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04:59.45 | muks | hi 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 ? |
05:00.17 | muks | i was instructed here before that not setting the transport= setting in an [endpoint] means it will listen on all transports |
05:01.26 | muks | but 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. |
05:02.14 | muks | so 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 |
05:04.32 | muks | this is asterisk 16 |
05:12.10 | drmessano | Nested includes? |
05:22.00 | muks | is include => syntax supported in pjsip.conf? and will it work to configure multiple transports? |
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06:00.06 | Samot | muks: Will different devices register to this endpoint? |
06:00.35 | Samot | muks: If so, will different devices use different transports? I.e. one is TLS and the other is UDP? |
06:11.09 | allizom | I'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? |
06:14.19 | allizom | they're using ZTE equipment, not sure if there's a way to have them use sip: URIs by crafting my requests somehow |
06:19.30 | muks | Samot: 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. |
06:20.02 | Samot | OK so you setup two transports. |
06:20.36 | Samot | Tell the SIP provider endpoint to use the UDP transport and the others to use the TLS transport. |
06:21.20 | muks | it 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 |
06:21.55 | muks | if i add a UDP transport, the internal endpoint will listen on that too unless it is restricted to just the TLS transports |
06:22.11 | muks | that's what i'm not able to figure out :) |
06:23.01 | Samot | If you set a transport, it uses that transport. |
06:23.03 | Samot | Just that one. |
06:23.27 | Samot | If you don't set a transport it should select the first one that matches based on the contact uri |
06:23.49 | Samot | So if the contact uri shows TLS over X port then it tries to match the transport for it. |
06:24.22 | muks | i want to prevent the staff to connect over SIP over UDP |
06:24.35 | muks | but allow IPv4 and IPv6 transports for them |
06:24.40 | muks | how can i achieve that? |
06:24.52 | Samot | So a single endpoint needs both transports? |
06:25.08 | muks | give me a moment, let me paste an edited pjsip.conf |
06:25.15 | muks | it'll be easier with that |
06:27.47 | drmessano | I was thinking more templates that include other templates |
06:28.04 | Samot | Well that's the beauty of PJSIP. |
06:28.18 | Samot | Don't tell the endpoint a transport= it will match the one needed. |
06:28.55 | Samot | But you cannot do like chan_sip and do transport=udp,tcp,tls |
06:29.43 | muks | https://mukund.org/static/tmp/pjsip.conf |
06:30.01 | muks | (a snippet of the full pjsip.conf) |
06:30.59 | muks | if i add the commented-out [transport-udp-ipv4] and another endpoint for the incoming number, will [endpoint-internal] also be able to use it? |
06:31.44 | Samot | OK so I'm going to ask this again, does a single endpoint need to use both transports? |
06:32.00 | Samot | Because, as I asked before, device A is doing X and device B is doing Y? |
06:32.10 | Samot | But both registering to Z endpoint. |
06:32.55 | muks | these 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 |
06:33.11 | muks | but they would use only 1 transport at a time |
06:33.28 | Samot | OK and do they have control over their softphones? |
06:33.34 | muks | yes |
06:36.13 | Samot | Do they know the UDP port? |
06:36.22 | Samot | They need to know how to get to it over UDP |
06:36.51 | muks | staff are not going to access it via udp. they will connect to it via TCP (and TLS). |
06:37.09 | Samot | Right but your concern is that they'll get the wrong transport |
06:37.11 | muks | the udp transport (commented out in this config) is for the incoming number |
06:37.19 | muks | yes |
06:37.22 | muks | exactly |
06:37.26 | Samot | Transports are based on a few things including the port they are bound to. |
06:37.34 | drmessano | Why offer TCP at all if the devices can do TLS? |
06:37.40 | muks | well staff are programmers |
06:37.44 | Samot | So your users would need to know the details for TLS for IPv4/IPV6 |
06:37.53 | Samot | OK |
06:37.59 | muks | drmessano: i meant SIP over TLS over TCP.. i.e., not UDP |
06:38.27 | Samot | So if they are programmers they hack it? |
06:38.29 | muks | they can configure their devices properly (and there's a section in the handbook on config settings for the SIP service) |
06:38.42 | Zombie | If I can make time, I want to try and get this working again. |
06:38.59 | Samot | So now it's about the employees hacking at the phone system? |
06:39.15 | Samot | Why would they want to try and use UDP? |
06:39.18 | muks | but i don't want it to be possible to authenticate via UDP into the internal extensions.. |
06:39.24 | drmessano | TLS and UDP sounds like 2 transports to me |
06:39.39 | Samot | Again the phones would need to know the PORT |
06:39.48 | Samot | TLS and UDP can't listen on the same PORT |
06:40.00 | Samot | Same IP sure, not the same port. |
06:40.07 | muks | staff 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 |
06:40.20 | drmessano | omg |
06:40.22 | Samot | There's no plain text in SIP. |
06:40.30 | Samot | Passwords are not passed in SIP |
06:40.47 | Samot | I'm not sure I'm following the problem. |
06:40.57 | Samot | You make a transport for each that you need. |
06:41.01 | muks | sorry if i'm getting away from the point |
06:41.09 | Samot | TLS IPv4, TLS IPv6, IPv4 UDP |
06:41.26 | muks | by plain-text, i don't mean passwords.. i want staff to use the TLS transport |
06:41.27 | Samot | The SIP provider endpoint uses IPv4 UDP |
06:41.35 | muks | yes |
06:41.39 | Samot | Everything else is TLS either IPv4 or IPV6. |
06:41.43 | Samot | Done. |
06:41.48 | Samot | So what's the issue? |
06:41.52 | muks | 06:41 < Samot> TLS IPv4, TLS IPv6, IPv4 UDP |
06:42.12 | muks | these 3 transports are in the pjsip.conf file at: https://mukund.org/static/tmp/pjsip.conf |
06:42.17 | Samot | OK |
06:42.26 | muks | the UDP is commented, but it can be uncommented :) |
06:42.29 | Samot | So phones using TLS will not get on the UDP transport. |
06:42.37 | muks | ok.. with these 3 transports |
06:43.05 | muks | and the [endpoint-internal] not having a transport= option, a staff member could misconfigure their client to use plain UDP, correct? |
06:43.12 | Samot | HOW? |
06:43.16 | Samot | Do they know the PORT? |
06:43.36 | Samot | You are giving them the details they need to connect with |
06:43.41 | Samot | It will be the TLS details. |
06:43.43 | Samot | Correct? |
06:44.11 | Samot | So if they don't have the information about UDP they can't use it. |
06:44.23 | muks | in a client (e.g., linphone), they pick a radio button for "UDP", "TCP", or "TLS" |
06:44.32 | Samot | They still need the PORT |
06:44.37 | Samot | IP:PORT |
06:44.46 | muks | you mean i should configure non-standard ports? |
06:45.01 | Samot | Yes, don't use 5060 |
06:45.03 | muks | i.e., so that they are forced to pick a unique port |
06:45.03 | muks | ah |
06:45.12 | Samot | It's only for the SIP provider. |
06:45.18 | muks | ah |
06:45.19 | Samot | They have to use a port for TLS |
06:45.20 | muks | good point |
06:45.34 | Samot | 5061 or whatever you've chosen. |
06:45.45 | Samot | If you don't want them on 5060 UDP then don't make a transport for it. |
06:46.02 | muks | maybe i can give the SIP provider a non-standard UDP port |
06:46.08 | muks | good point |
06:46.13 | Samot | You sure can. |
06:46.20 | muks | thank you for this tip |
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11:33.12 | allizom | I'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? |
11:46.36 | file | any progress would be posted on the ASTERISK issue |
11:47.43 | allizom | ok file |
11:48.10 | allizom | so am I right in assuming the only way right now would be to use chan_sip? |
11:48.23 | file | as chan_pjsip has no support for it, yes |
11:48.37 | allizom | thanks |
11:49.03 | allizom | I started out using asterisk with pjsip, so now I have to learn backwards ;P |
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14:21.57 | allizom | file: can I mangle SIP headers with asterisk? My idea would be to convert the URI my provider hands out into the sip: scheme |
14:22.30 | allizom | it's predictable |
14:22.30 | file | no |
14:23.30 | allizom | do you happen to know any simple enough software which could do that? |
14:23.31 | Samot | You'll need something like Kamailio or OpenSISPs |
14:23.43 | Samot | "Simple enough" is also relative. |
14:24.00 | allizom | I don't need bells and whistles, that's it |
14:24.17 | Samot | Well they are like Asterisk. |
14:24.27 | Samot | You have to give them the bells and whistles. |
14:25.11 | Samot | But Asterisk is not designed to manipulate headers like that. |
14:25.41 | Samot | Nor is it designed to let you act off Replies or various request types. |
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16:06.31 | Zombie | I'm just fucked. |
16:08.12 | Samot | Pardon? |
16:15.03 | Zombie | I've fucked over my config again. |
16:20.48 | Samot | How so? |
16:35.17 | drmessano | Samot: What part of that didn't you fucking understand? |
16:35.26 | drmessano | Jeez man |
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16:36.55 | Samot | I'm dense. |
16:36.59 | Samot | Sue 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-Fender | Stop adding extra spaces |
19:15.44 | _pepo_ | sorry... my paste ! |
19:15.49 | _pepo_ | tnx |
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21:06.54 | Aristide | Hello. I have a big problem with Asterisk and sounds. I explain : |
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21:07.35 | Aristide | When 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) |
21:07.56 | Aristide | I use « exten => 4000,1,Goto(addtoblacklist,s,1 » for forward |
21:08.14 | Aristide | (I have forget a « ) » in paste :D) |
21:08.47 | Aristide | And part of configuration file https://pastebin.com/HNkH1Yst :x |
21:26.55 | Aristide | And if I launch asterisk in debug mode, I don't have any errors |
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22:41.21 | retentiveboy | Trying 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? |
22:52.42 | retentiveboy | phoneprov/LINE=2 works... Note to self... Always look in the code. |
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