| 00:49.17 | velix | Is it better to play silence or wait() ? |
| 00:51.32 | *** join/#asterisk scampbell (~scampbell@mail.scampbell.net) |
| 01:00.32 | Kobaz | depends what you want to accomplish |
| 01:00.59 | Kobaz | I've had issues with certain systems where doing an Answer(), Wait(1) would not start the audio stream, but doing Playback(silence/1) did |
| 01:01.06 | Kobaz | depends what's on the other end |
| 01:01.22 | Kobaz | I haven't needed to use those sorts of hacks since Asterisk 1.8/11 though |
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| 08:29.08 | _abc_ | I made the androids work with the * on lan, there was one more firewall on which I had not accounted for. It's still not possible to qualify local (non nat) clients in asterisk. |
| 08:29.30 | _abc_ | It must be a version thing, it works with another version of * at another location. |
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| 08:36.25 | sibiria | velix: you can't really customize anything with cdr_csv. that module has a more or less fixed set of records it stores per call. cdr_custom is what you want to use for customized CSV-based CDRs |
| 09:18.36 | _abc_ | Has anyone got a setup for lumicall app settings with plain SIP? Non sip_srtp, i.e. port 5060 instead of 5061? |
| 09:19.04 | _abc_ | It seems the apk I got from f-droid.org is broken, does not want to honor 5060 at all |
| 09:37.56 | velix | sibiria: Okay, I thought, this would have been possible. |
| 09:38.05 | velix | hmm, Anyone with an idea, why i get hundrets of "Received SSL traffic on RTP instance '0x5649d9b53f40' without an SSL session" ? |
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| 10:35.11 | Cyrillax | velix: http://lists.digium.com/pipermail/asterisk-users/2015-February/285720.html |
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| 10:48.58 | velix | Cyrillax: Yeah, I've read this yesterday night. So I need an TLS setup? |
| 10:49.56 | _abc_ | So, does anyone have a non SRTP settings set for lumicall and asterisk? On lan for now? |
| 10:50.34 | _abc_ | I tried v199 latest and v198 of lumicall and neither connects to the set registrar and sip server asterisk on lan. Trying to find out with tcpdump where the heck it is going. |
| 10:57.05 | Cyrillax | velix: If that's legitimate traffic, then yea. I would trace the traffic wireshark and make sure it's legitimate. |
| 10:58.07 | velix | Cyrillax: It's between the commercial SIP provider and my Asterisk PBX. Maybe he uses cryption. |
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| 12:09.01 | velix | Does "Locally RTP bridged" mean, my PBX mixes the RTP streams? |
| 12:09.50 | file | it forwards the RTP packets, yes |
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| 12:43.51 | velix | I've still have problems with one trunk (the PBX of our University). My PBX connects their PBX. I also have a commercial SIP trunk. The clients are connecting their softphone to my PBX. Then I'm originating a call to the other party and to the client. Asterisk does a simple or native bridge here. It works perfectly over the commercial trunk, but over the University, nobody can hear the other party. Kobaz did a great analysis and it's a fault of their |
| 12:43.52 | velix | PBX. Now the admin told me, I need a STUN at the clients side. That's totally wrong, isn't it? |
| 12:46.10 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:46.58 | Cyrillax | velix:Not necessarily. If the University PBX is behind a NAT or even just a firewall, and asterisk isn't in the media stream, then the device between the univ PBX and the client might not recognize the RTP packets going to the university PBX as authorized |
| 12:47.28 | Cyrillax | If you place a call directly from * to the university PBX, do you get 2-way audio? |
| 12:47.59 | velix | Sometimes. It sometimes works, sometimes doesn't. |
| 12:48.12 | velix | The admin says: coincidence, since the right port was hit. |
| 12:48.28 | Cyrillax | And is the university PBX NAT'd or just firewalled |
| 12:48.42 | Cyrillax | sounds like some kind of NAT based on what the admin says, so i'll rephrase |
| 12:48.50 | Cyrillax | does the university PBX have a public IP address |
| 12:48.52 | velix | university is NATed and firewalled. |
| 12:49.46 | velix | Cyrillax: even worse, I need a real. The host is 192.168.103.10, and the outboundproxy is via TCP (the public IP) |
| 12:49.51 | velix | realm* |
| 12:50.19 | velix | So it's my.pbx -> pbx.univer.city -> 192.168.103.10 |
| 12:50.36 | velix | directmedia is set to no |
| 12:50.44 | Cyrillax | Right, okay, well it sounds like you got the registration and calling authentication working |
| 12:51.01 | velix | yes |
| 12:51.10 | velix | It sometimes breaks... but that's another story :D |
| 12:51.22 | Cyrillax | okay so here's what you need to do |
| 12:51.34 | Cyrillax | you need to setup to use ice between itself and the university |
| 12:51.40 | Cyrillax | setup asterisk to use ice8 |
| 12:51.44 | Cyrillax | ice* |
| 12:52.03 | Cyrillax | then as long as your * box is on a public IP, your users shouldn't need to use ice |
| 12:52.15 | velix | mine is on public IP. |
| 12:52.24 | velix | icesupport=yes <-- set and now in rtp.conf |
| 12:53.02 | velix | let me look up, what rtp.conf needs for ice ;) |
| 12:53.07 | Cyrillax | you also need to setup the stun and turn server config |
| 12:53.12 | Cyrillax | https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+%28ICE%29+in+Asterisk |
| 12:53.30 | velix | yeah |
| 12:53.37 | Cyrillax | ICE basically uses STUN and then TURN if STUN fails, and ICE coordinates it all |
| 12:53.58 | velix | I've got a STUN server from my commercial trunk. |
| 12:54.03 | velix | But where do I get a TURN server? |
| 12:54.44 | file | ICE requires support on both sides, you can't just turn it on in Asterisk |
| 12:55.05 | velix | So the University's PBX also need it? |
| 12:55.18 | file | that is how ICE works, yes |
| 12:55.26 | velix | It's a innovaphone... I don't know ... let me look it up |
| 12:55.56 | Cyrillax | hm I thought you said the admin of the network said to use ice |
| 12:56.12 | Cyrillax | i see now he didn't actually say they used it |
| 12:57.31 | velix | He said, my clients (!) should use a STUN |
| 12:57.55 | file | STUN standalone provides discovery of external IP address |
| 12:59.55 | velix | But for which side? My clients might be behind a nat (router at home), but it works with my commercial SIP trunk. |
| 13:00.01 | velix | Several hundret calls without a failure. |
| 13:00.17 | Cyrillax | Right it sounds like the issue is between your * and the university |
| 13:00.20 | velix | But about every 2-3 calls through the University trunk fails. |
| 13:00.37 | Cyrillax | Fail means one-way audio? |
| 13:00.38 | Samot | Then here's an idea. |
| 13:00.42 | Samot | Stop using them. |
| 13:01.16 | Cyrillax | +1. May I ask what you need to connect to their PBX for anyway? |
| 13:01.18 | Samot | They have been problematic since the start, they lack proper support.... |
| 13:01.41 | Samot | They can't even tell you why they randomly send back 403 Forbidden errors. |
| 13:02.29 | velix | Cyrillax: no way audio :-) |
| 13:02.37 | velix | Cyrillax: Nobody can hear anone. |
| 13:02.55 | velix | Cyrillax: They've got a flatrate on mobile phone numbers. |
| 13:03.11 | velix | Samot: Yeah, funny thing. The 403 errors seem to be common for Innovaphone. |
| 13:03.18 | velix | Samot: Just wait 3-4 minutes, it was up again |
| 13:03.22 | Samot | F'ing Germany. |
| 13:03.27 | velix | 100% true :D |
| 13:03.40 | velix | f'ing German IT department. |
| 13:03.53 | velix | I'm half turk, half German ;) |
| 13:03.54 | Samot | The worst deployments of SIP I've ever seen come from that country. |
| 13:03.58 | velix | ^_^ |
| 13:04.19 | velix | But they say, it's my fault... that's makes me crazy. |
| 13:04.23 | sibiria | we've only had problems with one german ITSP; used a few of them |
| 13:04.35 | Cyrillax | over-engineering |
| 13:04.45 | Samot | Like you being on the same local network as the Universities network but using a PUBLIC PROXY |
| 13:05.22 | Cyrillax | omg he's really on the same lan? I thought he just had his local lan configured to be the same subnet |
| 13:05.25 | velix | Samot: No, I'm outside. |
| 13:05.27 | velix | No no no nooo |
| 13:05.28 | velix | nooo |
| 13:05.30 | velix | Noooo! |
| 13:05.36 | Samot | That's not what you said before. |
| 13:05.49 | velix | Samot: For _testing_ I ocne was on the same network. |
| 13:05.50 | Samot | You had a VPN/private connection from the DC with the schools networks |
| 13:05.55 | velix | My PBX is in a big datacenter. |
| 13:06.06 | velix | Yes, for local testing and dev @ home. |
| 13:06.26 | Samot | Oh so all that we helped you with in the beginning isn't even in play here? |
| 13:06.28 | Samot | Awesome. |
| 13:06.32 | Cyrillax | facepalm |
| 13:06.35 | Samot | So glad. |
| 13:06.50 | Samot | Cyrillax: This has been weeks. |
| 13:07.02 | Cyrillax | good lorf |
| 13:07.03 | Cyrillax | lord |
| 13:07.03 | velix | Samot: Nah, the OpenVPN stuff was a different problem. |
| 13:07.54 | Samot | So now there is NAT involved between you and the school? |
| 13:08.17 | velix | Only on the University's side. So yes. |
| 13:08.36 | Cyrillax | senses danger... |
| 13:08.36 | velix | I'm directly on the Internet without a nat. |
| 13:08.53 | Samot | So when you said you had no audio weeks ago you said there was no NAT between you and the school. You kept dismissing NAT based issues. |
| 13:09.01 | Samot | Because of VPNs and shit. |
| 13:09.04 | Samot | So we worked on that. |
| 13:09.07 | Samot | And around that |
| 13:09.11 | Samot | Now all that is gone. |
| 13:09.14 | velix | Samot: No, that was the client to my PBX side. I fixed that. |
| 13:09.20 | Samot | OK. |
| 13:09.22 | velix | Samot: I had a NAT on my side before. |
| 13:09.28 | Samot | OK. |
| 13:09.29 | velix | I removed it. Then all the clients worked. |
| 13:09.39 | Samot | So continue to use them because they have flat mobile rates. |
| 13:09.43 | velix | They've done more than 300 successful calls over a paid trunk. |
| 13:09.48 | Samot | You will continue to have your issues. |
| 13:10.13 | velix | Sure. But the IT guys says: "just tell them to enter a STUN in their client to fix all your issues". |
| 13:10.18 | Cyrillax | Even if you get it setup properly, it doesn't sound like they have a robus infrastructure in order to keep consistent calls |
| 13:10.37 | Samot | How's does a STUN server between you and your end users fix what happens between YOU AND HIM? |
| 13:10.37 | Cyrillax | That's not going to help |
| 13:10.44 | velix | Samot: That's the question. |
| 13:10.52 | Cyrillax | it won't |
| 13:10.55 | Samot | Which hasn't been answered. |
| 13:11.05 | velix | Kobaz said the same. The IT guy laught about this. I got angry |
| 13:11.07 | Samot | Because that solution doesn't do anything to address it. |
| 13:11.13 | velix | told the professor, who also got angry :) |
| 13:11.25 | Samot | OK. |
| 13:11.31 | Samot | So clearly they don't know what they are doing. |
| 13:11.39 | Samot | But hey, they got great mobile rates. |
| 13:11.43 | velix | Samot: That's what the old IT department chef said :) |
| 13:11.56 | velix | Samot: He quit the job some months ago because of stupidness |
| 13:12.05 | Cyrillax | it's been going on months |
| 13:12.06 | Samot | I guess having low rates allows you to not know what you're doing. |
| 13:12.30 | velix | You know, what I'd love to do? Install the VPN client on my PBX ... |
| 13:12.32 | Cyrillax | How much is it going to cost you to switch to a reliable carrier with support |
| 13:12.41 | Samot | He HAS one. |
| 13:12.50 | _abc_ | Suggest why 2 androids connected as sip clients with linphone on both, both on wifi same lan as *, can call 620 but not each other? One says the other is busy the other times out on call. |
| 13:12.55 | velix | Samot: But I'm not allowed to use it on unauthorized devices. |
| 13:13.15 | velix | And guess what? They're using 2013 shrewsoft client, which ended support and break with a "segfault" on current Linux :D |
| 13:14.03 | velix | Cyrillax: We've calculated this. About $2400 a year. |
| 13:15.20 | Cyrillax | I'm guessing you've probably used that much of Samot's time already :D |
| 13:15.45 | Samot | Cyrillax: Time doesn't count man. Don't you know that? |
| 13:16.08 | Samot | Because this is FOSS telephony baby, your time doesn't count. |
| 13:16.11 | Samot | It's all FREEEEEE |
| 13:17.24 | seanbright | it's a lot more than $2400 |
| 13:17.39 | seanbright | add a 0 |
| 13:18.01 | Samot | Student labor man. |
| 13:18.13 | Samot | I'm guessing their getting paid in credits and grades and not $$$ |
| 13:18.31 | Cyrillax | The professor will keep all the profit |
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| 13:22.10 | _abc_ | Is the "chat" feature in linphone and other clients routed through SIP messages or not? It seems to use a server too for some reason. |
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| 13:32.21 | velix | It's a student project. So all the university can offer is the free acount. They're also not interessted (or keen) at doing support. |
| 13:32.40 | velix | Cyrillax: Nah, it's no university initiated research project. |
| 13:32.51 | velix | Cyrillax: part of a bigger student project |
| 13:33.05 | velix | And as a Geographer, we don't get any $$ from 3rd parties. |
| 13:33.09 | velix | Nobody is interested in rocks. |
| 13:33.16 | _abc_ | https://www.linphone.org/snapshots/docs/liblinphone/multilang/guides/chatroom.html ah ok, so 1:1 chat does not require a server. Great. |
| 13:33.21 | velix | We better should do "rock". |
| 13:33.43 | _abc_ | velix: call yourself a prospector in oil gas and energy and suddenly money. |
| 13:34.03 | _abc_ | (emphasis on rocks and layout thereof) |
| 13:34.20 | velix | <3 |
| 13:34.26 | [TK]D-Fender | https://www.linphone.org/news/group-chat-now-available-linphone |
| 13:34.43 | _abc_ | [TK]D-Fender: yes was reading the API docs above |
| 13:35.09 | _abc_ | Trying to make that work with *, no success yet, version hell of linphone must be sorted 1st |
| 13:36.05 | _abc_ | Ok so the latest version of linphone which works on android 4.4 (my lowest support level) is 4.0.1 |
| 13:36.19 | _abc_ | 4.0.1 only has peer to peer chat as far as I can see |
| 13:37.54 | _abc_ | So now I have trouble with no video call and no call client to client, with no directmedia and no reinvite in * with linphones. |
| 13:37.57 | _abc_ | Fun. |
| 13:37.57 | _abc_ | bbl |
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| 13:41.58 | Kobaz | do de do |
| 13:43.06 | velix | Kobaz: ;) |
| 13:44.23 | Kobaz | velix: i really think it's a firewall issue, because you're having similar problems with another site |
| 13:44.36 | Kobaz | and you're passing data on the same interface, right? |
| 13:44.44 | velix | Uh wait? Firewall? |
| 13:44.47 | Kobaz | yes |
| 13:44.52 | Kobaz | SIP ALG, that sort of thing |
| 13:44.58 | velix | Kobaz: That's not my side. |
| 13:45.03 | Kobaz | especially since you do SIP OPTIONS, and the other site does not respond half the time |
| 13:45.12 | velix | Kobaz: That's the university's PBX. |
| 13:45.13 | Kobaz | not your side, correct |
| 13:45.16 | Kobaz | firewall on the university |
| 13:45.26 | velix | sure, I told them, they laught again |
| 13:45.46 | Kobaz | you need to set up a meeting and face to face say, yer shit's broken |
| 13:46.46 | velix | Kobaz: Not allowed, because of Corona :D |
| 13:46.52 | velix | More than 2 people :) |
| 13:47.01 | Kobaz | well not literally in the face |
| 13:47.15 | Kobaz | you can all stand in a field 10 feet apart and yell at each other |
| 13:47.16 | velix | Hehe, phone conference work on my PBX. Just had one with 7 people. |
| 13:47.31 | velix | googles up "feet" :) |
| 13:47.37 | Kobaz | 1/3 of a meter |
| 13:47.48 | velix | My barbier has closed, I'm looking like a werwolf |
| 13:48.34 | velix | Kobaz: Sorry for asking this, but he asked me to: how can I force re-registration every 60 seconds? |
| 13:48.34 | Kobaz | 3.3 meters apart |
| 13:48.47 | velix | Kobaz: 2 meters are enough, but more is better, of course. |
| 13:48.59 | velix | Problem is, I need to travel to them by public transport (no car) |
| 13:49.31 | Kobaz | a sneeze can travel further than 2 meters |
| 13:50.05 | velix | I can set the expiry... but is this enough to trigger a re-registration? |
| 13:50.59 | Kobaz | retry interval |
| 13:51.12 | Kobaz | type=registration retry_interval=60 |
| 13:51.16 | Kobaz | it's right in the example conf |
| 13:51.21 | Kobaz | pjsip.conf.sample |
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| 13:52.44 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 13:53.31 | velix | Kobaz: I'm still running this on chan_sip right now, since i didn't have time to migrate all other settings to pjsip. Need to do it tonight. |
| 13:54.10 | Kobaz | that's also in the example config: sip.conf.sample |
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| 14:01.46 | velix | Kobaz: I've used ...~60 |
| 14:01.59 | velix | Since default & maxexpiry would affect the commercial trunk, too |
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| 16:36.16 | velix | hmm, doesn't "stun show status" exist anymore? |
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| 16:38.15 | velix | I've set res_stun_monitor.conf of course |
| 16:38.49 | velix | okay, I've got res_stun_monitor module |
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| 17:08.16 | _abc_ | This is drving me nuts. I managed once to get a chat message passed between 2 androids running linphone connected in an asterisk call on the same wifi lan. Once. And never again. |
| 17:09.00 | _abc_ | Getting information literally requires reverse engineering the source of linphonelib. Only noise online. What does SIP use for usual message passing? Unsolicited MESSAGE ? |
| 17:09.36 | [TK]D-Fender | Thre is no concept of soliciting a message |
| 17:09.47 | [TK]D-Fender | so that'd be an obvious "yes" |
| 17:10.53 | _abc_ | in SIP debug I see "Unable to set format because channel Message/ast_msg_queue supports no formats" |
| 17:11.05 | _abc_ | I don't see a channel Message in messagetypes |
| 17:13.09 | [TK]D-Fender | Show the actual full interation |
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| 17:37.13 | velix | Interesting, res_stun_monitor.so has been buikld, I can see it in the modules folder, but I cannot load it. |
| 17:37.16 | velix | build* |
| 17:37.31 | velix | "module show like stun" doesn't show it. |
| 17:37.34 | velix | Oh come on... |
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| 17:39.03 | velix | ah, core reload did it. |
| 17:40.54 | velix | grrrr. PJ ICE Rx error status code: 370401 'Unauthorized'. |
| 17:40.55 | velix | lord! |
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| 18:02.49 | *** join/#asterisk sekil (~sekil@178-223-3-39.dynamic.isp.telekom.rs) |
| 18:04.04 | _abc_ | Sorry, long phone call |
| 18:08.47 | *** part/#asterisk sekil (~sekil@178-223-3-39.dynamic.isp.telekom.rs) |
| 18:09.50 | *** join/#asterisk mcdonali__ (~imcdona@65-100-46-166.dia.static.qwest.net) |
| 18:10.44 | velix | At least you *can* call |
| 18:18.51 | pabe | lol |
| 18:19.07 | _abc_ | not over sip... |
| 18:26.21 | _abc_ | m=audio 19164 RTP/AVP 0 100 ;; m=video 0 RTP/AVP 96 97 ;; suggests video is trying for port 0 ?! |
| 18:26.37 | *** join/#asterisk eharris (~eharris@unaffiliated/eharris) |
| 18:26.59 | igcewieling | suggests no video supported |
| 18:27.08 | _abc_ | This is an INVITE |
| 18:27.15 | igcewieling | right |
| 18:27.18 | _abc_ | Huh. Why no video supported? Just a second |
| 18:28.37 | _abc_ | Something is wrong. The androids support video. This is from an android to the server. |
| 18:30.04 | _abc_ | sip debug can be enabled for more than one peer at a time in the cli? |
| 18:33.13 | _abc_ | how does one unset a set debug mode on sip? |
| 18:34.05 | _abc_ | got it |
| 18:41.00 | _abc_ | Why does a registered phone get Unauthorized messages from the server in response to REGISTER messages? I find this is sticking out: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", ... |
| 18:41.16 | _abc_ | realm "asterisk" is an internal auth thing? It does not seem to be in my config, |
| 18:41.36 | file | because SIP uses a challeng response mechanism for authentication |
| 18:47.16 | _abc_ | that uses the nonce etc, no? "asterisk" is hard wired? I have a doubt? |
| 18:47.59 | file | it is not hardcoded, it is configurable as to what is used |
| 18:48.03 | file | the default is "asterisk" |
| 18:48.17 | _abc_ | this is the realm= param in config, right? |
| 18:48.52 | file | yes |
| 18:56.07 | velix | Hmm... writing DB via manager needs "system" rights. Is this secure? |
| 18:57.08 | _abc_ | * should pass video codec VP8 in non translating mode, right? |
| 18:58.21 | *** join/#asterisk dacod (~dacod@179.180.170.66) |
| 19:02.59 | *** join/#asterisk ruhnet (~ruel@unaffiliated/lvlinux) |
| 19:04.22 | velix | Why do I need ICE for STUN? My softphone doesn't have ICE |
| 19:04.46 | _abc_ | codec=vp8 works fine, the problem is not in * |
| 19:08.00 | _abc_ | Right, after crashing each android once or twice I have video calls going with media through * |
| 19:09.56 | _abc_ | Ok, now the chat. |
| 19:10.25 | _abc_ | What the heck could make the chat not work right. Is there a specific media/codec type which must be enabled? |
| 19:16.27 | velix | yells at cloud |
| 19:18.24 | [TK]D-Fender | waits till there is something worth commenting on... |
| 19:30.51 | igcewieling | [TK]D-Fender: you'll be waiting for days. |
| 19:31.43 | [TK]D-Fender | igcewieling, I've noticed the pattern. Squirrels... nothing but squirrels.... |
| 20:15.43 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 20:50.38 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:10.10 | _abc_ | plays red squirrel |
| 21:10.14 | _abc_ | save me save me |
| 21:12.36 | _abc_ | Ok, are there specific settings needed to make chat peer to peer (no server in between) on linphone latest (4.2?) and linphone 4.0? Note the latter does not have the group chat thing, 4.2.3 seems to have it, there's a chat server url in settings. Both clients are on a lan, the same as the * server, no nat, no firewalls (all turned off). * call talk and video work in all directions. |
| 21:13.56 | *** join/#asterisk imcdona (~imcdona@adpp107.ds.adp.com) |
| 21:17.26 | [TK]D-Fender | grabs some popcorn |
| 21:18.04 | _abc_ | you are having too much fun. I just cleaned the house a bit to discourage viruses on the floor. |
| 21:18.41 | _abc_ | Is this in the asterisk online book/wiki or not? |
| 21:18.50 | _abc_ | accept_outofcall_message=yes etc? |
| 21:22.58 | _abc_ | The strange part is, got it to work earlyer and don't know why. |
| 21:23.13 | _abc_ | w |
| 21:30.26 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 21:35.00 | _abc_ | Now it works in only one direction, clearly version issues in linphone. |
| 21:56.29 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 22:00.58 | _abc_ | Downgraded both to the same version which I assumed to work, now both don't. Giving up, will try other clients. * setup seems okay. |
| 22:01.14 | _abc_ | Implemented this: http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html |
| 22:01.28 | _abc_ | Seems to do the job apart from the client's weirdness |
| 22:01.39 | _abc_ | thanks for listening, especially [TK]D-Fender :) |
| 22:01.44 | *** part/#asterisk _abc_ (~usre@unaffiliated/ccbbaa) |
| 22:20.13 | *** join/#asterisk deavmi (~quassel@41.164.64.226) |
| 22:31.45 | Cyrillax | velix: module load res_stun_monitor # looks like you got it though |
| 22:32.32 | Cyrillax | oh missed that line, where you did it with a core reload, disregard |
| 22:45.46 | Cyrillax | velix: Do you have any kind of firewall at all in front of your PBX? iptables/pfsense? |
| 22:52.26 | Cyrillax | Can you get a packet capture (in wireshark) of the call + rtp path of a call with 1-way audio? This will help me tremendously velix |
| 22:52.48 | Cyrillax | must include the sip traffic and the RTP packet, and it has to be a call with 1-way audio that you're tracing |
| 23:03.28 | *** join/#asterisk javi404 (~quassel@unaffiliated/javi404) |
| 23:05.05 | *** join/#asterisk acovrig (44a9bb97@host-68-169-187-151.JENOLT2.epbfi.com) |
| 23:05.47 | acovrig | Is it possible to Dial() a Queue? I would like to send a call to a queue, but play a message to the callee before connecting them to the caller |
| 23:06.30 | *** join/#asterisk Janos__ (~Janos@201.204.94.76) |
| 23:26.08 | Cyrillax | Yes definitely |
| 23:28.22 | acovrig | how? if I try to Dial(Queue/###) it says Queue isn't a valid technology, if I try Queue(###,,,,,,macro-speak) it doesn't ring static members, and doesn't play the message on answering; I tried Dial(Local/quque) also, but IDK what the context should be |
| 23:30.16 | Cyrillax | You would have to do Local/3333@ext-queues/n |
| 23:30.33 | Cyrillax | Local is the TECH |
| 23:31.51 | Cyrillax | the extension "ext-queues" would be whatever context your Queue runs in |
| 23:33.07 | Cyrillax | So if you haven't already, you would want to centralize your Queue access to a specific context, like [ext-queues] exten => 5555,1,Queue(5555,${QOPTIONS}) |
| 23:33.33 | Cyrillax | then you can dial into the queue like I showed above |
| 23:40.09 | *** join/#asterisk acovrig (44a9bb97@host-68-169-187-151.JENOLT2.epbfi.com) |
| 23:40.11 | acovrig | Cyrillax thanks, I can dial the queue, but how do I have it play a message before the caller connects with the callee? I did something similar to this with Dial(,,,U(context^arg1^arg2)) which does a Playback() and Return() - I presume there's a way to do that with a queue? |
| 23:42.26 | acovrig | I tried a macro in the Queue() but it didn't seem to go, docs says "... once the parties are connected", which I presume isn't want I'm looking for? |
| 23:44.17 | Cyrillax | That's significantly harder. It's been a while since I've done something similar, so I don't have a straight answer for you |
| 23:46.11 | Cyrillax | I'll think it over for a bit, but off the top of my head, conferences are extremely useful for doing odd things like this |
| 23:46.37 | Cyrillax | What's the full call path like here? Tell me how the user would end up talking to the agent in the queue |
| 23:47.15 | Cyrillax | example: user calls the phone number, presses 3, etc |
| 23:47.27 | acovrig | user calls an IVR, it gathers some info and drops them in a queue, my idea is for it to Flite() some of the info to the agent when the agent answers |
| 23:48.30 | [TK]D-Fender | acovrig, then Macro is your way |
| 23:48.41 | acovrig | I'm guessing an agi would work/be better? user calls, agi sends info to a separate UI for the agent, then the agent can answer normally (it wouldn't need to read them any of the info as it would be displayed in a different medium) |
| 23:53.22 | acovrig | [TK]D-Fender where would I put the macro? in Queue()? |
| 23:53.46 | [TK]D-Fender | acovrig, If you need to played to the member, yes |