IRC log for #asterisk on 20200418

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04:59.45mukshi everyone; in pjsip.conf, how can i set an endpoint to use [transport1] an IPv4 TLS transport, and [transport2] an IPv6 TLS transport, but *not* [transport3] a plain UDP transport ?
05:00.17muksi was instructed here before that not setting the transport= setting in an [endpoint] means it will listen on all transports
05:01.26muksbut i want internal endpoints to specifically not use [transport3] (to force use of TLS). however [transport3] is used for an external endpoint that doesn't support TLS.
05:02.14muksso 3 transports are configured.. but i want internal endpoints to use 2 of them only.. how do i achieve this? multiple transport= lines in the [endpoint] doesn't seem to do it
05:04.32muksthis is asterisk 16
05:12.10drmessanoNested includes?
05:22.00muksis include => syntax supported in pjsip.conf? and will it work to configure multiple transports?
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06:00.06Samotmuks: Will different devices register to this endpoint?
06:00.35Samotmuks: If so, will different devices use different transports? I.e. one is TLS and the other is UDP?
06:11.09allizomI'm trying to set up asterisk to use my ITSP account. Received calls use the tel: URI and my asterisk is giving them "SIP/2.0 416 Unsupported URI Scheme". I'm using pjsip. Unfortunately my provider is not going to change their configuration, and I'd prefer to keep using pjsip. Is there anything I can do about this?
06:14.19allizomthey're using ZTE equipment, not sure if there's a way to have them use sip: URIs by crafting my requests somehow
06:19.30muksSamot: currently our staff connect via SIP over TLS and SRTP media encryption is forced for them; i'm trying to get our "incoming number" SIP provider to also route calls to this PBX. the SIP provider only supports SIP over UDP.
06:20.02SamotOK so you setup two transports.
06:20.36SamotTell the SIP provider endpoint to use the UDP transport and the others to use the TLS transport.
06:21.20muksit would be 3.. as we currently have two transports, one listening on IPv4 and another IPv6, both doing TLS and SRTP. to support these, currently our internal endpoints don't have a transport= setting configured
06:21.55muksif i add a UDP transport, the internal endpoint will listen on that too unless it is restricted to just the TLS transports
06:22.11muksthat's what i'm not able to figure out :)
06:23.01SamotIf you set a transport, it uses that transport.
06:23.03SamotJust that one.
06:23.27SamotIf you don't set a transport it should select the first one that matches based on the contact uri
06:23.49SamotSo if the contact uri shows TLS over X port then it tries to match the transport for it.
06:24.22muksi want to prevent the staff to connect over SIP over UDP
06:24.35muksbut allow IPv4 and IPv6 transports for them
06:24.40mukshow can i achieve that?
06:24.52SamotSo a single endpoint needs both transports?
06:25.08muksgive me a moment, let me paste an edited pjsip.conf
06:25.15muksit'll be easier with that
06:27.47drmessanoI was thinking more templates that include other templates
06:28.04SamotWell that's the beauty of PJSIP.
06:28.18SamotDon't tell the endpoint a transport= it will match the one needed.
06:28.55SamotBut you cannot do like chan_sip and do transport=udp,tcp,tls
06:29.43mukshttps://mukund.org/static/tmp/pjsip.conf
06:30.01muks(a snippet of the full pjsip.conf)
06:30.59muksif i add the commented-out [transport-udp-ipv4] and another endpoint for the incoming number, will [endpoint-internal] also be able to use it?
06:31.44SamotOK so I'm going to ask this again, does a single endpoint need to use both transports?
06:32.00SamotBecause, as I asked before, device A is doing X and device B is doing Y?
06:32.10SamotBut both registering to Z endpoint.
06:32.55muksthese are mobile SIP endpoints.. they have the sip.example.org domain configured in the clients and they travel.. so they could access the service via IPv4 or IPv6
06:33.11muksbut they would use only 1 transport at a time
06:33.28SamotOK and do they have control over their softphones?
06:33.34muksyes
06:36.13SamotDo they know the UDP port?
06:36.22SamotThey need to know how to get to it over UDP
06:36.51muksstaff are not going to access it via udp. they will connect to it via TCP (and TLS).
06:37.09SamotRight but your concern is that they'll get the wrong transport
06:37.11muksthe udp transport (commented out in this config) is for the incoming number
06:37.19muksyes
06:37.22muksexactly
06:37.26SamotTransports are based on a few things including the port they are bound to.
06:37.34drmessanoWhy offer TCP at all if the devices can do TLS?
06:37.40mukswell staff are programmers
06:37.44SamotSo your users would need to know the details for TLS for IPv4/IPV6
06:37.53SamotOK
06:37.59muksdrmessano: i meant SIP over TLS over TCP.. i.e., not UDP
06:38.27SamotSo if they are programmers they hack it?
06:38.29muksthey can configure their devices properly (and there's a section in the handbook on config settings for the SIP service)
06:38.42ZombieIf I can make time, I want to try and get this working again.
06:38.59SamotSo now it's about the employees hacking at the phone system?
06:39.15SamotWhy would they want to try and use UDP?
06:39.18muksbut i don't want it to be possible to authenticate via UDP into the internal extensions..
06:39.24drmessanoTLS and UDP sounds like 2 transports to me
06:39.39SamotAgain the phones would need to know the PORT
06:39.48SamotTLS and UDP can't listen on the same PORT
06:40.00SamotSame IP sure, not the same port.
06:40.07muksstaff must use TLS. we have conference room conversations about security vulnerabilities.. and it would not be a good idea to allow a plain-text way to login into the system
06:40.20drmessanoomg
06:40.22SamotThere's no plain text in SIP.
06:40.30SamotPasswords are not passed in SIP
06:40.47SamotI'm not sure I'm following the problem.
06:40.57SamotYou make a transport for each that you need.
06:41.01mukssorry if i'm getting away from the point
06:41.09SamotTLS IPv4, TLS IPv6, IPv4 UDP
06:41.26muksby plain-text, i don't mean passwords.. i want staff to use the TLS transport
06:41.27SamotThe SIP provider endpoint uses IPv4 UDP
06:41.35muksyes
06:41.39SamotEverything else is TLS either IPv4 or IPV6.
06:41.43SamotDone.
06:41.48SamotSo what's the issue?
06:41.52muks06:41 < Samot> TLS IPv4, TLS IPv6, IPv4 UDP
06:42.12muksthese 3 transports are in the pjsip.conf file at: https://mukund.org/static/tmp/pjsip.conf
06:42.17SamotOK
06:42.26muksthe UDP is commented, but it can be uncommented :)
06:42.29SamotSo phones using TLS will not get on the UDP transport.
06:42.37muksok.. with these 3 transports
06:43.05muksand the [endpoint-internal] not having a transport= option, a staff member could misconfigure their client to use plain UDP, correct?
06:43.12SamotHOW?
06:43.16SamotDo they know the PORT?
06:43.36SamotYou are giving them the details they need to connect with
06:43.41SamotIt will be the TLS details.
06:43.43SamotCorrect?
06:44.11SamotSo if they don't have the information about UDP they can't use it.
06:44.23muksin a client (e.g., linphone), they pick a radio button for "UDP", "TCP", or "TLS"
06:44.32SamotThey still need the PORT
06:44.37SamotIP:PORT
06:44.46muksyou mean i should configure non-standard ports?
06:45.01SamotYes, don't use 5060
06:45.03muksi.e., so that they are forced to pick a unique port
06:45.03muksah
06:45.12SamotIt's only for the SIP provider.
06:45.18muksah
06:45.19SamotThey have to use a port for TLS
06:45.20muksgood point
06:45.34Samot5061 or whatever you've chosen.
06:45.45SamotIf you don't want them on 5060 UDP then don't make a transport for it.
06:46.02muksmaybe i can give the SIP provider a non-standard UDP port
06:46.08muksgood point
06:46.13SamotYou sure can.
06:46.20muksthank you for this tip
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11:33.12allizomI've seen there is no support in asterisk for incoming calls which use tel: URIs, but the bug ( https://issues.asterisk.org/jira/browse/ASTERISK-26894 ) links to a private issue: SWP-10585. Is there any ongoing work on this?
11:46.36fileany progress would be posted on the ASTERISK issue
11:47.43allizomok file
11:48.10allizomso am I right in assuming the only way right now would be to use chan_sip?
11:48.23fileas chan_pjsip has no support for it, yes
11:48.37allizomthanks
11:49.03allizomI started out using asterisk with pjsip, so now I have to learn backwards ;P
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14:21.57allizomfile: can I mangle SIP headers with asterisk? My idea would be to convert the URI my provider hands out into the sip: scheme
14:22.30allizomit's predictable
14:22.30fileno
14:23.30allizomdo you happen to know any simple enough software which could do that?
14:23.31SamotYou'll need something like Kamailio or OpenSISPs
14:23.43Samot"Simple enough" is also relative.
14:24.00allizomI don't need bells and whistles, that's it
14:24.17SamotWell they are like Asterisk.
14:24.27SamotYou have to give them the bells and whistles.
14:25.11SamotBut Asterisk is not designed to manipulate headers like that.
14:25.41SamotNor is it designed to let you act off Replies or various request types.
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16:06.31ZombieI'm just fucked.
16:08.12SamotPardon?
16:15.03ZombieI've fucked over my config again.
16:20.48SamotHow so?
16:35.17drmessanoSamot: What part of that didn't you fucking understand?
16:35.26drmessanoJeez man
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16:36.55SamotI'm dense.
16:36.59SamotSue me.
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18:31.33_pepo_Hello friends, I am working with ODBC and I have a problem because I cannot do an INSERT in a table; could you please guide me how an INSERT function should be declared in both func_odbc and dialplan?
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18:57.43_pepo_It worked with:
18:57.43_pepo_same => n, Set (ODBC_MYFUNCTION () = $ {myVariable})
18:57.43_pepo_Thanks anyway!
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19:00.44[TK]D-FenderStop adding extra spaces
19:15.44_pepo_sorry... my paste !
19:15.49_pepo_tnx
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21:06.54AristideHello. I have a big problem with Asterisk and sounds. I explain :
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21:07.35AristideWhen i forward call without response to 4000, I get sound and its work. But if I answer call, forward to 4000, caller don't get sounds but instructions are runned (because number is blacklisted)
21:07.56AristideI use « exten => 4000,1,Goto(addtoblacklist,s,1 » for forward
21:08.14Aristide(I have forget a « ) » in paste :D)
21:08.47AristideAnd part of configuration file https://pastebin.com/HNkH1Yst :x
21:26.55AristideAnd if I launch asterisk in debug mode, I don't have any errors
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22:41.21retentiveboyTrying to migrate from ancient instance that used sip and users to pjsip_wizard.  Used to use linenumber in a user do I could use phoneprov to configure multiple lines on a station.  Is there an equivalent?
22:52.42retentiveboyphoneprov/LINE=2 works...  Note to self...  Always look in the code.
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