IRC log for #asterisk on 20170618

00:23.35*** join/#asterisk infobot (~infobot@rikers.org)
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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04:46.43adonaroskunwon1: thank you for giving me the CID number matching idea!
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06:47.06kunwon1adonaros: you're quite welcome
06:48.12drmessanonp
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16:09.41KorolevHi, 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
16:09.49Korolevgoogle is not being helpful, at all
16:09.56Korolevhas anyone had this issue?
16:10.55[TK]D-FenderYou haven't described the other end of the call or shown any debug
16:13.46Korolevdebug doesn't show any errors or warnings
16:14.59Korolevthe call path is linphone->asterisk using chan_mobile->some other cellphone
16:15.06[TK]D-FenderJust because it doesn't say "error" doesn't mean it isn't showing something USEFUL in dtermining the cause
16:15.09[TK]D-FenderAnd we still see nothing
16:15.13[TK]D-Fenderyou are showing us nothing
16:15.24SamotSo both cell phones are on the same network as the PBX?
16:15.41Korolevlinphone and the pbx are on the same local network
16:15.52[TK]D-FenderShow us the call
16:15.55SamotAnd the PBX is on a public IP?
16:16.10Korolevno, private ip
16:16.14Samot12:16:03 PM <[TK]D-Fender> Show us the call <-- That needs to happen
16:16.15SamotOK
16:16.21SamotSo NAT is 100% involved.
16:16.32Korolevlet me get some logs for you guys
16:16.59Korolevhow would NAT be involved?
16:17.16[TK]D-FenderHe didn't read all the back-story
16:17.22[TK]D-Fenderjust get the call
16:17.25Korolevokay
16:18.05SamotUhm. 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-Fenderno nat involved
16:20.45[TK]D-Fender<Samot> So NAT is 100% involved.
16:20.52Samot12:15:10 PM K<Korolev> the call path is linphone->asterisk using chan_mobile->some other cellphone
16:20.57Samot12:15:52 PM K<Korolev> linphone and the pbx are on the same local network
16:21.05[TK]D-FenderWhat part of that says ITSP <---?
16:21.07Samot12:16:06 PM S<Samot> And the PBX is on a public IP?
16:21.07Samot12:16:21 PM K<Korolev> no, private ip
16:21.14SamotDoesn't mater
16:21.17[TK]D-FenderNO INTERNET SERVICE PROVIDER
16:21.18SamotREMOTE endpoint
16:21.22Korolevsame local network
16:21.26SamotALL THREE?
16:21.31Korolevno packet is going across a router
16:21.48SamotI asked if ALL endpoints and PBX where on the same local network
16:21.53SamotYou said Linphone and PBX
16:21.57SamotWhat about the other cellphone?
16:22.06Korolevis connected using bluetooth
16:22.08SamotIs it on the same local network?
16:22.17Korolevit doesn't matter where it is
16:22.32SamotIs it on the same local network?
16:22.40[TK]D-Fender<Korolev> same local network
16:22.41Korolevis not on the network at all
16:22.55Korolevthe cellphone has no network connection
16:23.20Korolevapart from bluetooth and wcdma
16:23.24[TK]D-Fenderget the call
16:23.26[TK]D-Fenderstop now
16:23.43Samot12:16:25 PM <Samot> 12:16:03 PM <[TK]D-Fender> Show us the call <-- That needs to happen
16:23.47SamotYup.
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16:47.21Korolevokay, this is the call http://www.pastebin.com/YhEbtHSn
16:50.34Korolevchan_mobile.conf http://www.pastebin.com/kdixcmxv
16:52.03[TK]D-Fenderwe see no proof on the SIP end
16:52.05Korolevrtp debug shows no rtp from asterisk to linphone after the call is answered
16:52.14Samotsip set debug on
16:52.18Samotthen make the call
16:52.26Korolevok
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17:11.16Korolevhttps://pastebin.com/ekSWhBpg
17:11.30Korolev^ thats with sip debug enabled
17:13.08Korolevagain, for context, rtp debug shows rtp packets going in both directions during ringing, I'm guessing asterisk is generating those
17:13.27Korolevand then rtp packets stop in the direction asterisk->sipphone after the call is answered
17:13.32SamotSo you're calling from Linphone?
17:13.36Korolevyes
17:13.45SamotAnd you're calling a local extensions?
17:13.47SamotAnd you're calling a local extension?
17:13.55SamotWhich is the other cell phone?
17:14.02SamotConnected via bluetooth?
17:14.27KorolevI'm using the phone connected via bluetooth as a gateway
17:14.44SamotSo you're calling out over the Internet to the PSTN?
17:14.50Korolevyes
17:14.52SamotINVITE sip:18004321000@192.168.1.13:50060 SIP/2.0 <-- This is an outbound call?
17:14.53SamotOK
17:14.55SamotSo then yes.
17:14.57SamotAs I said before
17:15.03Samot100% NAT is involved.
17:15.10SamotYour call is going over the Internet
17:15.23Korolevwhen I said yes to "over the internet" I meant "over ip"
17:15.34Korolevthere is no internet involved here
17:15.39SamotINVITE sip:18004321000@192.168.1.13:50060 SIP/2.0 <-- WHERE is that number?
17:15.42SamotIs that yours?
17:15.49Korolevat 192.168.1.13
17:15.57Korolevthe call is coming from 192.168.1.12
17:16.02Korolevneither is on the internet
17:16.04SamotThe PBX is .13?
17:16.08Korolevyes
17:16.17Samot18004321000 <-- is that your number?
17:16.34SamotOr are you calling someone?
17:16.39Korolevthat's bank of america, I used it as a placeholder for the actual number
17:16.45SamotFFS.
17:16.52KorolevI'm calling a cellphone that I have in my hand
17:17.06SamotSo you're calling a LOCAL extension on the PBX?
17:17.09[TK]D-FenderSamot, HE'S USING CHAN_MOBILE TO TALK THAT OTHER PHONE VIA BLUETOOTH.
17:17.11Korolevfrom another cellphone that is connected via chan_mobile to asterisk
17:17.16SamotOK.
17:17.21SamotSo there's no audio.
17:17.23[TK]D-Fender<PROTECTED>
17:17.29SamotI get what is happening..
17:17.31Korolevno audio
17:17.41SamotI'm trying to figure out if the call is going to the PSTN or not.
17:17.48SamotVia the mobile carrier or an ITSP
17:18.03Korolevit is going to the pstn via a mobile carrier
17:18.06[TK]D-FenderKorolev, Prove the SIP end independently <-
17:18.07SamotOK
17:18.14SamotThere you go.
17:18.20[TK]D-FenderAnswer
17:18.34[TK]D-FenderPlayback.  Record.  Playback the recording
17:18.45Korolevthat works
17:19.01Korolevthe sip end is working well
17:19.10Korolevthe problem is only with chan_mobile
17:21.13Korolevthat 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
17:21.45Korolevbut I'm very unfamiliar with how bluetooth works or chan_mobile for that matter to know where to look next
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19:41.46tehgooch[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.
19:44.32tehgoochWell, technically if any dest endpoint is "INUSE"
19:50.40[TK]D-Fendersame => n,While($["${SET(i=${SHIFT(extensions)})}"!=""])
19:50.53[TK]D-FenderI fail to see how SET() returns anything
19:58.42[TK]D-FenderI also wouldn't loop twice to do this
19:58.45[TK]D-Fenderyou can do it in one
19:59.10[TK]D-Fenderbuild the dialstring once, checking as you add, and you can abort the moment one that you check comes up busy
20:16.25tehgooch[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.
20:18.13SamotThe while function makes sense.
20:18.32SamotThere's nothing wrong with the while loop. I think he's just saying you don't need to loop twice.
20:19.11tehgoochSamot, 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.
20:19.32SamotIt basically says "While I can SET $i do this"
20:19.56tehgoochSamot, it's comparing the result to the empty string though.
20:20.10SamotI get it.
20:20.12tehgoochOh
20:20.12SamotYou want data
20:20.16tehgoochIt probably returns 0
20:20.21tehgoochwhich is not the empty string
20:20.23SamotThere is that.
20:20.25tehgoochas long as it works
20:21.18[TK]D-FenderI suppose the function returns what it also sets to that var
20:21.35[TK]D-FenderNot really announced in its instructions and feels odd to use
20:21.53SamotOh, I'm not sure it's right for this.
20:22.06SamotBut the while loop, itself, should function just fine.
20:22.09[TK]D-Fenderprobably is fine give the sample it was lifted from
20:22.26[TK]D-Fenderbasically keep going until you hit a blank value
20:22.40Samottechgooch: Now you have to do what everyone does with examples lifted from the Internet.
20:22.45SamotTweak it to your own needs.
20:22.53SamotYou're going to need to refine it.
20:23.05tehgoochof course :)
20:23.39SamotBut I agree with [TK]D-Fender on just checking the INUSE state and building the dialstring then.
20:23.46SamotNot doing it as two different things.
20:23.48tehgoochThe Internet provides and the Internet takes away.
20:24.24tehgoochI thought modularizing it might prove beneficial in the future.
20:25.20SamotI'm a fan of modular programming.
20:25.36SamotBut the only thing I would see you'd want modular is the INUSE check...
20:25.56SamotBut even then, depending on what you are doing it might not fit.
20:26.01SamotDown the road.
20:26.02tehgoochtrue
20:26.27tehgoochWell at least I learned something.
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20:26.40SamotThat's always a plus.
20:29.00tehgoochJust did a test and Set() does return what it set the thing to.
20:30.21tehgooch<3 undocumented functionality
20:30.30SamotWell just make sure the whatchamacallit fires properly.
20:31.37tehgoochI was thinking the 'b' or possibly 'B' Dial() option might work better for my use case, but I don't think it's appropriate.
20:32.45tehgoochI'm not sure how to get the endpoints sent to the Gosub.
20:33.00tehgoochOf course I could just pass them as args, but then I need to specify them in two places.
20:37.02tehgooch'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.
20:37.34tehgooch'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.
20:40.09tehgoochAnyway. 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-Fenderjust What about them?
20:43.34[TK]D-FenderI don't see how those apply to your requirement
21:02.19tehgooch[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.
21:02.28tehgoocher, s/macro/gosub
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21:04.33[TK]D-FenderThat's backwards
21:05.04[TK]D-Fenderyou are ACTUALLY dialing them with that feature
21:05.13[TK]D-Fenderbefore bridging means you both called... and they ANSWERED
21:05.17[TK]D-Fenderit's ABOUT to bridge
21:05.34[TK]D-FenderYou want to make your decision BEFORE attempting to call
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21:34.19tehgooch[TK]D-Fender, Hm, I guess I misunderstood the docs. I'll have to continue reading.
21:39.08tehgoochIt doesn't work if I do the state check in the loop because I removed the first endpoint before the loop.
21:39.38tehgoochI could check before the loop and again in it, but that's duplicating code.
21:41.51[TK]D-Fenderpop the next value.  check it.  Jump if busy, otherwise add.
21:41.52[TK]D-FenderLoop
21:42.10[TK]D-FenderYou don't need 2 loops
21:43.30[TK]D-FenderIn fact.... you shouldn't even HAVE to build a new string.  You're just dialing all of the things being passed.
21:43.45tehgoochok I'll give it a try
21:43.58[TK]D-Fenderit's not like you're EXLUDING busy from the call.  You're aborting if ANY are busy
21:43.59tehgoochyou're right I could just replace the '.'s with '&'s
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22:53.37filefalls over
22:59.29fileoffering multiple audio streams to phones is fun
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23:10.08SamotI'm sure it makes the call interesting.
23:12.23fileSamot: some do a 488, some accept the SDP negotiation but decline the extras
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