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| 00:22.44 | lvlinux | ok i need some help here---i have faxdetect=incoming in chan_dahdi.conf |
| 00:23.00 | lvlinux | i have the fax machin connected to an FXS port |
| 00:23.41 | lvlinux | receiving faxes work great---Asterisk detects them, sends them to the fax extension, which calls the FXS line (where the fax machine is connected) |
| 00:24.19 | lvlinux | but i went to send a fax using the fax machine, and of course that's an incoming call, so Asterisk detected the fax and sent it to the fax extension, before the fax machine could dial the number. |
| 00:25.08 | lvlinux | so can i disable fax detection within a context or something? or is there a correct way to handle outgoing faxes? |
| 00:25.16 | WIMPy | So switch off fax detection for that FXS channel. |
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| 00:25.37 | lvlinux | oh ok that should work |
| 00:27.18 | lvlinux | i didn't know i could enable/disable it on certain channels, thoght it was the whole card for some reason. |
| 00:27.40 | WIMPy | I don;t ssee why it should be. |
| 00:28.01 | WIMPy | But as always: Any idea is pure hope that needs to be verified. |
| 00:28.16 | lvlinux | verifying... |
| 00:37.44 | lvlinux | yeah it worked fine. Thanks! |
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| 00:57.49 | lvlinux | how do i NOT send ringing tone back through to the caller? |
| 00:58.05 | lvlinux | is there an option to Dial() that i can use for that? |
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| 01:06.14 | ChannelZ | you don't necessarily have a choice, if via SIP for instance asterisk reports ringing to the device, it its self might be generating the ringing tone. |
| 01:07.34 | lvlinux | it's between two dahdi channels---pstn line answers, then dials FXS channel. I'd like silence while it's dialing if possible. |
| 01:07.39 | ChannelZ | I guess you could use a silent MOH and pass m to Dial |
| 01:08.03 | lvlinux | i thought about that but hoped there was a little easier way lol. |
| 01:08.12 | lvlinux | or rather an "official" way. |
| 01:08.24 | Selloper | is there a way to pass DID from 1 pbx system to another? |
| 01:08.51 | ChannelZ | well I guess you could fudge up indications.conf and make a silent ring but that will affect everything |
| 01:09.04 | lvlinux | yeah definitely don't want that... |
| 01:10.05 | ChannelZ | Well you could make a 'nothing' indication I guess and use r(nothing) |
| 01:10.23 | lvlinux | hmm |
| 01:10.38 | ChannelZ | assuming you can make a silent indication. Hmm. |
| 01:11.08 | ChannelZ | 0/1000 or something probably |
| 01:11.19 | ChannelZ | or just 0 might work. |
| 01:11.38 | lvlinux | i'll need to read about indications.conf---never messed w it before |
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| 03:10.31 | spicyramen_ | hi all, im trying to setup and outbound dialer, that dials contact center agent and customer at specific time. Is there some module available today ? or needs dev work |
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| 12:21.55 | zafu | hi, is there a .gsm audi player for android? (to listen * voicemail) |
| 12:22.02 | zafu | s/audi/audio |
| 12:24.02 | [TK]D-Fender | VLC should be able to play them |
| 12:24.30 | zafu | nope |
| 12:25.33 | zafu | just found a non-free app that does it WavPlayer |
| 12:26.13 | zafu | supports all voip codecs, nice |
| 12:27.37 | zafu | on another subject, does latest * support shared sip line registration? |
| 12:27.51 | zafu | several devices on the same sip accounts? |
| 12:28.29 | [TK]D-Fender | chan_sip = no, pjsip = yes |
| 12:28.37 | [TK]D-Fender | for multiple registration. |
| 12:28.42 | [TK]D-Fender | neither supports SLA |
| 12:28.57 | zafu | sla? |
| 12:29.02 | [TK]D-Fender | ~slacareful on your us |
| 12:29.06 | [TK]D-Fender | ~sla |
| 12:29.06 | infobot | i guess sla is service level agreement, or shared line appearances |
| 12:29.11 | [TK]D-Fender | ^^^ |
| 12:29.31 | zafu | is pjsip stable? |
| 12:29.44 | [TK]D-Fender | having a registration on another device that can track held calls, etc |
| 12:29.53 | [TK]D-Fender | seems to be from the sound of things. |
| 12:30.15 | [TK]D-Fender | Not sure what shortcomings it may still have at this point but it is looking like it's going to be the future... |
| 12:30.24 | zafu | aha, interesting |
| 12:30.47 | zafu | is it a transparent migration from chan_sip? (sip.conf ) |
| 12:31.01 | zafu | or new conf format |
| 12:33.58 | [TK]D-Fender | quite new |
| 12:34.10 | [TK]D-Fender | it breaks things into pieces... which is a good thing |
| 12:34.38 | [TK]D-Fender | I've only glossed over it... but it's simple enough.. |
| 12:38.18 | leifmadsen | zafu: totally different channel driver with different approach to configuration |
| 12:38.36 | leifmadsen | so it'll require a migration plan. It's not a straight drop in replacement |
| 12:38.59 | zafu | ok, thanks |
| 12:42.53 | zafu | just installed WavPlayer in his android phone and it reads .gsm files from email attachements fine |
| 12:44.49 | file | there IS a Python script in the contrib/scripts/sip_to_pjsip directory which will try to configure a sip.conf configuration into a pjsip.conf configuration |
| 12:45.30 | zafu | nice |
| 12:45.40 | file | configure? convert |
| 12:47.45 | leifmadsen | file: orly, I was unaware of this script. Very cool |
| 12:47.59 | leifmadsen | if I had been writing a book I'd probably have known this :) |
| 12:48.12 | leifmadsen | file: want to write a book with me? |
| 12:48.40 | file | sure! because I don't have tons to do as it is |
| 12:48.45 | leifmadsen | me either |
| 12:48.57 | leifmadsen | it's obvious since we're chatting in #asterisk |
| 12:48.59 | carrar | need some work? |
| 12:49.08 | carrar | assigns some tickets |
| 12:49.26 | leifmadsen | lottery tickets? |
| 12:49.29 | carrar | heh |
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| 12:49.36 | leifmadsen | concert tickets? |
| 12:49.47 | file | speeding tickets? |
| 12:49.51 | leifmadsen | ohnoez |
| 12:49.54 | carrar | I get those dropped |
| 12:50.01 | carrar | lawyer++ |
| 12:50.01 | leifmadsen | I just don't speed |
| 12:50.06 | leifmadsen | heh |
| 12:50.25 | carrar | yeah not speedign tends to help too |
| 12:52.04 | leifmadsen | for some reason I keep adding stuff to my car to make it go faster though |
| 12:52.16 | leifmadsen | I just end up accelerating to the speed limit faster and faster :) |
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| 13:02.18 | neeby_goosey | Hello. |
| 13:04.43 | Kattyroo | hi |
| 13:07.50 | dan_j | Hi. I'm working on an auto dialler that integrates with SalesForce. Ideally, I want the agent to have an active channel already open, and receive audio from the auto-dialled calls as soon as they start ringing. Is that possible? If yes, what manager API commands should I be looking at? |
| 13:09.44 | [TK]D-Fender | dan"bridge" |
| 13:10.00 | carrar | hi KattyRooo |
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| 13:16.43 | Kattyroo | hugs carrar |
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| 13:18.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:22.22 | carrar | hugs Kattyroo back!! |
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| 13:25.50 | dan_j | [TK]D-Fender: This question is based on the fact that I think that a channel must always be connected to another channel. When using bridge to join two channels, what happens to the old channels? |
| 13:26.59 | [TK]D-Fender | dan_j: there is also Redirect". |
| 13:27.08 | [TK]D-Fender | dan_j: Read the instructions for each |
| 13:30.02 | dan_j | Finally, at the end of a successful call, i dont want the agent's channel to be hung up. I want it to go back to a waiting state. Is there any way to do that? |
| 13:30.19 | [TK]D-Fender | dialplan. |
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| 13:30.50 | dan_j | dialplan still applies even if the call path has been altered by the manager api? |
| 13:31.00 | Ice_Strike | exten => _60.,1(Name),Chanspy(Agent/${EXTEN}) |
| 13:31.05 | Ice_Strike | What does _60. mean? |
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| 13:31.28 | dan_j | Ice_Strike: anything beginning with 60 |
| 13:31.33 | Ice_Strike | Ah |
| 13:32.24 | [TK]D-Fender | Ice_Strike: Need to read up on your dialplan basics..... |
| 13:32.26 | [TK]D-Fender | ~book |
| 13:32.26 | 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 |
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| 13:34.45 | Kattyroo | read?! ain't nobody got time for that |
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| 13:42.27 | WIMPy | dan_j: No, channels need not be bridged to another channel. And if you Bridge a cahnnel via AMI, it will return to the dialplan where it left off, when the bridge is broken. |
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| 13:47.08 | dan_j | WIMPy: thank you |
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| 13:52.11 | dan_j | Whats the easiest way to retrieve a specific channel id if i know what the sip peer name is? |
| 13:54.26 | file | a SIP peer may have multiple channels |
| 13:54.44 | dan_j | True, however if the client is instructed only to have one? |
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| 14:00.33 | [TK]D-Fender | Instructed means nothing. |
| 14:00.40 | [TK]D-Fender | Scan the channel list. |
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| 14:00.56 | dan_j | Channel id always contains the sip peer name? |
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| 14:10.55 | Greenlight | Howdy folks. So say I did something silly and tried to MixMonitor with the argument File: / (Just the slash) then any chance the recording is going to be kicking around somewhere (I've checked on / and also in the spool dir)... I'm perhaps clutching at straws.. |
| 14:12.49 | file | just the slash? doubt it |
| 14:13.41 | Greenlight | Yea, I'm following app_mixmonitor.c through at the moment, and that seems to be the case ;( |
| 14:14.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:16.12 | file | although it doesn't fail, huh |
| 14:16.33 | Greenlight | Yea, it checks for zero length, a check that it passes.. |
| 14:17.33 | file | gimme a sec and I'll see where it goes... |
| 14:17.38 | Greenlight | It may have thrown an error further down. I called it via AMI |
| 14:18.13 | file | Greenlight, what is your Asterisk running as? |
| 14:18.18 | file | user wise |
| 14:18.26 | Greenlight | asterisk user |
| 14:18.37 | file | hum it probably failed due to permissions |
| 14:18.40 | Greenlight | Heh, this the time I wish it was running as root ;) |
| 14:18.55 | file | check if /.raw exists though |
| 14:19.09 | file | if it doesn't then it is gone |
| 14:19.26 | Greenlight | Nope, unfortunatly not |
| 14:20.58 | Greenlight | Thanks for taking the time to check it out though |
| 14:21.08 | file | you're welcome |
| 14:21.44 | Greenlight | I'll give the customer the news, and fix the race condition in my code |
| 14:30.16 | jameswf | malcolmd: is http://forums.digium.com/viewtopic.php?p=187179 still valid |
| 14:30.53 | file | yes |
| 14:31.08 | jameswf | okie dokie :) |
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| 14:37.54 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:49.24 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 14:49.36 | anonymouz666 | I am in |
| 14:50.08 | anonymouz666 | WIMPy: very nice stuff from polycom called micro browser and web kit |
| 14:50.10 | anonymouz666 | very powerful |
| 14:51.27 | WIMPy | Don't konw the Polycom stuff. |
| 14:51.51 | WIMPy | Apart from the conference phons they seem virtually non-existant on this continent. |
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| 14:54.07 | anonymouz666 | WIMPy: what do you have there? |
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| 14:55.00 | WIMPy | I prefer the old Snoms. |
| 14:55.12 | anonymouz666 | Snoms are very rare here |
| 14:55.18 | anonymouz666 | never saw one |
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| 15:13.43 | vlad_starkov | QUESTION: Is it possible to check whether an inbound call has been answered already? (to prevent unnecessary Answer() call) |
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| 15:14.10 | leifmadsen | vlad_starkov: I pretty much never use Answer() |
| 15:14.14 | leifmadsen | and it won't hurt having extras |
| 15:14.46 | leifmadsen | if you plan to playback a prompt, then just use Playback(silence/1) before your initial prompt to answer the call |
| 15:14.50 | leifmadsen | and get the RTP flowing |
| 15:15.12 | file | calling Answer() multiple times won't do anything |
| 15:18.56 | vlad_starkov | leifmadsen: I'm currently implementing call queues. It turned out that calling Queue(support) on incoming call doesn't provide RTP flowing, so that I added Answer() before Queue(...) |
| 15:19.16 | vlad_starkov | file: thanks |
| 15:19.23 | leifmadsen | vlad_starkov: right, you do need that |
| 15:19.33 | leifmadsen | queue won't answer for you |
| 15:19.37 | vlad_starkov | leifmadsen: ok, thanks |
| 15:21.48 | anonymouz666 | vlad_starkov: the only side effect not calling answer before queue, is that telco will hangup in 90 seconds (at least here) |
| 15:22.28 | leifmadsen | anonymouz666: that's probably because you're getting early media, in most of the carriers I've used I don't get early media, so the result is just no media at all (no-way audio) |
| 15:22.54 | vlad_starkov | anonymouz666: I got no media at all as leifmadsen said |
| 15:23.29 | vlad_starkov | Question: How rare should I reconnect to #asterisk so that it won't be considered as too often and I'm not to be banned? |
| 15:23.47 | leifmadsen | huh? |
| 15:23.56 | leifmadsen | never disconnect? |
| 15:24.10 | vlad_starkov | that would be awesome :-) |
| 15:24.21 | leifmadsen | unless your link is flapping, you'd have to be trying pretty hard to connect/disconnect/connect/disconnect |
| 15:25.33 | Qwell | vlad_starkov: Your connection was very bad that particular day. It's usually bad, and it would be preferred that you just not idle, if your connection just can't handle it. |
| 15:25.43 | vlad_starkov | leifmadsen: I'm using IRC on my macbook so it reconnects too often as I think now. |
| 15:25.59 | vlad_starkov | Qwell: oh, I see |
| 15:26.38 | vlad_starkov | Considering to setup ZNC http://wiki.znc.in/ZNC to be never disconnected :-) |
| 15:26.46 | leifmadsen | setup a server somewhere and use a bouncer like znc |
| 15:26.51 | leifmadsen | something with a more stable connection |
| 15:26.52 | Qwell | That would be a good alternative. |
| 15:26.57 | leifmadsen | like on digital ocean |
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| 15:28.03 | vlad_starkov | leifmadsen: that a good idea to use digital ocean for this kind of stuff |
| 15:28.29 | Greenlight | Damn your connection must be bad if it can't even keep an IRC session open |
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| 15:30.36 | vlad_starkov | Greenlight: I think it was that day when I was working in coffeeshop with broken wifi |
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| 15:44.09 | navaismo | morning! |
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| 16:09.21 | ipengineer | When trying to send a call to a queue I am getting “Unable to join queue queue_1”. Could it be because: “agent_5 (PJSIP/204 from PJSIP/204) (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet”? I am not sure what the INVALID means |
| 16:10.19 | navaismo | is not available and propably you have the joinempty setting =no |
| 16:10.27 | [TK]D-Fender | odds are pjsip was loaded after app_queue so it doesn't know about that type at all |
| 16:10.44 | [TK]D-Fender | And nothing to do with "unavailable" |
| 16:10.44 | ipengineer | I do have join empty set to no |
| 16:10.51 | [TK]D-Fender | nor that |
| 16:10.57 | [TK]D-Fender | the device itself is not valid |
| 16:10.59 | ipengineer | [TK]D-Fender: Let me look at the load order |
| 16:11.32 | navaismo | ipengineer, so fix your way to add your agent and then you can join the queue |
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| 16:12.10 | ipengineer | ahh yea ok that makes sense |
| 16:13.49 | [TK]D-Fender | Nothing wrong with the way the agent is added... |
| 16:14.07 | [TK]D-Fender | in terms of the tool |
| 16:14.33 | [TK]D-Fender | <PROTECTED> |
| 16:15.28 | navaismo | if he can show us the dialplan and thec li |
| 16:16.12 | ipengineer | It was the way I was adding agents the device is PJSIP/200_a not just PJSIJP/200 cant believe I missed that |
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| 16:47.34 | file | wobbles |
| 16:55.21 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 16:57.57 | ChkDigit | Who can recommend a decent 4-port FXO to ethernet SIP device? |
| 16:58.27 | ChkDigit | I need to replace some Mediatrix 1204s that are... misbahaving. |
| 16:59.00 | [TK]D-Fender | They hve newer models. Audiocodes is pretty solid as well. Many other broands out there as well |
| 17:01.01 | ChkDigit | Thanks! |
| 17:01.36 | Kobaz | ChkDigit: audiocodes |
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| 17:01.44 | Kobaz | ChkDigit: if you want cheaper but still good, cisco |
| 17:01.54 | Kobaz | less features |
| 17:02.19 | Kobaz | ChkDigit: did you firmware update the mediatrix? |
| 17:02.43 | ChkDigit | One popped in an electrical fault... |
| 17:02.55 | ChkDigit | The other randomly makes calls in. |
| 17:03.43 | ChkDigit | Which means I have no spares left. |
| 17:05.53 | Kobaz | yay |
| 17:05.54 | Kobaz | fun |
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| 18:01.19 | navaismo | how a good promising technology like webrtc convert into a PITA for all users :'( |
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| 18:06.08 | marceloamorim | guys, I have a problem with chan_sip.c, __sip_autodestruct autodestruct on dialog .... in place Method BYE, Rescheduling destruction for 10000ms |
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| 18:07.18 | marceloamorim | I call from PSTN, when I hangup the call the cli show me this warning and the SIP wasn't hangup |
| 18:08.49 | marceloamorim | the command hangup request shows me 2 options SIP/a-000000027 and Local/100@context |
| 18:12.20 | marceloamorim | but I couldn't hangup with this CLI>hangup request. |
| 18:14.23 | [TK]D-Fender | Show us |
| 18:14.25 | [TK]D-Fender | ~pb |
| 18:14.26 | 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. |
| 18:14.27 | [TK]D-Fender | ^^^^^ |
| 18:14.34 | [TK]D-Fender | This means a complete call with full SIP debug. |
| 18:14.42 | [TK]D-Fender | that warning line is basic verbose only... |
| 18:16.35 | marceloamorim | but the sip side does´t hangup |
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| 18:35.03 | marceloamorim | I´m try to rebuild the problem |
| 18:36.25 | marceloamorim | I called sometimes and didn´t happen again |
| 18:37.54 | navaismo | MAGIC |
| 18:38.10 | navaismo | thats why im try to be always here |
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| 19:00.00 | ipengineer | Isnt the timeout for queues supposed to be in seconds? Queue(queue_1,tn,,,300)? The queue times out at 30 seconds with this setting |
| 19:17.04 | [TK]D-Fender | What timeout are you describing? |
| 19:17.09 | [TK]D-Fender | There are multiple. |
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| 19:39.37 | ipengineer | [TK]D-Fender: The 5th option here says timeout that the queue will fail after X seconds. https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_Queue?src=search |
| 19:41.39 | [TK]D-Fender | will depend on other things to including retries, agents being available, etc |
| 19:52.25 | ipengineer | [TK]D-Fender: Ok. So if I have one agent in the queue it wont ring that agent for 300 seconds it will stop at 30? I have retry set to 5 |
| 19:53.44 | [TK]D-Fender | pastebin the whole thing..... |
| 19:54.40 | marceloamorim | I used tail -f /var/log/asterisk/full and I have http://pastebin.com/ynWy1euW |
| 19:55.20 | marceloamorim | maybe is a clue about that sip destroying problem |
| 19:57.27 | navaismo | marceloamorim, the destroying message is a normal message that notice about that, its not an error |
| 19:58.25 | marceloamorim | ok |
| 19:58.54 | marceloamorim | Iĺl ignore, thx |
| 20:03.06 | ipengineer | [TK]D-Fender: I am going to get you the queue config from the db real quick too.. https://gist.github.com/zconkle/da19e000de75b5c798c4 |
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| 20:06.08 | navaismo | what happen if you add the following 5 commas? |
| 20:07.23 | ipengineer | I just updated the config |
| 20:07.34 | ipengineer | Or the gist with the realtime info |
| 20:07.46 | ipengineer | Let me try to add the rest of the commas |
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| 20:08.06 | anonymouz666 | I am back |
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| 20:14.43 | ipengineer | I see there is a timeout in the queue table.. I guess that overrides the timeout on app_queue. I need to test that theory |
| 20:18.41 | marceloamorim | I copy the /var/log/asterisk/full and paste on pastebin, can you guys see any problem with those informations? http://pastebin.com/D8FxnBdg |
| 20:20.36 | navaismo | do you have an specific question |
| 20:20.51 | marceloamorim | I look to this var log asterisk full and can´t see anything wrong. but I never analyze those files |
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| 20:22.20 | navaismo | well the debug information is like a very detailed diary |
| 20:22.37 | navaismo | so you will find when the pbx farts and the color of the shit |
| 20:23.00 | marceloamorim | this logs is about a call from pstn, I wish to see something wrong that I could understand and help me with the pbx in general |
| 20:23.06 | navaismo | but if you disable the debug and only leave the verbose you will find only stuff like "Oh i popped" |
| 20:23.19 | navaismo | whats your issue |
| 20:23.22 | navaismo | ? |
| 20:24.13 | marceloamorim | the issue that I talk early about destroying channels, I afraid that happen again |
| 20:24.41 | marceloamorim | Iĺl disable the debug |
| 20:25.41 | navaismo | i missed that so your pstn call is stcuk even if you hangup the phone, is this an analog or digital trunk? |
| 20:25.56 | marceloamorim | analog, fxo trunk |
| 20:26.10 | navaismo | ah that is a common issue |
| 20:26.34 | navaismo | try adding the busydetect=yes and busycount=4 into your chan_dahdi.conf |
| 20:26.54 | navaismo | mostly caused by the lack of answer supervision in the analog trunks |
| 20:27.11 | marceloamorim | uhmm, nice tip |
| 20:27.25 | marceloamorim | I didn´t think about it |
| 20:27.51 | marceloamorim | busycount=6 |
| 20:28.13 | marceloamorim | I changed for 4 |
| 20:28.34 | navaismo | and also setup your indications.conf and the system.conf to your country |
| 20:29.37 | marceloamorim | I did on system.conf but I didn´t realized about indications.conf |
| 20:32.38 | [TK]D-Fender | checkout time, BBL |
| 20:36.25 | marceloamorim | thx navaismo, I did those tips that you said, probably will help me, I totally forget about indications.conf |
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| 20:45.28 | ipengineer | Ok so that timeout on the realtime queues tables is for the agent timeout. So ring each agent for 30 seconds but ring the queue for 300 seconds. I am not sure why the queue is timing out at 30 seconds |
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| 21:31.41 | ipengineer | In queues when the ring strategy is set to leastrecent how do we set a timeout to progress to the next agent? |
| 21:32.12 | ipengineer | If I set a timeout it ends up timing out the entire queue instead of progressing to the next available agent. |
| 21:32.33 | navaismo | w00T |
| 21:34.32 | navaismo | pastebin or didnt happen |
| 21:38.55 | ipengineer | navaismo: Here is one example with ring all. When I send the calls to the queue I have the timeout set at 300 seconds but that seems to not be acknowledged. https://gist.github.com/zconkle/f88d13c6c78affcbe251 |
| 21:46.54 | ipengineer | navaismo: I think my issue with the ring all is that I was passin the “n” option in the queue. Amazing how I can search for something for 30 minutes post on here and it jump out and slap me in the face |
| 21:47.08 | navaismo | hehe |
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