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| 02:24.12 | nickgaw | Hi, Does asterisk support any type of voice to text where a user would speak to the system and it would execicute a command? |
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| 09:38.15 | redax | hi! is there any CID wildcard possibility in dialplan ? |
| 09:42.07 | redax | like exten => _12X./3630X.,1,.... |
| 09:42.33 | redax | but seems like the CID part is not accepting wildcards |
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| 10:43.26 | carrar | redax, what do you use to signal pattner matching? |
| 10:43.31 | carrar | pattern |
| 10:44.58 | carrar | redax: What you want to do works great, if you tell it to do it properly |
| 10:46.44 | redax | carrar: got it, I had to add an underscore to the CID part as well. |
| 10:47.07 | redax | carrar: exten => _12X./_3630X.,1,.... is working |
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| 10:47.14 | carrar | :) |
| 10:48.54 | redax | so thanks :) |
| 10:49.18 | carrar | np |
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| 11:10.44 | marceloamorim | hello, I couldn't find how to set in queue, if nobody pickup even if they are available, how to leave the app_queue |
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| 11:11.55 | Zogot | afternoon |
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| 11:20.10 | _abc_ | Hello. I have some unistim trouble with certain phones. Years ago I talked to Cedric Hans by email about this, in 1.6.x times. I sent him 2 emails a few days ago and got no answer. Is he still active in * / unstim support and or on this channel? Does anyone know? Thanks. |
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| 11:25.24 | _abc_ | Cedric Hans is the author of chan_unistim.c |
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| 12:16.31 | dan_j | marceloamorim: you set the 'timeout' to something. then if no one answers, the call will leave the queue and continue in the dialplan. |
| 12:16.56 | dan_j | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Queue |
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| 12:18.11 | dan_j | I have a client that wants to put incoming sip calls into a queue but not answer them. Then after 20 seconds if they havent been answered, he wants them to be answered, played music on hold and put back into the queue. |
| 12:18.36 | dan_j | Is it possible to maintain the caller's queue position when doing this? If yes, how? |
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| 12:21.02 | kchehab | hi ppl , |
| 12:21.32 | kchehab | how can i use cmd monitor will callfile ,where i want to make an automated call with asterisk recording feature |
| 12:21.39 | marceloamorim | dan_j: the timeout on my side was setting to next extension and wasn't for queue, maybe option 'n - No retries on the timeout; will exit this application and go to the next step.' |
| 12:22.00 | marceloamorim | let me check, ty sir |
| 12:22.20 | dan_j | marceloamorim: timeout in queues.conf is what you are talking about |
| 12:22.30 | dan_j | I am talking about the timeout in the Queue() command |
| 12:22.37 | dan_j | Queue(queuename,[options,[URL,[announceoverride,[timeout,[AGI,[macro,[gosub,[rule,[position]]]]]]]]]) |
| 12:23.46 | dan_j | See the wiki link above |
| 12:23.50 | dan_j | "timeout - Will cause the queue to fail out after a specified number of seconds" |
| 12:25.03 | dan_j | kchehab: send the call through a dialplan that starts monitor? |
| 12:27.20 | kchehab | dan_j the call will start using web /perl script , my question ,how can start monitor for call using web/perl ...callfile |
| 12:28.01 | marceloamorim | dan_j: it worked, ty sir |
| 12:29.48 | kchehab | dan_j auto dial_out |
| 12:32.13 | dan_j | I've not dealt with callfiles yet. From what I understand, you specify the Channel which is the device you are calling, then you can specify a Context, Extension, Priority in your dialplan somewhere. |
| 12:43.32 | dan_j | You can either specify an application in the callfile, or a context/extension/priority in your dialplan. If you use the dialplan option, you can start monitor in the dialplan and then dialout |
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| 13:01.42 | kchehab | dan_j i test that but setting the context to test-contest then playback a file , it do the play back but not record teh call |
| 13:01.46 | kchehab | the* |
| 13:04.01 | kchehab | dan_j i think the DIAL command should be before the call |
| 13:04.27 | kchehab | dan_j sorry the monitor command should be before the dial ? |
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| 13:07.20 | dan_j | yes. I believe so |
| 13:10.07 | kchehab | dan_j how to make this happen , the call file cant prossed without Channel/ |
| 13:10.55 | dan_j | Keep channel, and put the monitor in the dialplan that you are using when you enter the Context and Extensions |
| 13:11.02 | dan_j | Extension |
| 13:15.20 | kchehab | dan_j i did and i set monitor command ,but its not recording |
| 13:16.12 | dan_j | You'll need to pastebin your callfile and dialplan. But i dont have experience with callfiles so maybe someone else can help. |
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| 13:30.35 | catphish | i'm looking to set up a SIP proxy, initially to migrate users from a legacy asterisk setup to a new one, and then in the future potentially loadbalancing between asterisk servers, are there any obvious choices for software to do this? |
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| 13:34.14 | dan_j | catphish: http://www.opensips.org/ |
| 13:34.21 | dan_j | But i dont have any experience with it |
| 13:35.47 | catphish | thanks |
| 13:38.52 | dan_j | Other option is to transfer calls between asterisk servers using IAX and DUNDi |
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| 13:57.57 | youtmon | is there a Fpbx plugin for Skype ? |
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| 13:59.21 | [TK]D-Fender | youtmon: this is not a #freepbx support channel |
| 13:59.29 | [TK]D-Fender | youtmon: Stop asking those questions here. |
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| 14:02.01 | youtmon | is there a plugin for Skype ? |
| 14:03.00 | catphish | http://lmgtfy.com/?q=skype+asterisk |
| 14:03.58 | [TK]D-Fender | ~freepbx |
| 14:03.59 | 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 |
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| 14:07.26 | _abc_ | Hello. I have some unistim trouble with certain phones. Years ago I talked to Cedric Hans by email about this, in 1.6.x times. I sent him 2 emails a few days ago and got no answer. Is he still active in * / unstim support and or on this channel? Does anyone know? Thanks. |
| 14:07.50 | [TK]D-Fender | _Never seen him here under tthat name |
| 14:07.52 | _abc_ | (repeat post since noon, will try one last time later in the day) |
| 14:07.58 | [TK]D-Fender | _You should be using the mail-lists for this |
| 14:08.09 | _abc_ | Well, I'm not on there now. Is he? |
| 14:08.16 | _abc_ | Thanka. |
| 14:08.18 | _abc_ | *s. |
| 14:08.26 | _abc_ | needs a break been working for 4 hours |
| 14:09.32 | [TK]D-Fender | Don't just nag ONE person. Maybe they're paying attention, maybe they aren't. Doesn't mean he's the ONLY person who might be able to help you. |
| 14:09.58 | [TK]D-Fender | Like everything else just ask that entire resources if ANYONE can't help |
| 14:10.11 | [TK]D-Fender | can* |
| 14:10.35 | _abc_ | Yeah I know the rules but I have good reason to try to continue the discussion I had with him 4 years ago or so ;) Besides that, I will ask others too, but my questions are rather specific. |
| 14:11.13 | _abc_ | As this is a nortel unsupported / ancient phone question, deep in the unistim protocol, unrelated directly to '*' |
| 14:11.19 | _abc_ | Thanks. |
| 14:12.02 | WIMPy | Maybe you can become the new maintainer of chn_unistim and ask yourself :-) |
| 14:12.23 | _abc_ | Given I have 4 patches against it so far... the answer is HELL, NO. But thanks... ;) |
| 14:13.18 | WIMPy | Sounds like there are good chances you already are the most active developer for it then. |
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| 14:19.49 | catphish | feels bad for being unhelpful to youtmon |
| 14:20.57 | [TK]D-Fender | catphish: This isn't the place for GUI support. He's been told several times. Still keeps asking. Nothing to feel bad about. |
| 14:21.20 | youtmon | catphish do not, its a holiday here for most in US. in retrospect i'm feeling good |
| 14:23.50 | youtmon | Been told several times .. when ..on the contrary boss man. I think you like to feel your one line statement is act of congress around here |
| 14:25.33 | [TK]D-Fender | It's a botlet and a well established norm here |
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| 14:26.38 | [TK]D-Fender | 3M plastics does not owe people support for their alarm clock-radio even though SONY used their plastic in it's construction |
| 14:26.54 | catphish | what holiday? |
| 14:27.15 | youtmon | most importatant one for many MLK |
| 14:28.43 | catphish | ah that guy :) |
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| 14:28.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:29.19 | youtmon | yeah most important |
| 14:29.57 | cemotyz09 | I've followed the tutorial https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial for setting up pjsip and tls. The issue is I couldn't find the ast_tls_cert script so I created one using openssl. Now when I try to register I get message to enter pem password and error adding private key but when I enter it nothing happens same error comes up about entering pem password. my question is is there somewhere to enter th |
| 14:29.59 | cemotyz09 | e key/cert passwords so this prompt doesn't return and hopefully have a working config |
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| 14:35.37 | BeachBall | is it true that we should use pjsip ? |
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| 14:37.10 | catphish | BeachBall: i believe so |
| 14:37.20 | [TK]D-Fender | BeachBall: It is the future. Chan_sip is in "slow death mode" |
| 14:37.50 | catphish | i'm redesigning right now with pjsip and lots of ARI |
| 14:37.56 | BeachBall | i love watching linux code fly across the screen, people that see it scratch there head, and it looks like i'm working hard |
| 14:38.04 | BeachBall | :} |
| 14:38.41 | dpilon | just compile a kernel in a loop...you will seem like the most productive person in the place |
| 14:39.21 | [TK]D-Fender | http://imgs.xkcd.com/comics/compiling.png |
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| 14:55.35 | BeachBall | does asterisk support h.323 ? |
| 14:58.46 | cemotyz09 | I've found newer versions of asterisk don't have the script I was looking for but still doesn't answer why certiificate being used isn't working. Is the CLI supposed to accept the passphrase being input? |
| 15:01.12 | [TK]D-Fender | BeachBall: There are clearly somple configs for a channel driver ... so I'd venture a guess and say "yes" |
| 15:02.51 | _abc_ | h.323 is supported in * but compiling / installing H323 libs on some systems is a b** |
| 15:04.03 | _abc_ | Also chan_sip will die real soon now, when the fat lady will sing. sip is mostly run through vpns in companies due to security. |
| 15:04.09 | catphish | downloads.asterisk.org is surprisingly slow |
| 15:04.10 | _abc_ | So no problems there so far. |
| 15:04.36 | catphish | i'm only pulling down files at 2-4Mbps |
| 15:04.51 | _abc_ | Moving on: is there a neat little trunking howto for * to * trunking? I seem to find only personal snippets. |
| 15:06.18 | catphish | _abc_: IAX is pretty easy to use, perhaps search for a iax tutorial |
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| 15:09.03 | BeachBall | i'm almost done installing |
| 15:09.04 | BeachBall | :) |
| 15:09.24 | _abc_ | I did see some iax*2* things but no direct tutorial... |
| 15:11.51 | [TK]D-Fender | Ther is no difference with "direct |
| 15:12.09 | [TK]D-Fender | Auth is the same on BOTH sides |
| 15:12.18 | [TK]D-Fender | It's no different that what you'd do with a provider |
| 15:12.33 | [TK]D-Fender | My side sends password? Other side better need it. |
| 15:12.54 | [TK]D-Fender | And it'd better match |
| 15:13.04 | [TK]D-Fender | Same with usernames if that's involved |
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| 15:14.07 | catphish | _abc_: yeah, its iax2 you want, maybe try http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
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| 15:28.26 | _abc_ | thanks. |
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| 15:53.59 | BeachBall | my registration to my provider is timing out |
| 15:58.34 | BeachBall | can someone look over my sip.conf? http://pastebin.com/h2s1aTtz |
| 15:58.37 | BeachBall | it's been a while |
| 15:58.42 | BeachBall | and i might have missed something |
| 15:58.46 | [TK]D-Fender | Don't lok at the conf alone |
| 15:58.49 | [TK]D-Fender | look at the COMMS |
| 15:58.53 | [TK]D-Fender | Prove what is going out |
| 15:59.02 | BeachBall | whats comms? |
| 15:59.17 | [TK]D-Fender | SIP <--------------- |
| 15:59.23 | [TK]D-Fender | Where is the SIP DEBUG? |
| 15:59.37 | BeachBall | oh i remember how to do that |
| 16:00.08 | [TK]D-Fender | Show us it registering. I don't see qualify being enabled on it so I'm not suspecting any status checks for it |
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| 16:05.00 | _abc_ | sip set debug on |
| 16:06.52 | BeachBall | for transport, i'm using udp, thats correct? |
| 16:07.09 | BeachBall | i'm getting an error of serious network trouble :( |
| 16:07.21 | _abc_ | How serious |
| 16:07.24 | BeachBall | it's red |
| 16:07.34 | _abc_ | Unfortunately some clever people decided sip tcp is also a thing |
| 16:07.43 | _abc_ | It almost works on local nets. Almost. |
| 16:07.58 | BeachBall | one think i know for sure, the problem lies in my config |
| 16:08.08 | _abc_ | And those clever people are ... Cisco. I am not joking. |
| 16:08.50 | _abc_ | chan_sip supports it and it works but I get shivers looking at connections in the firewall when that is going on |
| 16:09.20 | _abc_ | Is anyone using tcp sip on a wan at all? |
| 16:12.02 | [TK]D-Fender | wonder's what "red" means, and why he still has nothing to look at |
| 16:12.29 | [TK]D-Fender | _abc_Unfortunately some clever people decided sip tcp is also a thing <- That would be "the official RFC since day 1" |
| 16:13.04 | [TK]D-Fender | _abc_Is anyone using tcp sip on a wan at all? <- Plenty of people, and that the place you'll see it most regardless |
| 16:13.05 | _abc_ | Ah yes. Alas. |
| 16:14.00 | _abc_ | So transport=tcp should be default in sip.conf but isn't... how come? ;) </me is just being evil here -- disreagard> |
| 16:14.12 | _abc_ | -a |
| 16:14.27 | _abc_ | Anyway running tcp sip over a vpn is sort of wasteful. |
| 16:15.08 | [TK]D-Fender | It's YOUR sip.conf. There is no such thing as "default" |
| 16:15.18 | [TK]D-Fender | #responsibility |
| 16:16.02 | _abc_ | right |
| 16:16.19 | [TK]D-Fender | _abc_Anyway running tcp sip over a vpn is sort of wasteful. <- not wasteful, depending on what the goal is. If it's "I want my comms forcibly private" ... then TCP plays not inherent role as-is |
| 16:16.40 | [TK]D-Fender | the the role* |
| 16:16.41 | _abc_ | True. |
| 16:16.42 | [TK]D-Fender | gah |
| 16:16.57 | [TK]D-Fender | I'm just butchering sentences this morning. |
| 16:17.09 | _abc_ | No problem. The meaning came across perfectly. |
| 16:17.13 | BeachBall | http://pastebin.com/VLqWvSm6 |
| 16:17.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:18.41 | poolson | hey im seeing hundreds of failed attempts from the same IP |
| 16:18.46 | poolson | but fail2ban doesnt ban it |
| 16:18.51 | poolson | any idea why that might be ? |
| 16:19.54 | [TK]D-Fender | fail2ban being misconfigured would be the immediate thing |
| 16:19.58 | [TK]D-Fender | Sine ... that's its job |
| 16:20.11 | poolson | hmmm ive never modified it |
| 16:20.28 | [TK]D-Fender | BeachBall: [Jan 18 11:15:06] ERROR[6051]: chan_sip.c:4187 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 16:20.52 | [TK]D-Fender | that means you either have something fighting over the SIP port, or ahve a firwall or other similar issue |
| 16:20.56 | BeachBall | yes, i think that is a key clue |
| 16:21.27 | BeachBall | hmmm |
| 16:22.24 | [TK]D-Fender | BeachBall: First thing : Go prove how PJSIP is configured |
| 16:22.52 | BeachBall | that line makes me not feel well |
| 16:24.43 | [TK]D-Fender | * 13 has 2 completely separate SIP channel drivers. Up to you tto make sure they aren't fighting for the same port |
| 16:24.49 | [TK]D-Fender | Then check your networking |
| 16:24.57 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 16:28.33 | [TK]D-Fender | It is prettty apparent you're taking old configs to a new version pretty "blind" |
| 16:28.35 | [TK]D-Fender | context=inbound ;change to 'ext-did' or 'from-trunk' for asterisk@home |
| 16:28.56 | [TK]D-Fender | Considering "asterisk@home" as a name hasn't existed in ... 8 years maybe? |
| 16:30.53 | _abc_ | This shows how far back we go here ;) |
| 16:34.56 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:46.27 | *** join/#asterisk MRH3 (~Thunderbi@233.47.187.81.in-addr.arpa) |
| 16:47.13 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 16:47.21 | BeachBall | am i correct in thinking if using pjsip that i don't use sip.conf ? |
| 16:48.50 | catphish | BeachBall: correct, pjsip has its own config with different syntax, see https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
| 16:52.24 | BeachBall | how to debug pjsip ? |
| 16:55.15 | BeachBall | i don't see it trying to connect to my trunk |
| 16:59.56 | [TK]D-Fender | Did you configure it to? |
| 17:00.05 | BeachBall | i did a configuration |
| 17:00.08 | [TK]D-Fender | Based on what you've already shown .. you SHOULDN'T be. |
| 17:00.22 | BeachBall | i deleted my sip.conf |
| 17:00.26 | BeachBall | and using pjsip.conf |
| 17:00.35 | [TK]D-Fender | And that doesn't answer what I told you to look for |
| 17:01.13 | [TK]D-Fender | because now rather tthan focusing on just keeping things out of the way of what you used to know you are now starting fresh. |
| 17:01.22 | [TK]D-Fender | And we've seen nothing. So we're back to running blind |
| 17:01.28 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 17:02.26 | BeachBall | lunch time |
| 17:02.29 | BeachBall | be back in 1 hr |
| 17:07.52 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:09.55 | *** join/#asterisk membiblio (~membiblio@pool-71-112-149-68.pitbpa.fios.verizon.net) |
| 17:17.40 | *** join/#asterisk twanny796 (~user@195.158.67.164) |
| 17:21.46 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 17:21.48 | BeachBall | i'm back |
| 17:25.32 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:25.36 | catphish | that wasn't an hour |
| 17:25.50 | BeachBall | i know :{ |
| 17:25.51 | BeachBall | i'm sorry |
| 17:25.54 | BeachBall | i ate fast |
| 17:26.05 | BeachBall | now i got a belly ache |
| 17:28.01 | catphish | lol |
| 17:29.35 | BeachBall | i don't understand pjsip at all |
| 17:29.44 | BeachBall | does flowroute even work with it? |
| 17:34.21 | drmessano | Everything does |
| 17:34.26 | drmessano | It's just SIP |
| 17:34.39 | BeachBall | i really wish you had said that sooner :( |
| 17:34.50 | BeachBall | i just sent an email to flowroute support |
| 17:34.54 | BeachBall | now i'm gonna look stupid |
| 17:35.03 | drmessano | Pretty much |
| 17:36.22 | drmessano | Asterisk didn't invent SIP. Chan_sip is just one implementation. PJSIP is another and now Asterisk has decided to use it. It's all just SIP |
| 17:37.22 | drmessano | That's like asking if Postfix can talk to Bobos SMTP server. It's all SMTP |
| 17:39.19 | *** join/#asterisk MRH2 (~Thunderbi@233.47.187.81.in-addr.arpa) |
| 17:55.02 | BeachBall | now i get an error, unsupported transport |
| 17:57.16 | DanQuinney | best use a transport that's supported then |
| 17:57.25 | BeachBall | i tried udp and tcp |
| 17:57.28 | BeachBall | same results |
| 17:58.29 | DanQuinney | What does your pjsip.conf contain? |
| 17:59.35 | BeachBall | http://pastebin.com/iLZdKcy3 |
| 17:59.40 | BeachBall | don't judge me based on this |
| 18:10.10 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 18:12.47 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 18:12.47 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 18:18.41 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 18:18.41 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 18:22.48 | BeachBall | well? |
| 18:23.21 | [TK]D-Fender | Stop showing only the configs |
| 18:23.25 | [TK]D-Fender | This is stupid & useless |
| 18:23.34 | [TK]D-Fender | And you still aren't learning |
| 18:23.58 | BeachBall | i don't have anything else to show |
| 18:24.04 | BeachBall | debugging isn't giving anything |
| 18:24.54 | BeachBall | i have the pjsip logger on |
| 18:26.36 | [TK]D-Fender | SHOW US |
| 18:26.42 | [TK]D-Fender | We have no proof you turned it on. |
| 18:26.51 | [TK]D-Fender | You need to actually start showing things |
| 18:29.58 | BeachBall | http://pastebin.com/Gdcv6WGD |
| 18:31.35 | [TK]D-Fender | I still have no proof your channel drivers are not in conflict with one another |
| 18:31.45 | BeachBall | neither do i |
| 18:35.13 | *** join/#asterisk alsaiduq (~emerson@200.239.65.114) |
| 18:35.20 | alsaiduq | hello |
| 18:35.39 | alsaiduq | is anybody here? |
| 18:35.59 | seanbright | alsaiduq: just ask your question |
| 18:36.14 | seanbright | unless that WAS your question, in which case the answer is 'yes' |
| 18:37.42 | alsaiduq | hahaha no that wasnt my question. My question is: I'm trying to connect two asterisk via SIP but it's not working. |
| 18:37.44 | *** join/#asterisk spicyramen (~Adium@c-50-184-3-185.hsd1.ca.comcast.net) |
| 18:39.06 | alsaiduq | I'm following that tutorial http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 18:40.40 | alsaiduq | when i use 'sip show peers' to notice whether everything wents well, i realize that the two asterisk still dont know each other |
| 18:40.59 | [TK]D-Fender | Show us what you have actually done along with the full debug from * CLI |
| 18:42.18 | alsaiduq | i've set the following parameters into sip.conf of each server |
| 18:42.32 | alsaiduq | [general] |
| 18:42.33 | alsaiduq | register => toronto:welcome@192.168.1.101/osaka |
| 18:42.33 | alsaiduq | [osaka] |
| 18:42.33 | alsaiduq | type=friend |
| 18:42.33 | alsaiduq | secret=welcome |
| 18:42.34 | alsaiduq | context=osaka_incoming |
| 18:42.37 | alsaiduq | host=dynamic |
| 18:42.39 | alsaiduq | disallow=all |
| 18:42.42 | alsaiduq | allow=ulaw |
| 18:42.44 | seanbright | !pastebin |
| 18:42.53 | seanbright | hmmm... what used to work? |
| 18:42.58 | [TK]D-Fender | ~pb |
| 18:42.58 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:42.59 | [TK]D-Fender | ^^^ |
| 18:43.00 | seanbright | yes |
| 18:43.00 | seanbright | that |
| 18:43.15 | [TK]D-Fender | alsaiduq: PASTEBIN. Do NOT flood in here |
| 18:43.21 | [TK]D-Fender | alsaiduq: Show us BOTH SIDES |
| 18:43.24 | alsaiduq | oh sorry, i'm still learning |
| 18:43.31 | alsaiduq | just a second |
| 18:47.36 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 18:53.27 | *** join/#asterisk zaf (~zaf@76.72.92.37) |
| 18:56.11 | lordvadr | I've finally managed to get a working t.38<->ISDN gateway going on el5 with asterisk 10. However, while the re-invite is successful in both directions, the two endpoints (asterisk and ATA) only exchange packets in one direction. If the ATA is sending the fax, it works. If the fax comes in from ISDN to asterisk, asterisk produces no T.38 frames after the re-invite. I have no idea where to start loo |
| 18:56.11 | lordvadr | king. |
| 19:03.02 | *** join/#asterisk dimitry7 (~dimitry7@gate.aaamerica.com.mx) |
| 19:03.17 | [TK]D-Fender | There is no re-invite |
| 19:03.21 | [TK]D-Fender | * is always in the way |
| 19:03.26 | [TK]D-Fender | This isn't SIP passthrough |
| 19:03.29 | [TK]D-Fender | You are TERMINATING |
| 19:03.38 | [TK]D-Fender | And * has no support for T.38 Origination AFAIK |
| 19:06.25 | *** join/#asterisk kritzikratzi (~kritzikra@80.120.104.90) |
| 19:08.35 | lordvadr | I didn't say anything about pass-through, yes, I'm wanting orig and term. Term works. There's a re-invite in the media stream to change from g.711 to t.38. This works. So no Origination? |
| 19:09.14 | [TK]D-Fender | "asterisk produces no T.38 frames after the re-invite" |
| 19:09.25 | [TK]D-Fender | There is no re-invite if you're the gateway |
| 19:09.57 | [TK]D-Fender | With there being no Orignation |
| 19:10.04 | lordvadr | I'm looking at a capture now. About 4 seconds in, the ATA sends an INVITE changing the media channel from 711 to t.38, which asterisk accepts. |
| 19:10.33 | lordvadr | But it turns into the t.38 equivalent of one-way-audio |
| 19:10.54 | lordvadr | However, in the termination direction, both the ATA and * exchange T.38 frames and the fax goes through. |
| 19:12.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 19:13.40 | lordvadr | From the Wiki: "The limitation of these versions of Asterisk is that while 1.6 - 1.8 are fully capable of passing through or terminating T.38 sessions, there is no capability for T.38 gateway... Asterisk 10 overcomes this limitation and provides full T.38 Fax Gateway support." |
| 19:14.54 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 19:16.01 | [TK]D-Fender | Are you enabling gateway on calls from the other direction? |
| 19:16.31 | [TK]D-Fender | I'd show that for sure, but... if there is any other kind of bug or limitation * 10 is dead |
| 19:17.30 | lordvadr | In the direction that it doesn't work, I've got faxdetect=incoming in chan_dahdi.conf, then in the dialplan, before sending it to the ATA, I set FAXOPT(gateway), and the ATA's sip peer definition also has it set. So it's set about everywhere I can think of. |
| 19:18.20 | lordvadr | In re: 10 being dead, yeah, I know. This is just an attempt to band-aid some bs my telephony guys decided to do. If it doesn't work, fine, but until my new * 13 boxes are up and running, this is worth taking a shot at. |
| 19:18.59 | BeachBall | :{ |
| 19:19.05 | BeachBall | found an oopsie |
| 19:19.19 | BeachBall | i had another app running on port 5060 that i didn't know about |
| 19:20.01 | *** join/#asterisk CeBe (~CeBe@p5488F7A6.dip0.t-ipconnect.de) |
| 19:22.47 | BeachBall | all fixed |
| 19:22.48 | BeachBall | :) |
| 19:23.03 | BeachBall | thank you defender |
| 19:23.04 | *** join/#asterisk Khronos (~Khronos@72-238-20-238.res.bhn.net) |
| 19:23.58 | Khronos | I have a couple of questions about imap voicemail storage. |
| 19:24.14 | Khronos | Do I need a different user created on the imap server for each extension? |
| 19:25.02 | Khronos | And if I have say 1000 extensions checking for voicemail every 30 seconds won't this put a lot of load on both the Asterisk and imap servers? |
| 19:25.59 | Khronos | I am trying to figure out the best way to deploy a setup where a user can login to either of two servers, but have the same voicemail tree show up on each. |
| 19:30.15 | [TK]D-Fender | Normally phones don't poll for *, they subscribe for it, which means * should eb throwing out notify's for it, and only as needed |
| 19:33.44 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:59.34 | Khronos | If I use imap for the voicemail module won't the module poll the imap server for new mails though? |
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| 20:02.52 | *** join/#asterisk twanny796 (~user@195.158.67.164) |
| 20:05.47 | drmessano | Khronos: no |
| 20:05.58 | drmessano | It's doesn't care about your email |
| 20:06.11 | drmessano | It cares about the messages it manages |
| 20:14.28 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 20:17.29 | *** join/#asterisk mjordan (mjordan@nat/digium/x-nzinincsmxwhebcv) |
| 20:17.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:20.37 | Khronos | drmessano: I mean won't the asterisk checking for voicemails every 30 seconds put a load in the imap server asking for mesages? |
| 20:20.53 | Khronos | drmessano: The imap server I'm thinking of setting up will only be for this process. |
| 20:21.17 | [TK]D-Fender | I'm thinking * doesn't check for VM... as it puts them th4ere. It should only refresh if it has a reason to |
| 20:21.28 | drmessano | ^ |
| 20:21.28 | *** join/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-ohsedjxkniejztrr) |
| 20:22.23 | Khronos | The reason why I'm thinking of using imap for this is so that I can have a way of doing shared voicemail over two servers. |
| 20:22.44 | Khronos | Instead of sharing a single spool for both. |
| 20:23.00 | drmessano | You keep explaining that |
| 20:23.07 | drmessano | Nobody is telling you not to |
| 20:23.14 | drmessano | The load won't be significatn |
| 20:23.29 | drmessano | Yes there will be "a load" everything has a CPU cost |
| 20:23.29 | Khronos | This way it prevents two messages with the same number being left on the box. |
| 20:23.40 | Khronos | I mean beling left i nthe shared spool. |
| 20:25.00 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 20:31.26 | Khronos | Mmm, here's something ealse I'm reading that might work better. |
| 20:31.38 | Khronos | Puting voicemail into mysql. |
| 20:31.40 | Khronos | putting |
| 20:32.45 | *** join/#asterisk cemotyz09 (~cesqueret@cpe-70-121-147-107.satx.res.rr.com) |
| 20:40.29 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:46.33 | *** join/#asterisk PeteSB (~PeteSB@cpc80931-thor6-2-0-cust711.14-2.cable.virginm.net) |
| 20:46.46 | PeteSB | Evening |
| 20:57.18 | PeteSB | Would you recommend always installing asterisk from srouce over a package like deb? |
| 20:58.36 | [TK]D-Fender | Well it's guarante4ed to be the version you choose |
| 20:58.53 | [TK]D-Fender | complied with the options you want |
| 20:58.57 | [TK]D-Fender | compiled |
| 21:02.04 | [TK]D-Fender | If you trust where the pacgaaes come from and that they are well maintained, then that is your choice |
| 21:03.45 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 21:09.48 | PeteSB | Yes I see what you mean |
| 21:10.02 | PeteSB | That is a bonus |
| 21:11.20 | [TK]D-Fender | Packageks make upgrading easier. It could also make them slower, or not with all the options you need |
| 21:11.37 | [TK]D-Fender | You'll have to jusge how you manage your server |
| 21:11.49 | [TK]D-Fender | is DAMN sloppy typing today. |
| 21:12.42 | PeteSB | heh |
| 21:13.21 | PeteSB | Does the config change much with the minor versions? |
| 21:13.36 | PeteSB | Is it easy enough to upgrade it? |
| 21:13.47 | PeteSB | It has been so long since I used asterisk |
| 21:14.32 | [TK]D-Fender | Within a branch only improvements |
| 21:14.49 | [TK]D-Fender | Real structure changes require upgrading the whole branch you're on |
| 21:14.50 | PeteSB | That is fair enough. |
| 21:15.46 | PeteSB | I might spin up a VM and just test with a packaged version first. Get a look at the config again |
| 21:15.55 | [TK]D-Fender | 99% safe upgrading from general config purposes within a branch. |
| 21:16.04 | [TK]D-Fender | Keep an eye onv13 though |
| 21:16.17 | [TK]D-Fender | Because they have been adding actual functionality in those sub-releases |
| 21:16.34 | [TK]D-Fender | Especially with PJSIP, and this is a good reason to keep up to date |
| 21:17.08 | PeteSB | That is the new SIP? Any good? |
| 21:24.03 | [TK]D-Fender | Yes, and improving. |
| 21:24.17 | [TK]D-Fender | Superior in several ways. Missing a thing or two I really want though |
| 21:24.32 | [TK]D-Fender | it is the future. Chan_sip is on the "no new devel" list |
| 21:25.55 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 21:30.04 | PeteSB | I take it the devices have to support it to connect using that protocol? |
| 21:31.35 | *** join/#asterisk Echo6_ (~Echo6@64.136.247.50) |
| 21:33.09 | [TK]D-Fender | It's still just SIP |
| 21:33.11 | [TK]D-Fender | SIP is SIP./ |
| 21:33.15 | PeteSB | OH right |
| 21:33.29 | [TK]D-Fender | It's just a question about the backing framework's expandability, etc |
| 21:33.31 | PeteSB | So my old cisco 7940s with work then |
| 21:33.34 | [TK]D-Fender | yes |
| 21:33.37 | PeteSB | I see |
| 21:33.47 | [TK]D-Fender | It works differently on the backend giving you more options |
| 21:33.50 | [TK]D-Fender | go read up |
| 21:33.56 | [TK]D-Fender | packs up to head home |
| 21:34.01 | PeteSB | I will have too. |
| 21:34.27 | PeteSB | Are the frontends like freepbx anygood? |
| 21:36.35 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
| 21:39.18 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
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| 22:04.04 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 22:07.12 | drmessano | PeteSB: Yep, FreePBX is pretty awesome |
| 22:09.01 | PeteSB | drmessano, I am just playing with it running in a VM. It's impressive. I did try to do a manual install alot of modules do not work unless you use the distro |
| 22:09.14 | *** join/#asterisk gu1lle_ (~Thunderbi@181.167.195.114) |
| 22:09.25 | drmessano | There are some commerical modules that require the distro |
| 22:09.33 | drmessano | They are enhancements |
| 22:09.51 | drmessano | Far from necessary to build and run a fully functional, feature packed PBX |
| 22:11.57 | gu1lle_ | Hi, im looking for a proxy-outbound-server app, due the restriction on my isp voip connections, may someone hav e any recommendation ?, Thanks |
| 22:12.49 | drmessano | Explain what you're trying to do |
| 22:12.54 | gu1lle_ | me ? |
| 22:12.57 | drmessano | Yes |
| 22:13.12 | drmessano | a "proxy outbound server app" means nothing |
| 22:13.31 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:15.24 | PeteSB | drmessano, that is good to know. I might have to revisit my install. |
| 22:16.10 | gu1lle_ | I use VPN for VoIP, I always have the same issue ... whenever I connect to the VPN the internet speed throttled from 8 mb/s to 1 mb/s |
| 22:16.42 | drmessano | gu1lle_: ok, and? |
| 22:16.43 | gu1lle_ | the only one I found stable and did not make the internet speed slower is www.hide.me service |
| 22:17.46 | drmessano | You are better off asking in a channel or forum where VPN is more appropriate. |
| 22:17.56 | drmessano | Since Asterisk isn't really your issue |
| 22:18.12 | *** part/#asterisk gu1lle_ (~Thunderbi@181.167.195.114) |
| 22:18.37 | PeteSB | lol |
| 22:18.47 | drmessano | Also, the part where VoIP is illegal in his country |
| 22:18.49 | PeteSB | not even a thank you |
| 22:18.51 | drmessano | and I dont do that |
| 22:19.18 | drmessano | Nope.. but not suprising |
| 22:19.50 | drmessano | You get a lot of folks basically asking the same. They are in Dubai or Vietnam or something |
| 22:19.54 | drmessano | Want to start an ISP |
| 22:19.57 | drmessano | Want to start an ITSP |
| 22:20.06 | drmessano | or a calling card business |
| 22:20.06 | PeteSB | Ahh |
| 22:20.10 | PeteSB | Really |
| 22:20.37 | drmessano | So they need to VPN their PBX to a GSM channel bank or a wholesaler outside the country |
| 22:20.39 | PeteSB | It is sad to think that some of those really annoying calls I get could be powered by Asterisk :( |
| 22:20.39 | drmessano | Yep |
| 22:21.11 | PeteSB | Ahh well. |
| 22:21.33 | PeteSB | It has be around 10 years at least since I have used asterisk |
| 22:21.39 | *** join/#asterisk F2Knight (~F2Knight@173-8-193-173-Oregon.hfc.comcastbusiness.net) |
| 22:21.44 | PeteSB | I thought it was about time I used it again |
| 22:21.51 | PeteSB | Plus I need a pbx |
| 22:22.13 | PeteSB | I don't remeber any of the config, lol I wonder if it will come back to me. |
| 22:23.27 | drmessano | With FreePBX you won't be touching config, generally.. You CAN.. there are MANY ways to basically add and make tweaks, but FreePBX is designed to do it all |
| 22:23.56 | PeteSB | Yes, I see that. I am going to do it by hand for this one I think. |
| 22:23.57 | drmessano | But I guess that's the beauty of Asterisk.. options |
| 22:24.07 | PeteSB | I migth play with freepbx later. |
| 22:24.25 | PeteSB | It would be good for clients. eventually. |
| 22:24.34 | PeteSB | If I get that far, lol |
| 22:25.13 | PeteSB | I assume freepbx needs to be added to a vanilla system and cannot be added to a system thas has already been configure |
| 22:26.45 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 22:28.38 | *** join/#asterisk klow (~klong@c-24-19-57-176.hsd1.wa.comcast.net) |
| 22:29.35 | klow | Newb question - when compiling asterisk 13 (debian) .. needing TLS / PJSIP support - does one download from pjpsip.org and configure/make install that first, then compile asterisk, or can pjsip module be added after the compile |
| 22:30.05 | klow | I was surprised to not find the pjsip modules in my asterisk build after compiling - its a separate project I see? |
| 22:31.30 | *** join/#asterisk Subhi (~subhi@de.subhi.com) |
| 22:31.33 | Subhi | Hi All, |
| 22:32.05 | Subhi | All the sites I see online regarding asterisk on debian talk about compiling it, I am wondering, is there anything wrong with the debian packages? |
| 22:32.12 | Subhi | IS there something missing ? |
| 22:33.32 | drmessano | PeteSB: Correct.. because it takes over the config |
| 22:33.35 | [TK]D-Fender | All of the obvious things |
| 22:33.41 | drmessano | PeteSB: Writes its own |
| 22:33.50 | [TK]D-Fender | Is debian GOOD at maintaining their packages? |
| 22:33.59 | PeteSB | drmessano, good to know. |
| 22:34.01 | [TK]D-Fender | Do they leave anything out? |
| 22:34.43 | drmessano | klow: You need PJSIP first.. Asterisk cant compile a module for something that doesnt yet exist |
| 22:34.45 | PeteSB | I just installed the package on unbuntu server 15.10. The asterisk version is 13.1 |
| 22:35.09 | drmessano | 13.1 is kinda old |
| 22:35.18 | drmessano | I never use packages for Asterisk |
| 22:36.11 | PeteSB | Yea |
| 22:36.39 | Subhi | [TK]D-Fender: What's the problem with them ? (I am genuinely asking this question |
| 22:36.42 | PeteSB | hmm I might just remove it and compile it |
| 22:36.50 | PeteSB | I dunno, lol |
| 22:36.52 | [TK]D-Fender | I'm not saying there IS a problem. |
| 22:37.02 | PeteSB | The packages? |
| 22:37.24 | [TK]D-Fender | If you're interested, go look at what the forums say about them, what versions are available. If they update often enough for your comfort |
| 22:37.36 | PeteSB | If it is debian they are usually quite old. They take an age to update anything |
| 22:37.52 | PeteSB | Unbuntu is faster |
| 22:37.52 | klow | dangit. i compiled/installed project then ./configure asterisk and went to 'make' and "/usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC" |
| 22:38.14 | PeteSB | I guess it depends on what you want as you say |
| 22:40.50 | Subhi | wonder if I should go with the lazy option with packages or compile it from scratch |
| 22:41.06 | Subhi | I only want basic SIP trunk and voicemail |
| 22:41.10 | Subhi | nothing clever |
| 22:41.37 | drmessano | It isnt about "clever" |
| 22:41.42 | drmessano | Its about patch levels |
| 22:42.09 | drmessano | If you use 13.1 and they are only adding security fixes.. and a bug is present that makes you life hell, guess what? |
| 22:42.31 | drmessano | Chances are, that's what Debian is doing |
| 22:43.18 | drmessano | That goes with ANY packaged app.. Thats not specific to Asterisk |
| 22:43.42 | drmessano | However, Asterisk is a very active project |
| 22:44.09 | drmessano | So there are a lot of bug fixes and improvements are you move along |
| 22:48.21 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 22:48.57 | Subhi | I see |
| 22:49.01 | Subhi | something for me to think about |
| 22:53.39 | PeteSB | Yea |
| 22:53.47 | PeteSB | I am not sure they even add the sec fixes |
| 22:53.49 | PeteSB | I dont know |
| 22:54.05 | PeteSB | Is it easy enough to update the compiled version? |
| 22:56.18 | [TK]D-Fender | recompile, install |
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| 23:01.17 | *** join/#asterisk italorossi (~Adium@179.156.47.253) |
| 23:01.39 | drmessano | Debian seems to be adding the patches |
| 23:02.00 | PeteSB | thats good |
| 23:02.09 | PeteSB | I wonder if unbuntu do the same |
| 23:02.40 | PeteSB | Debian has always been good at keeping up with the sec updates, even if the packages are a little old |
| 23:02.53 | drmessano | Ubuntu would be downstream from Debian |
| 23:03.53 | drmessano | So they have the same package |
| 23:06.16 | PeteSB | That should not be too bad then. |
| 23:06.29 | PeteSB | How often are the patches released |
| 23:11.25 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
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| 23:27.52 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 23:32.14 | drmessano | As needed |
| 23:32.26 | drmessano | Updates is more my concern |
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| 23:41.36 | PeteSB | How often are the updates released? |
| 23:57.50 | drmessano | 13.1 is like a year behind |
| 23:58.20 | PeteSB | I am just compiling the new version... takes an age |
| 23:58.39 | PeteSB | Just going to give freepbx a play as well |
