IRC log for #asterisk on 20170208

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08:44.24tparcinaSometimes I have problem with attendant call transfer.
08:46.43tparcinaA calls B, B answers and they talk. B puts A on hold, calls C. C answers and talks with B. B hangs up. MOH on A stops and he doesn't hear C. C doesn't hear A.
08:47.15tparcinaI have done debug and I see messages like: Cannot use native DAHDI.  Channel 'SIP/302-00013337' not compatible.
08:47.33tparcinaor: Bridge technology native_dahdi is not compatible with properties of existing bridge.
08:47.44tparcinaBridge technology holding_bridge does not have any capabilities we want.
08:51.59tparcinaHere is the part of debug log: http://pastebin.centos.org/65816/
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09:32.47wasanzyhello
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09:34.01wasanzypart of my dialplan: https://paste.linux.community/view/c284c9a5
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09:36.48wasanzythe problem I have is because the audio has to be played whiles the loop iterates, There is sort of noise after each iteration. Reason I used loop is, when an amount is supplied, I need to join figures. eg. one.alw, two.alw, thousand.alaw    and this will be played individually whiles the loop iterate.
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09:38.13wasanzysupposed the amount is ten thousand and fifty  it means the ten will be played first, then the wiles loops again, play thousand, then and, then fifty.
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09:38.46wasanzyI realized this is not efficient. I need help with ideas how to make this better.
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09:58.48SamotWhy are you playing back every digit like that?
09:59.56SamotYou can just use SayNumber().
10:00.35SamotSayNumber(10050) will playback "Ten thousand fifty)
10:00.40SamotSayNumber(10050) will playback "Ten thousand fifty"
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10:03.09wasanzySamot: I actually tested the SayNumber() but it didn't sound professional. Is there a way to change the voice?
10:03.52SamotYeah, buy or download another voice package.
10:04.54wasanzywhere can I download from and how do I tell SayNumber() which voice to use?
10:04.56SamotOr invest in a TTS.
10:05.14SamotYou're going to have to do some googling on where to find a decent package.
10:05.19SamotI don't know why  you want.
10:05.33SamotApparently Allison isn't professional enough.
10:05.45SamotWhich is funny, because I would pay her.
10:06.06wasanzypay her?
10:06.17SamotAllison is a real person.
10:06.34SamotShe recorded pretty much all the standard prompts you use in Asterisk.
10:06.59wasanzyoh I see
10:07.03SamotShe is a profession voice over/IVR voice talent.
10:07.07SamotShe is a professional voice over/IVR voice talent.
10:07.16WIMPyThe american ones, that is. There are other english prompts available.
10:07.23SamotYes.
10:07.52wasanzywell so I can as well get someone to record voice then I use right?
10:07.59SamotYou can.
10:08.08SamotOr you can see what is out there..
10:08.24wasanzyI mean for SayNumber()
10:08.31SamotSure.
10:08.40SamotYou can have someone record all the numbers..
10:08.47WIMPySome sets can be dowenloaded via the Asterisk installer. But there are more out there.
10:08.51SamotAll the variations of saying the numbers...
10:09.08wasanzyI see
10:09.27SamotYou can replace the recordings or use any recordings youd like.
10:09.46wasanzyI will need to look at all the variations of numbers then
10:10.04SamotBut if you're going to generate playback on the fly for your callers....
10:10.15SamotYou might want to look at a TTS engine..
10:10.44wasanzySamot: I tried google tts, marytts but didn't workout well for me
10:11.01SamotA decent TTS engine isn't going to be free.
10:11.30wasanzywell,  I dont care buying one so far as it does what I want
10:11.40WIMPyI found espeak to be not that bad, but if you want somethign that really sounds good, you will have to take some money.
10:11.46SamotThen you should get googling...
10:11.55SamotYup, Cepstral.
10:12.47wasanzyok
10:13.06wasanzyI see the SayNumber() doesn't mention "and"
10:13.20wasanzyand "point"
10:13.54WIMPySayNomber has its configuration to do the magic for different languages.
10:13.55SamotNo it says a number.
10:14.21SamotYou want to say a monetary balance...
10:15.26SamotBut then again, why wouldn't it say something like "Your balance is one hundred twenty dollars and fifty cents"
10:15.54WIMPysay.conf
10:16.01wasanzySamot: that is the problem
10:16.07SamotWhy would you say "point"
10:16.21SamotUnless you're given back an actual decimal or float.
10:16.48wasanzySamot: I don't mind something like "Your balance is one hundred twenty dollars and fifty cents"
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10:19.10SamotOK
10:19.20SamotThere is a "current balance" recording
10:19.25SamotThere is a dollars recording
10:19.45Samota cent recording and an "and" recording.
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10:20.42SamotSayNumber(120) Playback(dollars/and) SayNumber(50) Playback(cents)
10:20.47SamotSomething like that.
10:22.02SamotYou can even start it with Playback(current_account_balance/is)
10:22.03wasanzyok, am going through the config to see what I can change
10:22.26wasanzythe "/" ?
10:22.55wasanzybut then, is it possible to call functions on one line like that? not tried it before
10:22.57SamotIt's early, I didn't want to use the brain power to remember what the syntax was for playback of multiple files.
10:23.18WIMPyNo, you can't.
10:23.19SamotNo, I was just spitballing.
10:23.25SamotI wasn't writing actual dialplan code.
10:23.49wasanzyis Playback(file1 & fle2) etc
10:23.55SamotSure.
10:24.12SamotSo Playback(dollars & and)
10:24.26SamotOh whatever files you want to string together.
10:24.29Samotor
10:24.37wasanzyyea sure
10:24.44wasanzylet me review the config and get back
10:25.01SamotHonestly if this was for paying customers..
10:25.19SamotI'd get a TTS engine that can playback things specifically for them on the fly.
10:27.17wasanzywell the asterisk server can't go internet so using external tts will be a problem
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11:14.22shilongasterisk 13 on hangup request sip bye is getting sent to only one participant on a bridge
11:14.28shilongany ideas?
11:37.35SamotShow a call that does this.
11:39.29andrelcmHi guys, due to a problem with a specific provider, I need to change the hangupcause of calls in cases where I get "NETWORK_OUT_OF_ORDER" as a hangup cause.
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11:58.17SamotSo your provider is sending you the error?
12:00.07SamotYou also can't change the hangup cause. That is something they send you, you're just getting the response.
12:00.37SamotNow if you want to handle the caller channel differently based on the causecode, you can do that.
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12:08.57andrelcmyes, unlike the other providers, this specific provider send me a wrong hangupcause in an specific case
12:16.38SamotOK. So why do you need to change the causecode?
12:16.47SamotCant you just deal with the code based on what it is?
12:17.30WIMPyI'd rather ask where you want to change it.
12:18.01SamotI'm guessing back to the caller.
12:18.21SamotWhich I'm also guessing isn't a phone.
12:18.28SamotBecause they generally don't care.
12:18.52WIMPyThat would be easily done in the dialplan.
12:18.59SamotYes.
12:19.16Samot7:00:36 AM <Samot> Now if you want to handle the caller channel differently based on the causecode, you can do that.
12:20.28WIMPyBut generelly speaking it doesn't really matter any more these days. Either you get the person you wan to answer or anythign else is just a general call failure.
12:20.43andrelcmit's not a phone. It's a third party soluction
12:20.58SamotThat's untrue.
12:21.43Samotandrelcm: You're going to have to check the causecodes from the provider..
12:21.52WIMPyIn the VoIP ages you need quite a bit of luck for anythign to work in a sensible way.
12:22.08SamotI disagree.
12:22.11SamotBut OK.
12:22.23WIMPyWhat does "third party solution" mean?
12:22.34andrelcmIn my case, the third party solution, receive the info and deal with it, but as they receive the "wrong" cause, the classification ends wrong, impacting on the final result
12:22.41SamotYes.
12:22.43andrelcmbut as you say Samot
12:22.56SamotYou'll have to send a HangUp back to the caller based on the code.
12:23.04WIMPyThen set another cose with Hangup().
12:23.16WIMPycause
12:23.18WIMPycode
12:23.26WIMPyWboth? whatever.
12:23.58andrelcmok guys, thks for the answers
12:24.13Samotandrelcm: The issue is with 500
12:24.14andrelcmi will try here
12:24.21SamotIt's generic.
12:24.41andrelcmno 503
12:24.41SamotIt means they aren't sending back any specific code to their error.
12:24.43SamotOK
12:24.47SamotThat's different.
12:26.07andrelcmtheir response is 503 for a range o problems, that others providers returns as separated problems
12:26.39Samot503 is basically "Service is unavailable"
12:27.01SamotThat is generally the response if you have a channel limit..
12:27.09Samotor you've hit a CPS limit.
12:29.05andrelcmYes, but this provider is sending 503, for example, in cases where "Network out of order", "Invalid Number" ...
12:30.01SamotSo when you call the same number out a different provider...
12:30.21SamotYou get back "Invalid Number", so like a 404 instead of the 503
12:30.53andrelcmyes
12:31.50SamotIf accuracy of the causecode is needed, you shouldn't use them.
12:32.03SamotBecause it's their choice to send back the 503.
12:32.11SamotBecause it's still valid.
12:32.28SamotWell they could actually send back a 603 but I can see why.
12:33.42SamotWild guess, they are getting the 4xx errors from their upstream..
12:33.47SamotBut sending you 5xx errors.
12:34.55andrelcmthat is what they answer me
12:35.17SamotPardon?
12:39.28andrelcm<Samot> Wild guess, they are getting the 4xx errors from their upstream.. <Samot> But sending you 5xx errors.
12:39.39SamotAnyways, like I said. If having accurate hangup reporting is needed, this provider isn't best option for you.
12:40.40SamotBecause if they send you a 503 for six different reasons, how can you guess what that reason is to send the proper response back to the caller?
12:41.04SamotThe best you can do is send them a causecode they accept.
12:41.15SamotBut it will always be the same.
12:41.24WIMPyMaybe they send additional headers. But probably not.
12:42.18Samotandrelcm: Show a SIP debug of a call. Let's see their reply message.
12:43.31SamotAnd what hangup code Asterisk is catching in ${CAUSECODE}
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14:24.29wonderworldi am looking for a way to check if an established call is actually using encryption for the mediastream or not. i thought i setup SRTP correctly but during the (working) call i get SSL errors like: tcptls.c:397 tcptls_stream_close: SSL_shutdown() failed: 5
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14:26.33SamotThat can be seen in the SDP of the SIP messages..
14:26.44SamotSo a SIP debug will show it.
14:26.57wonderworldSamot: thanks, I'll have a look
14:27.18SamotAre you using a self-signed or a public cert?
14:27.27wonderworldletsencrypt...
14:27.37SamotK.
14:34.49wonderworldwhat should i be looking out for? I found a=fingerprint:SHA-256 00:11:22:33:44:55.........
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14:54.40SamotThat usually indicates encryption.
14:54.45SamotShow the debug.
15:01.58wonderworldhttps://ghostbin.com/paste/97ezo
15:02.05wonderworldcheck line 556 pls
15:02.49wonderworldSDP stuff starts @ 170
15:03.57wonderworldsignalling is via WSS
15:04.39SamotYeah, I just saw that.
15:04.40Samot[2017-02-08 15:29:20] WARNING[7752][C-00000002]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
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15:22.23wonderworldyes, thats the second error that made me suspicious
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19:09.20blistovHey guys. How do I determine from a pcap, which RTP stream is associated with a given call?
19:10.09blistovLike, looking at wireshark SIP flow, and lots of calls don't have any associated RTP shown, but I'm positive the RTP stream exists (the call was successful and I can find the RTP stream manually).
19:10.21blistovBut what packet actually says where our RTP stream is going to be?
19:13.55[TK]D-FenderSIP obviously
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19:22.30blistovYea, found it. SDP. derp.
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19:23.30blistovSo I have an issue. Asterisk RTP port range is set to 10000-10100, but i'm ocassionally, on outbound calls, seeing RTP being setup outside of that range, which of course results in one way audio.
19:23.35*** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-lxxmnehmeexjeyrh)
19:23.49blistovFirewall NAT is set with static port for my PBX to any TRUNK IP.
19:23.52blistovsame range.
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19:24.10blistovin an outbound call, who determines the RTP port? Me or my trunk provider?
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19:32.22[TK]D-FenderThey have THEIR port, you have YOURS
19:32.33[TK]D-Fenderneither side dictates the other
19:33.00[TK]D-FenderYou have to forward your inbound range and you have to permit outbound to theirs
19:33.09blistovSo when your PBX is behind a firewall though, how do you configure the firewall to allow that RTP stream in?
19:33.18[TK]D-FenderAnd given you have no control, that means you need to allow ALL outbound
19:33.34blistovAll outbound is allowed;.
19:33.46[TK]D-Fenderthe INBOUND only targets YOUR port.  The SOURCE port they send from is irrelevant
19:34.47blistovRight, but that's what I'm not seeing though. My config says 10000-10100. SDP from the trunk ends up occasionally specifying that they're initiating their RTP to me on port 45000 or something.
19:35.18blistovUSUALLY, they stay in the configured range, but one in 50 calls or so will show up to some random port.
19:35.39blistovForgive me if i'm retarded here. I have an actual medical excuse :)
19:36.12blistovBut I just don't see how, if the SDP comes from the trunk provider and just randomly picks an RTP port they're going to send to me, how that could possibly work.
19:36.14[TK]D-Fendertheir rtp can be wherever but they should never send to anything but your range
19:37.05blistovRight. Which is what I figured. So if they're sending RTP to a port outside of my specified range, that's THEIR problem right?
19:38.20[TK]D-Fenderthat should be, unless you are allowing REINVITES for a device inside your LAN and that reinvite as initiated from the phone has its OWN ideas.
19:38.49[TK]D-FenderBut I was hoping your special condition wouldn't be allowing thatt to happen...
19:39.09blistovHeh. canreinvite is disabled on asterisk.
19:39.51blistovWhich is why I'm stumped, and have been stumped for 2 years. To make matters worse, wireshark has a bug where it's not correctly associating (sometimes) the RTP stream back to the SIP headers.
19:40.26blistovSo I have to manually read every goddamn SIP flow and manually inspect every SDP and match it up to the resulting RTP stream.
19:40.53blistovBut, this is what kills me. This problem goes away for a few hours or sometimes weeks, if I reboot my firewall.
19:42.26[TK]D-FenderGot another device that has any reason to talk directly out?
19:42.45blistov(medical issue being that I'm sorta like the guy from Momento, in that i very often cannot form short term memory, so retaining static data is very difficult, and I have to instead rely on conceptualization and logic for everything).
19:42.55blistovNothing I can think of.
19:43.01blistovIn terms of VOIP anyhow.
19:43.37[\\\]I can't login to a freepbx box.  Every time I click FreePBX Adminstration, it doesn't go anywhere.  UCP, Operator Panel links both work as expected.
19:43.52[\\\]I've already tried fwconsole firewall stop on the offchance that was it.
19:44.01[\\\]oops, wrong channel
19:44.35blistovLooking at the call flow for an affected call, I see I fire out SDP to the trunk, asking for them to respond back on port 10098. Then "giving a try, and then they come back with another SDP stating they'll be expecting me to send my RTP from port 26406.
19:45.03[TK]D-Fenderwell that's what you send.. which doesn't mater
19:45.12[TK]D-Fenderits what you LISTEN on that does
19:45.44blistovoh sorry. but then they say they're sending back to me on port 34714
19:45.57blistovwhich of course is outside my range specified, and my FW doesn't have NAT for that port.
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19:46.13blistovI'm just confused about who's in charge of determining the RTP ports I think.
19:47.04blistovI initiate an outbound call, why does the trunk come back with SDP specifying that they'll be sending RTP back to me on 34714? And what did my OUTBOUND SDP packet actually DO then? Are they just ignoring me?
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20:01.49[TK]D-FenderWhat they send FROM doesn't matter
20:02.13[TK]D-Fenderthey can send FROM 34714 on ttheir side... as long as it is TO a port in your rtp.conf range.
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20:22.21blistovRight, which they are not doing.
20:22.43blistovBut that's where it gets weird. I can hear the incoming audio in the cases where outbound does not work.
20:23.27blistovAnd I don't understand why. I'm not allowing RTP in on the port outside my range, but I'm clearly receiving anyway. But for some reason I'm not sending out.
20:23.36blistovIt's like the symptoms are backwards from what I'd expect.
20:27.39[TK]D-FenderI'd really need to see a whoe thing together in one place
20:27.48[TK]D-Fenderbut can't concentrate on that ATM
20:29.07blistovNp man. Thanks for your help. I'm waiting for the trunk provider to get back to me. maybe they know something. If not, i'll be back asking about this again soon :)
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22:35.03wonderworldwhat would be the correct way to verify, that a RDP stream is encrypted? i setup encryption, but i am unsure if it actually works.
22:35.53wonderworldi would need to verify it server-side, because the client wouldn't tell me
22:39.01SamotThat hasn't change.
22:39.09SamotIt's still in the SDP in the SIP messages.
22:39.23wonderworldSamot: hey....
22:39.40wonderworldOK, but I still don't know what I should be looking out for
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22:45.29SamotWhat version of Asterisk?
22:46.21wonderworld13.1.0
22:48.26SamotYou should upgrade.
22:48.37SamotWhat browser are you using?
22:48.54SamotYou're about 15 releases behind.
22:50.16wonderworldi know. it's just a dev box.
22:51.00wonderworldi am testing with chrome / sipjs atm
22:53.08SamotOK, well in the little googling I've done on the errors you've had..
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22:53.29SamotI see there were some on 13.1 that had issues like this with WebRTC and they said 13.2 fixed it.
22:53.53SamotSo you should update. 13.1 is like the first release after 13.x went from beta to live.
22:54.01wonderworldi manged to get rid of the errors by upgrading my client
22:54.07SamotOr that.
22:54.28wonderworldthey are gone, but i still don't know if the connection is encrypted or not :)
22:54.43SamotYou will see it in the SDP
22:55.22wonderworldyes, thats my question... what would the SDP say, if it was encrypted?
22:56.46SamotIt would have the fingerprint.
22:56.59SamotIf it's over WSS, it has it.
22:57.03SamotBecause that's the point.
22:57.30Samota=fingerprint:SHA-256 22:06:E7:C7:DA:D2:D9:FC:85:47:6C:6C:3F:45:56:90:10:36:C3:1E:25:4C:DA:DF:ED:FA:CD:A2:78:A2:62:9B
22:57.34Samot^^^ Something like this.
22:58.05wonderworldWSS is working, but thats only for the signalling part
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22:58.18wonderworldi want to make sure that the media transport is encrypted as well
22:59.00wonderworld(SRDP)
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23:00.04Samotm=audio 52086 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
23:00.07Samot^^^^^
23:01.39wonderworldok, so it's the TLS in that line?
23:03.29Samota=fingerprint:SHA-256 22:06:E7:C7:DA:D2:D9:FC:85:47:6C:6C:3F:45:56:90:10:36:C3:1E:25:4C:DA:DF:ED:FA:CD:A2:78:A2:62:9B
23:03.44SamotRTP isn't "encrypted"
23:04.00SamotIt's encapsulated because UDP doesn't do ecryption.
23:04.12*** join/#asterisk pa (~pa@unaffiliated/pa)
23:04.27SamotIt's encapsulated because UDP doesn't do encryption so that means RTP doesn't either.
23:05.00SamotSo it uses a crypt cipher to encapsulate it based on the cert.
23:07.41SamotThat fingerprint output shows the ecapsultion..
23:07.46SamotThat fingerprint output shows the encapsultion..
23:14.29wonderworldis there a place where i could read up on that? i really want to understand it but it seems complicated to get proper documentation....
23:15.28SamotGoogle
23:15.43Samotsecure RTP
23:15.47Samotsecure calling
23:15.53Samotencrypted SIP
23:15.57SamotWebRTC
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23:23.51blistov[TK]D-Fender: hey man. If you got time, here's what I've found. - http://pastebin.com/ftkUzTsQ
23:24.16blistovI see where the problem is now, but can't figure out why it's occurring. Seems like legit a PFsense issue to me.
23:27.53Samotblistov: Show a real SIP debug.
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23:38.52blistovSamot: in what format? I'm just looking at the PCAPs from the PBX and then again at the firewall. Definitely seeing that SDP agrees everywhere, that we should be using port 50776. PBX sends out via that port, but PFsense seems to be natting the port and firing it out as port 58308.
23:39.48Samotsip set debug on
23:44.24blistovThat quickly fills up the HD. It can take hours or weeks before the issue occurs again.
23:44.33blistovI've never managed to catch it with SIP debug from asterisk.
23:54.55SamotOK you realize that the RTP port that comes from the PBX..
23:55.00SamotIn the INVITE...
23:55.17SamotIsn't the RTP port provided in the 1xx replies?
23:55.27SamotYou call X
23:55.39SamotYou are saying my RTP port is 50776...
23:56.02SamotThey answer, they say "Hey my RTP port is 17744"
23:56.19SamotSo you send RTP to X on port 17744
23:56.30SamotAnd X sends RTP to you on port 50776

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