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| 00:02.49 | danfromuk | navaismo: any idea how fast they process the orders? |
| 00:05.29 | navaismo | very quick, same day whitin 2 hours if i recall |
| 00:06.37 | navaismo | just make sure to do the order in the right way, if you ask for 3 licenses you only get 1 serial for 3 licences, if you want split licenses you need to buy separately |
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| 00:21.36 | danfromuk | I'm not sure what just happened to my asterisk box. I'm getting "No such command 'dialplan reload'" |
| 00:21.49 | voar | reload dialplan |
| 00:22.52 | danfromuk | voar: no, its dialplan reload. |
| 00:22.58 | danfromuk | also 'help' isnt working |
| 00:23.13 | voar | I stand corrected. Just verified myself as well |
| 00:23.39 | danfromuk | weird. i just removed the g729 codec module and now its working fine again |
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| 00:25.29 | [TK]D-Fender | danfromuk: Call in. Leave a voicemail. Pick it up. The end. |
| 00:26.26 | danfromuk | [TK]D-Fender: I would but when i enable the g729 module, asterisk doesnt seem to fully load up properly. Half the CLI commands are missing. |
| 00:26.29 | [TK]D-Fender | danfromuk: You could also just Record() and Playback() just the same |
| 00:26.50 | [TK]D-Fender | maybe you've got a b0rked module |
| 00:27.02 | danfromuk | When I delete the codec module and try loading asterisk again, the CLI commands are there and it loads properly |
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| 00:28.34 | danfromuk | The module is downloaded from the digium site. |
| 00:28.50 | danfromuk | Tried a few different flavours. |
| 00:31.11 | danfromuk | Do you have any suggestions? |
| 00:31.23 | [TK]D-Fender | remove all and finish your passthrough test |
| 00:31.44 | [TK]D-Fender | Which includes going phone-phone |
| 00:32.34 | danfromuk | I dont have a phone to test it with. Its an incoming DID. A client wants to use g729 on their server. But before I can give them access, I want to check that g729 from the DID provider is working properly. |
| 00:33.11 | voar | Send it back to G729? |
| 00:33.16 | [TK]D-Fender | You wan to test ... and you have nothing to test |
| 00:33.17 | voar | Out* |
| 00:33.22 | [TK]D-Fender | (with) |
| 00:33.29 | danfromuk | I was going to make a small dialplan which includes Set(SIP_CODEC=g729) and then ECHO |
| 00:33.36 | [TK]D-Fender | This is just broken thinking |
| 00:33.50 | [TK]D-Fender | It isn't passthrough unless it goes THROUGH |
| 00:34.04 | [TK]D-Fender | A ---> * ---> B |
| 00:34.19 | [TK]D-Fender | As for G.729 working.. I told you can just Record() and Playback() |
| 00:35.11 | voar | danfromuk, Why don't you take the DID and just point it at an external number? G729 from Provider to you, And then G729 on the second channel from you to provider, To let's say.. your cell phone. Also speaking of cell phone. If you've got a smartphone maybe use a softphone that supports G729? |
| 00:35.31 | danfromuk | Firstly, I can't test passthrough yet because I dont want to give the client access until I know that the DID provider is correctly establishing calls using g729. So I need my asterisk servers to support and be licensed to test g729 between the DID provider and my servers. |
| 00:37.13 | danfromuk | voar, I'll give it a try and see if I can get that to work. Currently we only make outbound calls using alaw. |
| 00:37.42 | [TK]D-Fender | That makes no sense |
| 00:38.08 | [TK]D-Fender | You don't need a full codec to prove it establishes right |
| 00:38.16 | [TK]D-Fender | Record() |
| 00:38.22 | [TK]D-Fender | Playback() |
| 00:38.26 | [TK]D-Fender | that is all you need... |
| 00:38.30 | danfromuk | Record doesnt require a license? |
| 00:38.44 | [TK]D-Fender | TRANSCODING does |
| 00:38.50 | danfromuk | Ok. I understand |
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| 00:54.47 | danfromuk | Whats the file extension for recordings if I want to record in g729? |
| 00:58.01 | [TK]D-Fender | .... |
| 00:58.03 | [TK]D-Fender | .g729 |
| 01:10.10 | Maliuta | [TK]D-Fender: really? |
| 01:10.28 | Maliuta | wow! who'd have thunk? ;) |
| 01:10.38 | [TK]D-Fender | chan_bigprint.so :) |
| 01:11.23 | danfromuk | Hmm. Doesnt seem to want to record. http://pastebin.com/78Vd5KDS |
| 01:11.24 | Maliuta | [TK]D-Fender: do we have chan_cluebat.so yet? |
| 01:11.49 | [TK]D-Fender | danfromuk: -- Executing [005117185269@incoming_calls:2] Set("SIP/46.19.209.14-0000001d", "SIP_CODEC=g729") in new stack <- stop assuming this means anything |
| 01:11.52 | [TK]D-Fender | danfromuk: And include SIP DEBUG |
| 01:12.00 | [TK]D-Fender | danfromuk: Beacuse so far.... you're running blicd |
| 01:12.04 | [TK]D-Fender | blind* |
| 01:12.22 | [TK]D-Fender | [2013-05-14 02:09:32] WARNING[12591]: channel.c:5205 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) |
| 01:12.23 | danfromuk | I did sip show channel to check it was running g729 |
| 01:12.27 | Maliuta | danfromuk: normally record to .wav and then use sox to get the .g729 |
| 01:12.28 | [TK]D-Fender | meaningless |
| 01:12.31 | [TK]D-Fender | show the entire call |
| 01:12.59 | [TK]D-Fender | [2013-05-14 02:09:33] WARNING[12591]: file.c:137 ast_stopstream: Unable to restore format back to gsm |
| 01:13.04 | [TK]D-Fender | It clearly started as GSM |
| 01:13.11 | [TK]D-Fender | and you are not controlling your test |
| 01:13.16 | [TK]D-Fender | You want passthrough... FORCE IT |
| 01:13.19 | [TK]D-Fender | And actually do the job |
| 01:13.31 | [TK]D-Fender | Stop trying to hack it after the fact in the dialplan. |
| 01:13.35 | [TK]D-Fender | That is more broken logic |
| 01:14.02 | Maliuta | amen |
| 01:14.17 | danfromuk | How can I force it for specific calls and not for others? I dont want all calls from this DID provider to be g729. |
| 01:14.24 | danfromuk | Just this specific DID. |
| 01:14.31 | [TK]D-Fender | danfromuk: Doesn't work that way. |
| 01:14.59 | danfromuk | Thats a pain. |
| 01:15.17 | Maliuta | danfromuk: that's how things work |
| 01:15.17 | danfromuk | Ok. I'll have to re-think this. |
| 01:15.54 | danfromuk | I dont want to affect all the other calls coming from this DID provider. May have to scrap this order. |
| 01:16.13 | Maliuta | you can tell a device/provider what you support - in order of preference - and the software negotiates the rest |
| 01:17.03 | danfromuk | I was under the impression that i could use SIP_CODEC to select a codec before the call is answered. |
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| 01:18.10 | igcewieling | danfromuk: you are welcome to try. 8-) |
| 01:20.06 | Maliuta | did the impression come from one of those pin table things? |
| 01:20.08 | Maliuta | ;) |
| 01:20.38 | danfromuk | The SIP debug seems to show that the call is established as g729. |
| 01:21.16 | [TK]D-Fender | danfromuk: remember the magic word : thorough |
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| 01:21.53 | tessier_ | Teliax has just pissed me off. Usually they've been pretty good. But this is the second time they've done something dumb. |
| 01:22.19 | danfromuk | You said that I should be able to use record to test it. I wanted to see if I can get it to select g729 on a per-call basis using SIP_CODEC |
| 01:22.34 | tessier_ | I need to pay more for service. Who has service which costs enough that they will actually call me if there is an issue which would prevent them from being paid? |
| 01:25.28 | [TK]D-Fender | danfromuk: You took what I said to do and twisted your approach to negotiating to code |
| 01:25.30 | [TK]D-Fender | c |
| 01:25.52 | [TK]D-Fender | danfromuk: That is not "testing passthrough" that is testing "how can I break codec negotiation" |
| 01:26.01 | [TK]D-Fender | dancStop polluting your tests |
| 01:27.08 | danfromuk | Ok. I'm confused. But I'll take your word for it. I'll get hold of the client's SIP URI and then try just doing a passthrough and see if i can get SIP_CODEC to work once thats done. |
| 01:27.22 | Maliuta | [TK]D-Fender: but I wanted to break codec negotiation ... it means less annoying calls? |
| 01:34.19 | tessier_ | I also need to find a way to monitor my phone system to ensure that calls can be answered. Especially tricky when this number doesn't get calls very often but when calls do come they are important. |
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| 01:35.26 | tessier_ | Anyone care to recommend a company to transfer my DID to? |
| 01:36.12 | [TK]D-Fender | tessier_I also need to find a way to monitor my phone system to ensure that calls can be answered. Especially tricky when this number doesn't get calls very often but when calls do come they are important. <- you could try ... I dunno ... calling it? That's usually a good way to test to see if it answers... |
| 01:46.28 | tessier_ | [TK]D-Fender: That is not a useful suggestion. I can't call every hour. I don't want my system to be down without my knowing it for even an hour. |
| 01:46.38 | igcewieling | [TK]D-Fender: I hacked up a set of scripts to call a server, and have the CID name and number encoded in ascii and sent to the caller using dtmf. 8-) I should clean it up an release it. |
| 01:47.37 | [TK]D-Fender | tessier_: Sure you can.. that's what automation is for. |
| 01:48.18 | [TK]D-Fender | tessier_: I don't know any provider that goes around e-mailing you when calls aren't making it in. I know some that'll fail-over to some secondary place.. but that's different |
| 01:48.31 | [TK]D-Fender | tessier_: Detection != redirection |
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| 04:18.39 | igcewieling | [TK]D-Fender: Vitelity has that option (or had, I've not use it for a while). |
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| 06:03.34 | Addisk | anyone alive? |
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| 06:27.16 | Addisk | i'm getting this wierd [WARNING]: get_headers(http://mirror.freepbx.org/provisioner/v3/polycom/polycom.tgz): failed to open stream: HTTP request failed! |
| 06:27.20 | Addisk | any ideas?? |
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| 06:29.03 | Addisk | i can maually grab the file, but the endpoint manager just not doing it :/ |
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| 07:23.21 | Sling | what do the flags (S) and (T) mean in 'iax2 show peers' output after the IP's ? |
| 07:23.42 | Sling | sorry, (S) and (D) |
| 07:23.56 | Sling | the (T) flag is after the port |
| 07:25.18 | kaldemar | Sling: Static/Dynamic |
| 07:25.47 | Sling | ah, thanks |
| 07:26.08 | kaldemar | (T) is for trunk. |
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| 08:02.33 | BarthezZ | does anyone know how accurate the output of "sip show channelstats" is? I see percentages well above the 100 (like 5000% for example) |
| 08:02.52 | BarthezZ | this is on ast. 1.8.20.1 |
| 08:03.20 | BarthezZ | Strangely enough the packets lost coutner exceeds the packets sent |
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| 08:06.31 | eagles0513875 | hey guys :) if i setup an asterisk pbx can i have multiple clients and multiple numbers? on one setup |
| 08:07.01 | ChannelZ | Yes-ish |
| 08:07.14 | eirirs | hehe ofcourse |
| 08:07.28 | eirirs | and multiple vendors |
| 08:07.30 | Maliuta | umm it's PBX, that's what it's for |
| 08:07.32 | eirirs | multiple everything |
| 08:07.48 | ChannelZ | except for multiple SIP ports. BA-ZING! |
| 08:08.00 | eirirs | actually - yes you can |
| 08:08.00 | eagles0513875 | the next question is how can lets say people at home connect to the pbx |
| 08:08.00 | eirirs | :) |
| 08:08.06 | Maliuta | eirirs: even orgasms? because I haven't got that from my * setup yet |
| 08:08.20 | Maliuta | eirirs: do you have the code for chan_orgasm? |
| 08:08.24 | eirirs | Maliuta: make a phone sex thru your pbx, with a IVR ! |
| 08:08.27 | eirirs | an* |
| 08:08.46 | eirirs | needs to setup the IVR properly for phone sex though |
| 08:08.59 | eagles0513875 | lol guess nobody saw my 2nd question |
| 08:09.00 | ChannelZ | eagles0513875: through the interwebs |
| 08:09.04 | Maliuta | eirirs: but there isn't the vocab in the sound files for it |
| 08:09.11 | ChannelZ | It's all the rage |
| 08:09.19 | eirirs | you can actually create your custom sound files |
| 08:09.20 | eagles0513875 | ChannelZ: guessing they would need voip phones that one sets up to connect to the pbx |
| 08:09.21 | eirirs | :) |
| 08:09.26 | eirirs | or let other make it ;) |
| 08:09.33 | ChannelZ | Yeah, or softphones on their computer |
| 08:09.37 | Maliuta | eirirs: I don't want to have phone sex - not even with myself |
| 08:09.40 | Maliuta | :P |
| 08:09.44 | eagles0513875 | ChannelZ: does asterisk have a specific softphone |
| 08:09.50 | eagles0513875 | or any softphone software really |
| 08:09.51 | ChannelZ | Or you go the analog route |
| 08:09.56 | Maliuta | ChannelZ: now you're just making shite up ;P |
| 08:10.00 | ChannelZ | Any SIP softphone |
| 08:10.01 | eirirs | Maliuta: well, the possibility are there if you wants a multiorgasm from a PBX |
| 08:10.02 | eirirs | :P |
| 08:10.11 | eagles0513875 | ChannelZ: and i can forward calls to mobile devices as well right |
| 08:10.25 | ChannelZ | If you have service to do so, yes |
| 08:10.33 | Maliuta | sets up a phone sex line for eirirs that just plays tt-monkeys |
| 08:10.50 | eirirs | no, I got other... ahem, vendors for such things |
| 08:10.53 | eagles0513875 | ok :) im sure ill end up asking lots of questions soon |
| 08:11.07 | eagles0513875 | in terms of analogue lines wouldnt i need a special gateway for that |
| 08:11.15 | ChannelZ | Yes and no |
| 08:11.24 | Maliuta | eirirs: you should hear the tt-monkeys in the Australian sound set ... it sounds like a bunch of ducks |
| 08:11.31 | eirirs | lol |
| 08:11.42 | ChannelZ | You can do VoIP to an ITSP who then bridges your call onto the public telephone network |
| 08:11.46 | eirirs | if it quacks like a duck... |
| 08:11.58 | eagles0513875 | ChannelZ: woudl the 3cx softphone and app work for accepting and making calls :p |
| 08:11.59 | ChannelZ | It really depends on what you want these "clients" to be able to do |
| 08:12.05 | Maliuta | eagles0513875: you need an ATA of some sort for analogue, digium make cards for it tooo |
| 08:12.24 | eagles0513875 | Maliuta: im going to be setting this up on my vps at my provider so analogue is outa the questiono |
| 08:12.32 | Maliuta | eirirs: and it tastes like a duck ... |
| 08:12.49 | eirirs | Maliuta: woot? a PBX that gives you tastes? gimme |
| 08:13.04 | eagles0513875 | then im guessing the sip trunk providers charge by the min or can i come to some sort of monthly fee agreement? |
| 08:13.05 | Maliuta | eagles0513875: so why ask about analogue then? |
| 08:13.22 | Maliuta | eagles0513875: depends on the ITSP |
| 08:13.23 | eirirs | would love to set custom tastes at extensions, for ppl I don't like lol |
| 08:13.23 | ChannelZ | So you do all SIP |
| 08:13.32 | eagles0513875 | ok |
| 08:13.36 | Maliuta | or IAX |
| 08:13.47 | eagles0513875 | how do sip trunk providers charge usually? |
| 08:13.54 | ChannelZ | Depends |
| 08:13.57 | eagles0513875 | on what |
| 08:14.07 | Maliuta | eirirs: I think I have that code for chan_taste ... but it went bad ;) |
| 08:14.09 | ChannelZ | Some have monthly unlimited incoming and you pay per min for outgoing |
| 08:14.18 | Maliuta | eagles0513875: the proider |
| 08:14.22 | eagles0513875 | ok |
| 08:14.32 | ChannelZ | Some do per minute all around with a small monthly charge (like me) |
| 08:14.39 | Maliuta | s/proider/provider/ |
| 08:14.43 | eirirs | Maliuta: ah, you got fired for accidentally setting taste on your CEO's phone? |
| 08:14.49 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 08:14.58 | eagles0513875 | then im thinking just bill clients a monthly rental fee then bill them the costs of the calls as well |
| 08:15.07 | Maliuta | eirirs: no, it was just a bad taste |
| 08:16.00 | Maliuta | eirirs: and it kept playing "Air Supply" in the background |
| 08:16.48 | Maliuta | eagles0513875: so you are going to use a vps to onsell SIP calls from another provider? |
| 08:17.10 | Maliuta | eagles0513875: unless you're consulting and push clients to use your service I don't see the point |
| 08:17.11 | ChannelZ | I'M ALL OUT OF LOVE! |
| 08:17.24 | Maliuta | ChannelZ: we already knew that :P |
| 08:17.33 | eirirs | lol |
| 08:17.47 | Maliuta | ChannelZ: I'M NOT IN LOVE ... |
| 08:18.03 | Maliuta | 10cc, the size of my ex's bladder ;) |
| 08:18.14 | eirirs | TMI |
| 08:18.16 | eagles0513875 | Maliuta: i plan to push clients to do my service im trying to build up my services i have to offer |
| 08:18.41 | eagles0513875 | Maliuta: i have a linode 24gb of space 1gb ram and 2tb of bandwidth i dont think bandwidth wil be an issue cuz then i have 4 of those for a total of 8tb of bandwidth lol |
| 08:19.15 | eirirs | 8tb bandwidth? |
| 08:19.18 | aruntomar | my calls are recording, but b'cas of mistake of mine, the master.csv is not updated. now i want to search for specific call records. is there another way, wherein i could get the filename <filename>.wav for a specific call. |
| 08:19.31 | Maliuta | eagles0513875: to do it properly you need to actually connect to the PTSN and do deals with other VoIP providers aswell as PTSN carriers |
| 08:19.54 | eagles0513875 | Maliuta: pstn carriers you mean |
| 08:20.02 | Maliuta | them tooo |
| 08:20.15 | eagles0513875 | whats the difference between ptsn and pstn |
| 08:20.43 | Maliuta | I just get my ETLA's mixed up sometimes. |
| 08:21.16 | Maliuta | the only copper I deal with is to my portable handset, and my DSL line |
| 08:21.26 | ChannelZ | PTSN is Post Traumatic Stripping Nude |
| 08:21.52 | Maliuta | everything else is some kind of web related traffic |
| 08:22.06 | eagles0513875 | Maliuta: i deal wiht copper for the tv lol and internet |
| 08:22.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:22.12 | eagles0513875 | and network cabling in the house |
| 08:22.26 | Maliuta | eagles0513875: not _that_ copper |
| 08:22.42 | eirirs | haha |
| 08:22.51 | Maliuta | copper pairs for phone lines |
| 08:22.55 | eagles0513875 | oh haha |
| 08:23.02 | eagles0513875 | is pstn really needed |
| 08:23.15 | eagles0513875 | couldnt i get away with sip and then softphones for those that have mobile internet access |
| 08:23.18 | ChannelZ | If you plan to call anyone, sort of. |
| 08:23.26 | eagles0513875 | ok |
| 08:23.38 | ChannelZ | But not in the 'direct' sense as I think you think you mean |
| 08:23.55 | Maliuta | I think that in a few years most stuff will just be VoIP, people will just need to learn that a phone number is just like an email address |
| 08:24.33 | ChannelZ | I doubt a few years |
| 08:24.44 | ChannelZ | Phones and phone numbers aren't going away anytime soon |
| 08:24.57 | eagles0513875 | they are all going voip |
| 08:24.58 | Maliuta | eagles0513875: I'm talking about setting up a proper ITSP, with dial out capabilities to the pstn |
| 08:25.06 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:25.18 | eagles0513875 | Maliuta: in other words being able to dial from a voiip number to a pstn network number |
| 08:25.22 | *** part/#asterisk Sling (~Sling@unaffiliated/sling) |
| 08:25.47 | Maliuta | ChannelZ: with the NBN here in .au the copper is going to pretty much rot and then disappear |
| 08:26.04 | Maliuta | eagles0513875: or vice versa |
| 08:26.11 | eagles0513875 | gotcha |
| 08:26.17 | eagles0513875 | going to be a while before i can offer that service then |
| 08:27.05 | eagles0513875 | Maliuta: would i need pstn agreements with local carriers to be able to call pstn numbers or all pstns globally |
| 08:27.12 | ChannelZ | That's what ITSPs are for. |
| 08:27.20 | Maliuta | eagles0513875: so if I were to do that here in .au I'd be dealing with Telstra and Optus for multiple lines and phone numbers, then selling that time on to customers through VoIP |
| 08:27.42 | eagles0513875 | so in otherwords local psnt's |
| 08:27.45 | eagles0513875 | is all i would need |
| 08:27.59 | eagles0513875 | ChannelZ: what itsp are here in malta if any |
| 08:28.07 | ChannelZ | I have no idea what you're talking about that this point |
| 08:28.19 | ChannelZ | ?? ask The Google |
| 08:28.24 | eagles0513875 | ill google |
| 08:28.53 | Maliuta | ChannelZ: what eagles0513875 has been describing is basically setting up an ITSP that is SIP only ... which I see as pointless |
| 08:29.07 | eagles0513875 | im willing to go pstn as well |
| 08:29.19 | eagles0513875 | just wondering if making an agreement with the local pstn provider would be sufficient |
| 08:29.26 | eagles0513875 | or if it would need to be done on a global scale |
| 08:29.31 | ChannelZ | The short of it is, there are (basically) two ways to connect Asterisk to the phone network. You either do SIP to an ITSP and let them deal with it, or you put hardware in your system and get a T1 or two and connect to a telco with it |
| 08:29.32 | Maliuta | ChannelZ: I'm trying to educate him on what is required to set one up properly |
| 08:30.00 | Maliuta | ChannelZ: or you do both |
| 08:30.25 | ChannelZ | Or others but that's the simple explanation. |
| 08:30.29 | Maliuta | you could route some calls through pstn and others through an itsp ... cost based routing |
| 08:30.49 | eagles0513875 | problem is im not finding any itsp's here in malta |
| 08:31.04 | Maliuta | eagles0513875: you'd be needing to bring something big to the table to get decent rates |
| 08:31.31 | eagles0513875 | to who Maliuta |
| 08:31.43 | Maliuta | eagles0513875: itsp's don't have to be local |
| 08:31.50 | eagles0513875 | oh ok so any itsp |
| 08:32.13 | eagles0513875 | so an itsp would suffice to get me to make calls even to pstn based numbers |
| 08:32.21 | Maliuta | eagles0513875: to get decent call rates from a pstn carrier you'd need to be buying time in bulk, lots of bulk |
| 08:32.25 | ChannelZ | though termination rates to your own region will vary.. |
| 08:32.39 | eagles0513875 | so what would be the first thing to do |
| 08:32.44 | ChannelZ | Your ITSP can be in the USA but it's going to cost you more to call across the street. |
| 08:32.46 | eagles0513875 | setup astrisk and then work on other things |
| 08:33.03 | eagles0513875 | i could ask vodafone here not sure if they provide that kind of service |
| 08:33.24 | ChannelZ | Yeah. learn Asterisk first, it's not something you "just install" and suddenly offer phone service to people you want to pay you for said service. |
| 08:33.36 | eagles0513875 | ChannelZ: :) i know |
| 08:33.41 | Maliuta | eagles0513875: yes. You could use, for example, Pennytel who are here in .au as your ITSP. They have decent call rates to a number of countries |
| 08:33.57 | eagles0513875 | Maliuta: there is no one itsp that will cover the globe |
| 08:34.30 | ChannelZ | Just depends on who your customers want to call. |
| 08:34.52 | eagles0513875 | humm ok |
| 08:35.29 | Maliuta | eagles0513875: most of them do, because they connect to local pstn providers who provide the OS trunks. Unless they have a deal with a ITSP in another region and route those calls via VoIP |
| 08:35.42 | eagles0513875 | man this is complex |
| 08:35.43 | eagles0513875 | lol |
| 08:35.47 | ChannelZ | If they're going to call tons of people in China, you'd get better termination rates from an ITSP in the country. |
| 08:35.50 | Maliuta | eagles0513875: also depends on if you need to provide DID's to your customers |
| 08:36.00 | ChannelZ | that too |
| 08:36.03 | eagles0513875 | did's remind me what those are the term seems familiar |
| 08:36.26 | Maliuta | Direct In Dial ... a phone number |
| 08:36.37 | eagles0513875 | ahh right then i would need to find a local sip trunk provider for that no? |
| 08:37.02 | Maliuta | if you want the numbers to be local then sure |
| 08:37.32 | eagles0513875 | ok then if the client is making ots of calls or wants to deal with a market in another country then i would need to find a sip trunk provider from that country |
| 08:38.16 | Maliuta | I have a Canadian provider and a DID from them (for talking to my parents) and Pennytel here in .au with another DID. Then I also get a DID and VoIP service with my DSL plan |
| 08:38.46 | Maliuta | and the two .au numbers are "local" to different states |
| 08:39.22 | Maliuta | eagles0513875: or one that has decent rates to that country. |
| 08:39.37 | eagles0513875 | exactly :) |
| 08:39.43 | Maliuta | eagles0513875: have a look at the pricing plans on a number of ITSP sites and compare them |
| 08:39.59 | eagles0513875 | Maliuta: any ones you recommend |
| 08:40.07 | eagles0513875 | or a site with a listing of them cuz wikipedia has a few listed |
| 08:40.37 | Maliuta | sometimes it might be worth using a provider in another country, if you're not worried about lag, just to save some money on a pstn terminated call |
| 08:41.10 | Maliuta | eagles0513875: I think there is a list somewhere on the * wiki, wikipedia should have a list too |
| 08:41.15 | Maliuta | ~wiki |
| 08:41.48 | eagles0513875 | lets say i have a DID for a client and as well he makes calls through an itsp what happens is that ok or would there be issues with that |
| 08:44.40 | eagles0513875 | Maliuta: poor freenode and this DdoS attack :( |
| 08:47.24 | eagles0513875 | hey guys will any softphone work with asterisk or are there specific softphones for asterisk |
| 08:47.28 | eagles0513875 | as well as hardphones |
| 08:48.13 | Maliuta | now I'm confused. The client has a DID(or an in-dial range), through which they receive calls, and which they set as their outgoing callerid. The outgoing VoIP calls can be sent anywhere, even directly to another SIP/IAX server |
| 08:48.45 | eagles0513875 | and even to a itsp |
| 08:49.02 | Maliuta | if the softphone supports SIP or IAX then it should work with *, hard phones are different kettle of fish |
| 08:49.35 | Maliuta | well if it goes to an itsp then it's going via SIP/IAX - which is VoIP |
| 08:50.09 | Maliuta | I have to run off to a meeting |
| 08:50.19 | eagles0513875 | Maliuta: thanks for all your help :) |
| 09:03.14 | *** join/#asterisk admin0 (~admin0@2001:1af8:4100:1128:1dc8:e327:17d:46d5) |
| 09:04.17 | admin0 | i would like to make asterisk build very small .. just iax2 support and ulaw support .. rest everything can go .. no voicemail, no playback etc .. how do I set this in the config flags |
| 09:04.22 | admin0 | so that i get a very small base |
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| 09:16.50 | bombev | hi all |
| 09:17.23 | bombev | I am looking to intigrate web based click to call function |
| 09:17.31 | bombev | is there any tutorials? |
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| 09:59.02 | aruntomar | bombev: for which platform and browser you are looking for? |
| 10:01.53 | bviktor | any ideas why the fop2buttons table doesn't get updated when an extension's name's changed? |
| 10:03.01 | bviktor | fop2buttons/label displays the name with which the extension was created, but after i change an extension's details, it still displays the old name |
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| 10:29.50 | racho | how does deny/permit actually work? it seems completely useless for sorting incoming calls based on ip |
| 10:33.59 | bombev | aruntomar Mozilla or Chrome it does not matter |
| 10:37.58 | emk | How do I make sure that a variable persists (and is useable) when a call jumps to a different extension? I have the following config in my extensions.conf: http://www.bpaste.net/raw/IQx9EgmtJVrwjXEJNUFn/ and when I enter the context the variable ${MY_CUSTOM_VARIABLE} is set but when the call jumps to extension one or two then the variable ${MY_CUSTOM_VARIABLE} is no longer set. |
| 10:40.13 | aruntomar | bombev: windows or linux? |
| 10:40.22 | bombev | aruntomar windows |
| 10:41.35 | aruntomar | bombev: well on windows there is zoiper sip client, and it has some active x plugin for click to call kind of functionality |
| 10:42.50 | aruntomar | bombev: we use linux, + twinkle sip client, firefox plugin called telify which calls twinkle at the backend to dial calls for our click to call requirements. |
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| 12:33.45 | Assid | heya |
| 12:34.37 | Assid | so .. i figured out why my NEC has issues talking with me.. apparently if i enable enbloc sending on the NEC; the call conference etc starts working. HOWEVER, i am unable to "simulate" a dialtone in that case |
| 12:38.19 | Assid | however; the system more or less works if i use overlap.. but my NEC acts a bit funny |
| 12:55.04 | jeffspeff | is there a max number of characters for the name field in sippeers? |
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| 13:05.08 | carrar | jeffspeff, I would probably keep it 31 characters or less |
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| 14:29.31 | eagles0513875 | hey guys |
| 14:29.45 | eagles0513875 | is it better to use the version of astrisk that is in ones distro repos or compile from source? |
| 14:30.22 | mmlj4 | that depends |
| 14:30.27 | mmlj4 | which distro? |
| 14:30.35 | WIMPy | If you want to save a minute. |
| 14:31.19 | mmlj4 | but generally your distro will maintain stable software, while compiling from source will result in folks telling you to recompile every time a version tick is released |
| 14:31.28 | eagles0513875 | mmlj4: ubuntu 12.04 |
| 14:31.57 | eagles0513875 | im looking as well at the getting started documentation all i need to do is just install the asterisk packages and that is it or is some configuration required |
| 14:32.09 | mmlj4 | you would choose one I have no knowledge of |
| 14:32.31 | eagles0513875 | ? |
| 14:32.37 | mmlj4 | oh, some, or actually a lot of configuration is required, regardless of what you decide to do |
| 14:32.57 | mmlj4 | asterisk isn't plug-and-pray software |
| 14:33.04 | WIMPy | Asterisk has a configuration before make if that means anyhting to you. |
| 14:33.14 | eagles0513875 | WIMPy: im not going source route |
| 14:33.24 | eagles0513875 | im suprised that the configuration isnt done after install via a web interface |
| 14:33.50 | WIMPy | There are web interfaces available, but very limited in what they can do. |
| 14:34.02 | file | there are projects built around that, but Asterisk itself isn't meant to be a complete package out of the box |
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| 14:34.59 | eagles0513875 | ok im trying to see if ubuntu has it documented and it sadly doesnt |
| 14:34.59 | mmlj4 | web interface? that's funny |
| 14:35.15 | eagles0513875 | well the competition has at least an application to manage the pbx |
| 14:35.18 | jmetro | ubuntu instead of debian? *gigglefit* |
| 14:35.50 | WIMPy | Asterisk is not a PBX. Some people say you can use it to build one, however. |
| 14:36.07 | eagles0513875 | WIMPy: what is asterisk exactly |
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| 14:36.36 | [TK]D-Fender | eagles0513875: Asterisk is a PBX and telephony toolkit. What you make out of it is up to you. |
| 14:36.50 | [TK]D-Fender | eagles0513875: For me it's a coffee-timer and jukebox... |
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| 14:37.02 | eagles0513875 | [TK]D-Fender: so the sky is the limit |
| 14:37.12 | [TK]D-Fender | eagles0513875: For some... |
| 14:37.21 | eagles0513875 | :D and im that some hehe |
| 14:37.22 | [TK]D-Fender | eagles0513875: What do you want to accomplish? |
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| 14:37.44 | eagles0513875 | i woudl like to provide voip services and like line rentals etc as part of my business |
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| 14:38.05 | WIMPy | Are are a lot of limits very close to earth. |
| 14:38.38 | eagles0513875 | ? |
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| 14:39.38 | WIMPy | There are... |
| 14:39.58 | igcewieling1 | was there a question from eagles0513875? |
| 14:40.14 | eagles0513875 | been answered already |
| 14:40.21 | [TK]D-Fender | igcewieling1: All meta. |
| 14:41.41 | igcewieling1 | [TK]D-Fender: Ah. Q: How many Buddhist Monks does it take to change a lightbulb? A: None, the lightbulb contains the seeds of its own enlightenment. |
| 14:41.51 | igcewieling1 | There meta question and meta answer |
| 14:43.42 | coppice | the buddhist monasteries around here seem to light the place from the grid |
| 14:44.50 | WIMPy | Then go and tell them that it's perfectely possible to light your rooms using Linksys or Digium phones. |
| 14:44.58 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 14:45.55 | coppice | how many phones does it take to light this http://www.discoverhongkong.com/eng/images/see-do/highlight-attractions/large/1.1.1.10-Giant-Buddha_03.jpg |
| 14:47.40 | *** join/#asterisk slidesinger (~slidesing@c-69-141-23-216.hsd1.nj.comcast.net) |
| 14:48.19 | WIMPy | With thos models not too many, I guess. |
| 14:49.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:49.09 | *** mode/#asterisk [+o malcolmd] by leguin.freenode.net |
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| 14:52.04 | drmessano | What about the Buddhist Monk that went through the drive through at a burger joint.. When they asked him what he wanted on his burger he said "Make me one with everything" |
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| 15:24.41 | coppice | I saw some idiot journalist tell that joke to the Dalai Lama |
| 15:27.19 | mmlj4 | idiot journalist? why repeat yourself? |
| 15:27.37 | *** join/#asterisk carrar (~tim@osburn.com) |
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| 15:30.49 | drmessano | What about when the guy gave him his burger and told him "$4"... so he hands him a $5. After a long pause, he says to the burger guy "Where is my change?" and he replies "Change only comes from within" |
| 15:31.21 | drmessano | Double ba-dump, ching! |
| 15:32.26 | eirirs | then you don't have a badass look |
| 15:32.32 | igcewieling1 | I have created a monster |
| 15:32.43 | coppice | if they try that on me when I have lunch at the monastery, someone's getting a spanking |
| 15:32.45 | eirirs | oppenheimer |
| 15:32.54 | eirirs | coppice: monastery? lol |
| 15:38.12 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 15:40.20 | slav3_kitten | next time i think i'm going to just easily change routers.... i need to be bitch slapped |
| 15:41.43 | jmetro | changing routers is ezmode |
| 15:41.56 | *** part/#asterisk leedm777 (~leedm777@nat/digium/x-kafaadvbfowokmdq) |
| 15:42.26 | slav3_kitten | jmetro, for no apparent reason my asterisk server can not reach the sip or iax providers |
| 15:43.02 | slav3_kitten | and there is some screwy shit works fine on x vlan, but gives an error 206 on the two other vlans |
| 15:43.34 | *** join/#asterisk kusznir (~kusznir@76.178.145.28) |
| 15:46.51 | jmetro | slav3_kitten: you are always having issues...=p |
| 15:47.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 15:47.40 | igcewieling1 | jmetro: I think his internet uses rubber bands, bobby pins, dixie cups and string. |
| 15:48.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.180) |
| 15:48.09 | slav3_kitten | jmetro, i am lately. shit was running fine before the isp without warning changed my static IP |
| 15:48.12 | jmetro | igcewieling1: and not real rubberbands, those plastic ones. |
| 15:54.08 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
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| 16:17.06 | joesmoe_ | I'm buying BTC for OkPay if anybody is interested. I'm verified and all with OkPay, already have the balance @ OkPay. |
| 16:20.30 | jmetro | you came here to buy bitcoins? |
| 16:22.42 | Qwell | jmetro: Obviously. What else would you do in #asterisk? |
| 16:23.37 | jmetro | Qwell: though you could definitely finance a 'vette by daytrading bitcoins |
| 16:23.51 | Qwell | or lose your life savings. |
| 16:23.57 | Qwell | but either way |
| 16:24.04 | joesmoe | bahhh |
| 16:24.06 | joesmoe | sorry |
| 16:24.10 | joesmoe | my channel order got messed up |
| 16:24.13 | joesmoe | how's everybody |
| 16:24.44 | joesmoe | i buy and sell BTC daily |
| 16:25.15 | joesmoe | OkPay stopped working with BTC-e so now i've got a bunch of money tied up with OkPay (which isn't a problem as it's an irreversable payment provider, so most people don't mind taking it for BTC) however they are located in Cyprus and i don't want my money sitting there too long. |
| 16:26.09 | jmetro | dwolla imo |
| 16:26.51 | jmetro | even if you bought btc right when it crashed its already recovered |
| 16:27.37 | jmetro | holy s.... i should have bought more when it was 7$ a coin |
| 16:35.03 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 16:42.25 | *** join/#asterisk mviera (~mviera@84.122.57.125.dyn.user.ono.com) |
| 16:42.29 | mviera | Hello all |
| 16:45.29 | mviera | I have a question, if someone could help me would be really appreciated :-) I have an asterisk 1.6, and I have a limit of 7200 secs (2 hours) in meetme, but I can't find a proper solution. I've googled it but I haven't found anything... |
| 16:46.29 | Qwell | ~upgrade asterisk |
| 16:46.29 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:47.34 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
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| 16:52.22 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:52.26 | saint_ | hi all |
| 16:54.59 | saint_ | Does anyone have asterisk work with 1 line or 1 DID on a multi line, and is able to detect fax ? |
| 16:55.23 | saint_ | My stuff is not working correctly. I thought when a fax was detected, it would go to the fax extension of the same context .. |
| 16:55.32 | [TK]D-Fender | what is a "multi line"? |
| 16:56.13 | saint_ | [TK]D-Fender: like if you have more than 1 channel coming into the same number.. not a T1 , but like 2 or 3 chans to the same number ..' |
| 16:56.27 | saint_ | [TK]D-Fender: i probably used the wrong terms |
| 16:56.32 | saint_ | but you know what i mean ... :D |
| 16:56.47 | [TK]D-Fender | No, I really don't... |
| 16:56.57 | [TK]D-Fender | Show us what's happening and be clear about what you ARE using |
| 16:58.17 | saint_ | I have a fax extension in my dialing plan. To make sure it works correctly (the setup), I change my dialing plan so if a call comes in, it goes directly to the fax. By doing this, and sending myself a fax from another number, it works, and I have the .tiff that I can see as a fax. |
| 16:58.23 | saint_ | That is over google voice |
| 16:58.25 | saint_ | and it works |
| 16:58.39 | saint_ | now, I heard that asterisk can detect fax after Answer() |
| 16:58.58 | saint_ | so I changed my dialing plan, Answer(), and Dial() an extension -which is working- |
| 16:59.11 | saint_ | In the context where I answer, I still have my fax extension . |
| 16:59.13 | igcewieling1 | weren't you ask in the same question before I went on vacation? |
| 16:59.27 | saint_ | igcewieling1: nah, before your vac I had issues with DPMA |
| 16:59.34 | saint_ | which is almost fixed now |
| 17:00.00 | saint_ | unless if you call the past week end your vacations .. i just tried this fax stuff yesterday.. |
| 17:00.58 | saint_ | [TK]D-Fender: so when I am back to normal with a context that answer and dial , and had a fax extension , if I call with a fax , my phone rings, and the fax is never detected / sent to the fax extension .. |
| 17:01.12 | Qwell | Why are you answering? |
| 17:01.39 | saint_ | Qwell: because the wiki I read (i need to find it) said that you need to answer in order for asterisk to detect the fax |
| 17:07.29 | saint_ | ha, here : miscarriage |
| 17:07.38 | saint_ | ooops: http://nerdvittles.com/?p=88 |
| 17:07.56 | saint_ | When Asterisk answers the call, it listens for a fax tone. If it hears one, it reroutes the incoming call to a context which then processes the incoming fax. |
| 17:08.24 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 17:08.47 | *** part/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:08.54 | saint_ | same here: http://www.voip-info.org/wiki/view/Asterisk+fax |
| 17:09.19 | [TK]D-Fender | saint_: Your description about "going to fax" is too vague. Clean this up a lot. Be clear about exactly WHAT your call is coming over. |
| 17:09.52 | [TK]D-Fender | saint_: And show what you're actually doing, not some guide link. |
| 17:15.52 | saint_ | [TK]D-Fender: as simple as that: http://pastebin.com/mJiUtsSC |
| 17:15.57 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:15.58 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:16.29 | tL- | ™1 |
| 17:17.53 | saint_ | [TK]D-Fender: If a fax is detected during the answer / dial , it's supposed to go to exten => fax , and it does not . |
| 17:19.22 | [TK]D-Fender | saint_: DAHDI has fax detection. SIP has T.38 You seem to be using Google Voice. Who told you that * detects faxes over that? |
| 17:19.52 | saint_ | nobody. but nobody said it did not. |
| 17:20.45 | [TK]D-Fender | saint_: Instrustions say what things do, not what they do not. |
| 17:20.51 | *** join/#asterisk Big_Boss (~None@cpe-67-247-177-212.rochester.res.rr.com) |
| 17:21.14 | [TK]D-Fender | saint_: unless to specifically prevent you from making a dangerous open ended assumption |
| 17:21.19 | saint_ | I assume google voice is sip, isn't it ? |
| 17:21.24 | [TK]D-Fender | NO |
| 17:32.11 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:32.19 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
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| 17:43.41 | *** join/#asterisk sweeper (~sweeper@softcheese.xen.prgmr.com) |
| 17:44.36 | sweeper | hey folks. I have a gstreamer application I need to connect to a conference call. I can make gstreamer talk rtp or raw udp. any suggestions on how I would go about hooking this up? |
| 17:44.42 | *** part/#asterisk tL- (gtgt@gateway/shell/sh3lls.net/x-coudsfpifwgssilc) |
| 17:45.44 | *** join/#asterisk asilva (~asilva@2801:88:1000:2::12) |
| 17:48.35 | *** join/#asterisk LeLutin (~lelutin@vps-1036922-2706.manage.myhosting.com) |
| 17:52.52 | asilva | I'm having a problem using Asterisk 11 and might be a BUG in IAX2 peer comunication. Can someone check this out with me ? |
| 17:52.53 | asilva | 2 servers, ubuntu 12.04.02 LTS 32bit Asterisk 11.0.2 in one box and Asterisk 11.1.0 in the other, same configurations, same network(no firewalls in between), IAX2 peers doesnt communicate, |
| 17:52.53 | asilva | if i downgrade the box to from 11.1.0 to 11.0.2 it works just fine every version from 11.1 up presents the same issue for more info |
| 17:52.53 | asilva | http://pastebin.com/VhwwLehs (THIS ONE) |
| 17:53.25 | Qwell | asilva: Where do you want help? Pick a channel. |
| 17:54.32 | asilva | Qwell: Don't know, could be here, because is the one i got an answer! |
| 17:57.31 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 18:00.47 | asilva | Qwell: care to help !? |
| 18:02.25 | *** join/#asterisk navaismo (~navaismo@189.241.84.20) |
| 18:03.09 | Kobaz | mmm |
| 18:07.02 | asilva | any thoughts ? |
| 18:09.19 | asilva | Qwell: ? |
| 18:09.28 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 18:09.37 | Qwell | I didn't say I would help, no. |
| 18:10.28 | asilva | I didn't ask if you would help, i'm asking if you know the problem! care to check it out and give me a hand ? |
| 18:12.51 | Kobaz | does it make sense to see multiple INVITEs during a dialog when you don't have canreinvite enabled? |
| 18:13.41 | jmetro | does it make sense that it was 55° yesterday and its 85° today |
| 18:13.57 | Kobaz | ? |
| 18:14.18 | jmetro | temp outside |
| 18:14.28 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 18:14.56 | Kobaz | i dunno |
| 18:15.04 | Kobaz | it was 65 yesterday and now it's like 50 |
| 18:15.17 | jmetro | yeah you suck, its 85 here |
| 18:15.42 | Kobaz | so anyway |
| 18:15.45 | Kobaz | invites? |
| 18:16.41 | sweeper | Kobaz: sounds like good fermentation weather. get some cider going \o |
| 18:16.53 | Kobaz | hehe |
| 18:17.03 | navaismo | asilva, can you psatebin the iax2 debug when you do a iax2 reload on both servers and maybe attach a tcpdump capture? |
| 18:17.33 | sweeper | has 16 liters of apple cider and 4 liters of peach/apple wine going |
| 18:18.07 | Kobaz | ah nice |
| 18:18.31 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 18:18.32 | asilva | navaismo: sure hold one |
| 18:18.33 | jmetro | sweeper: how does that taste compared to normal wine? |
| 18:19.14 | sweeper | jmetro: dunno, first time trying it. saw some white grape/peach juice at the grocery store and figured why not |
| 18:20.38 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:20.56 | sweeper | jmetro: the apple cider/wine (I add sugar to get 10-13$ abv) tastes like applie cider you'd get at the store, although better than some ciders I've had |
| 18:21.14 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 18:21.30 | jmetro | there's always that one apple cider / one eggnog you buy that tastes freaking delicious but you dont remember the name / where you got it. |
| 18:22.11 | sweeper | and that's why I keep brew notes :3 |
| 18:22.35 | asilva | navaismo: server 1 - http://pastebin.com/ir3BvBVQ -====== Server 2 - http://pastebin.com/PwQrsaYM |
| 18:22.53 | sweeper | jmetro: https://workflowy.com/shared/7ca20496-4410-48e6-d10e-5acbd815244c/ :D |
| 18:25.11 | sweeper | anyways, I'm really baffled with this thing. I've got webrtc clients sending/receiving audio to/from my server application, and I need to get those channels hooked up to a conference call |
| 18:26.18 | sweeper | can I get asterisk to listen/send RTP without any SIP negotiation? I could do ami calls or whatever from my app |
| 18:26.40 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:29.21 | navaismo | asilva, try modify the iax2.conf with an enter or add a ; at top save and reload again with the iax2 debug enable, i cant see in that debug the register stuff |
| 18:29.23 | navaismo | only pokes |
| 18:33.05 | asilva | navaismo: server 1 - http://pastebin.com/ecFGEcAw ==== Server 2 - http://pastebin.com/rMy7U24B |
| 18:42.07 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 18:45.16 | asilva | navaismo: connectivity between machines are OK, pings OK, same switch, same vlan, same network, not firewalls in between or in the machines! if i dowgrade the version to 11.0.2 or use 1.8s works 100% fine |
| 18:47.06 | asilva | navaismo: ignore the 6.150 btw inf the pastbin's! |
| 18:47.23 | navaismo | try removing serverlookup & iaxcompat |
| 18:48.07 | pabelanger | asilva: likely a bug. There was recent commits to chan_iax2, might want to open a bug and label it a regression |
| 18:48.07 | navaismo | and finally try with friends instead user/peer I cant see anything wrong with your config but maybe worth to test another way |
| 18:48.43 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 18:49.21 | asilva | navaismo: compat and lookup no changes! |
| 18:49.24 | navaismo | asilva, I have 11.2.0 and 11.3.0 connecting fine via iax2 have your tried 11.3.0 in both servers |
| 18:49.43 | asilva | navaismo: yes after 11.1 the problem persists, up to 11.3 |
| 18:50.14 | asilva | from 11.1 to 11.3 i have problem, 11.0.2 and below i don't |
| 18:50.35 | navaismo | weird, mine 11.2.0 to 11.3.0 connecting fine |
| 18:51.24 | asilva | 11.3.0 is even worse starting as OK ( X ms ) then turns to UNREACHABLE |
| 18:51.41 | asilva | navaismo: using user and peer or just friend ? |
| 18:52.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:53.07 | navaismo | using friend |
| 18:53.09 | asilva | navaismo: friend account also UNREACHABLE |
| 18:53.32 | navaismo | servers are in the same network right? |
| 18:53.37 | asilva | yes |
| 18:53.46 | navaismo | have you tried with the lan IP |
| 18:53.49 | navaismo | instead public? |
| 18:53.53 | asilva | yes |
| 18:54.28 | navaismo | and persist? |
| 18:54.39 | asilva | when using version 11.1 and up yes |
| 18:54.48 | asilva | when using 11.0.2 or 0.1 or 0 works 100% |
| 18:54.52 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:55.17 | navaismo | weird<goat voice> Maybe its time to open a bug |
| 18:55.18 | asilva | or even 1.8s |
| 18:55.24 | asilva | hehehe |
| 18:56.50 | navaismo | Do you have the original iax.conf? Try restoring and see if the demo user is actve |
| 18:56.59 | asilva | let me check |
| 18:57.11 | *** join/#asterisk maxhbp204 (~chatzilla@115.248.240.58) |
| 18:58.31 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 18:58.37 | maxhbp204 | Hi, how can i pass argument to pass on mailcmd command in voicemail.conf, actually i have added mp3 conversion script in mailcmd and i need to put mailbox number in script, but i am not getting how can i put there, I have tried to put ${VM_MAILBOX}, but it is coming empty on script, can anybody help on this |
| 18:59.12 | navaismo | asilva, a make samples will genertae but backup the folder(/etc/asterisk) first, that will override all files |
| 18:59.34 | maxhbp204 | i have pass like this mailcmd=/path/script ${VM_MAILBOX} |
| 18:59.48 | maxhbp204 | but it is not working and it is coming as blank, so can anybody help me on this please |
| 19:02.42 | asilva | navaismo: i'm pretty sure i have block for the outgoing 4569 it wont work. eheh between servers on version 11.0.2 works correctly.. do you have a working scenario ? like 11.0.2 and others versions working ? |
| 19:04.07 | navaismo | my working scenario is LAN to LAN, from asterisk 11.2.0(Vanilla) to asterisk 11.3.0(FreePBX) |
| 19:04.15 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.132) |
| 19:04.37 | asilva | this is a LAN TO LAN the only different thing is a PUBLIC ip settings eheheh |
| 19:05.26 | Qwell | public IP on a LAN? what? |
| 19:05.30 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.134) |
| 19:06.04 | asilva | Qwell: this is a public university and we don't use private IPs even for LAN. |
| 19:06.22 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:06.36 | Qwell | then it's not a LAN. |
| 19:06.45 | asilva | a public IP doens't define a LAN or WAN |
| 19:06.53 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 19:07.29 | asilva | the only difference is that one is routed thru the internet and the other has to go out thru a NAT! |
| 19:08.31 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
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| 19:11.05 | asilva | navaismo: going to do more tests and collect data and open a ticket for a bug! thkz for the help! |
| 19:11.23 | maxhbp204 | Hi, how can i pass argument to pass on mailcmd command in voicemail.conf, actually i have added mp3 conversion script in mailcmd and i need to put mailbox number in script, but i am not getting how can i put there, I have tried to put ${VM_MAILBOX}, but it is coming empty on script, can anybody help on this |
| 19:11.27 | asilva | navaismo: i really wanted to make sure that my configuration was correct |
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| 19:13.01 | jmetro | public ip on the local side = wrong. |
| 19:13.43 | Qwell | maxhbp204: Nowhere does it say that mailcmd can take variables. |
| 19:13.44 | seanbright | maxhbp204: what version of asterisk? |
| 19:15.51 | seanbright | actually, it doesn't matter. it's not possible. |
| 19:16.15 | maxhbp204 | i am using asterisk 11, thanks for reply seanbright |
| 19:16.51 | maxhbp204 | if i put some name instead of variable then it is coming to script, like mailcmd=/path/script test123 then i will get test123 over there |
| 19:17.09 | maxhbp204 | but if i put mailbox number or some asterisk variable then it is not |
| 19:17.45 | seanbright | yeah, asterisk variables are not replaced |
| 19:17.57 | maxhbp204 | do we have any method, how can i pass variable or mailbox number to mailcmd script |
| 19:19.45 | navaismo | maxhbp204, create a scrip which catches the output of the emailbody and then process it like here--->http://tonylandis.com/uncategorized/asterisk-voicemail-sending-emailing-custom/ |
| 19:20.19 | seanbright | perfect |
| 19:21.13 | maxhbp204 | ok thanks for this information |
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| 20:14.47 | ruben231 | hi guys i ahve installed on my cento with dahdi and it say you do have kernel source ---> my current kernel is 2.6.21.7-2.ec2.v1.2.fc8xen <--------------anyone can help please |
| 20:16.25 | navaismo | try yum install kernel-devel-`uname -r` |
| 20:18.53 | ruben231 | No package kernel-devel-2.6.21.7-2.ec2.v1.2 available. |
| 20:20.41 | ruben231 | navaismo: any idea..? |
| 20:21.18 | navaismo | check your repo source for that package |
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| 20:21.45 | navaismo | do you really need dahdi, seems a Cloud server right? |
| 20:22.49 | igcewieling1 | ruben231: pastebin the output of "rpm -qa | grep kernel" |
| 20:23.01 | ruben231 | <PROTECTED> |
| 20:23.08 | Chainsaw | ruben231: You don't need DAHDI for timing. |
| 20:23.11 | navaismo | asterisk version? |
| 20:23.12 | igcewieling1 | ruben231: no you don't. |
| 20:23.21 | navaismo | above 1.8 you dont need dahdi for timing |
| 20:23.24 | Chainsaw | ruben231: Unless you are trying to get 1.2 going, in which case, please explain to me what year it is. |
| 20:23.31 | igcewieling1 | you might need it for meetme mixing, but you don't need it for timing |
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| 20:24.13 | ruben231 | http://pastebin.com/3HR6KcHE |
| 20:24.33 | ruben231 | <PROTECTED> |
| 20:24.41 | Chainsaw | ruben231: Use Asterisk 11 and ConfBridge, and that requirement goes away. |
| 20:24.46 | navaismo | use confbridge instead mettme |
| 20:25.19 | ruben231 | <PROTECTED> |
| 20:25.20 | igcewieling1 | kernel-headers-2.6.26.8-57.fc8 doesn't match your running kernel |
| 20:25.47 | ruben231 | <PROTECTED> |
| 20:25.57 | igcewieling1 | ruben231: install it |
| 20:26.12 | navaismo | or just ignore us and keep installing dahdi |
| 20:26.15 | ruben231 | navaismo: yes i have 1.8 asterisk only |
| 20:26.25 | Chainsaw | ruben231: You should use 11. |
| 20:26.44 | jmetro | ~upgrade asterisk |
| 20:26.45 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:26.55 | ruben231 | hmmm sorry fro some application i wanted to used it does 1.8 at present only |
| 20:27.00 | igcewieling1 | 1.8 is still a supported branch |
| 20:28.02 | Chainsaw | igcewieling1: It's never going to do a confbridge without requiring DAHDI, which is not going to work on EC2. |
| 20:28.40 | igcewieling1 | Chainsaw: Ah. I never been crazy enough to install Asterisk in a VM. |
| 20:28.48 | Chainsaw | igcewieling1: So yes, pedantic instead of pragmatic means you and ruben231 can spend ages on your beautiful 1.8 install. |
| 20:29.04 | pabelanger | Asterisk works fine inside a VM |
| 20:29.10 | Kobaz | yeah it works fine |
| 20:29.14 | jmetro | asterisk works great in a vm |
| 20:29.41 | Chainsaw | Until you expect to scale it up, sure. |
| 20:29.45 | igcewieling1 | Chainsaw: pragmatic is waiting until I stop having to say "Gee, I'm glad we didn't run a version with THAT bug" when I look through Asterisk 11 changelogs |
| 20:30.05 | pabelanger | Chainsaw: you scale out with VMs, not up |
| 20:30.13 | Chainsaw | igcewieling1: Ah yes, because early 1.8 was a cakewalk. |
| 20:30.33 | igcewieling1 | Chainsaw: I did not upgrade to 1.8 until I stopped saying that too. |
| 20:30.41 | Kobaz | heh yeah |
| 20:30.58 | ruben231 | pabelanger: guys how you did te 1.8 on vm..im having challenge somehow now |
| 20:31.05 | igcewieling1 | Generally Asterisk is "good enough" for my use around .20 release. |
| 20:31.28 | pabelanger | Still using 1.8.7.1 for production installs, with a few backports |
| 20:31.36 | pabelanger | don't plan on upgrading any time soon |
| 20:31.42 | Kobaz | that was me on 1.8.12 |
| 20:31.46 | igcewieling1 | pabelanger: we tend to use a recent release of whatever branch is on the server. |
| 20:31.57 | pabelanger | ruben231: re-phrase your question |
| 20:32.13 | Kobaz | with some of my own bug fixes and lots of my own features before they were put in officially |
| 20:32.19 | jmetro | why would you scale up one box? |
| 20:32.20 | igcewieling1 | thankfully we now have 3 asterisk boxes so when we discover some weird issue it is usually localized to 1/3 of our calls. |
| 20:32.22 | jmetro | make multiple boxes. |
| 20:32.32 | ruben231 | pabelanger: hi guys i ahve installed on my cento with dahdi and it say you do have kernel source ---> my current kernel is 2.6.21.7-2.ec2.v1.2.fc8xen |
| 20:32.40 | ruben231 | http://pastebin.com/3HR6KcHE |
| 20:33.00 | pabelanger | Kobaz: Ya, I try not to include custom patches, only once I get them into trunk, do I consider backporting them :) Hopefully easier when I need to move the client forward |
| 20:33.19 | Chainsaw | VMs don't scale. Let's have more VMs and multiply that 10% overhead to more workloads. I'll never understand you virtualised people. |
| 20:33.30 | Kobaz | it's hard not to do that when you write your whole application layer to rely on them |
| 20:33.43 | igcewieling1 | Chainsaw: there are lots of reason to virtualize, running a PBX is not one of them. |
| 20:34.11 | jmetro | i like my separated boxes. |
| 20:34.34 | Chainsaw | jmetro: And having some margin in the design. You know, for when it gets busy. |
| 20:34.38 | pabelanger | Chainsaw: the whole purpose of VMs is to scale |
| 20:34.46 | pabelanger | you scale across more hardware |
| 20:35.08 | igcewieling1 | pabelanger: that is no different than installing more servers without using VM |
| 20:35.20 | pabelanger | live migrations |
| 20:35.24 | ruben231 | pabelanger::-[ |
| 20:35.26 | pabelanger | can't do that on bare metal |
| 20:35.33 | igcewieling1 | We use VMs for low volume servers like DNS and SMTP |
| 20:35.44 | igcewieling1 | pabelanger: you can't migrate a live PBX |
| 20:35.47 | Chainsaw | pabelanger: Two PSUs, a UPS and a trolley. |
| 20:36.54 | pabelanger | EIther way, bare metal vs vms both do the same thing. Saying VMs don't scale, is entirely correct |
| 20:37.25 | navaismo | ruben231, you need to ask to your EC2 provider where is the kernel-devel for that or donwload the sources and compile against the sources |
| 20:38.39 | ruben231 | ok |
| 20:38.46 | *** join/#asterisk polysics (~Adium@95.236.26.218) |
| 20:38.51 | polysics | hi! |
| 20:39.00 | Chainsaw | Or you use Asterisk 11 and bypass the need for DAHDI entirely. |
| 20:39.14 | Chainsaw | But roll that boulder up that hill if you must. |
| 20:40.49 | navaismo | ruben231, not sure if this url can help you to find your kernel devel but check it http://kojipkgs.fedoraproject.org/packages/kernel-xen-2.6/ |
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| 20:47.06 | polysics | so, I have this ConfBridge where agents sit idle while logic brings in customers one at a time |
| 20:47.24 | polysics | agents complain they can hear their own ambient noise in the conference when alone |
| 20:47.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:48.11 | polysics | anyone has any idea if a) that can be and b) there is something to mitigate/remove that? |
| 20:48.27 | igcewieling1 | mute them while nobody else is in the room |
| 20:48.45 | polysics | through what? AMI? |
| 20:49.11 | igcewieling1 | Are you bringing in calls using AMI? |
| 20:49.52 | polysics | yes |
| 20:50.00 | igcewieling1 | then using AMI is the easiest way |
| 20:50.36 | igcewieling1 | is there a confbrige option to auto mute when there is only a single caller? |
| 20:51.04 | polysics | I was looking but I think now |
| 20:51.06 | polysics | *not |
| 20:51.23 | Chainsaw | Most use hold music in that situation (only 1 caller on bridge). |
| 20:51.50 | Chainsaw | Which also causes complaints if the second caller doesn't show up quickly and they end up learning the melody by heart. |
| 20:51.54 | Chainsaw | It's always something :) |
| 20:53.03 | igcewieling1 | interesting idea, use silent hold music |
| 20:53.15 | polysics | we could use very low music |
| 20:53.19 | polysics | or a silence sample |
| 20:53.58 | polysics | moh plays instead of audio, genius :-) |
| 20:54.25 | polysics | at worst I just turn the music very low on Audacity |
| 20:54.37 | Chainsaw | Just don't do that super-annoying thing that HP do on their bridges. |
| 20:54.42 | polysics | which is? |
| 20:54.43 | Chainsaw | Silence, with an announcement every minute. |
| 20:54.57 | Chainsaw | Making you think "ah good, they're her... oh". |
| 20:55.02 | igcewieling1 | "get to work" "stop slacking" "a productive worker is a happy worker" |
| 20:55.11 | polysics | sounds like that Simpsons episode when Homer invents the "ALL IS GOING WELL" alarm |
| 20:55.16 | jmetro | protip: get the Cisco Call Manager hold music and put it on your system |
| 20:55.26 | polysics | jmetro: why so? |
| 20:55.38 | polysics | is it good? |
| 20:55.40 | jmetro | because $!@# everyone has it |
| 20:58.08 | jmetro | http://www.youtube.com/watch?v=6g4dkBF5anU |
| 20:58.39 | polysics | hey! I have heard this! |
| 20:58.50 | Chainsaw | Everybody has. |
| 20:59.01 | Chainsaw | We have the Uplink theme tune (because it's royalty free; Blue Valley). |
| 20:59.09 | drmessano | Anyone have a trick for playing MoH where we're not starting the same piece of audio from the beginning each time? |
| 20:59.24 | Chainsaw | https://www.youtube.com/watch?v=KUn9SYdPF4A |
| 20:59.26 | polysics | split up music in sections? |
| 20:59.43 | _Corey_ | drmessano: You could always use the (old) mp3 type setup and mpg123 |
| 20:59.50 | Chainsaw | (We're a networking company so it's doubly appropriate) |
| 20:59.55 | drmessano | I thought about that |
| 21:00.07 | _Corey_ | or stream an MP3 source |
| 21:00.28 | Chainsaw | drmessano: We use a single instance that keeps on looping, and it just connects people to that simple stream. It's a setting. |
| 21:00.39 | jmetro | ^ |
| 21:00.54 | Chainsaw | drmessano: cachertclasses=yes |
| 21:01.26 | Chainsaw | drmessano: And my "custom" application is dumbout TheBlueValley.s3m -m -s 8000 -r 2 -v 0.2 -o - |
| 21:01.52 | polysics | question 2: when a customer is put in the room, only the agent should hear the "person joined" sound |
| 21:01.57 | Chainsaw | Downside is you trigger this whenever your stream wraps around: [2013-05-14 21:59:18] WARNING[1268] res_musiconhold.c: poll() failed: Interrupted system call |
| 21:02.05 | polysics | instead, the customer hears that too, even though he is on quiet |
| 21:03.13 | polysics | is that possible or do we have something configured wrong? |
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| 21:03.17 | sweeper | can anyone tell me what the state of the webrtc support in asterisk is? |
| 21:04.27 | navaismo | sweeper, supported in asterisk 11 |
| 21:04.34 | Chainsaw | polysics: You should be able to enforce that with two different "user" types, agent & customer. |
| 21:05.08 | Chainsaw | polysics: Where "agent" has announce_join_leave=yes and customer has announce_join_leave=no. |
| 21:05.38 | drmessano | Ok.. I will try that. Thanks |
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| 21:13.56 | polysics | Chainsaw: thanks again |
| 21:14.29 | Chainsaw | polysics: Any time :) |
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| 23:05.38 | forgotmynick | hello |
| 23:07.52 | cusco | hi |
| 23:08.17 | forgotmynick | my current version of asterisk doesn't support google voice (if it's not broke..) so for some years I've been using a third party providor but for some reason which they can't explain, after a certain time the google voice trunk doesn't seem to take any incoming calls and all I get is missed call emails. |
| 23:08.57 | cusco | upgrade asterisk? |
| 23:08.59 | forgotmynick | I rarely get calls through the Google Voice number but nonetheless I'm thinking of deploying the latest version and was wondering if these disconnections/timeouts/whatever it is will happen or is known to happen? |
| 23:09.12 | forgotmynick | with Google Voice |
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| 23:13.38 | jagster` | probably a full logfile or something? |
| 23:13.51 | jagster` | sounds like you have to contact the vendor |
| 23:20.27 | [TK]D-Fender | forgotmynick: chan_motif is pretty solid these days |
| 23:27.24 | forgotmynick | [TK]D-Fender ok thanks |
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| 23:29.56 | fireman_biff | whats the difference between peerip and recvip in SIPCHANINFO? |
| 23:30.49 | fireman_biff | or in CHANNEL |
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