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00:23.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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04:46.43 | adonaros | kunwon1: thank you for giving me the CID number matching idea! |
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06:47.06 | kunwon1 | adonaros: you're quite welcome |
06:48.12 | drmessano | np |
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16:09.41 | Korolev | Hi, I'm trying to get chan_mobile to work. I'm able to dial out and receive calls, but there is no audio in either direction. No NAT involved. Asterisk 13.13 using a Galaxy S5 |
16:09.49 | Korolev | google is not being helpful, at all |
16:09.56 | Korolev | has anyone had this issue? |
16:10.55 | [TK]D-Fender | You haven't described the other end of the call or shown any debug |
16:13.46 | Korolev | debug doesn't show any errors or warnings |
16:14.59 | Korolev | the call path is linphone->asterisk using chan_mobile->some other cellphone |
16:15.06 | [TK]D-Fender | Just because it doesn't say "error" doesn't mean it isn't showing something USEFUL in dtermining the cause |
16:15.09 | [TK]D-Fender | And we still see nothing |
16:15.13 | [TK]D-Fender | you are showing us nothing |
16:15.24 | Samot | So both cell phones are on the same network as the PBX? |
16:15.41 | Korolev | linphone and the pbx are on the same local network |
16:15.52 | [TK]D-Fender | Show us the call |
16:15.55 | Samot | And the PBX is on a public IP? |
16:16.10 | Korolev | no, private ip |
16:16.14 | Samot | 12:16:03 PMÂ <[TK]D-Fender>Â Show us the call <-- That needs to happen |
16:16.15 | Samot | OK |
16:16.21 | Samot | So NAT is 100% involved. |
16:16.32 | Korolev | let me get some logs for you guys |
16:16.59 | Korolev | how would NAT be involved? |
16:17.16 | [TK]D-Fender | He didn't read all the back-story |
16:17.22 | [TK]D-Fender | just get the call |
16:17.25 | Korolev | okay |
16:18.05 | Samot | Uhm. Yes I did. |
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16:20.34 | [TK]D-Fender | <Korolev> Hi, I'm trying to get chan_mobile to work. I'm able to dial out and receive calls, but there is no audio in either direction. No NAT involved. Asterisk 13.13 using a Galaxy S5 |
16:20.38 | [TK]D-Fender | no nat involved |
16:20.45 | [TK]D-Fender | <Samot> So NAT is 100% involved. |
16:20.52 | Samot | 12:15:10 PM K<Korolev> the call path is linphone->asterisk using chan_mobile->some other cellphone |
16:20.57 | Samot | 12:15:52 PM K<Korolev> linphone and the pbx are on the same local network |
16:21.05 | [TK]D-Fender | What part of that says ITSP <---? |
16:21.07 | Samot | 12:16:06 PM S<Samot> And the PBX is on a public IP? |
16:21.07 | Samot | 12:16:21 PM K<Korolev> no, private ip |
16:21.14 | Samot | Doesn't mater |
16:21.17 | [TK]D-Fender | NO INTERNET SERVICE PROVIDER |
16:21.18 | Samot | REMOTE endpoint |
16:21.22 | Korolev | same local network |
16:21.26 | Samot | ALL THREE? |
16:21.31 | Korolev | no packet is going across a router |
16:21.48 | Samot | I asked if ALL endpoints and PBX where on the same local network |
16:21.53 | Samot | You said Linphone and PBX |
16:21.57 | Samot | What about the other cellphone? |
16:22.06 | Korolev | is connected using bluetooth |
16:22.08 | Samot | Is it on the same local network? |
16:22.17 | Korolev | it doesn't matter where it is |
16:22.32 | Samot | Is it on the same local network? |
16:22.40 | [TK]D-Fender | <Korolev> same local network |
16:22.41 | Korolev | is not on the network at all |
16:22.55 | Korolev | the cellphone has no network connection |
16:23.20 | Korolev | apart from bluetooth and wcdma |
16:23.24 | [TK]D-Fender | get the call |
16:23.26 | [TK]D-Fender | stop now |
16:23.43 | Samot | 12:16:25 PMÂ <Samot>Â 12:16:03 PMÂ <[TK]D-Fender>Â Show us the call <-- That needs to happen |
16:23.47 | Samot | Yup. |
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16:47.21 | Korolev | okay, this is the call http://www.pastebin.com/YhEbtHSn |
16:50.34 | Korolev | chan_mobile.conf http://www.pastebin.com/kdixcmxv |
16:52.03 | [TK]D-Fender | we see no proof on the SIP end |
16:52.05 | Korolev | rtp debug shows no rtp from asterisk to linphone after the call is answered |
16:52.14 | Samot | sip set debug on |
16:52.18 | Samot | then make the call |
16:52.26 | Korolev | ok |
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17:11.16 | Korolev | https://pastebin.com/ekSWhBpg |
17:11.30 | Korolev | ^ thats with sip debug enabled |
17:13.08 | Korolev | again, for context, rtp debug shows rtp packets going in both directions during ringing, I'm guessing asterisk is generating those |
17:13.27 | Korolev | and then rtp packets stop in the direction asterisk->sipphone after the call is answered |
17:13.32 | Samot | So you're calling from Linphone? |
17:13.36 | Korolev | yes |
17:13.45 | Samot | And you're calling a local extensions? |
17:13.47 | Samot | And you're calling a local extension? |
17:13.55 | Samot | Which is the other cell phone? |
17:14.02 | Samot | Connected via bluetooth? |
17:14.27 | Korolev | I'm using the phone connected via bluetooth as a gateway |
17:14.44 | Samot | So you're calling out over the Internet to the PSTN? |
17:14.50 | Korolev | yes |
17:14.52 | Samot | INVITE sip:18004321000@192.168.1.13:50060 SIP/2.0 <-- This is an outbound call? |
17:14.53 | Samot | OK |
17:14.55 | Samot | So then yes. |
17:14.57 | Samot | As I said before |
17:15.03 | Samot | 100% NAT is involved. |
17:15.10 | Samot | Your call is going over the Internet |
17:15.23 | Korolev | when I said yes to "over the internet" I meant "over ip" |
17:15.34 | Korolev | there is no internet involved here |
17:15.39 | Samot | INVITE sip:18004321000@192.168.1.13:50060 SIP/2.0 <-- WHERE is that number? |
17:15.42 | Samot | Is that yours? |
17:15.49 | Korolev | at 192.168.1.13 |
17:15.57 | Korolev | the call is coming from 192.168.1.12 |
17:16.02 | Korolev | neither is on the internet |
17:16.04 | Samot | The PBX is .13? |
17:16.08 | Korolev | yes |
17:16.17 | Samot | 18004321000 <-- is that your number? |
17:16.34 | Samot | Or are you calling someone? |
17:16.39 | Korolev | that's bank of america, I used it as a placeholder for the actual number |
17:16.45 | Samot | FFS. |
17:16.52 | Korolev | I'm calling a cellphone that I have in my hand |
17:17.06 | Samot | So you're calling a LOCAL extension on the PBX? |
17:17.09 | [TK]D-Fender | Samot, HE'S USING CHAN_MOBILE TO TALK THAT OTHER PHONE VIA BLUETOOTH. |
17:17.11 | Korolev | from another cellphone that is connected via chan_mobile to asterisk |
17:17.16 | Samot | OK. |
17:17.21 | Samot | So there's no audio. |
17:17.23 | [TK]D-Fender | <PROTECTED> |
17:17.29 | Samot | I get what is happening.. |
17:17.31 | Korolev | no audio |
17:17.41 | Samot | I'm trying to figure out if the call is going to the PSTN or not. |
17:17.48 | Samot | Via the mobile carrier or an ITSP |
17:18.03 | Korolev | it is going to the pstn via a mobile carrier |
17:18.06 | [TK]D-Fender | Korolev, Prove the SIP end independently <- |
17:18.07 | Samot | OK |
17:18.14 | Samot | There you go. |
17:18.20 | [TK]D-Fender | Answer |
17:18.34 | [TK]D-Fender | Playback. Record. Playback the recording |
17:18.45 | Korolev | that works |
17:19.01 | Korolev | the sip end is working well |
17:19.10 | Korolev | the problem is only with chan_mobile |
17:21.13 | Korolev | that asterisk stops sending rtp packets to linphone after the call is answered tells me that it is not receivign any rtp from the chan_mobile side |
17:21.45 | Korolev | but I'm very unfamiliar with how bluetooth works or chan_mobile for that matter to know where to look next |
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19:41.46 | tehgooch | [TK]D-Fender, I think I solved the problem I was asking about the other day. https://pastebin.com/bBLJuL8P - if any destination endpoint is busy the caller gets a busy signal. If you would be so kind as to take a look and let me know if you see any problems with it I would be appreciative. |
19:44.32 | tehgooch | Well, technically if any dest endpoint is "INUSE" |
19:50.40 | [TK]D-Fender | same => n,While($["${SET(i=${SHIFT(extensions)})}"!=""]) |
19:50.53 | [TK]D-Fender | I fail to see how SET() returns anything |
19:58.42 | [TK]D-Fender | I also wouldn't loop twice to do this |
19:58.45 | [TK]D-Fender | you can do it in one |
19:59.10 | [TK]D-Fender | build the dialstring once, checking as you add, and you can abort the moment one that you check comes up busy |
20:16.25 | tehgooch | [TK]D-Fender, That while function string was copied almost verbatim from the docs https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHIFT I see your point about the two loops. |
20:18.13 | Samot | The while function makes sense. |
20:18.32 | Samot | There's nothing wrong with the while loop. I think he's just saying you don't need to loop twice. |
20:19.11 | tehgooch | Samot, He said he didn't see how it returned anything. I'm not sure either, but it works and I lifted it from the docs. |
20:19.32 | Samot | It basically says "While I can SET $i do this" |
20:19.56 | tehgooch | Samot, it's comparing the result to the empty string though. |
20:20.10 | Samot | I get it. |
20:20.12 | tehgooch | Oh |
20:20.12 | Samot | You want data |
20:20.16 | tehgooch | It probably returns 0 |
20:20.21 | tehgooch | which is not the empty string |
20:20.23 | Samot | There is that. |
20:20.25 | tehgooch | as long as it works |
20:21.18 | [TK]D-Fender | I suppose the function returns what it also sets to that var |
20:21.35 | [TK]D-Fender | Not really announced in its instructions and feels odd to use |
20:21.53 | Samot | Oh, I'm not sure it's right for this. |
20:22.06 | Samot | But the while loop, itself, should function just fine. |
20:22.09 | [TK]D-Fender | probably is fine give the sample it was lifted from |
20:22.26 | [TK]D-Fender | basically keep going until you hit a blank value |
20:22.40 | Samot | techgooch: Now you have to do what everyone does with examples lifted from the Internet. |
20:22.45 | Samot | Tweak it to your own needs. |
20:22.53 | Samot | You're going to need to refine it. |
20:23.05 | tehgooch | of course :) |
20:23.39 | Samot | But I agree with [TK]D-Fender on just checking the INUSE state and building the dialstring then. |
20:23.46 | Samot | Not doing it as two different things. |
20:23.48 | tehgooch | The Internet provides and the Internet takes away. |
20:24.24 | tehgooch | I thought modularizing it might prove beneficial in the future. |
20:25.20 | Samot | I'm a fan of modular programming. |
20:25.36 | Samot | But the only thing I would see you'd want modular is the INUSE check... |
20:25.56 | Samot | But even then, depending on what you are doing it might not fit. |
20:26.01 | Samot | Down the road. |
20:26.02 | tehgooch | true |
20:26.27 | tehgooch | Well at least I learned something. |
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20:26.40 | Samot | That's always a plus. |
20:29.00 | tehgooch | Just did a test and Set() does return what it set the thing to. |
20:30.21 | tehgooch | <3 undocumented functionality |
20:30.30 | Samot | Well just make sure the whatchamacallit fires properly. |
20:31.37 | tehgooch | I was thinking the 'b' or possibly 'B' Dial() option might work better for my use case, but I don't think it's appropriate. |
20:32.45 | tehgooch | I'm not sure how to get the endpoints sent to the Gosub. |
20:33.00 | tehgooch | Of course I could just pass them as args, but then I need to specify them in two places. |
20:37.02 | tehgooch | 'b' does a Gosub() for each created channel, but I'm not sure if Busy() would affect the original Dial() and I'd need to get the name of the endpoint somehow which I'm not clear on. |
20:37.34 | tehgooch | 'B' does a Gosub() before creating new channels, so I'm fairly sure Busy() would affect the original Dial(), but I still don't know how to get the names of the endpoints. |
20:40.09 | tehgooch | Anyway. Thanks [TK]D-Fender and Samot. If you have any thoughts on the Dial() options I'd be glad to hear. I'm going to take a break for now. |
20:43.23 | [TK]D-Fender | just What about them? |
20:43.34 | [TK]D-Fender | I don't see how those apply to your requirement |
21:02.19 | tehgooch | [TK]D-Fender, they run a macro before the channels are bridged so I could detect the device state there and make it busy instead of looping and such. |
21:02.28 | tehgooch | er, s/macro/gosub |
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21:04.33 | [TK]D-Fender | That's backwards |
21:05.04 | [TK]D-Fender | you are ACTUALLY dialing them with that feature |
21:05.13 | [TK]D-Fender | before bridging means you both called... and they ANSWERED |
21:05.17 | [TK]D-Fender | it's ABOUT to bridge |
21:05.34 | [TK]D-Fender | You want to make your decision BEFORE attempting to call |
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21:34.19 | tehgooch | [TK]D-Fender, Hm, I guess I misunderstood the docs. I'll have to continue reading. |
21:39.08 | tehgooch | It doesn't work if I do the state check in the loop because I removed the first endpoint before the loop. |
21:39.38 | tehgooch | I could check before the loop and again in it, but that's duplicating code. |
21:41.51 | [TK]D-Fender | pop the next value. check it. Jump if busy, otherwise add. |
21:41.52 | [TK]D-Fender | Loop |
21:42.10 | [TK]D-Fender | You don't need 2 loops |
21:43.30 | [TK]D-Fender | In fact.... you shouldn't even HAVE to build a new string. You're just dialing all of the things being passed. |
21:43.45 | tehgooch | ok I'll give it a try |
21:43.58 | [TK]D-Fender | it's not like you're EXLUDING busy from the call. You're aborting if ANY are busy |
21:43.59 | tehgooch | you're right I could just replace the '.'s with '&'s |
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22:53.37 | file | falls over |
22:59.29 | file | offering multiple audio streams to phones is fun |
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23:10.08 | Samot | I'm sure it makes the call interesting. |
23:12.23 | file | Samot: some do a 488, some accept the SDP negotiation but decline the extras |
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