| 00:17.01 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 00:18.26 | hardwire | also.. can you prioritize contacts in an aor configuration to deal with failover outside of the dialplan? |
| 00:18.47 | hardwire | if they are ordered top to bottom that would be sweet. |
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| 00:29.56 | hardwire | https://inbox-tester.red/ |
| 00:30.02 | hardwire | fun example of datauri encoded background. |
| 00:30.11 | hardwire | I need to find the nerf widget for that sort of thing. |
| 00:30.30 | ilius | hardwire, instead of prioritizing contacts, just ring them all. |
| 00:30.32 | hardwire | Instead of manually setting those up I bet there is a nice purgecss function for that. |
| 00:30.36 | hardwire | oh.. wrong window. |
| 00:30.43 | hardwire | ilius: solid! |
| 00:31.34 | ilius | The PJSIP_DIAL_CONTACTS() function will give all of the endpoints in format for Dial. e.g.: |
| 00:31.40 | ilius | Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) |
| 00:31.58 | ilius | Er, all of the contacts in Dial() format |
| 00:32.57 | ilius | hardwire, https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_DIAL_CONTACTS |
| 00:33.53 | Samot | ilius: This is about failover. |
| 00:34.04 | Samot | Call contact 1 if contact 1 fails call contact 2 |
| 00:34.06 | hardwire | I mean.. it would handle that. |
| 00:34.17 | hardwire | just have some excess.. |
| 00:35.05 | ilius | Oh. He also said "outside of the dialplan" so I have no clue what to do there... |
| 00:35.15 | ilius | For failover, you just look at the DIALSTATUS and try another... But that's in the dialplan. |
| 00:36.51 | hardwire | yar. |
| 00:37.06 | hardwire | thanks. I figured this may end up being a dialplan based solution. |
| 00:37.26 | Samot | Well it's Asterisk, that's the only way. |
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| 00:37.31 | Samot | Unless you send the call to a proxy |
| 00:39.32 | Samot | You would basically need to pull the contacts and then loop through them one by one Dial()ing each. |
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| 01:09.06 | lembron | and what on earth does one do with a "SIP-UUI" - like i think i understand what it does, kinda --- but "my provider gave me one to use" - '>sip:+123456@172...<' - where and what do i put that? anything in google is just talkn about setting that on a call basis by "code" - i "want it for the trunk as config" or so, i think... bit lost :P |
| 01:09.24 | lembron | any pointers aprecciated |
| 01:10.19 | lembron | (the 123 is my number, but that LAN-ip, no clue where thats from) |
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| 01:14.00 | igcewieling | Are you sure that isn't a SIP-URI? |
| 01:14.14 | igcewieling | which is something you can google |
| 01:14.38 | lembron | 100% yep https://i.imgur.com/wKYSViC.png |
| 01:16.18 | igcewieling | I've not heard of it. |
| 01:16.27 | lembron | all in all how i got the details is a little confusing... - got a "SIP-User (AOR)" thats a "number@domain" -- "username" what a DIFFERENT number@domain -- and "SIP-Domain" again -- lookn at wireshark, one has to strip the domains from the first two.... |
| 01:19.31 | lembron | asterisk shows as "registred" in the peer status - but when i try to call out, i get "all is busy" |
| 01:19.41 | lembron | "user agent not registered here" https://i.imgur.com/Rk8Zk2o.png |
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| 01:20.17 | lembron | (that is, asterisk to trunk) |
| 01:25.08 | lembron | my config: https://paste.nginx.org/7b27c587e8c9a73e |
| 01:27.27 | lembron | asterisk itself just says "Received response: "Forbidden" from '<sip:987654@voice.UPSTREAM>;tag=as417d3007'" -- but given the register worked, i assume something is right, something is wrong, just no clue what :D |
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| 01:40.48 | lembron | and when comparing the register(that okays in the 2nd try with proxy) - with a call-out (that fails) -> reisters Authorization header says "uir="sip:UPSTREAM"" on okay - but then "sip:01...@UPSTREAM:5060" on the call -- what i guess technically makes perfect sense, but why does it break everythign :D |
| 01:41.15 | lembron | or actually, hold on... |
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| 01:44.22 | lembron | it kinda seems to whipe out the "From:"... |
| 01:45.18 | lembron | or is that expected as i have "defaultuser" and "fromuser"? - (and yep, i tried flipping the two around) |
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| 01:52.30 | lembron | and it appears as if my peer is a "innovaphone Virtual Appliance" |
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| 11:40.48 | Ast001 | Hello, I'd like to get only ready for calls agents from function QUEUE_MEMBER. Manual says this is syntax QUEUE_MEMBER(queuename,option), where option can be logged,free,ready or count. Problem is fact that with any option I get same result which is number of logged in agents. I think option ready should return only "agents ready to immediately receive calls". Am I doing something wrong ? |
| 11:42.02 | Ast001 | Is there any other function which can return this with interface names ? (Only not in use and not paused agents only agents which can receive calls in that queue according to asterisk engine) |
| 11:47.39 | Ast001 | my queue members are static agents from realtime queue_member table. |
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| 12:29.55 | adamp524 | Hi, I've got a Digium AEX2400 card that's DAHDI compatible populated with FX0 cards. I can see all channels listed withdahdi show channels |
| 12:30.37 | adamp524 | I'd like to try and interact with them using the ARI interface, but I'm not sure how to start. |
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| 12:31.58 | adamp524 | Since they are FXO cards, they can simulate an analogue telephone. I'd like to be able to use ARI to lift the receiver and dial a number. |
| 12:36.07 | Samot | When you say receiver you just mean take the lines off-hook? |
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| 12:48.06 | adamp524 | Samot - yes, take the lines off hook and dial |
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| 13:01.58 | jsmith | adamp524: Sure... you probably want to look at the Channel.originate command in ARI |
| 13:03.11 | jsmith | (Well, post to /channels to create a new channel) |
| 13:06.45 | adamp524 | @jsmith: I see POST channels and POST channels/create. What's the difference between these? |
| 13:09.31 | jsmith | adamp524: Honestly, I don't remember... the POST to channels/create seems to be new since I last did a lot of work on ARI |
| 13:10.18 | file | https://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ |
| 13:11.04 | file | that talks about the difference |
| 13:11.20 | jsmith | Ah, perfect! |
| 13:15.14 | adamp524 | Great, I think channels/create is what I'm looking for. thanks! |
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| 13:16.24 | file | the normal one is usually what people use |
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| 13:36.53 | Ryushin | Just curious, I have not seen [TK]D-Fender here. Is he still around? |
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| 13:55.47 | sibiria | been a while since i saw him |
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| 14:15.12 | seanbright | he's joined as recently as yesterday |
| 14:16.47 | seanbright | sorry, not yesterday, sunday |
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| 17:16.16 | ilius | I'm working on migrating to pjsip and one thing I'm stuck on is that when the dialed extension rejects the call with a "603 Decline", instead of getting a DIALSTATUS of BUSY, I'm getting a dialstatus of CHANUNAVAIL, so my voicemail always does the unavailable message even on reject. |
| 17:17.14 | ilius | Is this new/expected behavior? Or am I supposed to set a flag or something to make it work as I expect it to? |
| 17:19.20 | Samot | A 603 Decline isn't a BUSY |
| 17:21.00 | ilius | That's what the polycom 650 sends when I push the reject button. Hrm, let me do a debug again. |
| 17:21.14 | Samot | But that's not a BUSY |
| 17:21.18 | Samot | That's a REJECT |
| 17:21.50 | Samot | That could be the only call you have hitting the phone, you hit reject... |
| 17:22.00 | Samot | You're not busy, you're just declining the call. |
| 17:23.36 | ilius | Okay. I mostly just want to get the old behavior back where I can give them the busy voicemail message when they reject. If my memory serves me, chan_sip would not report CHANUNAVAIL on reject, I guess I should test that. |
| 17:24.21 | Samot | Chan_PJSIP follows RFCs more closely |
| 17:25.08 | Samot | That's the thing most people don't get when they say "I tried Chan_PJSIP, it didn't work. Moved back to Chan_SIP, it just worked" |
| 17:25.29 | Samot | Probably because Chan_SIP allows more non-compliant and wrong things. |
| 17:26.19 | ilius | Oh, I'm deep down the migration rabbit hole... No going back. I just need to restore behavior that my users expect. Maybe there's a config parameter for polycom to send the right sip busy. |
| 17:27.51 | ilius | Thank you kindly Samot. |
| 17:28.33 | Samot | Maybe. I know some offer DND choices, either 486 or 480 |
| 17:29.25 | ilius | Yeah, I restrict my agents from using the DND. The DND button goes to an app that pauses their queues instead. |
| 17:32.30 | Samot | voIpProt.SIP.use486forReject |
| 17:32.52 | ilius | Yes! I was just looking at that! https://community.polycom.com/t5/VoIP-SIP-Phones/VVX-reject-button-5-3-0/td-p/73459 |
| 17:33.35 | Samot | Well, you need to look at the admin guide for your firmware to verify. |
| 17:38.18 | ilius | Right, that page is for VVX, I'm still using these piece of crap SPIP650s. :) |
| 17:42.18 | ilius | WOO! It works! Default behavior is 603 (sigh), but that parameter made it become 486! Awesome. One more step closer to pjsip! :) |
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| 18:04.35 | igcewieling | Has anyone here used "Polycom Device Management Service for Service Providers" Looks to be a Polycom managed provisioning service. |
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| 18:11.19 | ilius | Wow, I hadn't heard of that, igcewieling. I've always just run my own web server for it. |
| 18:15.58 | igcewieling | ilius: I use an FTP server, but I sometimes consider "cloud based solutions". |
| 18:16.39 | ilius | Yeah, I don't even use Asterisk's internal phoneprov stuff. :D |
| 18:16.52 | ilius | I did FTP for a while, but that protocol was too slow compared to HTTP. |
| 18:20.35 | igcewieling | I wrote our own polycom provisioning GUI about 10 years ago, it is simple, but does what we need it to. |
| 18:21.27 | ilius | there is beauty in simplicity :) |
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| 18:24.00 | igcewieling | At some point I'll convert it to generate configs on the fly from a database holding the info and use https, but it isn't a priority. |
| 18:31.11 | ilius | I have all my polycom phones uploading their logs (which is very convenient for troubleshooting) and persist user customizations too (by logged-in extension, not device). |
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