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| 01:04.22 | scv | ah power outage |
| 01:04.25 | scv | hdon you still around? |
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| 07:48.55 | Haris | how can one check loaded asterisk mods ? |
| 07:51.57 | Haris | found it |
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| 07:56.06 | Haris | tlsenable=yes goes into http.conf for webrt / ws calls ? |
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| 07:56.30 | Haris | I'm on https://www.youtube.com/watch?v=GHFduPTNE1Q at 18:08 |
| 07:57.44 | Haris | its in sip.conf |
| 07:59.55 | Haris | I have tlsbindaddr=0.0.0.0:5061 where-as in this video it says to have the port be 8089 in case tls is enabled. is that correct ? |
| 08:07.33 | Haris | it says asterisk 11/12 has spotty support for webrtc without patches ? |
| 08:07.47 | Haris | around 20:00 |
| 08:10.39 | Haris | how to enable https server for ws calls on server side ? |
| 08:11.25 | Haris | http show status doesn't mention the https part. its not enabled ? |
| 08:11.33 | Haris | I have ssl cert installed |
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| 09:05.46 | HCorn | Hi! Does anyone know if it is possible to pass options to the Dial application in Asterisk call files (I am looking to pass the h and H options to the Dial that is executed by the call file)? |
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| 09:36.52 | sehh | hey there |
| 09:38.03 | sehh | q: I recently got a new Grandstream phone (GPX2135) which supports "HD Voice". From what I understand, its just support for wideband codecs. Does anyone know which codec I can use in asterisk with this phone to take advantage of this feature? |
| 09:56.02 | TandyUK | sehh: https://www.opus-codec.org/comparison/ |
| 09:56.10 | TandyUK | that page has a nice chart showing the different codecs |
| 09:58.10 | sehh | thanks for that, looks interesting. I always thought that g711 (alaw/ulaw) and g722 had similar quality, their graph shows g722 as being a lot better... hmmm |
| 09:58.35 | sehh | plus, g722 seems like the only real wideband choice |
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| 12:10.09 | gingitsune | Is it possible to forward a call via the REST api? |
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| 12:21.10 | file | what does "forward a call" mean? |
| 12:22.07 | gingitsune | Redirect a incoming call from 1 recipient to another |
| 12:22.42 | gingitsune | We would like to define some rules that route calls to certain recipients or give a recieving user the chance to route a call to someone else |
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| 12:26.03 | file | ARI only allows you to "redirect" if the channel is in your ARI application |
| 12:26.24 | file | AMI allows you to grab any channel and send it elsewhere in the dialplan |
| 12:27.49 | gingitsune | Would it be crazy to all of the channels inside of one application? |
| 12:28.24 | file | depends on what you're trying to achieve... |
| 12:28.36 | file | but from an ARI application you can't execute dialplan applications... |
| 12:29.12 | gingitsune | Well I just started reading the doc just as I asked the questin |
| 12:29.39 | gingitsune | So to be frank, I'm not quite sure whats a dialplan |
| 12:30.22 | file | "The dialplan is the heart of Asterisk. All channels that arrive in the system will be passed through the dialplan, which contains the call-flow script that determines how the incoming calls are handled." |
| 12:31.02 | file | dialplan applications being the things that are executed which encompass specific logic - so dialing something, playing back a sound prompt, others |
| 12:31.28 | gingitsune | so thats the AGI interface, correct? |
| 12:31.41 | file | no. |
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| 12:32.27 | file | AGI is a method for executing an outside program, or connecting to one over TCP, and accepting commands from it to do things on a channel |
| 12:33.08 | gingitsune | I'm srry, you did say that once |
| 12:33.48 | gingitsune | Pardon my ignorance, but what excatly is a channel? |
| 12:35.51 | [TK]D-Fender | a call in progress |
| 12:36.24 | [TK]D-Fender | ARI & AGI are more advanced things. You need to learn your basics first. |
| 12:36.47 | gingitsune | Yeah, it seems I have spin it up 1st |
| 12:37.27 | gingitsune | I just wanted to know if achieving the "forwarding" functionality from a web interface is feasible, before getting too invested |
| 12:37.50 | [TK]D-Fender | can be done via AMI |
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| 13:15.25 | stefan27 | I recall there's a document somewhere explaining the scope, ups and downs of AGI, AMI and ARI |
| 13:15.39 | stefan27 | but not the location... probably somewhere on the wiki |
| 13:15.41 | file | there's a wiki page |
| 13:15.57 | file | tells you where stuff fits |
| 13:16.01 | file | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
| 13:16.18 | Haris | ok, so ws connection works on port 8088 |
| 13:16.43 | Haris | I checked with ws tool installed through npm |
| 13:16.55 | stefan27 | That's the one! |
| 13:17.13 | Haris | found out about it on the youtube video |
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| 13:19.06 | gingitsune | That was the one I stumbled on to |
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| 13:22.02 | blackstar | Hi, I am a newbie so please let me know of any protocol breaches. I have asterisk 13.7.2 on Ubuntu 14.04 and experiencing problem of some users being unable to recieve calls (the caller recieves busy tone), outbound calls are unaffected. Router upgrades help and it seems to occur mostly where there is a VDSL connection passing through network provider's modem. Anybody had similar/know why this happens? |
| 13:23.42 | shido6 | how many on VDSL? |
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| 13:27.53 | [TK]D-Fender | Look at the call |
| 13:28.04 | [TK]D-Fender | "busy tone" doesn't say anything meaningful. |
| 13:28.10 | [TK]D-Fender | You need actual call debug |
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| 13:39.58 | Haris | is there a tool or way to telnet ws port on asterisk and conduct manual ws registration and call attempt like one can do smtp test through telnet |
| 13:40.38 | Haris | like there's ws tool, I found out, installable through npm |
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| 13:44.30 | Haris | I can connect to ws port (8088) on asterisk box via the ws/wscat tool. but don't know how to conduct exchange of headers to manually check how it works |
| 13:45.43 | [TK]D-Fender | Go read the RFC's |
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| 15:05.53 | NinjaJon | Hi guys, I am upgrading from 1.8 to 13, and using PJSIP for the first time. I'm getting the error "digest_create_request_with_auth_from_old: Unable to create request with auth.No auth credentials for any realms in challenge." when making an outbound call via my (PJ)SIP trunk, and I am a bit stuck with what I need to check next. Can anyone give me any pointers? I have found example configs and such, but nothing that looks as though |
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| 15:14.48 | NinjaJon | I used the sip_to_pjsip.py script to migrate from sip.conf to pjsip.conf - the generated config looks reasonable but I'm not sure what would be missing :( |
| 15:16.55 | NinjaJon | I have a 'type=auth' section in pjsip.conf; it contains auth_type=userpass, username=xxx, password=yyy and realm=zzz |
| 15:21.55 | *** join/#asterisk polysics (~polysics@88.147.57.236) |
| 15:22.08 | polysics | hello! I have a pesky issue to troubleshoot |
| 15:22.22 | polysics | my init script results in Asterisk not creating asterisk.ctl |
| 15:22.32 | polysics | running asterisk or safe_asterisk manually work |
| 15:22.52 | polysics | and the fun thing is that I copied the script to my home folder, and running it from there works. |
| 15:23.07 | polysics | it has to be something the service handler itself does |
| 15:23.11 | polysics | I am on CentOS |
| 15:23.30 | polysics | verison 7 |
| 15:23.52 | polysics | anyone ever seen that or can point me in a direction? it's puzzling to say the least |
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| 15:51.11 | alemstrom | Morning everyone, any one had an issue with music on hold music playing fast and playing slow |
| 15:52.14 | alemstrom | its in a virtualized vmware container |
| 15:56.26 | rrittgarn | just upgraded to 13.10 from 13.4 and ami redirect is now behaving a bit differently. Specifically we use an app to AMI redirect things to a parking extension, that calls the park app. Calls come in through app_queue are not dropping the original channel and when I try to channel request hangup the original channel (shown by core show channels) the system says the channel does not exist. |
| 15:56.59 | rrittgarn | calls that don't come through app_queue behave 'normally' or as expected |
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| 16:57.20 | hdon | scv, yes |
| 16:57.39 | hdon | well, i am now |
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| 18:38.10 | scv | am i correct in understanding there's no way to list mwi/dialog subscriptions under pjsip (similar to chan_sip sip show subscriptions) |
| 18:45.29 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 18:45.42 | jrun | TEST_FRAMEWORK is missing! |
| 18:45.51 | jrun | any idea how to get it back in menuselect? |
| 18:46.04 | jrun | i do configure with --enable-dev-mode btw |
| 18:49.34 | file | scv, not currently - issue is open at https://issues.asterisk.org/jira/browse/ASTERISK-23828 |
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| 18:59.15 | jrun | all test_* modules have XXX in menuselect |
| 18:59.46 | jrun | and make distclean doesn't do anything either |
| 18:59.48 | jrun | any idea? |
| 19:00.47 | file | where have you looked for it? |
| 19:01.06 | file | it's in Compiler Flags all the way at the bottom |
| 19:01.29 | jrun | yep, it's missing |
| 19:01.49 | jrun | i have another system (gentoo) which has that option but not the other system (slackware) |
| 19:01.56 | file | what version of Asterisk? |
| 19:02.07 | jrun | 11.23 |
| 19:02.11 | jrun | on both machines |
| 19:02.28 | jrun | in fact i have a dev branch i push from gentoo to slackware |
| 19:02.59 | file | do you have a cflags-devmode.xml file in the build_tools directory? |
| 19:03.37 | jrun | yes |
| 19:03.47 | file | and is TEST_FRAMEWORK in it? |
| 19:03.48 | jrun | and it does have TEST_FRAMWORK element |
| 19:04.07 | file | hrm! nothing else springs to mind |
| 19:04.18 | file | are there any other missing options? |
| 19:04.57 | jrun | dunno but lots of XXXX options |
| 19:06.39 | jrun | is there a *yesall* to the config system? |
| 19:08.00 | file | um menuselect/menuselect --enable-all |
| 19:11.20 | jrun | could be because of libasteriskssl ? |
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| 21:13.29 | scv | file: any alternative method to pull that data? or should i start looking at implementing it myself |
| 21:13.44 | file | it may be accessible over AMI |
| 21:14.49 | scv | ouch, open for 2 years and no updates :( must not be in high demand |
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| 22:19.20 | SpaceInvaders | How do you stop a run-away CLI when it's throwing up lines so fast you can't access the CLI? |
| 22:19.41 | SpaceInvaders | I tried just typing in core stop now with lines flying by but no luck |
| 22:20.46 | robmal | Set messages to verbose in logger.conf |
| 22:24.11 | SpaceInvaders | I looked at logger.conf but don't understand. do you mean change it from |
| 22:24.12 | SpaceInvaders | messages => notice,warning,error |
| 22:24.17 | SpaceInvaders | to messages => verbose |
| 22:24.18 | SpaceInvaders | ? |
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