| 00:01.38 | ChannelZ | as in calling in to Asterisk and having it dial back out somewhere else with no authentication? sure.... |
| 00:06.09 | [TK]D-Fender | Asterisk does whatever you configure it to do |
| 00:07.01 | limpc | k wanted to make sure before i dived back into it |
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| 00:07.47 | limpc | digium has sure changed alot since i last saw it -- last time I was in the asterisk field, all they sold were 1-4 port digium cards |
| 00:08.06 | limpc | now they've got switches, analog cards, etc |
| 00:09.20 | jpsharp | Ask not what Asterisk can do for you, ask what you cannot do with Asterisk. It's a much shorter list. |
| 00:10.00 | jpsharp | hears JFK rolling over in his grave. |
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| 00:19.38 | ectospasm | Asterisk sadly doesn't keep me warm at night. Yet. |
| 00:19.49 | jpsharp | You're just not running it on a big enough server. |
| 00:20.07 | ectospasm | true enough, that machine is in another bedroom |
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| 00:28.00 | ChannelZ | nothing some clothes dryer conduit won't fix |
| 00:29.08 | limpc | how does the digium 8-span t1/e1/j1/pri card work? I only see 4 ports on it. do you use a cable for the other 4? |
| 00:32.44 | jpsharp | You use an adapter dongle |
| 00:33.32 | jpsharp | Each port has two spans on it, but you use the included adapter dongle to get to the second port. |
| 00:33.38 | jpsharp | second span rather. |
| 00:34.09 | limpc | i see. |
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| 00:41.14 | limpc | how many cpu cores do you think i should get in order to handle 600 lines simultaneously? |
| 00:41.36 | jpsharp | A lot. |
| 00:41.54 | limpc | like 2x6 cores or 4x6 cores? |
| 00:41.58 | jpsharp | But it really depends on whatelse you want to do. ARe you transcoding? Are you recording? Or just passing audio? |
| 00:42.14 | limpc | just passing audio. its a pinless calling card solution |
| 00:42.32 | limpc | probably do some transcoding |
| 00:43.02 | jpsharp | If you're goint to do any transcoding, you'll not likely be able to squeeze it into one box. |
| 00:46.00 | limpc | so a user calls into their account, places a call, im getting g711 compressed data already right? |
| 00:47.23 | jpsharp | g711 isn't compressed. |
| 00:49.57 | jpsharp | If you're going from a T1 to G711 voip, there's no transcoding. |
| 00:51.48 | limpc | hmm right, i think i was thinking of g726 or g729, its been a long while sorry :) |
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| 01:06.19 | limpc | so calls received from a teleco will be g726? |
| 01:06.30 | limpc | also, does asterisk support db's like mongo? |
| 01:06.39 | limpc | or do i need to develop my own extension for it |
| 01:07.44 | jpsharp | If it comes in via a T1, it is 8-bit 8Khz ulaw encoded audio. If it comes in via a voip ITSP, then its is anyones guess and only they will be able to tell you. |
| 01:08.08 | limpc | it'll be via T1 |
| 01:08.21 | jpsharp | Then it'll basically be G711. |
| 01:08.57 | limpc | so i should keep it g711 outbound as well (it'll be terminated back to another T1 channel) |
| 01:09.15 | jpsharp | Yep, if you've got the bandwidth between servers. |
| 01:09.52 | jpsharp | And as far as databases go, asterisk supports mysql natively and then numerous databases via ODBC. |
| 01:10.35 | limpc | i dont think theres an odbc driver for mongo. |
| 01:10.43 | limpc | its basically a nosql db |
| 01:12.03 | jpsharp | then you'll be writing your own driver. |
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| 02:03.06 | NerdNation5000 | hello - i was wondering in the asterisk console - if there's a command that will show stats on my asterisk box - i'm looking to see the total # of inbound calls that have been processed |
| 02:09.34 | acidfoo | NerdNation5000, if you do 'core show channels' |
| 02:09.45 | acidfoo | you'll see the number of calls processed |
| 02:10.11 | NerdNation5000 | love it! |
| 02:10.13 | NerdNation5000 | thanks so much |
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| 03:10.34 | limpc | hrm when a site says "international termination rates", it means its terminating at a US location and charging an international rate yes? or is it meaning it'll terminate it overseas? |
| 03:11.22 | limpc | nm. durr moment |
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| 05:40.01 | hebber | Hi, has anyone experienced troubles with MOH in queues - even with a fully working Music On Hold. |
| 05:42.09 | hebber | Using Asterisk 1.8 the CLI says MOH in queue is playing - but its silent. If the function MusicOnHold is called outside a queue is called MOH is always playing nicely. |
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| 06:41.46 | enki | has anyone played around with google's speech to text engine for voicemail transcription? |
| 06:42.43 | enki | (specifically with asterisk voicemails) |
| 06:48.29 | Tim_Toady | enki: thers a problem with the duration of the speech google accepts, its about 20-30 sec so you have somehow to find a way to chop the voicemail in smaller chunks if they are bigger than 20 sec |
| 06:49.31 | enki | yeah, I've been using 'head -c 250k' |
| 06:49.58 | enki | it gives me about 10-15 seconds worth of audio, which is enough to get the gist of the message |
| 06:50.19 | enki | i'm trying to figure out the best approach to get that put in the email asterisk sends out |
| 06:50.34 | enki | this: http://www.pbxinaflash.com/community/index.php?threads/howto-transcribe-asterisk-vm.4886/ seems to be a good template |
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| 07:16.50 | enki | woohoo, I got it working |
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| 07:30.59 | enki | it works about as good as google voice's translate |
| 08:12.21 | ChannelZ | make me |
| 08:12.26 | ChannelZ | oops |
| 08:15.27 | Tim_Toady | enki: how u got there? based on pbxiaf howto? |
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| 09:23.33 | hebber | Found a solution on the above - just need to have some playback before entering the queue - then it will work |
| 09:24.54 | sebastianpersic | Do anyone know if there's any how to guide to install GUI for asterisk 1.8 |
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| 10:31.58 | sebastianpersic | Do anyone know if there's any how to guide to install GUI for asterisk 1.8 |
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| 10:38.57 | Igneous | sebastianpersic: which GUI were you looking to use, specifically? |
| 10:47.41 | sebastianpersic | I thought about the asteriskNOW? gui, or are there any other which works better? |
| 10:59.14 | Igneous | the majority of those projects just use FreePBX |
| 10:59.26 | Igneous | I'd suggest finding a freepbx installation guide and just following that |
| 10:59.36 | Igneous | (if you already have an existing asterisk installation, that is) |
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| 11:03.06 | sebastianpersic | I've installed asterisk yes |
| 11:11.14 | sebastianpersic | #asterisk +o file |
| 11:11.19 | sebastianpersic | what's that |
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| 11:28.32 | fulcan | is there a feature on 'any' voip phone that will send an 'end call timestamp for billing' to a/the server at the end of the call even if the rtp stream has been release from the server (aka a switch call)? All voip, no hardwire. |
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| 11:47.07 | WIMPy | The audio stram and the signalling are two different things, yes. |
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| 14:12.39 | sebastianpersic | Mode #asterisk +o file by ChanServ |
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| 14:38.14 | dijib | ok ive scapped the v10 upgrade because the LTS support. anybody know of a guide or the steps and stages of setting up digium_fax? |
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| 15:04.10 | Molodus | i think there was a guide on digium's site that i used |
| 15:07.01 | Molodus | http://www.digium.com/en/docs/FAX/fax_faq.php |
| 15:07.28 | Molodus | there's also a documentation link on the right hand side that you can use to look up fax related docs |
| 15:07.52 | Molodus | think you just have to register and log in |
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| 15:38.24 | enki | Tim_Toady, yeah I used that as a blueprint for how asterisk would integrate with it, but then wrote my own transcription script based on the google pseudo-api. |
| 15:43.17 | Tim_Toady | enki: :) |
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| 15:59.26 | enki | When I was running 1.4, if I had a new VM and either deleted or moved it to 'old' through some outside mechanism (ie: web gui), Asterisk would update subscribing phones that there was no longer any new VM and the MWI would go off. However, in 1.8, that doesn't seem to be the case anymore. Asterisk is aware that there aren't any new VMs, but it doesn't push that out to the extension(s). |
| 15:59.41 | enki | any idea how to re-enable that feature (if possible)? |
| 16:00.36 | WIMPy | Can't remember the name, but there's a parameter to enable polling the mailboxes every ... seconds. |
| 16:01.50 | enki | WIMPy, thanks, found it. "pollmailboxes=yes" |
| 16:02.04 | enki | was disabled by default. enabled it,and that seems to work. |
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| 16:32.52 | kresp0 | Hi all! I have a problem with DUNDi, again. I have 3 hosts on the same VPN. lookup A from B works. lookup A from C dont. Here is the DUNDI debug on host A: http://pastebin.com/bg0B9tVj |
| 16:33.15 | kresp0 | show keys says the public keys are LOADED |
| 16:33.30 | kresp0 | and dundi show peers says that all peers are connected OK |
| 16:34.34 | kresp0 | here is the DUNDi debug on host A, when it works: http://pastebin.com/GcgTP4H3 |
| 16:37.23 | WIMPy | Have you compared the checksums with 'keys show'? |
| 16:39.09 | kresp0 | nop, let me see.. |
| 16:39.44 | kresp0 | WIMPy: they have the same sum |
| 16:41.08 | WIMPy | Those pastebins don;t seem to exist. |
| 16:41.18 | kresp0 | upss sorry |
| 16:41.25 | kresp0 | again, |
| 16:42.11 | kresp0 | the first one: http://pastebin.com/uXvsHx5f |
| 16:42.38 | kresp0 | the other one: http://pastebin.com/wECwVs7D |
| 16:43.39 | kresp0 | they are the dundi debug on host A |
| 16:43.50 | WIMPy | ENCREJ looks like the encryption is failing. Do you have the right keys in all peers? |
| 16:44.20 | kresp0 | "keys show" says it |
| 16:44.32 | kresp0 | on all peers |
| 16:44.48 | WIMPy | The peer configurations. |
| 16:45.15 | kresp0 | let me check it |
| 16:45.21 | WIMPy | Each peer in your dundi.conf on the hosts. |
| 16:49.00 | kresp0 | thank you again, WIMPy. I have bad entityid on host A dundi.conf, in peer config for host C |
| 16:49.13 | kresp0 | that was the problem :) |
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| 17:04.30 | mjordan | enki: pollmailboxes=yes, with some pollfreq set |
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| 17:15.33 | enki | i'm curious why it's disabled by default. It seems, to me, like most people would want that enabled. |
| 17:20.10 | WIMPy | It only wastes time if the mailboxes aren't manipulated by other software. |
| 17:29.08 | enki | ah ok. I am really liking the "answered elsewhere" flag on canceled calls. The lack of that was always a PITA on 1.4. |
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| 17:30.55 | WIMPy | yes |
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| 18:34.06 | fulcan | WIMPy you still around my friend? |
| 18:34.54 | WIMPy | yes |
| 18:37.08 | fulcan | You had responded to my question early. what is the technology that will send an end call time/date stamp from the phone? I thought it was 'can reinvite' but I am not finding things I found years ago when presented with this same question. :/ |
| 18:37.59 | WIMPy | For SIP it's a BYE. |
| 18:38.33 | fulcan | WIMPy no kidding. never thought of that. |
| 18:40.09 | fulcan | wait, is there a BYE to the server if the rtp stream has simply been switched? |
| 18:41.23 | WIMPy | BYE comes when the connection ends. What do you mean with "has switched"? |
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| 18:42.02 | fulcan | no longer occupying processor power from the asterisk cpu. |
| 18:42.59 | WIMPy | If Asterisk sets up directmedia, it will know. |
| 18:43.07 | WIMPy | What exactely are you looking for? |
| 18:43.14 | fulcan | but the call still exist between the 2 two enduser devices. |
| 18:43.28 | WIMPy | Sure |
| 18:43.54 | WIMPy | And the BYW at the end will still go to your server. |
| 18:44.07 | WIMPy | BYE |
| 18:45.50 | fulcan | if you switch a call "like SER' you are releasing the rtp stream. You say 'you go there' and point. But when you route (like a router) the connection is maintained at the router processor till the connection ends. I am address switched VoIP. nothing Zaptel. |
| 18:46.33 | WIMPy | SIP and RTP ar two different things. |
| 18:46.40 | fulcan | I know they are |
| 18:46.49 | fulcan | but they work hand in hand |
| 18:47.03 | fulcan | sip is the authentication and rtp is the audio data. |
| 18:47.24 | WIMPy | SIP is everything signalling. |
| 18:47.29 | fulcan | release rtp and still get SIP BYE. I need the howto. :) |
| 18:47.51 | fulcan | WIMPy binary signaling not analog. |
| 18:47.56 | WIMPy | I still dont know what you're looking for. |
| 18:47.57 | fulcan | its a packet. |
| 18:48.25 | fulcan | signalling is layer 1 on an ip network. :P |
| 18:48.26 | WIMPy | Analog doen't have much signalling and is evil. |
| 18:48.42 | fulcan | I am not addressing either! :) |
| 18:48.43 | WIMPy | Err, no. |
| 18:48.51 | fulcan | rtp and sip. |
| 18:49.10 | WIMPy | What are you looking for? |
| 18:49.48 | fulcan | 10,000 simulatainious on a single box. |
| 18:50.12 | WIMPy | What is your question? |
| 18:50.16 | fulcan | if they are switched, it isn't an issue. but I need the last packet BYE. |
| 18:50.29 | WIMPy | You will always get it. |
| 18:50.37 | WIMPy | Unless the peer just vanishes. |
| 18:56.41 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-11-238.bois.qwest.net) |
| 19:00.37 | fulcan | so there are no issues with getting the BYE packet if using 'can reinvite = no'? |
| 19:01.22 | WIMPy | You mean "yes"? |
| 19:01.30 | fulcan | I mean no. |
| 19:01.32 | WIMPy | And that's called directmedia now. |
| 19:01.37 | fulcan | get that rtp away! |
| 19:01.49 | fulcan | can reinvite = no |
| 19:02.13 | WIMPy | That meas no reinvite means RTP will stay at Asterisk. |
| 19:02.15 | fulcan | and still get BYE |
| 19:02.25 | fulcan | I have it backwards? |
| 19:02.30 | fulcan | o |
| 19:02.35 | fulcan | :) |
| 19:02.49 | WIMPy | And if you set up directmedia that will only set up a direct RTP link. |
| 19:03.01 | *** join/#asterisk aaronjaque (~aaron@76.9.63.65) |
| 19:03.02 | WIMPy | SIP stays where it is. |
| 19:03.14 | aaronjaque | hi |
| 19:03.26 | fulcan | and you with still get the BYE with no issues? |
| 19:03.27 | WIMPy | Directmedia only means P2P RTP. |
| 19:03.42 | WIMPy | If you want P2P SIP, that's a ENUM or DUNDi thing. |
| 19:04.33 | fulcan | WIMPy ok. now I see. |
| 19:05.02 | fulcan | thank you so much for clearing that up. |
| 19:05.06 | aaronjq | I am trying to register new polycom phones on an asterisk system but I'm getting a password error |
| 19:05.53 | fulcan | aaronjq polycom are nice phones. Did you check your password? |
| 19:06.02 | aaronjq | [Aug 12 13:05:31] NOTICE[2744]: chan_sip.c:15593 handle_request_register: Registration from '<sip:28@172.19.34.30>' failed for '172.19.34.206' - Wrong password |
| 19:06.02 | fulcan | :) |
| 19:06.39 | aaronjq | hah, funny thing is I'm not sure where to set the password that is causing the error |
| 19:06.48 | fulcan | aaronjq that is typically broke in one of two places. your phone or your sip.conf |
| 19:06.48 | WIMPy | Don't use numeric usernames if you're connected to the internet. |
| 19:07.36 | aaronjq | I'm a newb so forgive any dumb questions :) |
| 19:10.50 | aaronjq | the phones are IP330. I have the admin password for the phone (it is default), and I have root access for the linux server, but neither seem to help me with this error |
| 19:11.42 | WIMPy | Make sure username and password are set identically on both ends. |
| 19:12.03 | WIMPy | ... and that that user is allowed to connect fro its address. |
| 19:14.17 | aaronjq | I'm not sure where the username on the asterisk side is set...I mean in what config file |
| 19:14.40 | drmessano | 15:06] <fulcan> aaronjq that is typically broke in one of two places. your phone or your sip.conf |
| 19:21.19 | aaronjq | my sip.conf has the following: |
| 19:21.22 | aaronjq | [phone](!) |
| 19:21.23 | aaronjq | type = friend |
| 19:21.23 | aaronjq | host = dynamic |
| 19:21.23 | aaronjq | secret = 456 |
| 19:21.23 | aaronjq | context = users |
| 19:21.23 | aaronjq | setvar=UserID=UniqueID |
| 19:21.23 | aaronjq | canreinvite=yes |
| 19:21.39 | aaronjq | so secret is my password for registration I believe? |
| 19:24.40 | WIMPy | For all users using the phone template. That doesn't look very sensible. |
| 19:24.56 | WIMPy | For all users using the "[phone]" template. That doesn't look very sensible. |
| 19:26.14 | WIMPy | Oh, and next time, please |
| 19:26.16 | WIMPy | ~pb |
| 19:26.16 | 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. |
| 19:27.01 | aaronjq | hah, I'm not too surprised...this system was built a while ago and the administrator is long since gone...I'm trying to pick up the pieces... |
| 19:27.33 | aaronjq | sorry for the 'paste'...will look at the links |
| 19:30.14 | *** join/#asterisk trumee (~parul@188-222-165-248.zone13.bethere.co.uk) |
| 19:31.30 | trumee | i have asterisk built with srtp, and 'module show' does show res_srtp.so. However, 'module load res_srtp.so' says - Command 'module load res_srtp.so' failed |
| 19:31.44 | trumee | any idea why this might be so? |
| 19:34.32 | trumee | ah, maybe srtp is already present. in the log i see 'loader.c: Module 'res_srtp.so' already exists' |
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| 19:46.40 | lorsungcu | anyone have trouble with iPhones and dtmf? |
| 19:47.29 | lorsungcu | or does anyone have an iPhone they can test some dial plan with? |
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| 20:22.39 | aaronjq | exit |
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| 20:43.17 | lorsungcu | Qwell: you around? |
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| 21:14.26 | iulhk | hi , can i use goto command with dial option ? |
| 21:15.30 | iulhk | i am receiving DID calls at default context, now i want to send this call to some other context ? how is this possible ? |
| 21:33.42 | WIMPy | You can goto any place you like, but you sholuld configure your channel to send the calls to the right context. |
| 21:44.06 | iulhk | <WIMPy> i have context test1 where all sip to sip calls working fine, i hv another context test2 where all DIDs mapped , i hv written a simple agi where calls come to DID i can dial any sip number when user hits on DID, but i want to send these DIDs call to test1 ? |
| 21:45.56 | WIMPy | It's a little hard to understand, but I think you want to include test1 in to test2. |
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| 21:59.30 | General_z0d | how do i get flite app to work on latest astersisk |
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