
| 23:35.36 | drynish | :P |
| 23:36.14 | *** join/#asterisk zz_tm1000 (~tm1000@li251-245.members.linode.com) |
| 23:36.23 | drynish | My sound is not working in DISA |
| 23:36.30 | l1nuxman | oh I see I think lol |
| 23:36.33 | drynish | I don't know why I have no idea |
| 23:36.48 | drynish | :P |
| 23:36.48 | p3nguin | What are you doing with DISA? |
| 23:37.18 | drynish | My ATA is really bad, I need to use DISA to make outgoing calls |
| 23:37.36 | drynish | My ATA is dialing in my asterisk box as soon as I take the line |
| 23:37.42 | p3nguin | It sounds to me that you're doing it wrong. |
| 23:38.19 | p3nguin | Misconfiguration in the ATA, I guess. That's not what DISA is for. |
| 23:38.39 | drynish | The ATA is configured the right way, it was working before, when I was on debian. |
| 23:38.44 | drynish | I took the same config, put in gentoo |
| 23:38.48 | drynish | and right now it's not working |
| 23:38.56 | drynish | core and extra sounds are installed |
| 23:39.04 | drynish | I'm just wonedering what I could be missing |
| 23:39.24 | p3nguin | Your operating system on your computer has nothing to do with the ATA. If your ATA is working like you've described, it is configured wrong. |
| 23:41.09 | drynish | http://www.voip-info.org/wiki/view/Zoom+5801 |
| 23:41.14 | drynish | it is a really bad ATA |
| 23:41.18 | drynish | but that's the one I have |
| 23:42.30 | p3nguin | Even thw WORST ATA can work correctly. |
| 23:43.18 | drynish | What are you insinuate? |
| 23:43.25 | p3nguin | It's configured wrong. |
| 23:43.28 | drynish | Please read this page |
| 23:43.33 | drynish | if you don't believe me |
| 23:43.34 | p3nguin | I really don't want to. |
| 23:44.10 | p3nguin | I know how ATAs work, and I know how they work with Asterisk. |
| 23:44.38 | drynish | Wow ! |
| 23:44.47 | drynish | Are you sitting next to god or what? |
| 23:44.54 | drynish | :P |
| 23:44.58 | p3nguin | Next to? |
| 23:45.10 | drynish | on ? :P |
| 23:45.17 | p3nguin | You're confused. |
| 23:45.20 | p3nguin | I am God. |
| 23:46.17 | drynish | Having an FXO port makes you want to use it for Asterisk! Well it can to a point. You can have the ATA act as a single line SIP FXO port (in and out) or an ATA, but not concurrently - at this time |
| 23:46.21 | p3nguin | Unless you can cite where this ATA is said to be hotline only, I'm going to assume it works like any other ATA. |
| 23:46.41 | p3nguin | FXO ports are not for phones. |
| 23:47.04 | WIMPy | o.O |
| 23:47.22 | drynish | dunno |
| 23:47.28 | p3nguin | FXO ports are to have your line connected to them. |
| 23:48.06 | drynish | I know what are fxo |
| 23:48.38 | p3nguin | So you didn't plug your phone into it? |
| 23:50.30 | drynish | no |
| 23:50.41 | drynish | I'm plugging my phone on the FXS port of my ATA |
| 23:51.13 | p3nguin | Okay, great. |
| 23:51.32 | p3nguin | Now where's the setting for dialing from the phone? |
| 23:56.39 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |