| 00:02.34 | *** join/#asterisk BCrookAtRA (~Adium@63.226.136.253) |
| 00:02.56 | *** part/#asterisk ibercom (5320c26f@gateway/web/freenode/ip.83.32.194.111) |
| 00:14.10 | *** join/#asterisk newtonr (~RustyNewt@173-21-90-203.client.mchsi.com) |
| 00:14.10 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 00:16.50 | d1gital | Is there a Dial(,,U()) equivalent to execute a sub for the /calling/ channel, when the called channel answers? |
| 00:20.48 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 00:25.24 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 00:27.27 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-mszfwcbvnqbdvipq) |
| 00:28.19 | newtonr | d1gital, I think.. one second |
| 00:30.24 | newtonr | Oh, hmm, maybe not |
| 00:32.03 | d1gital | hmm... maybe I can use some combination of the G() option and Bridge() ? |
| 00:33.35 | newtonr | d1gital, yeah.. that would be weird, but may work. |
| 00:34.20 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 00:35.36 | d1gital | presumably bridging the calls later is the intended use-case for G()? how would I go about getting the appropriate channel string to pass into Bridge()? |
| 00:37.37 | newtonr | probably https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL |
| 00:38.22 | newtonr | and i've got to run! good luck! |
| 00:48.34 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 00:50.10 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 00:58.56 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 01:00.20 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:04.13 | *** join/#asterisk BCrookAtRA (~Adium@63.226.136.253) |
| 01:15.37 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 01:35.09 | AnonGirl | Is 11c1:0440 compatible with dahdi? |
| 01:38.38 | *** join/#asterisk w3rt (~jasonwert@71.89.137.28) |
| 01:41.29 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:06.41 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:17.13 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 02:17.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:33.29 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:48.39 | babak_ | hi, is it possible to use originate command in media bypass mode? I want Asterisk dials and connect two remote sip account just by signaling. |
| 02:51.30 | mjordan | babak_: The determination of how two channels are bridged is made when both channels go into the bridge. |
| 02:52.13 | mjordan | babak_: So yes, if you have two RTP capable channels, and both allow for direct media, then the media will be negotiated away from Asterisk when they are both in a bridge together, regardless of how the channels were created. |
| 02:57.08 | babak_ | mjordan: thx, is there any channel variable setting for media bypass for times we do not want globally set it ? |
| 02:58.02 | mjordan | no. A peer or endpoint either supports direct media, or it doesn't. Different options with direct media help to inform Asterisk whether or not it can actually send media to that peer or endpoint. |
| 02:58.41 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:58.47 | babak_ | thank you |
| 03:15.18 | *** join/#asterisk D30 (~D30@222.127.13.226) |
| 03:17.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:18.59 | *** join/#asterisk CrashHD (~na@104.220.5.4) |
| 03:25.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:25.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:27.01 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 03:29.00 | talntid | What is the current best way to, externally query Asterisk for like... queue information, or if someone is currently on a phone call, or things like that? |
| 03:29.17 | talntid | AMI seems to do some things, but not really all |
| 03:30.09 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 03:31.41 | mjordan | talntid: the concept of "someone being on a phone call" doesn't really map well to Asterisk. There aren't "someones", nor is there really a definition of a call. |
| 03:31.56 | mjordan | As for someone being in a Queue, there's lot of AMI commands to inspect the state of queues provided by app_queue. |
| 03:31.57 | *** join/#asterisk CrashHD (~na@104.220.5.4) |
| 03:32.01 | talntid | yeah, I'm finding this.. |
| 03:32.07 | mjordan | beyond that, you'll have to be a bit more specific about what you're looking for. |
| 03:32.16 | talntid | yeah, seeing if someone is in a queue is easy |
| 03:32.26 | talntid | but knowing if they are on a call seems to not be |
| 03:32.57 | talntid | and, if they are on a call, who are they on a call with? |
| 03:33.24 | mjordan | If you are trying to build out something that shows call state, I'd highly suggest using Asterisk 13. |
| 03:33.33 | mjordan | That answer is a bit more complex in Asterisk 11. |
| 03:33.49 | mjordan | But a 'call' is whatever you define it as. |
| 03:34.02 | talntid | I'm fine with using Asterisk 13. I'm currently on it's documentation, because I was having a lot of trouble doing this in 11 |
| 03:34.08 | mjordan | Asterisk is based around 'channels' and 'bridges'. |
| 03:34.29 | talntid | right. can you explain what those are? |
| 03:34.32 | mjordan | You can determine what channels have the capability to talk to other channels based on their presence in the same bridges. |
| 03:34.48 | mjordan | A channel is a path of communication from Asterisk to a device, or between Asterisk and itself. |
| 03:34.56 | talntid | ok |
| 03:35.15 | mjordan | A bridge shares media between one or more channels, where the sharing of media is based on its mixing type |
| 03:35.27 | mjordan | goes to get an example |
| 03:35.47 | mjordan | https://wiki.asterisk.org/wiki/display/AST/AMI+v2+Specification#AMIv2Specification-Bridging |
| 03:36.24 | mjordan | really, most of the examples there will help you a lot with the lifetime and model of channels/bridges |
| 03:36.45 | mjordan | keep in mind that the spec there only applies to 12+. In 11, things were a lot more complicated. |
| 03:37.09 | talntid | roger that |
| 03:37.16 | talntid | basically, what I'm trying to do is this: |
| 03:37.48 | talntid | have many queues. show a webpage with the queues, who's in them, and when someone is on a call, show that they are on a call, and who they are speaking with, and what queue the call came in on |
| 03:38.32 | talntid | I'm down to parsing the queue_log right now and watching the AMI over sockets...... and it mostly works, just... sucks. seems like there shoulda been an easier way. |
| 03:38.47 | mjordan | Generally, you would use AMI for most of that. |
| 03:39.04 | mjordan | Keep in mind that even the concept of a "Queue" is based on channels/bridges and other stuff. |
| 03:39.04 | talntid | and not being able to know if, for example... SIP/PolycomXXX is on hook, off hook, etc |
| 03:39.19 | mjordan | That doesn't really make much sense in a SIP world |
| 03:39.24 | talntid | yeah |
| 03:39.34 | mjordan | phones don't typically send a SIP request when someone picks up a handset. |
| 03:40.11 | talntid | right, but in asterisk, when a phone is ringing, versus when they pick up.. seems asterisk knows they picked up |
| 03:40.17 | talntid | because it then says bridging channels and such |
| 03:43.58 | [TK]D-Fender | Yes, basic presence cal already track that |
| 03:44.12 | [TK]D-Fender | based on channel states using the device you define |
| 03:44.20 | [TK]D-Fender | can* |
| 03:45.15 | [TK]D-Fender | <talntid> AMI seems to do some things, but not really all <- can get all of this from AMI |
| 03:46.23 | [TK]D-Fender | I have a MicroBrowser script running for my Polycom & Aastra phones tracking the presences of our 4 CSRs, and a dump of 2 queues with agent login status, on call, etc |
| 03:57.30 | talntid | Hmm |
| 03:58.09 | talntid | using the HINTS? |
| 03:58.42 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 03:59.29 | [TK]D-Fender | yes |
| 03:59.45 | [TK]D-Fender | For that half. Queue dumps for the rest |
| 04:14.04 | *** join/#asterisk danjenkins (~anonymous@24.96.140.21) |
| 04:28.57 | *** join/#asterisk BCrookAtRA (~Adium@63.226.136.253) |
| 04:51.42 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 04:54.39 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:39.31 | *** join/#asterisk gryphon_ (~gryphon@82.140.120.164) |
| 05:54.41 | *** join/#asterisk tzafrir (~tzafrir@24.57-65-87.adsl-dyn.isp.belgacom.be) |
| 06:06.31 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 06:33.19 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 07:26.26 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 07:29.11 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:30.21 | *** join/#asterisk ayrjola (~ayrjola@89.18.237.137) |
| 07:39.18 | *** join/#asterisk tzafrir (~tzafrir@78.129.111.57) |
| 07:47.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 07:49.23 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:18c1:8acf:1f50:1e17) |
| 07:56.15 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 07:57.33 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 08:12.12 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 08:15.51 | *** join/#asterisk evil_gordita (robert@ip70-188-56-139.rn.hr.cox.net) |
| 08:16.25 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 08:23.16 | *** join/#asterisk mlhess_ (~digiv@drupal.org/user/102818/view) |
| 08:35.36 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 08:45.57 | *** join/#asterisk madax (~madax@c42-156.i07-11.onvol.net) |
| 08:55.36 | *** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 08:56.54 | *** join/#asterisk jhlavacek (~jirka@avencall-gw-vbo.alionis.net) |
| 09:03.30 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9bd:6377:5954:5edb) |
| 09:14.00 | *** join/#asterisk XATRIX (~xatrix@77.88.209.171) |
| 09:14.23 | XATRIX | Hi guys, any idea why i can't call to my mobile handset via 'dongle' and play audio files ? |
| 09:14.25 | XATRIX | http://fpaste.org/178238/09223142/ |
| 09:14.49 | XATRIX | i'm trying to use a callscript.sh to create a callfile in a spool directory |
| 09:15.08 | XATRIX | And use my 3G dongle as a voice modem |
| 09:15.51 | XATRIX | Actually it does work. I can call from my Phoner app to my handset if i call directly '0662516287' |
| 09:16.04 | XATRIX | It's a shortage of +380662516287 |
| 09:16.13 | ChannelZ | well Context: outgoing_to_0662516287 seems a bit odd |
| 09:16.39 | XATRIX | What's wrong with it ? |
| 09:16.46 | ChannelZ | Not impossible, if you really have a context named that |
| 09:17.15 | ChannelZ | Anyway, the answer is what it's telling you: "sent to invalid extension" |
| 09:17.25 | XATRIX | Do i have to check 'dialplan show' for ? |
| 09:17.47 | ChannelZ | Seems like you have no extension 120 in that crazy context name, which is where your script is telling it to go. |
| 09:17.52 | XATRIX | and the second question is why does it try to call to @120 |
| 09:17.58 | XATRIX | it's my local extension in office |
| 09:18.25 | XATRIX | 120 - is a Phoner app on my PC |
| 09:18.58 | ChannelZ | In that context? "outgoing_to_0662516287" ? |
| 09:19.28 | XATRIX | Nope, context 'outgoing....' only consists of audio files to play |
| 09:19.49 | ChannelZ | ...so there you go... |
| 09:20.09 | ChannelZ | you're telling it to call extension 120 in a context called outgoing_to_0662516287 which doesn't exist apparently |
| 09:20.26 | XATRIX | Not sure what do i have to do ? |
| 09:20.35 | XATRIX | I need it to call to my cell phone |
| 09:21.00 | XATRIX | Channel: Dongle/i:351911049927240/$number |
| 09:21.15 | XATRIX | that should call to $number, which it takes from the diallist.txt |
| 09:22.28 | ChannelZ | Oh I see you do have that crazy context, at the bottom of your paste. But the only extension I see there is 's', not '120' |
| 09:23.15 | XATRIX | crap, can i put a cell phone number in there instead of 's' ? |
| 09:24.02 | XATRIX | The main idea is: i start a script from CLI, it dials my cell phone, plays some music, and hangs it down |
| 09:24.13 | XATRIX | gotta be simple |
| 09:24.24 | ChannelZ | yes and no. You make a pattern extension |
| 09:25.07 | XATRIX | what's the proper way ? |
| 09:26.10 | ChannelZ | well from all I see you just need to make your script call extension s, not 120 |
| 09:26.35 | XATRIX | ah, ok |
| 09:26.52 | ChannelZ | because that extension isn't actually doing anything but playing those sounds |
| 09:27.44 | XATRIX | http://fpaste.org/178246/42261005/ |
| 09:27.52 | XATRIX | Looks like i simply has no files to play |
| 09:28.01 | XATRIX | But at least it started to look for |
| 09:28.34 | ChannelZ | Yes.. or if the files do exist the permissions are wrong and asterisk can't access them |
| 09:28.42 | XATRIX | Yea |
| 09:32.36 | XATRIX | ChannelZ: also, what's wrong with it ? http://fpaste.org/178248/10337142/ |
| 09:33.00 | XATRIX | wrong format to play ? |
| 09:33.27 | ChannelZ | Well you don't put the file extension on the name you tell it |
| 09:33.52 | ChannelZ | Asterisk searches for different file types its self |
| 09:33.59 | XATRIX | exten => s,n,Wait(1) |
| 09:33.59 | XATRIX | exten => s,n,Background(/var/lib/asterisk/sounds/custom/Privetstvie.wav) |
| 09:33.59 | XATRIX | xten => s,n,hangup |
| 09:34.24 | ChannelZ | Leave off the .wav |
| 09:34.39 | XATRIX | Ah |
| 09:34.58 | ChannelZ | And also what are the specs of the actual file? Needs to be 8khz 16-bit mono |
| 09:35.40 | XATRIX | Yes, not it works as a charm! |
| 09:36.11 | XATRIX | /var/lib/asterisk/sounds/custom/Privetstvie.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 09:37.41 | XATRIX | Thanks a lot for the help! |
| 09:37.47 | ChannelZ | sure |
| 09:37.49 | ChannelZ | I'm off to bed |
| 09:38.18 | XATRIX | Yea, good night! |
| 09:49.22 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 09:59.11 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 10:13.10 | *** join/#asterisk tzafrir (~tzafrir@78.129.111.57) |
| 10:25.41 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 10:30.43 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 10:37.15 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-uoamatibaukugalv) |
| 10:41.37 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 10:52.00 | XATRIX | Can you advice me, where can i read about voice generation in asterisk ? |
| 10:52.26 | XATRIX | I need to feed it some text, and take wav file or directly to the call |
| 10:52.28 | *** part/#asterisk madax (~madax@c42-156.i07-11.onvol.net) |
| 10:56.33 | *** join/#asterisk CeBe (~CeBe@port-92-200-33-241.dynamic.qsc.de) |
| 10:56.45 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 11:05.13 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 11:08.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:12.32 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 11:14.08 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:22.52 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 11:52.15 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:02.30 | *** join/#asterisk Akuma (~Akuma@apn-46-169-22-104.dynamic.gprs.plus.pl) |
| 12:04.15 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 12:06.17 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:19.12 | *** join/#asterisk axp (~axp@80.121.172.38) |
| 12:19.20 | axp | hi all |
| 12:20.59 | axp | i do have some extension conf issues with my isdn line: |
| 12:21.03 | axp | -- Executing [s@capi-in:1] Wait("CAPI/ISDN1#02/-9", "1") in new stack -- Executing [s@capi-in:2] Dial("CAPI/ISDN1#02/23-b", "SIP/gitti&SIP/linphone&SIP/bb") in new stack |
| 12:22.04 | WIMPy | And where is the issue? |
| 12:23.25 | axp | the problem here is that if there is an inbound call from a landline , extensions are not cathed immediatly, so if someone dials my number + extension to make a call to me, the first things which matches is the s extension |
| 12:23.29 | axp | \hi WIMPy |
| 12:23.44 | stefan27 | the timer on which the rules in queue_rules.conf act on, seems to be started at 0 whenever a caller joins the Queue()... I want it to start at 0 whenever the caller becomes the head caller |
| 12:24.05 | stefan27 | would that require editing source code ? |
| 12:24.19 | WIMPy | Put a WaitExten at extension s. |
| 12:24.56 | stefan27 | (using 13.1.0) |
| 12:25.53 | axp | WIMPy: so waitexten(2) and if there is no exten i need to handle extension with 0, or can i still use the s extension? |
| 12:26.57 | WIMPy | If nothing is entered, the dialplan continues. |
| 12:27.08 | axp | ok thx |
| 12:27.19 | WIMPy | Or IIRC it jumps to t if it exists. |
| 12:28.19 | phix | WIMPy WIMPy WIMPy! HEFTy! HEFTy! HEFTy! |
| 12:37.57 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 12:40.06 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:00.49 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:05.20 | axp | thanks WIMPy |
| 13:08.32 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 13:09.08 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 13:09.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:16.11 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-lizcrfwvmdzapuie) |
| 13:18.28 | *** part/#asterisk mirela666 (~mirko@212.200.146.247) |
| 13:20.39 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:25.11 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 13:25.18 | *** part/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 13:30.08 | *** join/#asterisk blee (~blee@50-88-244-98.res.bhn.net) |
| 13:30.34 | *** join/#asterisk Xaviertoor (~jerson.ju@177.99.205.154) |
| 13:40.05 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 13:45.37 | *** join/#asterisk areski (~areski@109.236.131.114) |
| 13:54.25 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:54.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:57.45 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 14:12.54 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 14:13.11 | *** join/#asterisk w3rt (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:16.03 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:16.20 | *** join/#asterisk danjenkins (anonymous@nat/digium/x-axjokjonmdosxdad) |
| 14:20.49 | pa | WIMPy, hi |
| 14:21.26 | pa | i have a question for you :) do you know why, after i place a call through LCR, subsequent dialed numbers are not recognized? |
| 14:21.44 | pa | did i forget to configure something? |
| 14:22.36 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:26.34 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 14:26.48 | *** join/#asterisk iceslice (3b98672a@gateway/web/freenode/ip.59.152.103.42) |
| 14:27.10 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 14:27.25 | iceslice | Greetings everyone! |
| 14:30.32 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 14:31.05 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:31.05 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:39.56 | *** join/#asterisk areski (~areski@109.236.131.114) |
| 14:51.33 | overyander | can anyone tell me if hylafax+ has t.38 support? i see that the regular hylafax project wants you to purchase their entterprise product for t.38, but i can't find any documentation regarding hylafax+ and t.38 |
| 14:55.36 | AnonGirl | hi |
| 14:58.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:01.41 | *** join/#asterisk BCrookAtRA (~Adium@2620:11a:5000:0:9086:857d:f1d6:1031) |
| 15:04.08 | *** part/#asterisk pa (~pa@unaffiliated/pa) |
| 15:04.13 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:04.32 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-dkvczotsebjqppnq) |
| 15:04.33 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:10.11 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:20.39 | *** join/#asterisk mjordan (mjordan@nat/digium/x-marxcpdetxdpggah) |
| 15:20.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:23.49 | *** join/#asterisk lirakis (~lirakis@50-198-204-238-static.hfc.comcastbusiness.net) |
| 15:23.53 | lirakis | helo |
| 15:24.02 | lirakis | i have a question about a very old version of asterisk ... ha ha |
| 15:24.18 | lirakis | is g722 available in asterisk 1.2.9.1 ? |
| 15:24.24 | lirakis | im looking in the codec directory |
| 15:25.05 | lirakis | and i dont see anything related to g722 ... just codec_pcmu .... codec_g726 ... etc |
| 15:25.09 | [TK]D-Fender | no |
| 15:25.21 | [TK]D-Fender | * only got G722 transcode in 1.6+ |
| 15:25.37 | lirakis | so ... even if im not looking to transcode |
| 15:25.44 | lirakis | but i want to playback |
| 15:25.57 | lirakis | a g722 announcement to a user who is g722 capable |
| 15:26.37 | [TK]D-Fender | 1.4 added passthrough |
| 15:26.53 | lirakis | ok .. so totally SOL with 1.2.9.1 and g722 |
| 15:26.57 | file | yes. |
| 15:26.58 | [TK]D-Fender | yes |
| 15:27.01 | lirakis | [TK]D-Fender, thanks ;) |
| 15:27.09 | drmessano | 1.2 OMG |
| 15:27.16 | [TK]D-Fender | That looks like one of the oldest versions used by trixbox |
| 15:27.37 | lirakis | its not trix box |
| 15:27.45 | lirakis | its a ancient ... announcement server |
| 15:27.46 | lirakis | lol |
| 15:27.50 | lirakis | but vanilla |
| 15:27.57 | [TK]D-Fender | Burn it |
| 15:27.59 | lirakis | ... which is why i was looking in the src dir |
| 15:28.06 | lirakis | ya .. going to move it to freeswitch :P |
| 15:28.28 | lirakis | thanks again for the info/verification |
| 15:36.42 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-bohxzvlijkxqkrla) |
| 15:37.12 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hjmujlsbipeowgvu) |
| 15:37.37 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-plrayldgcjvbqsxg) |
| 15:37.59 | *** join/#asterisk tristan-mei (tristan@unaffiliated/tristan-mei) |
| 15:41.38 | *** join/#asterisk Shariff (~chatzilla@2001:980:167b:1:718b:cb72:2f67:5d17) |
| 15:41.40 | Shariff | Hi there |
| 15:42.57 | Shariff | I have just set up a basic asterisk server for voip using the pjsip libraries. I can make phonecalls to my extensions and voicemail. Now I'm trying to get MWI to work on the endpoints. Unfortunatley asterisk appears to be sending wrong notify packets. In the packets it states 0 messages... how do I fix this? |
| 15:44.49 | *** join/#asterisk tristan-mei (tristan@unaffiliated/tristan-mei) |
| 15:46.28 | *** join/#asterisk areski (~areski@109.236.131.114) |
| 15:46.32 | *** part/#asterisk lirakis (~lirakis@50-198-204-238-static.hfc.comcastbusiness.net) |
| 15:46.45 | Shariff | In fact when I do "voicemail show users" I see there are 4 new messages waiting... but the notify sends (0/0) |
| 15:49.22 | Shariff | Any ideas? |
| 15:50.50 | newtonr | Are you sure you are looking at the NOTIFY for the correct endpoint? |
| 15:51.01 | newtonr | Also, do you have "mailboxes" configured for the endpoint, the aor, or both? |
| 15:52.52 | Shariff | I have a mailbox configured in the AOR |
| 15:53.08 | Shariff | and the notify source is my asterisk IP and destination is my softphone ip |
| 15:53.12 | *** join/#asterisk iceslice (3b98672a@gateway/web/freenode/ip.59.152.103.42) |
| 15:53.14 | Shariff | So it would appear that they are correct |
| 15:53.26 | newtonr | what version of 13? |
| 15:53.28 | Shariff | 12 |
| 15:53.46 | Shariff | The extension is however my normal extension 202 and not my voicemail extension 902 |
| 15:54.06 | Shariff | Could that be part of the issue? That asterisk is sending the number of new messages for extension 202, rather than mailbox 902? |
| 15:54.37 | newtonr | 12 no longer receives bug fixes, you'll want to test with 13, and make sure you test with the latest of the 13 SVN branch |
| 15:55.06 | newtonr | The notify should be for whatever mailbox you have configured |
| 15:55.28 | newtonr | the endpoint/aor name does not have to match the mailbox |
| 15:55.57 | Shariff | ok |
| 15:56.31 | Shariff | Nowhere in the packet do I see 902 (voicemail box) only the extension 202.. which is associated with that mailbox.. is that correct? |
| 15:57.39 | newtonr | I don't think so, but I can't check at this moment. I'm about to run into a meeting. If no one else knows, then I'll take a look when I get back |
| 15:57.54 | Shariff | Well before you run off.. |
| 15:58.02 | Shariff | do you tink it better if I upgrade to 13 first? |
| 15:58.11 | Shariff | It's a fresh install, so nothing would be lost if I do |
| 15:58.16 | *** part/#asterisk mjordan (mjordan@nat/digium/x-marxcpdetxdpggah) |
| 15:58.37 | Shariff | Thanks a lot for the help! And have a good meeting! |
| 15:59.00 | iceslice | Shariff: are you author of "Elastix without tears" ? i.e. Ben Sharif |
| 15:59.06 | Shariff | Nope :D |
| 15:59.22 | iceslice | well :D |
| 15:59.28 | Synthase_ | Shariff: You'll want to be on 13 as a matter of principle. Might as well do it before temporary becomes long term. |
| 16:09.11 | WIMPy | pa: "not recognized"? What does that mean? |
| 16:13.16 | iceslice | WIMPy: use common sense :D |
| 16:13.55 | pa | WIMPy, it means that if some answering machine answers, asking me to press 1 /2 /etc.., and i press these numbers |
| 16:14.01 | pa | It's like if i press nothing |
| 16:14.09 | WIMPy | iceslice: Not recognized. |
| 16:14.16 | *** join/#asterisk sames (~sames@205.233.8.141) |
| 16:14.50 | WIMPy | pa: Yes you need to enable DTMF generation. The option depends on the version. That was also changed. |
| 16:15.03 | pa | hm i see.. and how do i do it on your version? |
| 16:15.11 | WIMPy | see lcr_config same options apply to Dial. |
| 16:15.12 | pa | the non existing 1.12something? :) |
| 16:15.38 | pa | thanks |
| 16:16.21 | WIMPy | That would be "s". |
| 16:18.25 | *** join/#asterisk benlangfeld (~benlangfe@unaffiliated/benlangfeld) |
| 16:18.42 | benlangfeld | Hey nerds! How is everyone today? |
| 16:19.44 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:20.47 | benlangfeld | I have a question about an old Asterisk 1.4 Realtime deployment I'm trying to integrate with, and I wondered if anyone could tell me what I should call this apparently crazy situation where all config is in a single DB table, with a row for each attribute, essentially a key-value store. |
| 16:20.53 | *** join/#asterisk mjordan (mjordan@nat/digium/x-shdogiqderwhgcau) |
| 16:20.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:20.59 | pa | WIMPy, yes i found that in some mailing list now. it seems it's the "s" option.. so that means LCR/bla/bla/s ? |
| 16:21.23 | WIMPy | pa: yes |
| 16:21.46 | pa | thanks i try |
| 16:22.40 | WIMPy | I did include that on the "Usual configuration issues". |
| 16:24.06 | pa | another question, though.. on channels that support dmtf generation my local asterisk keep answering "sorry, this is not a valid extension, please try again" before i can hear again the other endpoint. is this possible to disable on invalid extensions? |
| 16:24.18 | pa | *to be disabled i mean |
| 16:25.18 | WIMPy | On bridged calls with "features" enabled? |
| 16:25.44 | pa | hm.. "features"? |
| 16:25.56 | WIMPy | As in features.conf. |
| 16:26.05 | pa | ah.. never touched that |
| 16:26.08 | pa | i can show an extension wher ei have such problem, moment |
| 16:26.16 | WIMPy | Or what's the exact situation? |
| 16:27.38 | pa | http://pastebin.ubuntu.com/9959018/ |
| 16:28.10 | pa | well ok itæs not the correct example |
| 16:28.20 | pa | because this one goes on LCR that right now doesnt work |
| 16:28.21 | pa | anyway |
| 16:28.47 | pa | should it work, i would hear that sentence before hearing back the called person/number answering the right stuff |
| 16:30.06 | pa | ok no wait |
| 16:30.16 | pa | with LCR i don't have the problem, as it seems now |
| 16:30.18 | WIMPy | I don't see anything that would handle an invalid number there. And I certainly don't see it *happening*. |
| 16:31.12 | pa | but sometimes i get the sentence "transfer!! this is not a valid extension! please tr again!" |
| 16:31.48 | pa | and i assume this happens on my asterisk |
| 16:31.52 | WIMPy | You have enabled "features" then. i.e. option t/T to Dial. |
| 16:31.58 | pa | since i actually get transfered on the other side |
| 16:32.10 | pa | i see |
| 16:32.12 | pa | i check |
| 16:32.37 | WIMPy | Do you use a phone connected to LCR? |
| 16:33.30 | *** part/#asterisk mjordan (mjordan@nat/digium/x-shdogiqderwhgcau) |
| 16:34.13 | WIMPy | I'd recommend adding dtmfthreshold=200 (or up to 400) when loading mISDN_dsp. With the default it is oversensitive. |
| 16:34.42 | pa | i have only asterisk connected directly to LCR |
| 16:34.50 | pa | then i have some phones/softphones connected to asterisk |
| 16:35.01 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 16:35.02 | pa | also, the only uncommented lines in features.conf are: http://pastebin.ubuntu.com/9959136/ |
| 16:35.48 | WIMPy | Do you use inband DTMF with SIP? Or are you pressing # accidentally? |
| 16:36.11 | pa | well i press # when they ask me to dial some number and press # at the end |
| 16:36.23 | pa | and at that point i get the sentence, and then the correct stuff |
| 16:36.59 | pa | like if the sequence + # is first intercepted by asterisk, that does not understand it, and also passed through |
| 16:37.04 | WIMPy | It tells you the extension is invalid but you get the disired result anyway? |
| 16:37.31 | pa | yes, because that's an extension on the dialed endpoint, not on my asterisk box, i suppose |
| 16:38.00 | AnonGirl | How do I patch KDE2 under FreeBSD? |
| 16:38.05 | pa | so i have phone --> asterisk --> LCR || gtalk --> other side |
| 16:38.16 | WIMPy | If it only happens to a certain destination, they might have a configuration issue. |
| 16:38.38 | pa | so you think it's there that it happens, and not on my side? |
| 16:38.52 | WIMPy | Otherwise your dilplan must have some interesting bits. |
| 16:39.12 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 16:39.19 | pa | ok i see. thanks! i will do some experiment on the next call to see if i can trigger this on any call :) |
| 16:39.22 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 16:39.33 | WIMPy | You need to have t and/or T enabled on your Dial() for it to happen on your side. |
| 16:39.44 | *** join/#asterisk superscrat (asanders@nat/digium/x-kxaaslyoliatajoc) |
| 16:39.57 | pa | that means to intercept sequences after the call has been placed, right? |
| 16:40.07 | pa | also, in that case, is it possible to intercept && forward? |
| 16:40.12 | pa | or rather |
| 16:40.19 | pa | intercept and forward if not defined? |
| 16:40.25 | WIMPy | The one for "feature" transfers, yes. |
| 16:41.11 | pa | for example i thought of defining something like that to move the call to another phone |
| 16:41.26 | WIMPy | That's the idea. |
| 16:41.26 | pa | say i pick up, and then i want to transfer it to a phone downstairs |
| 16:41.56 | pa | ah i see :) |
| 16:42.18 | WIMPy | But better let the phone do it via SIP messages, unless it's unable to do so. |
| 16:43.22 | pa | well my phones are generally speaking pretty "unable" |
| 16:43.34 | pa | they barely work 70% of the times |
| 16:43.52 | WIMPy | That is really bad. |
| 16:44.03 | pa | i even had to put a timer on one of them to have it rebooted every night, and gain some more uptime |
| 16:44.15 | pa | like a timer power plug |
| 16:44.37 | WIMPy | That does not sound very usable. |
| 16:44.44 | pa | indeed :) |
| 16:44.56 | pa | i got some cheap fxs though.. waiting to try it |
| 16:45.01 | pa | and see if that helps |
| 16:45.15 | pa | can't really afford to buy a new cordless voip phone.. |
| 16:47.08 | iceslice | ever tried gsm gateway ? |
| 16:47.38 | pa | nope.. i wanted to, though |
| 16:48.10 | iceslice | Looking for minimum hardware requirement for a complete Unified Communication server system |
| 16:48.59 | iceslice | including fax, gsm etc.. |
| 16:50.53 | *** join/#asterisk eren (~eren@unaffiliated/eren) |
| 16:50.57 | eren | hello folks |
| 16:51.15 | iceslice | hello |
| 16:51.46 | eren | I have Cisco 7941 phone and I installed numerous firmwares. I have setup debug environment and I sniff packets. Although I set sip auth parameters, the phone does not send authentication info in "Authenticate:" header when 401 error got from server |
| 16:51.55 | eren | all people say that this phone works with asterisk |
| 16:52.14 | eren | does anyone here have a test server to try this with asterisk? |
| 16:54.37 | iceslice | sorry, my asterisk installation server is off atm |
| 16:54.39 | eren | we try to use it on some cloud voip provider, the only difference with our provider and asterisk is (hopefully) that asterisk sends "WWW-Authenticate: Digest algorithm=md5" but our provider does not send it without algorithm |
| 16:55.09 | eren | does send it without algorithm=md5* |
| 16:55.13 | eren | that's my only hope :( |
| 16:55.31 | eren | since i cannot see what's inside the fimware, I suspect that it cannot understand whether auth is required |
| 16:55.48 | Synthase_ | eren: Version? |
| 16:55.56 | eren | I guess the auth protocol works like this: try to REGISTER without auth, get 401 reply, send Authenticate header |
| 16:55.58 | *** join/#asterisk mjordan (mjordan@nat/digium/x-nftyuxbjzsuhrvbz) |
| 16:55.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:56.23 | Synthase_ | eren: Correct |
| 16:56.23 | eren | Synthase_: currently 8.5(4) |
| 16:56.29 | Synthase_ | Nope, of Asterisk |
| 16:56.48 | eren | Synthase_: uh, I don't have asterisk setup, I tried to install set environment but it was way complicated for me |
| 16:56.58 | eren | s/set/test/ |
| 16:57.36 | eren | that's why I came here to ask. I would be glad if someone can provide test server |
| 16:59.18 | eren | I can install asterisk but there are a lot of files, I configured sip.conf with some guide but I got "method not allowed" on register. Any pointers? |
| 16:59.28 | eren | my host os is Ubuntu 14.04 |
| 17:00.43 | iceslice | eren: why aren't you trying a virtual machine or cloud server for your test server ? |
| 17:01.51 | eren | iceslice: I couldn't configure asterisk, that was why :) |
| 17:02.05 | *** part/#asterisk BCrookAtRA (~Adium@2620:11a:5000:0:9086:857d:f1d6:1031) |
| 17:02.07 | iceslice | you can try "AsteriskNOW" |
| 17:02.15 | iceslice | *distro |
| 17:02.41 | eren | iceslice: do I need a SIP proxy as well, btw? |
| 17:02.53 | eren | because I'm communicating with sip proxy with our provider, I need to replicate it a bit |
| 17:03.17 | iceslice | depends on your requirement |
| 17:04.30 | eren | iceslice: ok, with asterisk setup, why do I get "method not allowed" when the phone sends "REGISTER" ? |
| 17:06.12 | iceslice | probably you need to configure the setup appropriately, (sorry, can't help much atm, try search engines) |
| 17:07.59 | iceslice | *or relevant docs |
| 17:08.05 | *** part/#asterisk mjordan (mjordan@nat/digium/x-nftyuxbjzsuhrvbz) |
| 17:09.03 | Synthase_ | ~book |
| 17:09.03 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:09.39 | *** part/#asterisk benlangfeld (~benlangfe@unaffiliated/benlangfeld) |
| 17:10.37 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:28.44 | iceslice | eren: try above mentioned book's chapter 5: "Testing to ensure your devices have registered" section.. |
| 17:39.47 | *** join/#asterisk WangDang (~Wang@24.140.224.212) |
| 17:43.45 | WangDang | Hi, I'm looking at a sample configuration file where it talks about setting up an iax trunk. "context=*" I'm not sure if that asterisk has special meaning or if the example is assuming there is a context [*] in the dialplan |
| 17:44.21 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 17:46.13 | DarkHelmet | thats for the help, I have the right developer working on my DAhdi issues |
| 17:50.00 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 17:55.47 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 18:11.27 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 18:13.32 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 18:13.41 | *** join/#asterisk raymondhim (~raymondhi@jessica.totalsyssolutions.net) |
| 18:13.47 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 18:13.54 | *** join/#asterisk BakaKuna (~BakaKuna@82-169-251-128.ip.telfort.nl) |
| 18:14.33 | *** join/#asterisk mjordan (mjordan@nat/digium/x-lcsnzmfwbtpkipnj) |
| 18:14.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:18.44 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-bbkamchieikkpeag) |
| 18:21.27 | *** join/#asterisk zamba (marius@flage.org) |
| 18:23.35 | *** join/#asterisk eren (~eren@unaffiliated/eren) |
| 18:31.02 | *** join/#asterisk tristan-mei (tristan@unaffiliated/tristan-mei) |
| 18:33.59 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 18:35.16 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:35.42 | *** join/#asterisk lorenzo (lorenzo@unaffiliated/lorenzo) |
| 18:35.52 | lorenzo | hello |
| 18:36.04 | lorenzo | exten => _91.,2,Monitor(wav,,mb) |
| 18:36.13 | lorenzo | according to the doc this should both save to file and merge, right? |
| 18:36.16 | lorenzo | (the channels) |
| 18:39.31 | WIMPy | You might be looking for MixMonitor. |
| 18:48.07 | *** part/#asterisk mjordan (mjordan@nat/digium/x-lcsnzmfwbtpkipnj) |
| 19:01.48 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:10.17 | *** join/#asterisk mjordan (mjordan@nat/digium/x-mezgjmqpulifbklr) |
| 19:10.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:12.04 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:24.23 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:26.46 | *** join/#asterisk danjenkins (anonymous@nat/digium/x-cbjohjznssbsirsc) |
| 19:29.17 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 19:57.08 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 19:58.59 | *** join/#asterisk Akuma (~Akuma@apn-46-169-22-104.dynamic.gprs.plus.pl) |
| 20:00.40 | *** join/#asterisk Alex_Bkash (7c06ecf5@gateway/web/freenode/ip.124.6.236.245) |
| 20:02.00 | *** join/#asterisk Akuma (~Akuma@apn-46-169-22-104.dynamic.gprs.plus.pl) |
| 20:06.11 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:c846:6224:cfdb:361d) |
| 20:31.55 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 20:34.04 | *** join/#asterisk tzafrir (~tzafrir@78.129.111.57) |
| 20:58.57 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-onvebfhailtaikxg) |
| 21:02.41 | DivideBy0x0 | When using TALK_DETECT on a channel, the default is 2.5 seconds, does anyone have any experience with what it should be to determine when a voicemail greeting is Over? The it won't be turned on until the voicemail greeting is in progress. Otherwise, I'll just start at 2.5 secs and work up or down from there |
| 21:06.53 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-gosgwwmqaxiablmh) |
| 21:07.11 | WangDang | Noob here, I'm trying out a voip termination provider. They suggest the config option "insecure=port,invite". I'd like to know what it is that I'm implementing rather than just blindly using the suggested config. Any suggestions where to find a reference to the insecure config option other than the O'reilly book or the wiki, cause I can't find it in either of those places places. |
| 21:07.43 | WangDang | I've looked in iax.conf.sample, too |
| 21:07.43 | WIMPy | The sample configuration. |
| 21:08.03 | [TK]D-Fender | WangDang: that tells * to accept any call from the "host=" you specify without chellenging them. |
| 21:08.07 | WIMPy | It's not an iax option. It's a sip option. |
| 21:08.21 | [TK]D-Fender | WIMPy: it exists in both actually |
| 21:08.36 | WIMPy | Woot? |
| 21:09.33 | WIMPy | Not in the iax.conf.sample from 11.15. |
| 21:09.54 | [TK]D-Fender | WIMPy: I recall its use previously. |
| 21:10.17 | WangDang | [TK]D-Fender: thanks, but is there some reference available where I could have looked that up myself, 'cuz I didn't find it the O'r book or the wiki or the sample config file |
| 21:11.12 | WangDang | unless, as WIMPy suggests, it isn't available to iax |
| 21:11.46 | WIMPy | I just used grep on the source. I only find "insecure" in chan_sip. |
| 21:12.05 | Shariff | If I don't plan on using any PSTN or ISDN phones.. would I still need to compile libpri or dahdi? |
| 21:12.36 | WIMPy | Shariff: If you want to connect to a PSTN line. |
| 21:12.38 | *** part/#asterisk mjordan (mjordan@nat/digium/x-mezgjmqpulifbklr) |
| 21:13.32 | Shariff | So no? if I don't plan on connecting to a pstn line I wouldn't need them? |
| 21:14.02 | WIMPy | Shariff: If you don;t plan to use any hardware on the PC, then you don't need them. |
| 21:14.15 | Shariff | Thanks! |
| 21:14.36 | WIMPy | Or rather: Then you definitely don't need them. You might not need them with hardware, either. |
| 21:15.36 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ihcocztixxjfqiei) |
| 21:15.36 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:16.14 | Shariff | Great, thanks a lot! |
| 21:18.49 | WangDang | if I'm not planning on haviing incoming calls from a provider (outgoing calls only), I wouldn't need the "insecure=port,invite" then would I? |
| 21:19.38 | WIMPy | WangDang: Corect |
| 21:20.22 | WangDang | thanks for the help! |
| 21:22.27 | *** join/#asterisk mjordan (mjordan@nat/digium/x-lkhendgcfczfaiau) |
| 21:22.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:25.10 | [TK]D-Fender | ShariffIf I don't plan on using any PSTN or ISDN phones.. would I still need to compile libpri or dahdi? <- Still used by MeetMe, and often IAX trunk mode, etc |
| 21:25.15 | [TK]D-Fender | Shariff: "PAge", as well |
| 21:27.10 | WIMPy | I think IAX doesn't need dahdi timing any more. And AFAIK Page() has been moved to the new bridging stuff as well. |
| 21:28.02 | WIMPy | Yes, Page exists without dahdi. And Meetme, yes, but who would still want that? |
| 21:28.35 | Synthase_ | FreePBX apparently |
| 21:28.36 | [TK]D-Fender | People using legacy code or a feature that ConfBridge hasn't quite caught up to |
| 21:29.05 | WIMPy | Are there any? |
| 21:29.34 | WIMPy | I see tons of features missing in meetme. |
| 21:30.29 | [TK]D-Fender | Yes, ConfBridge has grown fast and is easily on par or better in general |
| 21:30.39 | [TK]D-Fender | Sometimes it's just that "one thing" you insist on... |
| 21:31.02 | WIMPy | Do you know such a thing? |
| 21:33.03 | [TK]D-Fender | not offhand... then again I never used either much at all... |
| 21:33.05 | [TK]D-Fender | Checkout time, heading home. BBIAB |
| 21:35.59 | babak_ | hi, is there an open source voice broadcast software based on Asterisk ? (not mixed with contact center) |
| 21:36.21 | WIMPy | What kind of broadcast? |
| 21:36.28 | babak_ | voice |
| 21:36.36 | WIMPy | You said so. |
| 21:36.48 | WIMPy | But still that could mean about anything, |
| 21:36.51 | babak_ | play a recorded message to 10,000 |
| 21:36.53 | WIMPy | . |
| 21:37.07 | WIMPy | 10000 what? |
| 21:37.14 | babak_ | people |
| 21:37.38 | WIMPy | I'd recommend a big PA then. |
| 21:37.44 | babak_ | for example inform people from their telephone bill |
| 21:37.57 | babak_ | big PA ? |
| 21:38.05 | WIMPy | "from"? |
| 21:38.40 | babak_ | my english is not good , inform bad payers to pay |
| 21:38.48 | babak_ | bills |
| 21:38.53 | AnonGirl | hi |
| 21:39.09 | WIMPy | Where is the "broadcast" in that? |
| 21:39.21 | WIMPy | You need to be more specific about what you want to do. |
| 21:39.22 | AnonGirl | Calling all at once |
| 21:39.41 | babak_ | broadcast means calling many people |
| 21:40.12 | babak_ | and giving them some information |
| 21:40.14 | WIMPy | "broadcast" usually meand an unspecific number of receivers simultaneously. |
| 21:40.28 | AnonGirl | WIMPy, they want to call everyone who isn't paying their telephony bill all at once |
| 21:40.57 | babak_ | yes system should have more than 1000 simultanous call capability |
| 21:41.06 | lvlinux | if they aren't paying their bill, how do they still have service to receive the call? lol sorry... |
| 21:41.07 | WIMPy | Well, that would require some auto-answer feature. |
| 21:41.19 | WIMPy | :-) |
| 21:41.20 | AnonGirl | babak_, to what? PSTN lines? VoIP lines? |
| 21:41.33 | babak_ | pstn lines |
| 21:41.41 | AnonGirl | babak_, are you a telco? |
| 21:41.47 | babak_ | yes |
| 21:41.53 | WIMPy | Ok, so you just want to call a lot of people. That's not broadcast. That's just calling. |
| 21:41.55 | AnonGirl | then do your own homework |
| 21:42.36 | WIMPy | There are some dialers out there also based on Asterisk. |
| 21:42.51 | babak_ | some times I want tell 20,000 people they have 24 hours service(telephone) cut |
| 21:42.53 | WIMPy | Vicidial (AKA victimdial) must be the most popular one. |
| 21:43.48 | babak_ | vicidial is mixed with contact center and complicate and may be less capacity |
| 21:44.12 | WIMPy | Maybe you should do your own then. |
| 21:44.36 | WIMPy | It's not hard to originate calls. |
| 21:44.48 | babak_ | newfies dialer is good but... |
| 21:50.37 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:41cb:8a0c:8fb9:9024) |
| 21:50.43 | babak_ | TeleYapper has someone experience with it ? |
| 22:07.05 | *** join/#asterisk Akuma (~Akuma@apn-31-1-130-84.dynamic.gprs.plus.pl) |
| 22:12.27 | PaybackTony | Just as a heads up to anyone else interested in webrtc stuff. I have seen the same issues as described in this ticket and it's related ticket: https://issues.asterisk.org/jira/browse/ASTERISK-24651 in a few flavors of linux (CentOS 6.6, Ubuntu 14.04, Ubuntu 14.10) and 5 versions of Asterisk (13-current, 12-current, 11-current, 11.11, 11.15 and 11-trunk). Considering webrtc is a huge part of our product it's been a pain. Attempting t |
| 22:13.10 | *** join/#asterisk Cynagen (~cynagen@ip72-208-223-60.ph.ph.cox.net) |
| 22:14.01 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:16.13 | *** join/#asterisk jkjkjjjjjjjjjj (archaic@gateway/shell/layerbnc/x-bzfavyraixiknriu) |
| 22:18.28 | *** join/#asterisk Primer (~daniel@www.ceregatti.org) |
| 22:23.06 | *** part/#asterisk Primer (~daniel@www.ceregatti.org) |
| 22:35.06 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 22:37.24 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 22:51.16 | newtonr | PaybackTony, you need to comment on the issues involved and provide/attach debug relating to your environment. If the issue is not resolved yet, then your data may help.. |
| 22:58.35 | *** join/#asterisk danjenkins (anonymous@nat/digium/x-xhupfbpjvdncvhpp) |
| 23:20.35 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 23:26.59 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 23:29.59 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 23:31.50 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 23:31.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:33.32 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 23:34.05 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 23:38.28 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |