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| 03:33.58 | josefig | on extensions.conf how can be set the prefix ? I'd like to receive an extension like 11XXXXXXXX where the XXXX will be the a phone number, something like: exten => _11XXXXX,1,Dial(H323/@service) .. or how is it ? |
| 03:46.15 | p3nguin | If the person will be dialing 11 followed by some numbers, that could be what you're looking for. |
| 03:46.46 | p3nguin | After the 11, how many other digits will be entered? |
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| 03:48.02 | josefig | p3nguin: 10 numbers |
| 03:48.23 | josefig | does extesions.conf support some regex match ? |
| 03:48.57 | p3nguin | Yes, but you don't use regex for extension matching. You just use patterns. |
| 03:49.34 | p3nguin | Will those 10 digits be north american phone numbers? |
| 03:49.38 | josefig | p3nguin, great. How can I match that? 11 after a number, ie. UK 44 … would be 1144XXXXXX |
| 03:50.05 | josefig | will be worldwide. |
| 03:50.53 | josefig | because I did this .. exten => _X.,1,Dial(ooh323,${EXTEN}@mainroute) |
| 03:51.05 | p3nguin | That is invalid syntax. |
| 03:51.45 | josefig | damn :( |
| 03:51.59 | p3nguin | _X. will match one digit followed by one or more characters. Dial() requires Tech/extension or Tech/peer/extension or Tech/extension@domain.com |
| 03:52.32 | josefig | I'm following this url http://forums.digium.com/viewtopic.php?f=1&t=22152 |
| 03:52.41 | josefig | but I guess is not as useful as I though. |
| 03:52.54 | p3nguin | ~patterns |
| 03:52.55 | infobot | somebody said patterns was a method to install groups with zypper. Use `zypper pt` to list patterns, use `zypper install -t pattern pattern-name` to install them. |
| 03:52.59 | p3nguin | ~dialplab patterns |
| 03:53.03 | p3nguin | ~dialplan patterns |
| 03:53.03 | infobot | dialplan patterns are explained here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 03:53.07 | p3nguin | There ^^^ |
| 03:53.14 | josefig | thank you p3nguin |
| 03:53.23 | josefig | let me make a review. |
| 03:54.15 | p3nguin | exten => _1144XXXXXXXX,1,Dial(H323/mainroute/${EXTEN:2}) |
| 03:54.30 | p3nguin | This extension will match 11 44 and eight more digits 0-9. |
| 03:54.56 | josefig | ok, after mainroute/${EXTEN:2} is that neccesary ? |
| 03:55.09 | josefig | the ${EXTEN:2} |
| 03:55.10 | p3nguin | It will use the h323 channel driver, sending via peer named mainroute to the number you dialed without the 11 on it. |
| 03:55.27 | josefig | oh I got it! |
| 03:55.30 | josefig | ^^ |
| 03:55.55 | p3nguin | So if you enter 11 44 8080 1234, it would send 4480801234 via peer mainroute. |
| 03:56.10 | josefig | yes, that's just what I needed! thank you really! |
| 03:56.15 | josefig | a beer for ya! ;) |
| 03:56.47 | p3nguin | If you have variable length numbers, you will need to have another pattern for that. |
| 03:57.04 | josefig | like in the very next line ? |
| 03:57.13 | josefig | or changing it ? |
| 03:57.52 | p3nguin | You could do it two ways. You could change that pattern to _1144XXXXXXX.,1,Dial() |
| 03:58.25 | p3nguin | or you could make a second one with _1144XXXXXXXX.,1,Dial() |
| 03:58.35 | p3nguin | Each X is a digit 0-9. |
| 03:58.50 | p3nguin | The dot means one or more characters (letters or numbers). |
| 03:59.10 | josefig | and if I don't know how many X's I need? sometimes could be 10 and sometimes could be 9 ? |
| 03:59.29 | p3nguin | Then you could remove one X and replace it with a dot. |
| 03:59.41 | josefig | oh great, I got it. |
| 03:59.52 | p3nguin | or even remote two Xs and replace with one dot. |
| 04:00.09 | p3nguin | Even using _1144X. will still match 10 more digits. |
| 04:00.29 | p3nguin | I just like to be as close to the number of digits as possible. |
| 04:00.35 | p3nguin | Better patterns match first. |
| 04:00.37 | josefig | yes, I understood. |
| 04:02.06 | josefig | I have one more question, after the Dial(H323.. <-- If I add H323 it means I need to setup the h323.conf instead of adding ooh323 and using ooh323.conf right ? |
| 04:03.02 | p3nguin | You'll have to determine your channel tech. I just used H323 because I know the tech name for the channel driver is H323. If ooh323, you may need to use Dial(ooh323/......) |
| 04:03.33 | p3nguin | I don't use either, so I don't know the tech name. |
| 04:03.39 | josefig | yep, thank you! |
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| 04:33.47 | josefig | p3nguin: when I try to make a call in my softphone says Call Rejected and on messages say: [Dec 2 22:32:09] WARNING[2378][C-00000000] app_dial.c: Dial argument takes format (technology/resource) this is how I added the exten, exten => _11X.,1,Dial(h323,mainroute,${EXTEN:2}) but is not working |
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| 04:44.46 | jpsharp | You have to dial with h323/mainroute, not h323,mainroute. |
| 04:45.21 | p3nguin | That's what I told him before. |
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| 04:56.26 | josefig | p3nguin, yep my bad, now is working ;) thank you! |
| 04:56.39 | josefig | and the chan is ooh323 instead of h323 |
| 04:56.40 | josefig | :D |
| 04:56.44 | josefig | I'm very happy now! |
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| 06:43.45 | unicron | o hy |
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| 07:04.35 | salz212 | Guys I want to Log Agi verbose in logs..this is not agi debugs I am talking about.. I need just the AGI->verbose to be logged in Logs. It was working in Asterisk 1.6 but now in Asterisk 11.0.1 it is not.. Any clue whats the change?. |
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| 07:11.44 | kaldemar | salz212: did you ever answer when you were asked how the global variable in your dialplan was used? |
| 07:13.11 | kaldemar | and there is no such thing as agi verbosity in asterisk. you're talking about something custom. |
| 07:13.21 | salz212 | nah i was gone at that time. Anyways what I am doinf in extensions.conf is : [globals] AGIVERBOSE=9 |
| 07:13.58 | kaldemar | and the rest? |
| 07:14.08 | kaldemar | setting a global variable alone does nothing at all. |
| 07:14.20 | salz212 | also in cli.conf |
| 07:14.36 | salz212 | point is it should log all agi-verbose logs thats it. |
| 07:14.55 | kaldemar | log where? |
| 07:15.12 | salz212 | messages |
| 07:15.33 | kaldemar | logger.conf |
| 07:15.40 | salz212 | I want all CLI logs to be logged in messages thats it. I can see AGI logs on CLI but they are not going to log |
| 07:15.47 | salz212 | yes I know about logger.conf |
| 07:15.54 | kaldemar | anyway, are you going to tell what you have used the AGIVERBOSE variable for? |
| 07:16.46 | kaldemar | are you having trouble configuring logger.conf? |
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| 07:23.05 | salz212 | No i dont think so, I have enabled all for consol and messages but it is still not logging AGI-Verbose logs |
| 07:24.01 | kaldemar | what do you mean by "AGI-Verbose"? |
| 07:25.01 | salz212 | I am using perl Asterisk::AGI in which I am verbosing to logs it is working fine on Asterisk 1.6 but not on 11 |
| 07:26.02 | kaldemar | do you see the verbose messages in CLI? |
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| 07:27.30 | kaldemar | if it is the "verbose" command Asterisk::AGI really uses and your logger has "verbose" on the "messages" line, then it should work. |
| 07:28.28 | salz212 | yes.. i do see them.. that is what bugging me |
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| 07:30.11 | kaldemar | salz212: do what does your logger.conf look like? |
| 07:32.17 | salz212 | currently : console => notice,warning,error,verbose,dtmf and messages => notice,warning,error,verbose, though I have done hit and trial. |
| 07:33.18 | kaldemar | the whole file |
| 07:33.34 | salz212 | ok |
| 07:36.25 | salz212 | here http://pastebin.com/cGee7cMK |
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| 07:41.19 | salz212 | found something.. I hope you do..It has wasted a lot of time. |
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| 07:47.54 | kaldemar | salz212: nothing. do you get any verbose messages in the file? |
| 07:48.22 | salz212 | from dialplan . yes.. from AGI no |
| 07:51.45 | kaldemar | enable agi debug to see what your AGI really does. |
| 07:56.19 | salz212 | did that and it work as smooth as it should be. |
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| 08:00.48 | kaldemar | salz212: obviously something is not working if you don't see messages in your file. |
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| 08:01.08 | kaldemar | what is the CLI output with AGI debug enabled? |
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| 08:02.45 | salz212 | everything.. all the logic set verbose everything is being displayed.. like it should be. |
| 08:02.57 | surferboy | I'm not very good with asterisk |
| 08:03.01 | surferboy | can someone lend a hand |
| 08:03.11 | surferboy | I love learning this stuff |
| 08:03.37 | surferboy | if I dial an extension directly I get through to the voicemail |
| 08:03.53 | surferboy | if I get transfered from another extension I don't go through to the voicemail |
| 08:04.06 | surferboy | I have the output from asterisk in verbose mode |
| 08:04.13 | surferboy | what should I look for in there? |
| 08:06.20 | ChannelZ | my guess is that the other extension you get transferred from is in another context and your dialplan doesnt behave the same way in both |
| 08:07.44 | kaldemar | salz212: i don't take your word for it. |
| 08:08.10 | surferboy | ChannelZ, how can I check that? |
| 08:08.18 | salz212 | okay don't, I don't have anything left to say. |
| 08:08.19 | surferboy | ChannelZ, or more importantly how can I fix that |
| 08:08.21 | surferboy | haha |
| 08:08.47 | ChannelZ | ...by fixing it... |
| 08:08.49 | salz212 | i'll eventually resolve it, do't worry. tc. |
| 08:09.21 | ChannelZ | make calls, pastebin the console. |
| 08:09.45 | kaldemar | salz212: you have left to use pastebin and show the output. |
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| 08:10.48 | kaldemar | salz212: if not, you're just wasting time here by saying "everything is correct but not working", since logger does write verbosity to file in 11.0.1. |
| 08:13.04 | surferboy | I don't think I should pastbin this info |
| 08:13.25 | ChannelZ | salz212: worked for me... [Dec 3 01:11:54] VERBOSE[29861][C-00000046] res_agi.c: callhandler.php: blarfffff |
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| 08:14.54 | ChannelZ | surferboy: well, good luck then. A transferred call is a new call by the phone you're transferring from. Something is probably different about your dialplan as a result giving the behavior you describe. |
| 08:15.23 | salz212 | okay i found the problem. its perl AGI which takes message and a verbose level ... in case of absense of verbose level I had added the verbose level to 9.... I changed it to 3 and it worked... Wonderful, still illogical. |
| 08:15.41 | surferboy | what should I check for in the dial plan? thanks a lot for your assistance ChannelZ |
| 08:16.06 | ChannelZ | Check that it's doing whatever you wanted it to do. |
| 08:16.47 | ChannelZ | I'm not clairvoyant |
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| 08:17.32 | ChannelZ | I get the feeling however you maybe didn't make this dialplan... do I smell FreePBX? |
| 08:20.04 | surferboy | haha |
| 08:20.08 | surferboy | elastix |
| 08:21.04 | kaldemar | surferboy: have you been to #elastix ? |
| 08:21.05 | ChannelZ | same difference |
| 08:21.49 | surferboy | #asterisk is better |
| 08:22.04 | surferboy | the console verbose output looks pretty similar |
| 08:22.09 | ChannelZ | But you need support with Elastix, not Asterisk. |
| 08:22.10 | surferboy | what should I check out though? |
| 08:23.26 | kaldemar | surferboy: it's not the question of which channel you consider better. elastix is not supported on this channel. |
| 08:23.42 | surferboy | it's asterisk under the gui |
| 08:23.53 | ChannelZ | yes but the GUI wrote all the config files |
| 08:23.55 | kaldemar | surferboy: read the following and apply it to elastix: |
| 08:23.58 | kaldemar | ~freepbx |
| 08:23.58 | 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 |
| 08:24.19 | ChannelZ | and the dialplans it makes are enormous. You have a problem with THEIR operation. |
| 08:25.10 | kaldemar | you're giving a black box here and asking why it does not do what you expect. no one will be able to give you an answer. |
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| 08:25.58 | surferboy | 185 peeps vs. 20 peeps |
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| 08:28.25 | ChannelZ | It's not a matter of numbers. |
| 08:28.51 | surferboy | more numbers |
| 08:28.54 | kaldemar | surferboy: did you read what we just said? |
| 08:28.55 | surferboy | more help |
| 08:29.00 | surferboy | yip |
| 08:29.07 | kaldemar | so you did not understand. |
| 08:30.18 | kaldemar | during the 27 minutes you've been here, you've basically been told to ask elsewhere. you're not even letting us help you help yourself. |
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| 08:43.03 | bombev | good morning |
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| 09:19.38 | fredericve | Hi, I have a sip trunk, and when the remote party puts me on hold, I hear my own music on hold. I'm using asterisk 1.8.13.0. |
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| 09:19.52 | fredericve | Any way to fix that? |
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| 09:21.30 | Trixboxer | Hi, Im trying to create a menu IVR, in which I would be doing an announcement, then ask for name then ask for contact no then ask for information and then hangup. My problem is Im not able to record the call since announcement |
| 09:22.11 | ChannelZ | What do you mean "not able" |
| 09:22.59 | Trixboxer | Im using elastix UI to create the IVR, Im unable to get a single recorded wav file which will have info since the call came in |
| 09:23.26 | Trixboxer | I think I need to do something in ivr-10-custom but not sure what exactly goes in |
| 09:23.32 | ChannelZ | #elastix or #freepbx |
| 09:24.05 | Trixboxer | its a custom dialplan and does not relate to elastix ui or freepbx ui |
| 09:24.43 | ChannelZ | Then why did you just say "Im using elastix UI to create the IVR" |
| 09:25.26 | Trixboxer | sorry about that, its only the initial ivr which is made by it and now I'm customising it with additional asterisk info |
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| 09:25.56 | ChannelZ | Well I guess pastebin what you're doing then. |
| 09:26.27 | ChannelZ | fredericve: that one's kind of complicated.. |
| 09:27.56 | Trixboxer | sure, allow me some time to put up |
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| 09:28.46 | bulkorok | hey... I have a sangoma voicetime ut50... dahdi 2.6 and asterisk 1.8.18 is installed.. do I just plug this thing in and reconfigure dahdi?! |
| 09:28.51 | bulkorok | reload module etc!? |
| 09:30.11 | bulkorok | does that thing could help with MixMonitor jitter!? |
| 09:30.55 | ChannelZ | assuming DAHDI sees it as some device it knows about, reloading the DAHDI drivers at minumum yes.. |
| 09:31.50 | ChannelZ | but if I had to guess I'd assume they have a custom driver |
| 09:31.58 | bulkorok | they have one... |
| 09:32.36 | ChannelZ | http://wiki.sangoma.com/Voicetime-USB-Sync |
| 09:33.10 | bulkorok | oh... great note :-/ |
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| 09:35.39 | ChannelZ | What version of Asterisk? |
| 09:35.47 | bulkorok | 1.8.18 |
| 09:36.08 | bulkorok | it seems like MixMonitor is causing some delay/jitter/dropped packets |
| 09:36.22 | bulkorok | so I thought it's a timing one... |
| 09:36.29 | ChannelZ | I don't think MixMonitor is dependent on DAHDI |
| 09:37.20 | bulkorok | it's heavenly jitter... |
| 09:37.57 | ChannelZ | Heavenly? |
| 09:38.30 | bulkorok | well... lots of... |
| 09:38.58 | ChannelZ | Oh. Heavily perhaps. |
| 09:39.13 | bulkorok | :-) |
| 09:39.46 | ChannelZ | What codec are either side of the call using? |
| 09:39.57 | bulkorok | alaw |
| 09:41.47 | fredericve | ChannelZ: In what way complicated? |
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| 09:43.13 | ChannelZ | As in, it may or may not be something you can control, and there are a lot of variables as to why it's happening. For instance see https://issues.asterisk.org/jira/browse/ASTERISK-19353 for some discussion |
| 09:44.27 | bulkorok | I need some traces I suppose... |
| 09:45.22 | ChannelZ | bulkorok: Have you tried/do you have issues with ChanSpy as well, or just Record()ing files coming out badly? |
| 09:46.08 | ChannelZ | And is this a VM, or a poopy server? |
| 09:47.20 | bulkorok | it's a hardware server... |
| 09:47.25 | bulkorok | I just make recordings... |
| 09:47.56 | bulkorok | and I see that calls going out are bad... I can not see if they are comming in bad (that's why I trace now...) |
| 09:48.58 | bulkorok | but I suppose they come in ok, because calls come from SNOM phones... |
| 09:50.48 | ChannelZ | Sounds like maybe this isn't a MixMonitor issue at all? |
| 09:51.06 | bulkorok | I don't think so too, but I have to be sure... |
| 09:51.26 | bulkorok | can be I/O-problem from hdd... |
| 09:52.23 | ChannelZ | but you're saying that "calls going out are bad" - or are you only judging this by what you hear in MixMonitor recordings? |
| 09:52.44 | bulkorok | I can see that with my monitoring tool |
| 09:53.42 | ChannelZ | which is/does what |
| 09:54.48 | ChannelZ | Ffffuuuuuu it's 3am I gotta go to bed |
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| 09:55.00 | bulkorok | it's voipfuture rtpmon which is tapping all sip/rtp streams |
| 09:55.19 | bulkorok | http://www.voipfuture.com/ |
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| 10:38.39 | salz212 | kaldemar: you there? |
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| 10:39.54 | salz212 | I found the problem and solution, thought you deserve to know as you also spent quite some time with me. So, it the verbose level which we set in agi->verbose(Logz,Verbose LEVEL) .. this is just an example.. so if the verbose level is 9 Agi logs are never logged; it work in level 3. Weird though but its a solution for me :) |
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| 10:49.42 | kaldemar | salz212: the real reason is that the verbose level on your asterisk is below 9. |
| 10:50.16 | salz212 | no, Its not.. its 99 |
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| 10:51.29 | salz212 | it only works at 3 ... for agi logs. |
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| 11:12.47 | kaldemar | salz212: well, works here. |
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| 11:51.44 | salz212 | this verbose level i am talking about is set in ASTERISK::AGI for each log. Its not "core set verbose VALUE". |
| 11:52.35 | kaldemar | it is the asterisk verbosity level eventually. the module may have bugs then. |
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| 12:46.09 | *** join/#asterisk mathi (Matthew@ip-62-235-193-96.dsl.scarlet.be) |
| 12:46.10 | mathi | hi |
| 12:47.46 | mathi | to connect the PSTN to my asterisk server, what kind of card do I need ? |
| 12:47.48 | mathi | http://www1.digium.com/en/products/telephony-cards |
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| 12:50.12 | Chainsaw | mathi: That depends on how your current telephone lines are delivered. |
| 12:50.26 | Chainsaw | mathi: Should they be analog, please promise me that you will avoid the bargain basement X100 cards. |
| 12:51.12 | mathi | Chainsaw, I don't know how that works honestly:( |
| 12:51.41 | Chainsaw | mathi: Well, you have telephone lines coming in. What country is this, and what type of telephone line is it. |
| 12:52.30 | mathi | Chainsaw, I live in Belgium, what types of telephone lines are there? |
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| 12:52.46 | Chainsaw | mathi: Analog & ISDN. Both likely supplied through Belgacom. |
| 12:53.20 | Chainsaw | mathi: Do you have one of their business exchange boxes with the piano tune hold music, or just direct lines into the wall? |
| 12:53.22 | Chainsaw | plays the 12 second hold loop on his air piano |
| 12:54.06 | mathi | :-) I think it's from the wall... |
| 12:54.07 | WIMPy | mathi: How many lines. |
| 12:54.12 | WIMPy | ? |
| 12:54.28 | mathi | I just need one line into my asterisk, it's to tests home on my server |
| 12:54.35 | carrar | Oh sure |
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| 12:54.38 | carrar | it starts out with just 1 |
| 12:54.47 | carrar | then you want more and more |
| 12:54.53 | WIMPy | No, how many do you have, not how many do you want to connect. |
| 12:55.32 | mathi | WIMPy, I have one line home but I need a second for my server |
| 12:56.05 | Chainsaw | mathi: Could you look on the wall whether you have an "NT1" box with a green light please? |
| 12:56.05 | Chainsaw | mathi: Is it a modem-style RJ11 connector, with 4 metal strips in a 6-hole connector, or an RJ45 connector with at least 4 metal strips in an 8-hole connector? |
| 12:56.31 | mathi | Chainsaw, ok i'm gonna check that :P i'll be back |
| 12:56.36 | mathi | (might take some time :-)) |
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| 13:16.00 | surferboy | I'm going to ctcp version all of you |
| 13:16.06 | surferboy | cause you are all assholes |
| 13:16.31 | p3nguin | I am devastated. |
| 13:16.50 | ectospasm | don't dewit |
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| 13:23.54 | carrar | OMG |
| 13:26.01 | kaldemar | surferboy: what was that for? |
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| 13:27.06 | carrar | Cause he's about to fall off the FISCAL CLIFF!!!! |
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| 13:33.58 | jacekowski | i'm using elastix/freepbx |
| 13:34.02 | p3nguin | I'm sorry. |
| 13:34.31 | jacekowski | and i'm trying to set it up so when someone calls my number it rings 5 phones (i've got that sorted with ring group) |
| 13:34.43 | jacekowski | but then if it's not picked up it goes to voicemail (that's sorted as well) |
| 13:34.47 | p3nguin | We don't support those products here. |
| 13:35.03 | jacekowski | but i want message to change based on time of day |
| 13:35.14 | p3nguin | If you want help with asterisk, that we can do. |
| 13:35.21 | jacekowski | ok, how to do it with pure asteris |
| 13:35.24 | jacekowski | asterisk* |
| 13:36.23 | p3nguin | You can use a combination of GotoIfTime and Playback/BackGround, or ExecIfTime and Playback/BackGround. |
| 13:38.36 | p3nguin | I personally use ExecIfTime and BackGround to play a sound file conditionally based on the time. |
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| 13:39.12 | p3nguin | I even use ExecIfTime and WaitExten after the sound file plays to give the caller some extra time to enter the extension. |
| 13:39.33 | [TK]D-Fender | p3nguin, func_iftime. No need for even a whole extra line of code |
| 13:39.48 | p3nguin | I'll look into that right now. |
| 13:41.04 | p3nguin | Now to implement the change and test it. |
| 13:42.35 | p3nguin | What is going to happen when I playback nothing because the timespec is false and I only specify a file for true or vice versa? |
| 13:43.30 | p3nguin | I don't see how this is going to be better than my current ExecIfTime. |
| 13:46.39 | kaldemar | when you have two lines with execif, you may end up playing back both files. |
| 13:48.17 | p3nguin | I don't quite understand. Here's what I have that currently works: |
| 13:48.23 | p3nguin | same => n,ExecIfTime(18:00-22:00,mon-fri,*,*?BackGround(if-u-know-ext-dial)); |
| 13:48.25 | p3nguin | same => n,ExecIfTime(18:00-22:00,mon-fri,*,*?WaitExten(3)); |
| 13:48.43 | p3nguin | If the time is outside that time range, the sound doesn't play and the waitexten doesn't wait. |
| 13:48.51 | [TK]D-Fender | p3nguin, There is a "false" for IFTIME |
| 13:49.04 | p3nguin | I saw that. |
| 13:49.18 | kaldemar | the tiem might do out of the spec during playback. |
| 13:49.33 | [TK]D-Fender | p3nguin, And since you're suggesting playing a different announcement based on time, just use the busy/unavail for that and you can do it all in the call to VM |
| 13:50.07 | p3nguin | It's part of the AA. |
| 13:51.01 | p3nguin | After 10pm, you don't get the "if you know the extension, dial it now" message; you just get the regular after hours messages. |
| 13:51.43 | ghost75 | is there any soundfile like: press bla to forward to mobile phone? |
| 13:51.50 | p3nguin | It works great, so I don't see a reason to change mine. |
| 13:52.02 | [TK]D-Fender | ghost75, Nope |
| 13:52.13 | p3nguin | You can easily make one. |
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| 14:00.13 | mathi | I made some pictures of my PSTN line, could someone tell me what card I should purchase absed on the pics? |
| 14:01.06 | mathi | this is the wall: http://oi49.tinypic.com/17bngl.jpg and here is the connection with modem: http://oi47.tinypic.com/24oaxvt.jpg |
| 14:02.21 | [TK]D-Fender | mathi, Looks like some non- North American POTS plugged into a DSL modem. |
| 14:02.32 | mathi | [TK]D-Fender, I'm from Belgium |
| 14:02.47 | WIMPy | mathi: Looks analog. |
| 14:02.48 | [TK]D-Fender | mathi, Which would mean any of the usual analog cards should work (give or take the regional signalling) |
| 14:03.04 | WIMPy | But if you want to add a 2nd line, you probably want to change that for a BRI. |
| 14:03.08 | leifmadsen | FXO is for phone company (wall) and FXS is for telephone (some phone) |
| 14:03.22 | leifmadsen | ya, Belgium probably has BRI capabilities |
| 14:05.52 | mathi | could I put an asterisk server instead of the DSL modem? or am I forced by my provider (Belgacom) to use the DSL modem for my phones? |
| 14:06.55 | WIMPy | Hmm. Is that an IAD your phone is connected to? |
| 14:07.28 | [TK]D-Fender | mathi, Typically the DSL is just for INTERNET access. how do you figure Asterisk has any role in getting rid of that? |
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| 14:07.56 | [TK]D-Fender | mathi, Probably the only reason the phone is plugged into it it is for convenience of it having a filtered passthrough jack on it. |
| 14:08.26 | WIMPy | Or he doesn't have a phone line at all and that thin just contains an ATA. |
| 14:08.55 | [TK]D-Fender | WIMPy, Yes, that could be it as well |
| 14:09.30 | [TK]D-Fender | WIMPy, So combo DSL modem on dry line w/ ATA built in... pretty masty, but I can imagine it. |
| 14:09.41 | [TK]D-Fender | nasty* |
| 14:10.02 | WIMPy | That's what the future has been for sever years in Europe. |
| 14:10.07 | WIMPy | several |
| 14:10.19 | [TK]D-Fender | Wouldn't that be the PAST few years? ;) |
| 14:10.23 | mathi | it is an IAD, it provides telephony and internet ... so how could I put an Asterisk server in between? without losing the internet. Sorry i'm totally confsed :P |
| 14:10.55 | mathi | or is it impossible and I need to buy a second PSTN line coming from the wall ? :-( |
| 14:10.55 | [TK]D-Fender | mathi, You shouldn't be. |
| 14:11.17 | [TK]D-Fender | mathi, you need to actually KNOW what the box is doing first |
| 14:11.19 | WIMPy | mathi: I depends on what your provider allows. Will they give you the credentials for your phone account? |
| 14:11.22 | [TK]D-Fender | mathi, Go find out. |
| 14:12.09 | WIMPy | What other connections does the IAD have? Here they are usually equipped with 2 FXS and one S0. |
| 14:13.48 | mathi | WIMPy, you can see on the picture, seems there are two FXS and one FXO |
| 14:13.49 | surferboy | what is huntloop=1 |
| 14:14.31 | mathi | [TK]D-Fender, how do I find out? and what questions am I looking for ? |
| 14:14.50 | WIMPy | mathi: I don't think there's an FXO. |
| 14:15.11 | mathi | WIMPy, well, there is one cable connected from the wall to the modem, this is FXO |
| 14:15.19 | WIMPy | Do you have a model? |
| 14:15.31 | WIMPy | That must be DSL. |
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| 14:17.37 | [TK]D-Fender | surferboy, Where do you see this? |
| 14:18.06 | mathi | WIMPy, it's a SAGEM vdsl2. I think it's DSL indeed... so how does the phone ring? :P |
| 14:18.08 | kaldemar | surferboy: elastix-specific stuff. ask in #elastix. |
| 14:18.46 | surferboy | this is asterisk |
| 14:18.48 | surferboy | no elastix |
| 14:18.54 | surferboy | ... |
| 14:19.00 | surferboy | .. |
| 14:19.00 | surferboy | . |
| 14:19.00 | surferboy | ... |
| 14:19.00 | surferboy | .. |
| 14:19.00 | surferboy | . |
| 14:19.07 | surferboy | . |
| 14:19.10 | surferboy | ... |
| 14:19.16 | surferboy | <PROTECTED> |
| 14:19.22 | surferboy | lol |
| 14:19.24 | surferboy | jobs |
| 14:19.27 | surferboy | jokes |
| 14:19.30 | kaldemar | leifmadsen: would you mind doing something for him? |
| 14:19.40 | surferboy | yip |
| 14:19.45 | p3nguin | mjordan: Hi. |
| 14:19.45 | surferboy | I forgot to tell you something |
| 14:19.56 | surferboy | you ready? |
| 14:19.58 | surferboy | wait for it |
| 14:20.00 | surferboy | wait for it |
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| 14:21.24 | mathi | WIMPy, this is what we bought: http://www.scarlet.be/fr/packs/internet-tv-telephonie/ |
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| 14:26.02 | WIMPy | mathi: Yes, clearly an IAD. H.323, SIP or MGCP |
| 14:26.14 | mathi | WIMPy, so how does the phone works if there is only incoming DSL from wall? |
| 14:26.43 | WIMPy | So if you can get hold of the credentials for your hpone"line" you can skip the IAD and use them with Asterisk. That would be the best solution. |
| 14:27.02 | WIMPy | There's an ATA in that modem. |
| 14:27.26 | WIMPy | I.e. you don't have a phone line. |
| 14:28.25 | mathi | WIMPy, what does the ATA looks like ? You mean this device ? http://oi49.tinypic.com/17bngl.jpg |
| 14:28.53 | WIMPy | That's just a plug. |
| 14:29.28 | mathi | ahhh you meant inside the modem |
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| 14:30.07 | WIMPy | Yes |
| 14:30.26 | WIMPy | IAD = Modem/router/VOIP gateway combo box. |
| 14:30.58 | mathi | WIMPy, and this ATA inside the modem connects my phone with it ? |
| 14:31.07 | WIMPy | yes |
| 14:31.29 | WIMPy | And it looks like you could connect a 2nd phone. |
| 14:31.36 | mathi | WIMPy, thanks! i need to go now, I hope we can talk later |
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| 15:21.54 | bombev | what does it mean that sip code: Got SIP response 480 "Temporarily not available" |
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| 15:23.42 | wdoekes | depends on the equipment.. I use that whenever I don't have a more sensible error code (like 404 or 486) |
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| 15:24.39 | wdoekes | i.e. the peer should be reachable through this uas/proxy but he isn't at the moment |
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| 16:10.34 | radic | why does the last extension only work if the 1st is commented? |
| 16:10.35 | radic | ;exten => _[015].,1,Set(num=${OUTNUM2}) |
| 16:10.35 | radic | exten => _X.,1,Goto(keinan,1) |
| 16:17.05 | WIMPy | Because you dialed something that started with 0, 1 or 5. |
| 16:17.20 | leifmadsen | +1 |
| 16:18.14 | *** join/#asterisk SuperNull (~FreeManof@24.148.101.238) |
| 16:19.09 | SuperNull | uhg. back for more fun :) |
| 16:19.27 | SuperNull | TK you alive ? |
| 16:22.30 | radic | WIMPy: in that case NUM ist set to OUTNUM* and asterisk goes to the next matching extension. for example 501. but if I dial anything where isn't an extension available It should use the _X. |
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| 16:29.00 | SuperNull | uhg, asterisk realtime needs better documentation. |
| 16:29.02 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 16:29.26 | SuperNull | or perhaps i need to upgraded these age old boxes.. which ever. |
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| 16:35.30 | [TK]D-Fender | SuperNull, Yes I'm alive, no I don't have any experience with realtime (beyond setting up CDR via ODBC) |
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| 16:40.26 | SuperNull | tis what im doing my good buddy. trying to get userfield to fill on ast 1.4.x works flawless everything on 1.6 |
| 16:42.25 | SuperNull | If i were to rebuild all things asterisk in my network which would be ideal version ? 10 ? |
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| 16:46.41 | fredericve | SuperNull: I recommend 11 |
| 16:47.09 | SuperNull | alright, any crazy dialplan changes between 11 and say... 1.6 heh. as far as syntax or anything drastic ? |
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| 17:18.08 | pabelanger | Anybody have any example code showing asterisk originating a call to my phone, then once I answer, it will dialing the 2nd leg of the call? |
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| 17:20.31 | moos3 | anyone running imap_tk on centos 6 ? |
| 17:24.53 | wltjr | are there any sip devices for ringning, like if the phone can't be heard, ring not loud enough |
| 17:26.17 | [TK]D-Fender | pabelanger, "Code"? This is a simple Originate from * CLI, or a call files... or an AMI Originate request (you could trigger from telnet if you cared) |
| 17:26.29 | [TK]D-Fender | pabelanger, No real "code" technically required |
| 17:27.29 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 17:28.40 | pabelanger | [TK]D-Fender: Ya, for some reason I _thought_ originate did not handle it. After looking at the source, I was wrong |
| 17:28.42 | pabelanger | testing it now |
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| 17:32.50 | moos3 | anyone have ideas The IMAP_TK installation appears to be missing or broken ?? I just downloaded on my centos 6 and did make slx EXTRACFLAGS="-fPIC -I/usr/include/openssl" what am I missing |
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| 17:39.15 | mathi | hi |
| 17:39.26 | mathi | WIMPy, you are still here? |
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| 18:44.18 | ghost75 | do you know any free softphone that can send fax over g711 ? |
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| 19:08.56 | moos3 | is there a reason why asterisk is still using usr/lib64/libmyodbc3.so ?? instead of 5 ? |
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| 19:49.52 | pabelanger | Anybody using monitor_filename and MEMBERNAME for recording their queues? |
| 19:50.17 | pabelanger | Or queue member extension? |
| 19:50.52 | pabelanger | I haven't tested it yet, but curious if I need to use any macros when the agent answers to format the filename for mixmonitor |
| 19:50.58 | pabelanger | will be testing tonight |
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| 20:06.24 | moos3 | anyone have a idea on this https://gist.github.com/1b7b6c5d951755858627 |
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| 20:11.46 | ghost75 | when i receive a fax, why is it going again to fax context after it finished? |
| 20:12.37 | ghost75 | http://pastebin.com/wbq14SE8 |
| 20:14.13 | Qwell | Because you told it to. |
| 20:14.53 | navaismo | too late again |
| 20:15.50 | ghost75 | http://pastebin.com/1RzwYJaQ <- i did something here? |
| 20:17.31 | navaismo | ¬¬ |
| 20:17.41 | navaismo | Did you wrote that diapan? |
| 20:17.59 | ghost75 | of course said the horse |
| 20:17.59 | [TK]D-Fender | ghost75, == Using SIP RTP CoS mark 5 -- Executing [08192998774@incoming-arcor:1] NoOp("SIP/arcor_in-00000002", "Fax from unavailable for 08192998774") in new stack |
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| 20:18.27 | [TK]D-Fender | ghost75, Sure looks like another SIP call coming in right after. Of course ... one would think we'd have this log with SIP DEBUG enabled to be really looking at it.... |
| 20:18.59 | ghost75 | mmhh |
| 20:19.01 | moos3 | can anyone help me with a odbc issue ? |
| 20:19.05 | p3nguin | Check out this great idea: http://www.youtube.com/watch?v=lG5cEik2ABY |
| 20:19.06 | ghost75 | the dialplan looks ok? |
| 20:19.34 | ghost75 | need hangup in fax context? |
| 20:19.36 | [TK]D-Fender | ghost75, What we see ... sure... why not... |
| 20:20.38 | asr33 | would like to apologize to [TK]D-Fender for being rude the other day! |
| 20:21.08 | leifmadsen | asr33: don't worry, he's rude all the time |
| 20:21.48 | asr33 | reguardless sorry |
| 20:22.04 | [TK]D-Fender | Direction fail... |
| 20:22.11 | [TK]D-Fender | asr33, No worries... |
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| 20:39.43 | Dovid | hi all |
| 20:40.54 | ghost75 | g711 fax for sure sucks big time |
| 20:41.49 | dijib | why? |
| 20:42.04 | dijib | ghost75: compared to g722? |
| 20:42.10 | ghost75 | to t38 |
| 20:43.35 | jpsharp | G711 fax works as long as the IP link is *solid*. |
| 20:43.51 | jpsharp | But any hiccups and things go to hell in a heartbeat. |
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| 20:50.01 | dijib | jpsharp: had this situation today, i need a business line. |
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| 20:50.32 | ghost75 | and as long there is no echo cancelation and so on |
| 20:50.37 | Dovid | dijib: So go buy one ;0 |
| 20:50.38 | Dovid | ;) |
| 20:51.03 | ghost75 | i dont understand why isp dont support t38 |
| 20:54.35 | jpsharp | ISP or telephony provider? |
| 20:55.21 | jpsharp | dijib: A business line wont help you. You're at the mercy of transit providers and the internet in general. |
| 20:55.35 | jpsharp | Unless by "business line", you mean a dedicated T1 to your ITSP. |
| 20:56.02 | ghost75 | (21:55:16) jpsharp: ISP or telephony provider? <- thats here the same |
| 20:57.23 | jpsharp | Supporting T38 requires extra infrastructure. |
| 20:57.39 | jpsharp | T.38 enabled VOIP->PSTN gateways. |
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| 21:02.29 | J4nus | hi, I have a issue with called number starting with a prefix "+", such as "+32 4778855" |
| 21:02.30 | ghost75 | until they will support t.38 i think fax is extinct |
| 21:03.18 | J4nus | is there a good practice to manage these numbers ? Now I have a rule with exten => _X. |
| 21:03.21 | J4nus | and it's not matched.. |
| 21:03.21 | leifmadsen | flowroute seems to support t.38 fine |
| 21:03.32 | leifmadsen | J4nus: because you need _+X. |
| 21:03.52 | jpsharp | I use Gafachi's T38 all the time. |
| 21:04.09 | jpsharp | It's a bit of a hokey T38 installation, but it works well. |
| 21:05.23 | J4nus | leifmadsen, ok but how can i do to remove the "+" if needed, so i don't need to duplicate the rules |
| 21:05.34 | J4nus | i image i can do it with a Set |
| 21:05.47 | leifmadsen | you have to have multiple lines, then just call a GoSub() for duplicate logic |
| 21:06.15 | leifmadsen | exten => _+X.,1,GoSub(routing,${EXTEN:1}) |
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| 21:07.36 | J4nus | leifmadsen, ok tks i will look about it :) |
| 21:08.02 | mdg | Hello, is it possible to pass multiple options to an AGI EXEC command? I am trying "EXEC Swift 'test test test test' 10 1" but its failing with -2 |
| 21:08.47 | mdg | the Swift app accepts 3 args, the text to speak, timeout, and max digits to accept before jumping to that extension |
| 21:09.17 | leifmadsen | Returns whatever the <application> returns, or '-2' on failure to find < |
| 21:09.17 | leifmadsen | application>. |
| 21:09.23 | leifmadsen | you don't have a Swift application |
| 21:09.49 | mdg | let me double check. .. (ive been tinkering with it so I may have my use case mixed up) |
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| 21:14.07 | mdg | leifmadsen: sorry - when I do "EXEC Swift '<my text>' 10 1" it speaks "<my text> 10 1" |
| 21:14.27 | mdg | how would you pass multiple args to EXEC? |
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| 21:26.06 | Qwell | mdg: use commas |
| 21:27.17 | Qwell | or don't |
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| 21:27.29 | SuperNull | TK in your realtime CDRs did you include any kind of quality info ? |
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| 21:31.15 | [TK]D-Fender | SuperNull, nope |
| 21:31.23 | [TK]D-Fender | BBL |
| 21:41.46 | SuperNull | is there a help bot in here? |
| 21:42.54 | pabelanger | infobot: cookie |
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| 21:50.24 | SuperNull | infobot: help |
| 21:51.58 | tonikasch | !help |
| 21:52.03 | tonikasch | :? |
| 21:52.43 | WIMPy | infobot uses ~ |
| 21:53.45 | paulc | ~ask |
| 21:53.45 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:54.39 | SuperNull | he worked dont worry ;) |
| 21:54.41 | SuperNull | he pmed me. |
| 21:55.08 | paulc | I love it when a plan comes together :) |
| 21:58.39 | SuperNull | anyone use 'VoipMonitor(.org)' ? |
| 21:58.59 | tonikasch | thx WIMPy :) |
| 21:59.20 | tonikasch | and paulc :p |
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| 22:08.29 | wasabi1 | Hi folks. I'm looking for a good asterisk based utility for call center purposes. Downloading FreePBX now to see how it's IVR and queue UIs are... but if there are any other recommendations, I'd be interested. |
| 22:11.14 | wltjr | anyone know of any alternatives to like the algo 8180 SIP Audio Alerter, cyberdata not an option even more expensive, or if anyone can recommend a phone with a really loud ringer :) only need a single line using spa301, but its not loud enough |
| 22:13.26 | _Corey_ | wltjr: Think analog and use an ATA... There are all manor of alarm-bell type thingies out there |
| 22:14.28 | wltjr | _Corey_: ok, came across stuff along those lines, I have a spa112 I likely won't be using so I guess I can do a shared sip account and have an analog phone ring, but then they will want to use the analog phone vs voip one :) |
| 22:14.52 | wltjr | _Corey_: unless you mean just an analog ringer, I might be able to make one of those :) |
| 22:15.42 | jpsharp | Cheap ATA + analog ringer |
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| 22:17.10 | wltjr | cool thanks, was hoping to avoid the analog route, but seems thats the way to go, short of spending lots of $ on a sip one |
| 22:17.59 | _Corey_ | I'm not aware of any benefit of using a SIP product for that other than the lack of ATA, so I think you end up paying much more than it's worth... |
| 22:18.32 | wltjr | _Corey_: just was wanting less devices, but I have an extra ata, so that works ;) |
| 22:19.10 | WIMPy | You could always use an IP socket or an arduino or whatever and control it from the dialplan. |
| 22:19.15 | wltjr | plus the ringer is kinda cool, can hook up to strobe or horn :) |
| 22:19.17 | WIMPy | If youre server is on the LAN at least. |
| 22:19.48 | wltjr | WIMPy: interesting, haven't played with those devices, but seen them used for many things |
| 22:20.20 | WIMPy | You could even just wire somethign up to the servers serial or printer port. |
| 22:21.07 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zujhzdlybekckxhz) |
| 22:21.52 | wltjr | WIMPy: well thats not an option it technically is a lan, but the one location is about 1500ft away, connected via fiber |
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| 22:34.47 | navaismo | wasabil asterisk+queuemetrics |
| 22:42.18 | p3nguin | If you have an ATA and an analog ringer device, simply do parallel dialing to the ATA with the ringer and to whatever phone you want them to talk on. |
| 22:45.41 | tonikasch | bye |
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| 22:59.15 | wltjr | bet i can find an analog strobe or flash light as well :) |
| 22:59.37 | wltjr | ata has 2 ports, or 2 ringer and really make noise |
| 23:00.31 | J4nus | leifmadsen, ok nice it's working perfectly |
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| 23:54.18 | janmate | hi everybody, please is there any way to use SIP templates (from sip.conf) in LDAP? … the goal is to create minimal LDAP peer accounts and set the global shared settings in template (in sip.conf) |
| 23:55.13 | SeRi | p3nguin: pkg delivered. |
| 23:55.34 | jpsharp | Nice pkg! |
| 23:57.01 | sruffell | giggles |
