| 00:00.39 | *** join/#asterisk jrabe (jrabe@janikrabe.com) |
| 00:02.25 | hdon | oh hm yes i am |
| 00:04.17 | hdon | or there's some kind of a race condition and my simple test doesn't expose it... |
| 00:04.46 | hdon | oh whoops... used Promise.join when i meant Promise.map :| |
| 00:05.23 | hdon | is there a way to just get all variables on a channel? |
| 00:10.35 | Samot | hdon: I believe there is an #asterisk-ari channel these questions might be more suited for. |
| 00:10.50 | hdon | ah thanks Samot |
| 00:11.25 | Samot | You may get faster, better and/or more in-depth answers there. |
| 00:12.06 | Samot | I mean ARI, sure some of those could be answered here |
| 00:12.16 | Samot | But node-ari is getting a wee specific. |
| 00:12.33 | Samot | That might be more a thing the ARI-heads know about. |
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| 01:43.01 | tupton | stupid question because i don't know what to google. so a did number aka telephone number a person can get for free or dirt cheap. but what about service to that phone number? do i have to pay so much for talking on that number? or is that just covered under my isp bandwidth? |
| 01:43.41 | tupton | i plan on setting up freepbx. |
| 01:44.19 | tupton | are there any services i have to pay for besides the did number? considering i'll be running my own pbx |
| 01:44.22 | Samot | That question is in no way related to the type of PBX |
| 01:44.45 | Samot | tupton: VoIP = Phone Service |
| 01:44.56 | Samot | A SIP Trunk = Line |
| 01:45.15 | Samot | You would have to pay for your service just like any other voice service. |
| 01:45.35 | Samot | What plans/rates that are offered is up to the provider you choose. |
| 01:46.13 | tupton | so i run an asterisk server, but then still need to have a voip provider provide me service? so that's what i'd look into? voip providers |
| 01:46.21 | Samot | Yes. |
| 01:46.27 | Samot | ITSP/SIP Trunk Providers |
| 01:46.34 | tupton | gotcha. thank you |
| 01:46.38 | Samot | Think of the Asterisk box as a really big phone |
| 01:46.42 | Samot | That does a lot of stuff. |
| 01:46.51 | tupton | not asterisk related entirely, but this is the closest related freenode channel |
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| 01:46.55 | Samot | You still need to plug it into the phone line, so to speak. |
| 01:47.05 | tupton | understood. thank you |
| 02:22.11 | [TK]D-Fender | Whoeve you get the DID from gives you the service. |
| 02:22.18 | [TK]D-Fender | And by "give" we mean "charge you for" |
| 02:22.31 | [TK]D-Fender | It's more like you're renting the number |
| 02:22.59 | [TK]D-Fender | Having a number assigned to you does not necessarily have any relation on how many calls you can take or place however |
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| 08:48.09 | zicada | I am currently in the process of troubleshooting some problems we have with leaking event fd's. Could anyone here answer if these are "normal" in the logs ? |
| 08:48.12 | zicada | [Sep 6 10:41:55] WARNING[16408] res_http_websocket.c: Web socket closed abruptly |
| 08:48.15 | zicada | [Sep 6 10:41:55] WARNING[16408] ari/ari_websockets.c: WebSocket read error: Success |
| 08:48.33 | zicada | What is successful avout the read error I guess is what I'm asking :) |
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| 09:12.09 | *** join/#asterisk chl_ (~chl@unaffiliated/chl/x-9330839) |
| 09:12.48 | chl_ | hello, having some issues with call forwarding - no sound is coming through, but calls are being forwarded |
| 09:12.59 | chl_ | i have set up firewall rules |
| 09:13.59 | chl_ | should a rule for rtp - 10000:20000/udp be sufficient? |
| 09:17.53 | zicada | chl_: sudo lsof -p <pid of asterisk> | grep LISTEN |
| 09:18.10 | zicada | Will tell you what ports and protocols to open |
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| 09:32.07 | dan_j | chl_: does your asterisk box have a public IP or are you behind a router? |
| 09:44.20 | chl_ | dan_j: public IP |
| 09:45.01 | sibiria | i have a bit of a basic question |
| 09:45.28 | sibiria | the X-Asterisk headers that pop up when receiving BYE |
| 09:45.36 | sibiria | are these determined and set by us, or the remote side? |
| 09:46.53 | sibiria | we're getting cause-id 1 (unallocated/404) on calls that were correctly connected, proceeding normally |
| 09:46.54 | chl_ | zicada: asterisk 1463 asterisk 18u ipv4 17901 0t0 TCP *:cisco-sccp (LISTEN) was the only thing returned |
| 09:57.04 | pawiecki | Hi, I have a mechanizm to handle sip peers' restriction, for example local =1, same-country=2, mobile=3, international=4 and so on. Right now I set variable for every peer in sip.conf, and then retrieve level of restrictions before outbound call. Now I'm curious if there's a simpler way of doing this. |
| 09:57.29 | pawiecki | mechanism* |
| 09:58.32 | pawiecki | I use agi script, which does 'sip show peer x' for every account and then search for variable 'restriction' |
| 10:00.26 | pawiecki | that way, when peer A calls peer B, and B redirects to external number, * uses restrictions for B, instead of A. |
| 10:04.43 | pawiecki | Is there any more elegant solution for assigning restrictions for outgoing calls? |
| 10:09.08 | dan_j | chl_: If asterisk has a public IP and there is no NAT router involved in the setup, the audio port numbers are contained in rtp.conf |
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| 11:42.17 | chl_ | thanks dan_j |
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| 15:18.38 | imcdona | Which modules are required for IAX2 support in Asterisk? I'm getting "Auto-congesting call due to slow response" on IAX2 calls. If I set auto-load modules to yes it works fine so I must be missing a module that IAX2 depeonds on. Anyonw have any ideas? |
| 15:18.57 | imcdona | obviously I've included chan_iax2.so |
| 15:19.22 | imcdona | or what module would control response to a channel being opened? |
| 15:26.51 | newtonr | DId you try turning on debug and turning the debug verbosity way up? Might get more clues. |
| 15:27.09 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 15:31.40 | imcdona | Yeah, I did that already. I found this in the debug log " Can't find native functions for channel 'IAX2/" Not sure if that's realted. All codecs are loaded. All bridging modules are loaded. I'm literally loading modules one-by-one and attempting a dial until I find the module resposible |
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| 16:04.38 | sibiria | g.722 - is it somewhat deployed and usable or is it recommended to be avoided in favor of g.711? |
| 16:06.29 | Samot | g722 is great for extension to extension calling |
| 16:06.38 | Samot | But there is no g722 over the PSTN |
| 16:07.09 | Samot | Your provider, if even they accept g722, will transcode it to g711/g729/gsm which every is best for them. |
| 16:10.08 | sibiria | oki, ty |
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| 17:23.44 | imcdona | The module resposible for the IAX2 calls getting "auto-congesting call due to slow response" is "res_timing_pthread.so". Asterisk fully loads with no complaints but won't complete IAX2 calls unless that module is loaded |
| 17:27.48 | [TK]D-Fender | You need something for timing for IAX in trunk mode |
| 17:28.05 | [TK]D-Fender | So either a kernel based or DAHDI based module loaded |
| 17:28.07 | imcdona | How about making res_timing_pthread.so a dependency of chan_iax2.so so unless res_timing_pthread.so is loaded then fail to load chan_iax2.so? |
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| 17:29.04 | imcdona | Or make a timing source a dependency on chan_iax2? As it stands now, there's no error, just lots of frustation should anyone make the same mistake. |
| 17:31.31 | [TK]D-Fender | If you're going to cherry-pick your modules you should be aware of what they require... |
| 18:10.03 | imcdona | Is there a list of module dependencies I'm not aware of? |
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| 19:09.04 | bluez_ | hey guys |
| 19:09.51 | bluez_ | is there an easy way to make a exntension/dialplan rule self destruct once dialed? |
| 19:09.55 | bluez_ | i.e. it can only be dialed once |
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| 19:12.41 | Kobaz | [TK]D-Fender: yo yo |
| 19:12.52 | Kobaz | it's my favorite fender |
| 19:13.47 | Kobaz | anyway... anyone know if you can... on a call by call basis, have a polycom phone not show a call as 'missed'. Like from a hunt group... phone A/B/C ring, A picks up, B/C get 'missed calls' |
| 19:14.00 | [TK]D-Fender | bluez_, Check some value as the first priority. If it's set then don't continue. If you continue, set the flag |
| 19:14.17 | [TK]D-Fender | bluez_, Or something else |
| 19:14.53 | [TK]D-Fender | bluez_, put that dialplan in a separate file and INCLUDE it. Then in your dialplan overwrite that sub-file. |
| 19:14.53 | bluez_ | ok but i want it removed permanently |
| 19:15.10 | Kobaz | i know polycoms have a feature... Since UCS 4.0.0 we do support RFC 3326 "Call completed elsewhere" and this would clear the missed call if your SIP server supports this. |
| 19:15.17 | [TK]D-Fender | These are text configs. |
| 19:15.19 | Kobaz | not sure how to use it from asterisk |
| 19:15.26 | [TK]D-Fender | fuck with the files however you want |
| 19:15.45 | [TK]D-Fender | Kobaz, "core show application dial" <----------- |
| 19:16.07 | Kobaz | interesting... so it's in there somewhere |
| 19:16.44 | Kobaz | oOOo c: If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere' |
| 19:16.51 | Kobaz | iiinterrressting |
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| 21:34.46 | cervajs2 | any hints why ARI DeviceStates shows nothing but DeviceStates/SIP/test shows correct output ? |
| 21:40.11 | sibiria | is there a way to run a series of Application/Data sets from a call file without referring to a context? |
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| 22:13.41 | reveal | is it possible with asterisk to do a recursive ldap search or a way of automating adding the members automatically? |
| 22:16.22 | file | sibiria: no. |
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| 23:50.32 | warewolf | anybody here an asterisk core coder type of person? |
| 23:51.02 | warewolf | I'm trying to figure out where I need to hack asterisk source to add caller ID *name* to calls coming in over xmpp/motif (google voice) |
| 23:51.40 | Samot | Why even bother? |
| 23:51.52 | warewolf | I see in the xmpp debugging a <nick:nick>CallerName IsHere</nick:nick> stanza, trying to track down where that comes in, and how I stash it in caller ID |
| 23:51.54 | Samot | And you don't need to hack anything for that. |
| 23:52.41 | warewolf | okay |
| 23:52.42 | warewolf | listens |
| 23:53.13 | Samot | Well... |
| 23:53.40 | Samot | 1) Google officially ended third-party XMPP connectors to Google Voice. |
| 23:54.07 | warewolf | hold on |
| 23:54.09 | Samot | 2) Google Voice is now a Google only thing. Have Hangouts, you got Google Voice. It's for Google to Google only stuff now. |
| 23:54.15 | Samot | June 26th 2017 |
| 23:54.17 | warewolf | newp |
| 23:54.21 | Samot | Yewp |
| 23:54.32 | Samot | Some of the connectors may still be out there.. |
| 23:54.37 | Samot | Some may still be working |
| 23:54.39 | warewolf | you're wrong, becuase I'm placing and receiving calls via asterisk over xmpp/motif |
| 23:54.40 | Samot | But it's done. |
| 23:54.45 | Samot | OK. |
| 23:55.00 | warewolf | I *know* google has said it's going away, and has *for ages* |
| 23:55.21 | warewolf | anyhoo |
| 23:56.08 | warewolf | rewind, back to my question: where/how could I hack in caller id names from the XMPP <nick:nick>Stuff here</nick:nick> xml node? |
| 23:56.46 | Samot | Don't know. |
| 23:56.53 | warewolf | okay then. |
| 23:56.54 | Samot | First is has to be sent. |
| 23:56.58 | Samot | Is it being sent? |
| 23:57.01 | warewolf | yep |
| 23:57.10 | warewolf | captured it while my fiancee called me. |
| 23:57.13 | Samot | How did you confirm that? |
| 23:57.21 | Samot | OK so Asterisk shows the callerid? |
| 23:57.23 | Samot | In the logs? |
| 23:57.33 | Samot | CDRs, etc... |
| 23:57.35 | Samot | ? |
| 23:57.36 | warewolf | no |
| 23:57.41 | warewolf | in XMPP debugging |
| 23:58.08 | warewolf | I switched xmpp debugging on in asterisk with 'xmpp set debugging on' or whatever the command is |
| 23:59.56 | Samot | Show a call that this happens on |