| 00:01.18 | djgerm | oh maybe it's not unix time in my version of asterisk. I think http://www.voip-info.org/wiki/view/Asterisk+cdr+csv says field 10 is unique! |
| 00:01.36 | djgerm | yes! thanks [TK]D-Fender! |
| 00:04.48 | igcewieling | waves to katty |
| 00:05.41 | Katty | o/ |
| 00:05.46 | Katty | igcewieling: how're you dear? |
| 00:05.48 | [TK]D-Fender | uniqueid = unixtime |
| 00:06.40 | igcewieling | Katty: my cat has cancer so you can imagine |
| 00:06.50 | Katty | :< |
| 00:06.52 | Katty | so sorry |
| 00:06.54 | djgerm | cat cdr-csv/Master.csv | awk -F , '{ print $10}' | sort | uniq -c seems to be giving me something that might represent the CPS |
| 00:07.09 | igcewieling | Katty: we are enjoying our remaining time together. |
| 00:09.08 | djgerm | yikes… hmm I don't think that's an accurate field to be uniq ing |
| 00:10.45 | cusco | why not store cdr in a data manipulation engine ? |
| 00:11.21 | djgerm | i don't know what that means =D |
| 00:12.02 | navaismo | gosh disabling TLS & using vpn seems to work fine... grrrrrr |
| 00:13.47 | igcewieling | navaismo: indeed. 8-) |
| 00:13.53 | Katty | igcewieling: good :> |
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| 00:15.01 | igcewieling | djgerm: Why not grep for Dial? |
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| 00:15.42 | djgerm | igcewieling: from cdr? that's not in my cdr.... |
| 00:15.55 | igcewieling | heh, sorry, I have CEL on my mind. |
| 00:16.06 | igcewieling | thought lasapp should have a Dial |
| 00:16.12 | djgerm | oh it does showup for some calls... |
| 00:16.14 | igcewieling | I'd have to check |
| 00:16.18 | djgerm | not the failed ones |
| 00:17.36 | djgerm | and not the machine detect calls… well anyway, not sure how that'd represent CPS (sorry I am so ignorant) |
| 00:18.39 | navaismo | any hints where to start troubleshooting? |
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| 00:22.55 | igcewieling | navaismo: Let go of TLS and embrace VPN. |
| 00:24.43 | *** join/#asterisk petris (~petris@ryanpetris.com) |
| 00:32.01 | navaismo | so TLS & OpenVPN can't be mixed? |
| 00:35.51 | igcewieling | Why would you want to, both secure SIP, but OpenVPN also secures RTP. |
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| 00:47.39 | navaismo | well im using srtp too |
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| 07:58.16 | izbushka | hi |
| 07:59.14 | ChannelZ | ahoy |
| 07:59.22 | izbushka | is there any applocation like MessageSend to send a text message to remote asterisk connected by iax2? MessageSend works only with sip.. |
| 08:00.50 | ChannelZ | not that I know of. IAX wasn't really designed for such frivolties :) |
| 08:02.43 | izbushka | well i'd like to send sms from sip clients to mobile by gsm modules on remote astersik. is there any way to do so? |
| 08:04.09 | ChannelZ | Is there a reason you can't use SIP, even for just that purpose? Or do XMPP. |
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| 08:04.57 | izbushka | I can use sip, but i'm using iax already. so i'm just looking for easiest way :) |
| 08:05.13 | izbushka | thanks anyway |
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| 08:19.37 | BeeBuu | how can i know which number answered by dial(sip/2222&sip/111&sip/333) command? |
| 08:19.50 | BeeBuu | anyone help please.. |
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| 08:35.54 | bulkorok | BeeBuu: check your CDR records |
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| 08:39.17 | BeeBuu | bulkorok:can i know that in AMI? |
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| 08:39.49 | bulkorok | don't know... check the messages... |
| 08:40.03 | skorzen | says hello. |
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| 08:42.43 | skorzen | Anyone here using Nagios to monitor Asterisk? |
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| 08:51.52 | duchman | hi all,calls between 2 internal exts hang up as soon as the receiver picks up.......any help pls |
| 08:51.54 | duchman | http://pastebin.ca/2429008 |
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| 10:38.07 | duchman | hi all,calls between 2 internal exts hangup as soons as the receiver picks the call.....http://pastebin.ca/2429037 |
| 10:38.11 | duchman | any help pls |
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| 11:51.25 | izbushka | how does message auth (auth_message_requests=yes) works? I've got "Failed to authenticate MESSAGE with host.." while sending messages from one asterisk to another using sip. but both are registered on each other as sip friends |
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| 11:52.33 | Rico | hi all |
| 11:52.41 | Rico | is asterisk able to send snmp traps on events ? |
| 11:53.00 | Rico | like on sip trunk registration failure ? |
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| 11:53.39 | phix | duchman: firewall? |
| 11:53.56 | phix | RTP set on the same port or port range as another service? |
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| 12:04.46 | qakhan | <PROTECTED> |
| 12:05.02 | qakhan | i am getting this message when i use Array function |
| 12:13.50 | izbushka | qakhan, Function names by practice are all capitalzed letters. The names ARE CASE SENSITIVE! |
| 12:16.26 | qakhan | Thanks izbushka |
| 12:18.16 | qakhan | izbushka can you tell me how can i get each record in each veriable from DB |
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| 12:38.02 | Chainsaw | Is DAHDI 2.7.0 out or not? The topic here doesn't mention it, and I can't find a ChangeLog for it either. |
| 12:38.17 | *** part/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 12:39.46 | banane_ | could somebody please explain to me why this happens: i jump to a sub via Dial M and set a global var there, which is used in the sub too and works fine there. after continuing the dialplan via the g option i try to access the aforementioned variable but it only returns an empty string |
| 12:40.34 | Greenlight | And it's defined in [globals] ? |
| 12:40.55 | banane_ | oh, does it have to be? :) i just set it with the g parameter |
| 12:41.15 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Global+Variables+Basics |
| 12:41.18 | banane_ | my fault then, thanks for the answer |
| 12:41.37 | banane_ | stupid me |
| 12:41.49 | Greenlight | The sub in Dial M runs under the other channel, iirc, so that's why it works okay in that context |
| 12:43.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 12:48.08 | banane_ | works fine now, interestingly i couldn´t set it via set(var=val,g) but only via set(GLOBAL(var)=val), setting it the other way still returned an empty string outside of the sub |
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| 12:49.43 | Greenlight | Perhaps the ,g is deprecated |
| 12:50.27 | banane_ | i just see that it only appended the ",g" to the saved string for the sub context... somethin wrong there :) |
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| 12:57.49 | leifmadsen | ya, appending stuff like that with Set() is definitely not the right way |
| 12:58.00 | leifmadsen | if it ever was, it was like in version 1.2 and earlier |
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| 13:02.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:20.23 | jmetro | mushroom mushroom. |
| 13:24.12 | qakhan | all i am using ODBC to get records from MSSQL |
| 13:24.14 | qakhan | http://pastebin.com/bruCQWFW |
| 13:24.36 | qakhan | i need to get each record in different veriable |
| 13:24.51 | qakhan | please see my pastbin |
| 13:25.58 | leifmadsen | qakhan: every record? |
| 13:26.07 | leifmadsen | like, every row? |
| 13:26.11 | qakhan | yes |
| 13:26.20 | leifmadsen | you're pretty much doing it right-ish |
| 13:26.39 | leifmadsen | gotta loop through the values and create unique values |
| 13:26.45 | qakhan | like if there is 3 records i need to get them in 3 different veriable |
| 13:26.47 | leifmadsen | s/unique values/unique variables/ |
| 13:26.58 | leifmadsen | qakhan: ok, so what you have looks like it does that |
| 13:27.02 | leifmadsen | so what is the issue? |
| 13:27.32 | qakhan | let me send you output |
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| 13:34.18 | qakhan | @leifmadsen here is my cli output http://pastebin.com/u3Lf4ww2 |
| 13:34.51 | leifmadsen | qakhan: ya you need to use the ARRAY() function if you're going to return multiple columns too |
| 13:35.13 | leifmadsen | Set(ARRAY(col_name_${COUNTER},col_name2_${COUNTER})=....) |
| 13:37.42 | qakhan | in my cli record are coming from diff columns |
| 13:38.03 | qakhan | name, address, city, state, zipcode |
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| 13:46.54 | leifmadsen | yes they are |
| 13:46.59 | leifmadsen | that was my point |
| 13:47.07 | qakhan | leifmadsen how can i use ARRAY() for multiple rows |
| 13:47.09 | leifmadsen | you need to use ARRAY to set separate variables for each column |
| 13:47.12 | qakhan | in while loop |
| 13:47.17 | leifmadsen | in the same way you're using it now |
| 13:47.37 | leifmadsen | Set(ARRAY(name_${COUNTER},address_${COUNTER})=...) |
| 13:47.41 | leifmadsen | like my example above |
| 13:48.05 | leifmadsen | it's the same thing, you're just setting multiple columns per row |
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| 13:56.19 | igcewieling | leifmadsen: qakhan has been working on this EXACT issue for over a day. [TK]D-Fender spent a long time with him. He appears to have not made any progress. |
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| 13:58.48 | [TK]D-Fender | 0 programming background apparent, doesn't even look at the dialplan processing to see what variables are being set to. |
| 13:59.50 | leifmadsen | igcewieling: that's sadness |
| 13:59.53 | Weezey | oh Samba, you miserable bitch. |
| 13:59.59 | leifmadsen | I have nothing else to offer, as I've provided the answer |
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| 14:03.49 | [TK]D-Fender | leifmadsen: even that is not enough... you'll have to specefically formulate the teaching int code he can drop in as-is... |
| 14:03.50 | igcewieling | I do prefer for format/syntax of Set(ARRAY(name[${COUNTER}],address[${COUNTER}])=...) but that is mainly cosmetic |
| 14:04.37 | [TK]D-Fender | leifmadsen: he eith cannot and/or des not want to learn. He needs to hire someone else to run his system for him. |
| 14:04.47 | [TK]D-Fender | either* |
| 14:06.00 | [TK]D-Fender | he could us alpha-numeric var names just fine afaict |
| 14:06.10 | igcewieling | when you've annoyed both igcewieling AND [TK]D-Fender ...... |
| 14:06.19 | jmetro | and left me without tacos |
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| 14:06.28 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:06.48 | [TK]D-Fender | CRY FOR THE TACOS! |
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| 14:30.23 | WIMPy | Is directmedia something that doesn't work per peer? |
| 14:30.44 | Greenlight | No, you can set it per peer |
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| 14:31.50 | WIMPy | It went wrong between a peer with =nonat (global) and one witn =no. |
| 14:33.34 | Greenlight | As in, it tried to do directmedia between the two ? |
| 14:34.11 | WIMPy | It must. I had audio for about half a second. |
| 14:34.19 | Greenlight | To be honest, in my experience, setting directmedia=no saves a world of pain... there just seems to be too many "issues" with it |
| 14:34.21 | WIMPy | After transferring the call it was ok again. |
| 14:34.43 | WIMPy | Looks like you have to. |
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| 15:04.08 | duchman | hi all,calls between 2 internal exts hangup as soon as receiver picks up .......http://pastebin.ca/2429106 is the sip/rtp debug file |
| 15:04.14 | duchman | any help pls? |
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| 15:05.46 | Greenlight | Lets see the dialplan and sip.conf |
| 15:06.23 | duchman | ok pls hold |
| 15:06.45 | Greenlight | Oh hmm this is a freepbx beta? |
| 15:08.19 | duchman | yea |
| 15:09.39 | duchman | http://pastebin.ca/2429115 this is sip.conf |
| 15:10.25 | Greenlight | ~freepbx |
| 15:10.25 | 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 |
| 15:10.25 | igcewieling | duchman: that issue is almost always a NAT or directmedia problem. |
| 15:10.33 | igcewieling | oh, freePBX. |
| 15:11.05 | duchman | http://pastebin.ca/2429118 is the ext.conf |
| 15:11.11 | igcewieling | duchman: how many g729 licenses do you have? |
| 15:11.15 | Greenlight | I'm thinking NAT or network issue, but freepbx configs try and hide all that |
| 15:11.21 | igcewieling | duchman: perhaps you missed the part about not being supported here. |
| 15:11.23 | duchman | using ulaw/alaw |
| 15:11.33 | igcewieling | duchman: then why do you have g729 enabled? |
| 15:11.33 | duchman | ok |
| 15:12.08 | duchman | <igcewieling> in sip.conf? |
| 15:12.14 | igcewieling | duchman: correct. |
| 15:12.23 | Greenlight | Can you try and reproduce in vanilla asterisk? Or even try using a non-beta version? or ask in #freepbx as am sure they'll know where to check for settings in the GUI |
| 15:12.46 | duchman | ok thanks |
| 15:12.57 | igcewieling | I wish FreePBX users were flagged somehow, then I could automatically put them on /ignore |
| 15:13.19 | Greenlight | lol, somewhat harsh igcewieling :) |
| 15:13.27 | igcewieling | Greenlight: not at all. |
| 15:14.13 | igcewieling | If I went to the power company office and started asking about my water service, they would ignore me or toss me out of the building. |
| 15:15.54 | Greenlight | duchman: Looking at that SIP trace, it does seem as if there's a 2nd invite being sent, so I suspect directmedia is enabled |
| 15:16.08 | Greenlight | How to remedy that though, is away in FreePBX land |
| 15:16.37 | Greenlight | Try the water company accross the road... (#freepbx) :) |
| 15:16.44 | duchman | <Greenlight>ok thanks.....checking the freepbx channel |
| 15:16.48 | igcewieling | mutters something which sounds like "feeding a stray cat" |
| 15:18.13 | [TK]D-Fender | not a freepbx issue |
| 15:18.29 | [TK]D-Fender | and the codecs offered are all basic |
| 15:18.38 | [TK]D-Fender | ulaw, alaw, gsm |
| 15:19.07 | [TK]D-Fender | His client is saying BYE instantly after the OK for the call |
| 15:19.20 | Greenlight | I see another INVITE *before* that BYE |
| 15:19.29 | Greenlight | Why I presumed directmedia attempt |
| 15:19.38 | Greenlight | Unless I'm misreading |
| 15:20.04 | duchman | <[TK]D-Fender>ok...do I set directmedia=no in sip.conf? |
| 15:20.20 | Greenlight | duchman: No, you use the weird and wonderful GUI |
| 15:20.40 | [TK]D-Fender | duchman: I see something suspicious |
| 15:20.53 | duchman | <[TK]D-Fender>? |
| 15:21.12 | [TK]D-Fender | duchman: ... the CALLER'S IPis the same as the one you DIALING |
| 15:21.22 | [TK]D-Fender | duchman: WTF is that? |
| 15:21.41 | Greenlight | He's using two softphons on one PC ? |
| 15:21.59 | duchman | <[TK]D-Fender>both test softphone sit on the same pc |
| 15:22.02 | [TK]D-Fender | sredgstgsdrtydfty |
| 15:22.08 | [TK]D-Fender | DUMB SHIT TEST |
| 15:22.25 | jmetro | =) |
| 15:23.18 | duchman | <[TK]D-Fender> worked on a couple of our other pc so it is not something new |
| 15:23.56 | [TK]D-Fender | and those apps fight over the PORT TOO |
| 15:24.06 | [TK]D-Fender | Stop doing retarded tests |
| 15:24.45 | igcewieling | never ever test from the server you are testing to. |
| 15:25.15 | duchman | <[TK]D-Fender>i don't think it is an IP/Ports but thanks for the suggestion |
| 15:25.48 | [TK]D-Fender | duchman: ... 3cs is the CALLER and the RECEIVER |
| 15:25.52 | [TK]D-Fender | 3cx |
| 15:26.09 | Greenlight | lol it's two *accounts* on the *same* softphone ?! |
| 15:26.26 | duchman | <[TK]D-Fender>3cx phones handle multiple sip accounts |
| 15:26.34 | Greenlight | facepalms |
| 15:26.48 | [TK]D-Fender | You think it will acfcept talking to itself? |
| 15:27.06 | Greenlight | You just forgot to mention this little fact previously?! |
| 15:27.18 | duchman | have done it multiple times in test environments and it works |
| 15:27.36 | Greenlight | Ok igcewieling, maybe you were right about feeding the cats |
| 15:28.15 | igcewieling | This is getting too weird for me. |
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| 15:28.29 | [TK]D-Fender | duchman: This is a dumb test, prves nothing since you can have both ACTIVE at the same time and has every reason to reject instantly as we see it doing |
| 15:28.44 | [TK]D-Fender | can't |
| 15:28.57 | [TK]D-Fender | It hangs up, as it should |
| 15:29.15 | Greenlight | I suspect that *without* directmedia it would work, since it's unaware whats on the other end of each call |
| 15:29.18 | [TK]D-Fender | this is a waste of time |
| 15:29.19 | duchman | <[TK]D-Fender> I think u are wrong on that but thanks anyways |
| 15:29.33 | [TK]D-Fender | evidence begs to differ |
| 15:29.46 | [TK]D-Fender | both clients did this to you |
| 15:29.57 | [TK]D-Fender | that is some serious denial |
| 15:30.07 | [TK]D-Fender | but go on wasting your time |
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| 16:03.11 | *** join/#asterisk Zopieux (zopieux@jerrycraft.tk) |
| 16:03.13 | Zopieux | hello |
| 16:03.20 | rdancer | what is the european equivalent of LRN dipping? I understand that there are service providers that provide LRN dipping in the US, but in the United Kingdom, I have been unable to find any -- but we have local number portability, so there must be a way to find which carrier currently services a given nmuber? |
| 16:03.28 | Zopieux | i've got a working queue with agents connected |
| 16:04.42 | Zopieux | thing is, when the only available agent suspends its line, asterisk continues to "ring" him and the callers are able to speak (but obviously the agent cannot hear the caller) |
| 16:05.20 | igcewieling | Zopieux: "suspend line"?? |
| 16:05.23 | Zopieux | is it a misconception of my softphone software or something to configure in Asterisk? I would like agents to be able to be "busy" even if they're not speaking |
| 16:05.57 | igcewieling | Zopieux: is your softphone sending back a busy? |
| 16:06.18 | wdoekes | rdancer: coin.nl maintains the lists in the Netherlands, but that's obviously not Europe-wide |
| 16:06.27 | Zopieux | igcewieling, when the agents is speaking and then goes to busy, asterisk reacts correctly by playing MOH to the caller |
| 16:06.30 | igcewieling | At my company we only suspend lines if the customer doesn't pay us. |
| 16:06.37 | [TK]D-Fender | What is this "suspend" you're talking about? |
| 16:06.50 | [TK]D-Fender | I think you are using a wrong term... |
| 16:06.55 | Zopieux | i mean "becomes biusy" |
| 16:06.58 | igcewieling | Zopieux: I have no idea what you are talking about. When an agent is talking, they don't just "go to busy". |
| 16:06.59 | Zopieux | busy* dammit |
| 16:07.03 | rdancer | wdoekes: thank you |
| 16:07.20 | [TK]D-Fender | What do you mean "becomes"? |
| 16:07.23 | igcewieling | Zopieux: exactly how is the phone "becoming busy" |
| 16:07.32 | igcewieling | a button press, a *code, voodo? |
| 16:07.37 | [TK]D-Fender | MAGIC |
| 16:07.38 | Zopieux | well, put on hold! |
| 16:07.48 | igcewieling | Zopieux: busy is not the same as hold |
| 16:07.49 | Zopieux | is this the right term? "on hold"? |
| 16:08.11 | igcewieling | Zopieux: be more careful with your terms or people still stop trying to help you. |
| 16:08.12 | Zopieux | let's say busy = the agent is already answering a call |
| 16:08.13 | [TK]D-Fender | Zopieux: you mean they are on a CALL and press the hold button on the phone? |
| 16:08.38 | Zopieux | yes, they active the "on hold" switch |
| 16:08.45 | igcewieling | Zopieux: Based on your addled and rather incoherent description it sounds to me like the phone has call waiting enabled. |
| 16:09.00 | rdancer | wdoekes: are there any resellers of that service, in Netherlands, that you would know of? |
| 16:09.07 | [TK]D-Fender | and is this 2nd call a queue call? |
| 16:09.20 | Zopieux | asterisk then plays MoH to the caller, so the feature works asterisk-side |
| 16:09.20 | wdoekes | rdancer: none that I'm aware of |
| 16:10.25 | Zopieux | but, when the agent is idle, and actives the "on hold" feature (maybe it's stupid, you tell me), asterisk keeps sending him new calls. |
| 16:10.54 | igcewieling | Zopieux: not really our problem. your user is doing the wrong thing. |
| 16:10.57 | Zopieux | i would like to be able, as an agent, to say "i'm online but not ready to take calls" |
| 16:11.02 | [TK]D-Fender | ... if he is IDLE ...what is there to HOLD? |
| 16:11.20 | igcewieling | Zopieux: perhaps "do not disturb" setting? |
| 16:11.35 | rdancer | wdoekes: that's been actually quite helpful, thank you |
| 16:11.37 | [TK]D-Fender | What is THIS "hold" you are talking about now? they aren't even on a call... |
| 16:11.47 | Zopieux | I'm not sure it's in the SIP protocol, that's why I'm asking |
| 16:12.05 | Zopieux | I'm gonna try the "do not disturb" setting, but to me it's more an IM thing |
| 16:12.10 | igcewieling | Zopieux: we are not asking about sip protocol, we are asking what buttons are pressed |
| 16:12.23 | [TK]D-Fender | What hold are you even talking about now?? |
| 16:12.35 | [TK]D-Fender | they aren't on a casll you said |
| 16:12.57 | [TK]D-Fender | call |
| 16:13.12 | Zopieux | yes, they are not |
| 16:13.23 | Zopieux | so they're ready to take incoming calls |
| 16:13.29 | [TK]D-Fender | so what is this HOLD you are talking about? |
| 16:13.34 | igcewieling | Zopieux: if you are not a call, then you cannot place a call on hold. |
| 16:13.52 | [TK]D-Fender | Thayou are misusing word very badly |
| 16:14.06 | igcewieling | Do not Disturb is the correct way. Do it and stop wasting your time. |
| 16:14.19 | Zopieux | I know, I known, but I'm trying to figure out if there is an asterisk feature to prevent online, idling agents, to take calls temporarly |
| 16:14.23 | [TK]D-Fender | What actulal ACTION are you doing? |
| 16:14.33 | Zopieux | one sec. please |
| 16:14.38 | igcewieling | Zopieux: queues are well documented. |
| 16:14.46 | igcewieling | such as QueuePauseMember or similar. |
| 16:15.03 | [TK]D-Fender | pretty sure thats the goal... |
| 16:15.40 | igcewieling | Zopieux: you should go read the Asterisk book so you understand Asterisk and the correct terms. |
| 16:15.44 | igcewieling | ~book |
| 16:15.44 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:16.09 | [TK]D-Fender | then again... anyone who can't describe actual actions being taken and the expectattion attached... are pretty damn lost |
| 16:16.14 | igcewieling | Oh, look, a whole section on Queues! |
| 16:16.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 16:16.45 | Zopieux | i've read the book dozens of times |
| 16:16.48 | [TK]D-Fender | I 'd want to know what he's DOING. |
| 16:16.51 | [TK]D-Fender | ^^ |
| 16:17.14 | [TK]D-Fender | so what is this actual ACTION you are doing on your phone? |
| 16:17.20 | igcewieling | Zopieux: then why are you not using queuepausemember? |
| 16:17.38 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:18.35 | [TK]D-Fender | If you want to temporarily NOT get queue calls, the is a dialplan app to call for that as igcewieling said |
| 16:18.43 | [TK]D-Fender | there* |
| 16:18.48 | [TK]D-Fender | USE IT |
| 16:18.55 | carrar | Zopieux, there is a feature to get your agents working. It's call management |
| 16:19.00 | carrar | called |
| 16:19.06 | [TK]D-Fender | "core show applications like queue" |
| 16:19.08 | [TK]D-Fender | ^^ |
| 16:19.11 | igcewieling | I don't even use queues and I managed to find it in 30 seconds |
| 16:19.12 | *** join/#asterisk navaismo (~navaismo@189.241.46.165) |
| 16:19.20 | carrar | WORK or get FIRED |
| 16:19.22 | carrar | heh |
| 16:19.54 | Zopieux | igcewieling, seems I missed it, I thought I would be able to control their availabilty status through the softphone interface, but I was mistaken |
| 16:20.12 | igcewieling | Zopieux: you really can't do much on a softphone |
| 16:20.37 | Zopieux | I can see now! |
| 16:21.04 | zamba | my last mile is using ip.. how does fax work through this? what are my options? |
| 16:21.25 | zamba | with "last mile" i mean internally in the office space.. |
| 16:21.29 | Zopieux | but do you find it *that* stupid that the "on hold" button may be used as a "not available" hint when the agent is waiting for calls? |
| 16:23.37 | igcewieling | zamba: for the most part fax doesn't work over IP. T.38 is a sad attempt to make it work. It often doesn't. |
| 16:23.53 | zamba | igcewieling: so there's really no options here? |
| 16:23.56 | igcewieling | Zopieux: stupid at a grand and unimaginable scale. |
| 16:24.07 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:24.27 | igcewieling | zamba: install an analog POTS line or try T,38 and hope all your devices and carriers support it. |
| 16:24.38 | Zopieux | it's a "pause" feature, it's just the context that is different, I don't actually see the problem |
| 16:25.04 | igcewieling | Zopieux: I think you'll find NO messages are sent from your sofrphone to asterisk when you press that button |
| 16:25.19 | igcewieling | therefore how could Asterisk know the button was pressed? |
| 16:25.20 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:25.39 | Zopieux | that where I'm confused igcewieling: |
| 16:26.05 | Zopieux | when this button is pressed during a call, asterisk plays MoH to the caller |
| 16:26.19 | igcewieling | Zopieux: correct. Pressing the hold button during a call is a valid thing to do. |
| 16:26.39 | igcewieling | Much like pressing the brakes when your car is running. |
| 16:26.47 | *** part/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 16:26.47 | Zopieux | but agents are on a continuous call with asterisk server once they're logged in! |
| 16:27.10 | Zopieux | that is why it makes sense to me |
| 16:27.11 | igcewieling | Pressing the hold button when there is no active call makes no sense -- just like pressing the brake when your car is shut off. |
| 16:27.21 | zamba | igcewieling: that's about it? |
| 16:27.29 | Zopieux | no, I think you don't get why I'm confused |
| 16:27.43 | igcewieling | zamba: I guess you could invent a totally new fax protocol |
| 16:27.59 | igcewieling | Zopieux: those are not really calls |
| 16:28.01 | WIMPy | igcewieling: I think it's called e-mail. |
| 16:28.04 | Zopieux | I've got a dialplan to log in agents, eg. they call 25@my-sip-asterisk.com |
| 16:28.15 | Zopieux | oh really? |
| 16:29.14 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:29.52 | Zopieux | when idling, there *is* a opened channel between the agent and asterisk though, so why couldn't the "hold" button be used for that "pausequeuemember" purpose? |
| 16:29.55 | igcewieling | Zopieux: you understand there are multiple ways to set up queues, right? We are assuming the agent logs in then waits for a call to ring on the phone. You can also set it up so the agent logs into the queue and STAYS ON THE LINE to have calls patched into their existing call with the queue. I'm pretty sure everyone here is assuming the first method, as the second method is fraught with issues |
| 16:30.15 | Zopieux | oh |
| 16:30.25 | Zopieux | I wasn't aware of the first option, sorry |
| 16:30.43 | igcewieling | The second option did not occur to me as it is so uncommmon |
| 16:30.57 | [TK]D-Fender | That old wait for beep one is gone... |
| 16:31.08 | [TK]D-Fender | which was AgentLogin |
| 16:31.20 | igcewieling | Seems to me that the solution to your problem is for the agent to HANG UP THEIR PHONE. |
| 16:31.34 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:31.40 | Zopieux | (I'm not running a real callcenter with physical phones in a physical building; instead, I've got people all around the world that connects to asterisk through SIP and waits for calls) |
| 16:31.50 | [TK]D-Fender | Zopieux: And you need to make a dialplan EXTENSION to call PauseQueueMember to pause them |
| 16:32.13 | [TK]D-Fender | there is no magical button on your phone for this |
| 16:32.19 | igcewieling | 47,000 items in the NOC outlook folder. |
| 16:32.25 | Zopieux | igcewieling, what I dislike in this "solution" is the fact they've to re-authenticate each time |
| 16:32.45 | [TK]D-Fender | Zopieux: how you auth them is up to you |
| 16:32.52 | igcewieling | Zopieux: correct. This is a management issue, not a technical one. |
| 16:32.56 | [TK]D-Fender | Zopieux: it's your dialplan |
| 16:33.08 | Zopieux | ok [TK]D-Fender, now I've got the right thing in asteriskdocs |
| 16:33.45 | igcewieling | You'll find most issues with queues are management issues. Things like agents all logging out of the queue with calls waiting. Agents walking away from their desk without disconnecting from the queue. Agents playing games with the queue to increase their call stats. |
| 16:33.47 | Zopieux | if I use the "calls patched into existing call" feature, I don't see other solution that the dialplan extension, is there? |
| 16:34.25 | Zopieux | than*, sorry for my approximate English |
| 16:34.47 | [TK]D-Fender | zop forget that |
| 16:34.59 | igcewieling | Zopieux: no idea. no sane person uses it that way |
| 16:35.38 | Zopieux | do I have |
| 16:36.28 | Zopieux | do I *have* to use this insane trick for my usecase, which is: agents on the Internet connecting to Asterisk through softphones? |
| 16:36.50 | [TK]D-Fender | Zopieux: insane? |
| 16:36.58 | Zopieux | igcewieling says it's insane. |
| 16:37.22 | [TK]D-Fender | Zopieux: you want to STOP getting queue calls? then PAUSE or REMOVE the member |
| 16:37.46 | igcewieling | [TK]D-Fender: how is that done when the member is on the active call waiting for calls to arrive? |
| 16:38.20 | Zopieux | removing the member is not a solution, I don't want agents having to auth every 10 minutes |
| 16:38.39 | [TK]D-Fender | do we have confirmation that he's using AgentLogin .... and no using horrible terminology again? |
| 16:38.45 | [TK]D-Fender | SHOW US |
| 16:39.09 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:39.12 | [TK]D-Fender | terms being flungf around are vague and useless... |
| 16:39.27 | [TK]D-Fender | "Hold" is an adverb for EVERYTHING |
| 16:39.35 | [TK]D-Fender | Trust level = 0 |
| 16:40.07 | Zopieux | "please ignore me when a new call arrives" |
| 16:43.41 | [TK]D-Fender | Zopieux: then PAUSE THRM |
| 16:45.02 | Zopieux | I think the real issue is the way I auth my agents; I should use the SIP credentials instead of a pin |
| 16:45.10 | [TK]D-Fender | no... |
| 16:45.37 | Zopieux | if I understand correctly, there are two ways: using the SIP authentication or using a digit code |
| 16:45.52 | [TK]D-Fender | you want to pause them from getting calls ... the problem is you aren' calling the pause application to do it |
| 16:46.06 | [TK]D-Fender | there is no "auth" |
| 16:46.24 | Zopieux | I know, there are two different things here |
| 16:46.30 | [TK]D-Fender | PauseQueueMember has such thing as auth. |
| 16:46.40 | Zopieux | I understand I have to manually change something in the dialplan tu call PauseQueueMember |
| 16:47.04 | [TK]D-Fender | MAKE AN EXTENSION so thr can choose to pause themselves |
| 16:47.43 | Zopieux | but I'm know wondering if I could ease the authentification process because at the moment, my agents register through a single SIP account/password and then auth. interactively with the dialpad |
| 16:48.10 | [TK]D-Fender | REGISTERING HAS NOTHING TO DO WITH QUEUE STATUS AND MEMBERSHIP |
| 16:48.19 | Zopieux | it's all right [TK]D-Fender I got that! |
| 16:48.21 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:48.22 | [TK]D-Fender | ]YOU ARE MIXING THINGS UP AGAIN |
| 16:50.16 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:11.07 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:31.22 | *** join/#asterisk kresp0 (~kresp0@182.Red-79-144-162.dynamicIP.rima-tde.net) |
| 17:38.18 | *** join/#asterisk The_Phil (~Phizzel@41-135-102-211.dsl.mweb.co.za) |
| 17:39.14 | The_Phil | Hi Guys, please can anyone tell me what this message means? As far as I can understand it means there isn't sufficient codec licences for g729 but this box has 30 licenses? |
| 17:39.32 | *** join/#asterisk felipealmeida (~user@177.98.67.233) |
| 17:39.41 | The_Phil | WARNING[1687]: codec_g729.c:390 g729tolin_new: g729tolin_new pool->g729_pvt_dec |
| 17:40.28 | igcewieling | The_Phil: "g729 show license" or similar |
| 17:40.54 | igcewieling | The_Phil: lack of codecs should generate a cannot find translation path sort of message |
| 17:41.01 | igcewieling | s/codecs/licenses |
| 17:42.42 | The_Phil | mhm |
| 17:43.05 | The_Phil | so where in asterisk can I confirm the amount of G729 licenses? |
| 17:43.41 | igcewieling | The_Phil: I just told you |
| 17:43.51 | The_Phil | there is no s directory though? |
| 17:44.02 | igcewieling | "s" directory? |
| 17:44.41 | igcewieling | doing it i the CLI verifies the number of licenses, the fact the licenses are active, and the fact the codec is loaded. What is the output of that command? |
| 17:44.48 | The_Phil | ahhh |
| 17:45.43 | The_Phil | Just says the command doesn't exist |
| 17:47.45 | igcewieling | try g729 <tabkey> like all other commands in the CLI |
| 17:48.07 | The_Phil | there is no g729 command |
| 17:48.13 | igcewieling | then you do not have the codec loaded. |
| 17:48.22 | igcewieling | "module show like g729" |
| 17:48.37 | The_Phil | module might work |
| 17:48.55 | igcewieling | Actually, either you don't have the codec loaded or you are running an unlicensed non-digium codec. |
| 17:49.00 | ChannelZ | if you updated * recently make sure you downloaded a new module if applicable |
| 17:49.13 | igcewieling | if you are running a non-digium codec then you will likely be banned from this channel. |
| 17:50.41 | The_Phil | Module Description Use Count |
| 17:50.41 | The_Phil | format_g729.so Raw G729 data 0 |
| 17:50.42 | The_Phil | codec_g729.so G.729 Coder/Decoder 22 |
| 17:50.42 | The_Phil | 2 modules loaded |
| 17:51.00 | The_Phil | I know it's a digium license key |
| 17:51.05 | jmetro | That's a great example of something that should be pastebinned. |
| 17:51.50 | igcewieling | I have no idea what the issue is then. codec_g729 provides the g729 CLI command. |
| 17:52.01 | The_Phil | it's not a purely asterisk box though |
| 17:52.15 | ChannelZ | which means what exactly |
| 17:52.29 | *** part/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 17:52.29 | igcewieling | ChannelZ: Asterisk running on a pumpkin powered by mice! |
| 17:52.40 | ChannelZ | guess it's a secret |
| 17:52.45 | igcewieling | indeed. |
| 18:03.23 | jmetro | igcewieling: i run asterisk on a citrix server and my users launch it from a web ui |
| 18:03.47 | igcewieling | jmetro: pervert |
| 18:04.56 | jmetro | :3 |
| 18:06.33 | igcewieling | 8-) |
| 18:17.37 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:17.38 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:22.19 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 18:22.45 | *** join/#asterisk Jinxed- (c663818a@gateway/web/freenode/ip.198.99.129.138) |
| 18:23.35 | asilva | A little help here regarding macros, i create a priority label called sec: , when executing the macro the sec label is executed even without a goto under the macro. is something changed in asterisk 11 regarding priority labels inside macros ? |
| 18:24.05 | asilva | meaning the label is always execute as part of the ~~s~~ context dialplan inside macro. |
| 18:24.06 | leifmadsen | asilva: no, because the macro stuff isn't supported any more, so nothing will change |
| 18:24.19 | leifmadsen | asilva: please show an example, I suspect a typo or misuse perhaps |
| 18:24.25 | asilva | hold on |
| 18:24.30 | leifmadsen | I'd be surprised if there were a bug in asterisk there |
| 18:25.21 | asilva | http://pastebin.com/H6HYyHT2 |
| 18:25.25 | igcewieling | asilva: using gosub is the correct correct way. |
| 18:25.37 | leifmadsen | oh AEL |
| 18:25.40 | leifmadsen | stops looking |
| 18:25.43 | asilva | i use gosub, i'm not meaning the MACRO APP |
| 18:25.45 | igcewieling | LOL! |
| 18:25.57 | igcewieling | asilva: you forgot to mention AEL. |
| 18:26.00 | asilva | i use & to refer a macro something.. |
| 18:26.01 | asilva | eheheh |
| 18:26.02 | asilva | yup |
| 18:26.08 | asilva | sorry!! |
| 18:26.33 | asilva | regarding the example if arg = 1 or 0 shouldn't execute sec: label |
| 18:27.01 | asilva | but that doens't happen, sec: label gets executed every time.. am I doing something wrong ? or something is changed ? or is it a bug ? |
| 18:27.21 | igcewieling | asilva: value comparisons are somewhat complicated. |
| 18:27.47 | igcewieling | first add a Noop(ARG is '${ARG}'); to verify ARG is what you expect. |
| 18:28.15 | asilva | that was just an example |
| 18:28.22 | asilva | let me try again |
| 18:28.26 | asilva | forget about that IF |
| 18:28.36 | asilva | sec: will still be executed without a goto |
| 18:28.48 | igcewieling | If I forget about the IF then you don't have a question |
| 18:29.26 | asilva | is that the way is it supposed to be ? like this - http://pastebin.com/vyFNhc4W |
| 18:29.39 | igcewieling | try this http://pastebin.com/ECra7JnY |
| 18:32.28 | asilva | that will work, but ain't that wrong ? i just coppied from 1.8 to 11.5 and testing stuff. and I came up to that! |
| 18:32.57 | asilva | under 1.8 the label ain't executed unless i call a goto to it. |
| 18:33.09 | asilva | even if its before return; |
| 18:33.15 | igcewieling | why is it wrong? |
| 18:33.20 | blehxor | looking into switching my dev environment to Mac, will I be able to run asterisk (dev environment) fine? |
| 18:33.44 | igcewieling | asilva: do a dialplan show to see how your AEL is translated into extensions.conf |
| 18:33.44 | jmetro | if you have an exten=>(label)1,stuff and exten=>(label2)2,stuff both lines will be executed. |
| 18:33.48 | jmetro | without goto's |
| 18:33.58 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:34.06 | jmetro | blehxor: no, macs are awful. |
| 18:34.17 | igcewieling | jmetro: he thinks labels are not executed unless gone to. Totally wrong, of course. |
| 18:34.23 | asilva | jmetro: yeah, but in 1.8 inside the macro that never happened |
| 18:34.35 | igcewieling | asilva: prove it. |
| 18:34.58 | asilva | ahahah let me see here in another machine hold on |
| 18:35.09 | jmetro | Labels are like checkpoints in Mario, you can always go back to them, but everything beforehand and the checkpoint itself was still part of the level. |
| 18:35.12 | blehxor | jmetro: awful for asterisk? or you don't like them in general |
| 18:35.38 | jmetro | blehxor: awful as development environments, and awful in general in terms of upkeep and productivity. Even our Apple fanboy uses windows at work. |
| 18:35.43 | igcewieling | blehxor: assuming you remove OSX and replace it with Linux I don't see why you would have issues. |
| 18:35.47 | *** join/#asterisk serafie (~erin@nat/digium/x-uytfflulvahlupwc) |
| 18:36.04 | jmetro | And of course, asterisk is linux based. |
| 18:36.32 | blehxor | I need either a mac or windows machine for other reasons, just wondering if its feasible to combine |
| 18:37.13 | jmetro | I use windows, and SSH into our asterisk box. Vim FTW |
| 18:37.13 | asilva | igcewieling: you're right.. |
| 18:37.16 | igcewieling | blehxor: Windows won't run Asterisk at all. On a MAC you might be able to get it running after spending countless hours trying to figure out why the docs (which are all for Linux) translate. |
| 18:37.24 | asilva | igcewieling: it runs thru it |
| 18:37.35 | igcewieling | But why not put Asterisk on a Linux box and access that from your mac? |
| 18:37.56 | asilva | igcewieling: never paid attention to that!! |
| 18:38.03 | blehxor | thats probably what I'll do |
| 18:38.03 | jmetro | You can also do something crazy and use Notepad++ with its built in SSH tool to push/pull the files on use. |
| 18:38.05 | igcewieling | asilva: almost nothing has changed in AEL between 1.8 and 11 |
| 18:38.24 | igcewieling | blehxor: that is how all the cool kids do it. |
| 18:38.24 | asilva | jmetro: tkz! |
| 18:38.36 | igcewieling | hugs jEdit |
| 18:38.43 | jmetro | igcewieling: none of the cool kids use a mac though *shudder* |
| 18:39.12 | igcewieling | jmetro: I have no opinion on that. |
| 18:39.27 | jmetro | All of my users that switch from PC to Mac are the ones who use their google toolbar to google google and then click on google and search for "aol.com" to get to their email. |
| 18:39.30 | igcewieling | Mac got my respect when they switched to OSX. I would never buy one, but I respect them. |
| 18:40.00 | igcewieling | Same thing with iPhones. I would never buy one (for several reasons) but they don't appear to be horrible. |
| 18:40.38 | igcewieling | jmetro: uh, Macs are perfect for those people. |
| 18:40.39 | jmetro | igcewieling: i had an iphone for about 3 months since it was left-over when i started. Had them upgrade me asap. |
| 18:41.00 | igcewieling | jmetro: you can't remove the battery from an iPhone. That is one of the main reasons I won't buy one. |
| 18:41.10 | asilva | igcewieling: return; before it is :) tkz for the help! |
| 18:41.16 | jmetro | They also are real-time tracking your GPS location, a feature which you cannot turn off. |
| 18:41.25 | jmetro | the logs of said tracking are uploaded to any PC you connect your iphone to. |
| 18:41.26 | igcewieling | What do people with iPhones do when they visit their mistress or go buy drugs? Leave their phone at home? |
| 18:42.09 | jmetro | nah, they just get caught and dont know how =p |
| 18:43.00 | igcewieling | I know of a guy who sneaked off to a gay campground for a few days without making up a story for his wife. She filed a missing person's report, police tracked the phone. Last I heard he was living in the guest house and she filed for divorse. |
| 18:43.13 | jmetro | Classy |
| 18:43.21 | igcewieling | I don't know if he had an iPhone or not. 8-| |
| 18:43.36 | jmetro | that kind of tracking was probably cell tracking. |
| 18:43.51 | igcewieling | jmetro: I strongly disapprove of such behavior. |
| 18:43.58 | *** join/#asterisk pensmit (~pensmit@nc-184-3-96-113.dhcp.embarqhsd.net) |
| 18:44.11 | igcewieling | (the cheating part, none of the rest) |
| 18:44.17 | blehxor | well thanks for the advice |
| 18:44.43 | pensmit | is there a way for meetme to listen for dtmf and then make a call or join someone else to the conference? |
| 18:44.57 | jmetro | Anyway, the moral of the story is, get a Win7 PC and your future self will be much more productive and more talented for the experience. |
| 18:45.01 | pensmit | say if I press 5 while in meetme it calls a an extension |
| 18:45.20 | igcewieling | pensmit: did you ask this before? maybe it was someone else. |
| 18:45.34 | pensmit | I'm not sure |
| 18:45.39 | jmetro | I think Confbridge does that |
| 18:45.52 | igcewieling | I'd go find the answer to your question by using "core show application meetme" and looking at the meetme.conf.sample file included with Asterisk, but I'm too lazy right now. |
| 18:46.09 | igcewieling | BTW, that was a hint. |
| 18:46.22 | jmetro | Iceweasel is good at hints. |
| 18:46.32 | pensmit | lol |
| 18:46.39 | pensmit | does anyone just know? |
| 18:46.42 | Jinxed- | would it be possible to dial a number, and have the result of the dialed number call other devices and place them in a conference |
| 18:47.35 | jmetro | Like when your phone rings and you pick up and get placed on hold by the machine? |
| 18:49.28 | Jinxed- | I have some phones set up to auto answer |
| 18:49.41 | Jinxed- | basically I want to dial a number and have it setup a conference of phones |
| 18:49.50 | Jinxed- | dial another number and have it kick everyone out of the conference |
| 18:50.06 | igcewieling | Jinxed-: yes, it is possible to dial an extension which runs an AGI script or script to generate .call files |
| 18:51.16 | igcewieling | Jinxed-: Yes, you can use .call files and dialplan to force people into a conference. |
| 18:52.44 | *** join/#asterisk serafie (~erin@nat/digium/x-pjfhjmimuwtsnwyj) |
| 18:56.24 | pensmit | I looked in those places and didn't see anything. Does anyone know if this is possible? |
| 18:56.25 | Jinxed- | interesting |
| 18:56.38 | Jinxed- | how do I check if my asterisk has conferencing |
| 18:56.47 | pensmit | Is there a way for meetme to listen for dtmf and then make a call or join someone else to the conference? |
| 18:56.48 | Jinxed- | I have a lofty goal but need to start at square 1 |
| 18:57.33 | jmetro | Jinxed look for the modules meetme or confbridge |
| 19:01.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.31) |
| 19:02.35 | Jinxed- | hmm |
| 19:02.45 | Jinxed- | so I did a core application show MeetMe |
| 19:02.49 | Jinxed- | and didn't see anything |
| 19:02.55 | Jinxed- | I also don't have a meetme.conf |
| 19:03.02 | WIMPy | pensmit: Confbridge allows you to call dialplan. |
| 19:03.35 | Jinxed- | Is meetme not a standard feature? Is there some other way to conference users together besides meetme that is standard? |
| 19:03.48 | WIMPy | Jinxed-: ConfBridge |
| 19:03.56 | WIMPy | MeetMe requires dahdi. |
| 19:04.08 | WIMPy | But Confbridge has more features anyway. |
| 19:04.21 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:04.34 | Jinxed- | WIMPy: why wouldn't more people use ConfBridge then? |
| 19:04.55 | Jinxed- | Not questioning you (believe you) just curious |
| 19:04.59 | WIMPy | Because they use Asterisk versions from the last century perhaps. |
| 19:05.15 | asilva | lol |
| 19:05.48 | pensmit | ok |
| 19:06.05 | pensmit | well we run 60+ users in a conference and meetme has worked and it's part of freepbx |
| 19:06.08 | pensmit | that's my reason |
| 19:06.14 | pensmit | so confbridge will be a shift |
| 19:06.30 | pensmit | thanks guys |
| 19:06.42 | WIMPy | Luckily we don;t suppor FreePBX in here, so that's not an argument :-) |
| 19:07.04 | Jinxed- | WIMPy: so should conference bridge be supported in Asterisk 1.8.5.0 |
| 19:07.38 | WIMPy | Jinxed-: Use a current version. |
| 19:08.03 | Jinxed- | I'm unable to update the version on the device I'm using |
| 19:08.39 | jmetro | Why? |
| 19:08.47 | WIMPy | Get someone who is. |
| 19:09.01 | yoavz | I'm now building a big script that's monitoring my servers and if one of them is down I want it to record a message using "echo "Server X is down" | text2wave".... and call the local pager office. The thing is, I can't foresee how much time it'll take the operators to pick up. Is there any indicator that'll allow me to play the message only after the operator has answered? |
| 19:09.07 | yoavz | Thanks :) |
| 19:09.21 | jmetro | Trying to add 100 phones to a conference and listen to them and then kick them out of the conference with a button push kind of sounds shady. |
| 19:10.04 | jmetro | yoavz: core show application dial, there is an option to dial that will let you play a macro to the dialed party. |
| 19:10.12 | WIMPy | thinks today is retro day. |
| 19:10.40 | Jinxed- | jmetro: I only want to conference max 3 to 4 users |
| 19:11.08 | Jinxed- | on sip devices that are so dumb all they do is answer when called, you can't actually call from them, hang up, or anything |
| 19:11.13 | yoavz | jmetro: Hmm, is there any way to determine if the operator has picked yet or I'm just listening to on hold music? |
| 19:11.14 | asilva | yoavz: check option G |
| 19:11.14 | Jinxed- | it must be done externally |
| 19:13.05 | yoavz | asilva: option G, as far as I understood, is used to connect to legs to one after one party has answered |
| 19:13.07 | Jinxed- | hmm sounds like I may sol for getting any type of conference to work with 1.8.5.0 and no meet me installed |
| 19:13.09 | jmetro | yoavz: not that i know of. You could loop a sound file like all the pro's do. |
| 19:14.34 | yoavz | jmetro: I wanted to avoid that... Those basterds at the paging company charge me for every second the opertaor is on the line... I wanted to make something that'll start playing only after the operator really answered... |
| 19:15.01 | yoavz | Maybe I can have them shut down the elevator music and just keep the line ringing till the operator can pick up |
| 19:15.14 | yoavz | BTW, WIMPy, thanks for your help from last week :) |
| 19:16.11 | jmetro | yoavz: unfortunately both could be considered as "answered" |
| 19:16.26 | jmetro | yoavz: think of your asterisk server. You "Answer()" the call, and then start ringing on someone on your side. |
| 19:17.49 | yoavz | jmetro: That's what I thought at first, I was hoping for a "miracle"... :) |
| 19:18.37 | jmetro | Even a "Confirmation" would extend your time since they have to listen to the message |
| 19:18.42 | Jinxed- | Do you need to load confbridge or something |
| 19:18.48 | Jinxed- | it says it started in 1.6 |
| 19:19.01 | jmetro | check to see if it exists in your modules bro |
| 19:20.08 | yoavz | Interesting... I found this: http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect |
| 19:20.10 | WIMPy | There was an old version on 1.8. The real ConfBridge started on 10. |
| 19:20.47 | Jinxed- | :( |
| 19:21.47 | Jinxed- | WIMpy So because I can't update or compile from source, and there is nothing in "core show application c<tab>" that results in confBridge or "core show application m<tab>" for meetme |
| 19:21.51 | Jinxed- | nothing I can really do |
| 19:21.54 | Jinxed- | ? |
| 19:22.05 | jmetro | why cant you update or compile? |
| 19:22.17 | WIMPy | Get a new system. |
| 19:22.32 | jmetro | A pi would be a cheap alternative thats programmable. |
| 19:25.45 | Jinxed- | WIMPy: well I have plenty of other systems that I have control over, but this one has features I need that I can't get on other systems (jmetro and I have a couple of pi's which are fun but I prefer reg linux boxes for the voip playing) |
| 19:25.58 | Jinxed- | I wouldn't have a way to recompile it unless the vendor did it |
| 19:26.06 | pensmit | hey with confbridge what would you have to allow someone to press the number 5 and go to a certain extension |
| 19:26.22 | WIMPy | Then put a RPi next to it. |
| 19:26.44 | WIMPy | pensmit: Configure a menu in confbridge.conf. |
| 19:26.54 | jmetro | So its a phone server from a third party in which you cant touch the asterisk because of the third party things that integrate with it basically? |
| 19:27.21 | pensmit | thanks wimpy |
| 19:27.34 | pensmit | i wonder if I can get that even with freepbx installed |
| 19:27.47 | Jinxed- | yes, the third party has additional configuration and capability integrated into the system that I would lose if I tried to put another asterisk system next to it |
| 19:28.08 | WIMPy | pensmit: Probably not as easy, but that question belongs to #freepbx |
| 19:28.17 | Jinxed- | which is the whole reason I wanted to do it on this system, because it has these extra features |
| 19:28.59 | WIMPy | Why do you have to do it on the same box? |
| 19:29.08 | pensmit | me? |
| 19:29.21 | WIMPy | Jinxed- |
| 19:29.26 | pensmit | oh |
| 19:29.33 | Jinxed- | If I have a call manager [A] that I have calls over, can it put people on conferences that are registered with this other system call manager[B] that I don't have control over? |
| 19:29.36 | pensmit | yeah...I've asked freepbx...those bastards won' |
| 19:29.40 | pensmit | t answer wimpy |
| 19:29.48 | pensmit | those dirty, dirty bastards |
| 19:30.22 | WIMPy | pensmit: Well, i guess you get an idea, how easy FreePBX is now :-) |
| 19:30.58 | pensmit | trust me friend |
| 19:31.01 | pensmit | i don't want to use it |
| 19:31.04 | pensmit | but |
| 19:31.09 | pensmit | I have no choice |
| 19:31.25 | Jinxed- | is that possible WIMPy ? |
| 19:31.27 | pensmit | i have told folks that there is going to come a time when we want to use more advanced features |
| 19:31.29 | pensmit | or |
| 19:31.45 | pensmit | do things that the freepbx layer will slow us down |
| 19:31.55 | WIMPy | Jinxed-: Have you ever made a call to a non-local device? |
| 19:32.10 | WIMPy | pensmit: Looks like it already does. |
| 19:35.14 | *** join/#asterisk Alex25 (~kvirc@bzq-79-180-215-188.red.bezeqint.net) |
| 19:35.20 | Jinxed- | yeah i think i'm good for now |
| 19:35.23 | Jinxed- | thanks WIMPy |
| 19:35.43 | Alex25 | hi |
| 19:36.57 | pensmit | yep |
| 19:37.52 | Alex25 | weird scenario: |
| 19:37.58 | Alex25 | I want to allow a spied channel to kill all calls on system by pressing some digits |
| 19:38.13 | Alex25 | what's the best wasy to achieve that? |
| 19:38.59 | pensmit | ok can you configure the menu in meetme |
| 19:39.06 | pensmit | in conferences |
| 19:39.38 | WIMPy | I don't think there was such an option. |
| 19:39.57 | Alex25 | how to configure it? |
| 19:44.54 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 19:47.34 | jmetro | Alex25: the same way you do anything else, you make an extension that executes some code. |
| 19:50.21 | Alex25 | but how can i configure it to execute the code by pressing a digit, in any phase of call? |
| 19:50.39 | WIMPy | "features" |
| 19:50.53 | jmetro | ^ |
| 19:51.01 | WIMPy | Which is obviousely a really bad name for a feature. |
| 19:51.12 | jmetro | "press 0 to crash the server" |
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| 19:51.39 | Alex25 | so "features" is a part of meetme application /? |
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| 19:51.49 | jmetro | now might be a good time to read the book |
| 19:51.50 | WIMPy | no |
| 19:52.06 | jmetro | because features is a standard set of asterisk. its the same way you park by dialing *70 or etc. |
| 19:52.12 | WIMPy | Meetme doesn't have such functionality. |
| 19:52.34 | Alex25 | can u link me to info about that? |
| 19:52.51 | WIMPy | Look at your features.conf. |
| 19:53.16 | Alex25 | I've never configured that file |
| 19:53.23 | Alex25 | but I'll look |
| 19:55.16 | Alex25 | ;disconnect => *0 |
| 19:55.21 | Alex25 | do u mean that one? |
| 19:55.36 | WIMPy | No, you will have to make one yourself. |
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| 19:55.47 | jmetro | Alex25: you have to read the book's chapter about features.conf |
| 19:55.50 | jmetro | or do research. |
| 19:56.03 | Alex25 | ok |
| 19:56.29 | Alex25 | but I'm just trying to understand if it will solve my problem |
| 19:56.46 | WIMPy | It's not going to be an easy one. |
| 19:57.01 | Alex25 | does it work for calls which are not bridged yet? |
| 19:57.16 | jkroon | guys, guessing I need to ask in #asterisk-dev, but just to confirm, I'm busy cooking a patch for format_g729a.c that'll detect buggy SIP peers that insists on sending CNG frames, warn about them and move on (instead of littering my logs with with hundreds and hundreds of useless warnings) |
| 19:57.32 | WIMPy | Alex25: Probably not. |
| 19:57.37 | jmetro | i dont know if its possible with meetme, but for any other situation you would be making a custom feature that executes your code. |
| 19:57.50 | jkroon | ideally I'd like to make mention of where the data comes from, but can't seem to find an easy way to do that from the info in format_g729.c - any ideas? |
| 19:58.56 | Alex25 | I need to get it work even if call isn't bridged.... any other idea? |
| 19:59.24 | igcewieling | ~book |
| 19:59.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:59.32 | WIMPy | Alex25: I'm not sure I got your question. Weren't you on about spied on calls? |
| 19:59.44 | Alex25 | yep |
| 20:00.06 | Alex25 | I want to let a spied channel to execute a code by pressing a digit |
| 20:00.13 | Alex25 | anywhere on his call |
| 20:00.14 | WIMPy | Maybe you should describe more precisely what you have and what you want to do. |
| 20:00.45 | Alex25 | ok |
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| 20:01.13 | Alex25 | so forget about the spy issue that's not so essential |
| 20:01.26 | Alex25 | a call starts with a progress() |
| 20:01.51 | Alex25 | and then playback some music with the noanswer flag |
| 20:02.28 | Alex25 | I want to allow someone who initiate that call to execute some code by pressing a * |
| 20:02.34 | Alex25 | or another digit |
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| 20:03.33 | Alex25 | how can I do this? |
| 20:08.55 | jmetro | if youre just dialing someone, no meetme involved |
| 20:09.01 | jmetro | youre looking for features.conf |
| 20:11.08 | jkroon | Alex25, use Background() ? |
| 20:12.08 | Alex25 | i tried it, but it's not working well when channel is spied on |
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| 20:14.23 | fullstop | I've been away from asterisk for a few years, since 1.6, and I'd like to think that I'm not that rusty. |
| 20:14.46 | fullstop | however, since I just have the one line in this setup, I was not going to use the stdexten macro. |
| 20:16.16 | fullstop | http://pastebin.com/raw.php?i=RJYdqsx8 |
| 20:17.04 | fullstop | What's wrong with the goto? Asterisk states that s-NOANSWER doesn't exist in context incoming-motif but it sure looks like it does. |
| 20:17.33 | jmetro | show us the real dialplan? |
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| 20:19.15 | fullstop | jmetro: http://pastebin.com/raw.php?i=LQw99TAL |
| 20:19.46 | jmetro | and the actual failed call? |
| 20:19.51 | fullstop | more logging revealed this, so I'm likely doing something wrong. |
| 20:19.53 | fullstop | Priority 's-NOANSWER' must be a number > 0, or valid label |
| 20:21.20 | fullstop | jmetro: http://pastebin.com/raw.php?i=8Ly6TdX5 |
| 20:21.55 | jmetro | Goto(s-${DIALSTATUS}) -> Goto(s-${DIALSTATUS},1) |
| 20:22.02 | fullstop | derp |
| 20:22.02 | fullstop | thanks |
| 20:22.37 | jmetro | No such label 's-NOANSWER'*** in extension 's'*** in context 'incoming-motif' |
| 20:22.38 | jmetro | was my hint |
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| 20:46.27 | jmetro | "cachertclasseS" dont work |
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| 21:50.55 | zamba | does anyone know of a rack-mountable several port analog ip adapter? |
| 21:51.10 | zamba | like 6-8 ports? |
| 21:53.11 | jmetro | 4 SPA112's strapped to a rack-mountable shelf would probably be best. |
| 21:53.50 | igcewieling | or you could use something like an Adtran TA9xx or NetVanta series boxes. 8 - 48 ports |
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| 22:04.44 | jmetro | igcewieling: oh. Cachertclasses did 0 things. |
| 22:04.48 | jmetro | i just had to stick sort=random in there. |
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| 22:16.01 | zamba | jmetro: SPA112 supports fax? |
| 22:16.15 | zamba | "Supports reliable faxing with simultaneous voice and data use" |
| 22:16.27 | zamba | igcewieling gave fax over ip thumbs down earlier today |
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| 23:52.44 | igcewieling | zamba: you are asking the wrong question. The question you should ask is "Does the SPA-112 support T.38, and if so, does it work well with Asterisk?" |
| 23:53.25 | igcewieling | None of this will make it work if your provider does not support T.38 |
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