| 00:26.03 | *** join/#asterisk Sam47 (~sam@197.18.53.146) |
| 01:00.23 | *** join/#asterisk sam_vagabond (~sam@197.18.53.146) |
| 01:14.43 | *** join/#asterisk troyt (zncsrv@2601:681:4100:8981:44dd:acff:fe85:9c8e) |
| 04:12.16 | *** join/#asterisk ircarcs (~quassel@169.9.159.77.rev.sfr.net) |
| 04:38.51 | *** join/#asterisk Janos__ (~Janos@201.204.94.76) |
| 04:47.35 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 05:24.36 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 05:35.41 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:54.15 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:11.04 | *** join/#asterisk Pasha (Cory@unaffiliated/cory) |
| 06:45.20 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 07:02.07 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 07:23.16 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:29.41 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 07:33.51 | *** join/#asterisk jkroon_ (~jkroon@165.16.203.52) |
| 08:00.04 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 08:01.57 | *** join/#asterisk aberrios (545cd2cd@gateway/web/freenode/ip.84.92.210.205) |
| 08:18.06 | *** join/#asterisk Downlots_ (~Downlots@athedsl-4477202.home.otenet.gr) |
| 08:46.21 | *** join/#asterisk sam_vagabond (~sam@197.18.181.49) |
| 09:10.56 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@176.98.158.106) |
| 09:32.40 | *** join/#asterisk sa02irc (~sa02irc@155-079-043-212.ip-addr.inexio.net) |
| 09:46.58 | *** join/#asterisk sam_vagabond (~sam@197.18.181.49) |
| 10:26.33 | *** join/#asterisk sam_vagabond (~sam@197.18.181.49) |
| 10:35.59 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 10:39.26 | *** join/#asterisk sam_vagabond (~sam@197.18.181.49) |
| 10:39.31 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 10:52.39 | aberrios | anyone had success with MS Edge and DTLS negotiation on Asterisk? |
| 11:00.48 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 11:08.34 | *** join/#asterisk emsjessec (~emsjessec@96.56.225.51) |
| 12:22.35 | *** part/#asterisk aberrios (545cd2cd@gateway/web/freenode/ip.84.92.210.205) |
| 12:22.39 | *** join/#asterisk aberrios (545cd2cd@gateway/web/freenode/ip.84.92.210.205) |
| 12:29.02 | *** join/#asterisk ruben23 (~Opendial@49.151.238.192) |
| 12:32.34 | ruben23 | hi there guys i have asterisk server but on the CLI> when i type 'sip show peers' -------> it says that No such command 'sip show peers' --- any idea guys.? |
| 12:33.49 | Samot | Yeah, Chan_SIP isn't loaded |
| 12:43.48 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:45.27 | *** join/#asterisk ghoti (~paul@glphon2233w-grc-14-50-101-208-68.dsl.bell.ca) |
| 13:06.27 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:26.28 | aberrios | I'm attempting to debug what is going on with Edge and Asterisk DTLS negotiation. So far I can see Edge is sending "DTLS 1.2" "Handshake: Client Hello" and Asterisk is responding Port unreachable. I've set dtls_setup for the pjsip peer to actpass (and tried both passive and active anyway) with the same results. |
| 13:27.09 | *** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl) |
| 13:27.50 | file | have you examined the ICE negotiation and looked at the SDP to see the ports that should be used? the only reason port unreachable would occur is if the wrong port was used/Asterisk wasn't listening |
| 13:30.07 | aberrios | asterisk is sending the same port number in the m=audio line as is being used for Edge sending its DTLS Client Hello packet |
| 13:30.52 | aberrios | I think i might try forcing Edge to be DTLS client which, if i understand DTLS correctly, should get Asterisk to send the Client Hello |
| 13:31.23 | file | what is Asterisk running on and where? |
| 13:31.50 | aberrios | 16.3.0 on AWS |
| 13:32.25 | file | have you set up things in AWS to properly forward all the required ports and also configured the ICE host mapping in rtp.conf? |
| 13:32.41 | aberrios | works in FF/Safari and Chrome |
| 13:33.01 | aberrios | it seems Edge *really* wants to be the DTLS server rather than client |
| 13:33.15 | aberrios | and in this scenario DTLS is not working. |
| 13:33.23 | file | alrighty |
| 13:34.04 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-kmpsxyoovknfpojj) |
| 13:34.04 | *** mode/#asterisk [+o bford] by ChanServ |
| 13:35.31 | aberrios | should Asterisk respond with Server Hello when receiving a DTLS Client Hello packet? I've not actaully seen that happen |
| 13:36.07 | file | if I recall correctly, yes |
| 13:51.27 | *** join/#asterisk somepoortech (~somepoort@72.12.70.165) |
| 14:02.45 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-mlvvoeuuniltxthg) |
| 14:02.45 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:30.20 | aberrios | hmm right, fixed the dtls Client Hello/Port Unreachable error.. however now getting "[Reassembly error, protocol DTLS: New fragment overlaps old data (retransmission?)]" |
| 14:30.39 | aberrios | from some earlier googling this seems related to an issue with large certs iirc |
| 14:43.15 | aberrios | looks like setting the MTU for DTLS might fix this. will report back after applying some patches to res_rtp_asterisk.c |
| 14:55.05 | *** join/#asterisk Downlots (~Downlots@2a02:85f:1536:d500:d4aa:315e:5988:c59d) |
| 15:03.44 | *** join/#asterisk cusco (~tralala@neptune.tretas.eu) |
| 15:14.18 | cusco | reliable sip trunk providers.. anyone care to make recomendations? to Switzerland... |
| 15:17.44 | aberrios | inbound/outbound? |
| 15:18.43 | cusco | sorry, outbound for now |
| 15:18.53 | cusco | in the future, possibly some inbound.. |
| 15:22.11 | aberrios | Voxbone are very reliable, and have great support, their outbound service is fairly new though. Worth checking them out. (I do not work for Voxbone, am a customer) |
| 15:22.38 | cusco | lol |
| 15:22.52 | cusco | going to check if pricing is available online |
| 15:37.31 | *** part/#asterisk igcewieling (~ewieling@speedy-02.nyigc.net) |
| 15:45.08 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 16:03.01 | *** join/#asterisk jkroon (~jkroon@mail.ksm.co.za) |
| 16:07.12 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-gbifspxtuyzwejoa) |
| 16:07.12 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:10.22 | rrittgarn | with chan_sip, can you use a # in the dial string? I keep seeing it getting html escaped when we dial with it - is it a special/reserved character? |
| 16:10.58 | rrittgarn | example of what we're trying: Dial(SIP/Peername/101#8003913000) |
| 16:11.11 | rrittgarn | (just using comcast's TN as an example TN) |
| 16:12.21 | [TK]D-Fender | It's not special |
| 16:14.58 | rrittgarn | interesting - we're seeing %23 in the output - wonder if maybe its a wireshark thing |
| 16:16.17 | file | I don't believe it's allowed in a user portion without escaping it |
| 16:17.38 | [TK]D-Fender | There is also "pedatic=" as an option to change the parsing... |
| 16:17.42 | [TK]D-Fender | pedantic |
| 16:18.32 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-xicofgabasjlgjaq) |
| 16:18.37 | Samot | Kind of odd they are using # for that. |
| 16:20.23 | rrittgarn | i don't disagree |
| 16:21.39 | *** join/#asterisk sam_vagabond (~sam@197.18.53.146) |
| 16:22.39 | Samot | So you're only seeing %23 in wireshark, right? |
| 16:22.54 | Samot | If you watch a sip debug on the server, it has # in the URI? |
| 16:28.36 | aberrios | dunno if this channel is archived, and searchable, but I managed to fix DTLS fragmentation with MS Edge and Asterisk |
| 16:28.52 | rrittgarn | pedantic=no fixed it |
| 16:29.03 | aberrios | will post a gist of the patch. |
| 16:30.01 | rrittgarn | next question is if from dialplan we can overwrite/modify the from header with chan_sip in dialplan. I always thought that variable was R/O when accessing from CHANNEL - but figured i should ask here too |
| 16:30.28 | Samot | Well the fromuser and fromdomain settings do that |
| 16:31.05 | Samot | Outside of that, no there's not much you can do to overwrite the full header. |
| 16:31.25 | rrittgarn | but can those be set from dialplan? i know those are in chan_sip |
| 16:31.32 | rrittgarn | err sip.conf |
| 16:31.33 | Samot | No. |
| 16:31.39 | Samot | You set them on the peer. |
| 16:32.02 | Samot | Chan_SIP can let you change them in the dialplan with a modified Dial() string. |
| 16:32.09 | Samot | But that's only Chan_SIP. |
| 16:32.22 | rrittgarn | which is what we're using for this particular nightmare |
| 16:32.23 | rrittgarn | err project |
| 16:32.36 | Samot | Check out the sip.conf.sample |
| 16:32.42 | Samot | Well wait.. |
| 16:32.46 | Samot | What version of Asterisk? |
| 16:33.10 | rrittgarn | 13.26 |
| 16:33.30 | Samot | Uhm, file did that little tidbit get backported in 13? |
| 16:33.38 | file | I have no idea? |
| 16:33.39 | Samot | I know it was introduced in 14. |
| 16:34.08 | Samot | 14 had the ability to change the To and From header in the Dial() for Chan_SIP. |
| 16:34.18 | file | we don't touch chan_sip and I generally forget it |
| 16:34.20 | Samot | Well at least the user/domain portions. |
| 16:34.36 | Samot | It's not really a bad thing... |
| 16:35.09 | Samot | rrittgarn: What are you looking to overwrite/modify? |
| 16:35.31 | rrittgarn | customer is looking to set the from to a value received from a curl call |
| 16:35.43 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample |
| 16:35.45 | rrittgarn | which ends up being an account code, which then the carrier parses |
| 16:36.55 | rrittgarn | of that sample, guessing you're in reference to line 28/29 |
| 16:37.11 | Samot | The use of !! in the dial string. |
| 16:37.24 | Samot | You can try in in 13.26 and see if that works. |
| 16:37.34 | rrittgarn | awesome thanks |
| 16:37.35 | Samot | I'm just not sure it got back ported.\ |
| 16:37.43 | Samot | It was a v14 thing. |
| 16:37.51 | rrittgarn | well i'll try and report back |
| 16:38.10 | Samot | Well it also exists in 16 |
| 16:38.14 | Samot | So there is that. |
| 16:44.57 | *** join/#asterisk LoKoMurdoK (~LoKoMurdo@fedora/LoKoMurdoK) |
| 17:44.08 | *** join/#asterisk orn (~orn@nova-078-040-250-100.corp.nova.is) |
| 17:44.28 | *** join/#asterisk miralin1 (~Thunderbi@host-46-45-217-207.stavropol.ru) |
| 17:44.35 | orn | hey peeps! anyone know of a way to get the channel ID of the last outgoing channel from Asterisk? |
| 17:45.07 | orn | so I have an inbound call, which will be my channel's ${CHANNEL} variable, but what about the channel ids of subsequent Dial commands? |
| 17:46.35 | orn | I should note that this will be happening after said channel has been hung up |
| 17:49.38 | [TK]D-Fender | <orn> hey peeps! anyone know of a way to get the channel ID of the last outgoing channel from Asterisk? <- go store it earlier in the processing |
| 17:50.05 | orn | [TK]D-Fender: sounds like a plan -- is there a way to do that using chan-sip with those pre-channel routines? |
| 17:50.11 | [TK]D-Fender | Shove it in a global variable, in an AstDB value, some other external data format, some local file you will read in.... wherever |
| 17:50.37 | [TK]D-Fender | Pre-dial hook if you're testing for outbound, yes |
| 17:51.05 | orn | ok, thanks a lot for the tip -- i'll see if i can't figure it out |
| 17:56.36 | orn | beautiful -- works like a charm. thanks a lot for always being here :) |
| 17:58.37 | [TK]D-Fender | You're welcome |
| 18:12.36 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 18:35.00 | *** join/#asterisk LoKoMurdoK (~LoKoMurdo@fedora/LoKoMurdoK) |
| 18:37.12 | *** join/#asterisk K0HAX (~michael@gateway/tor-sasl/k0hax) |
| 19:07.54 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 19:20.58 | *** join/#asterisk MIsAn (~yuljk@unaffiliated/yuljk) |
| 22:04.49 | ruben23 | hi there guys how do i create a call bridging to my asterisk coming from another voip platform |
| 22:07.36 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:08.24 | Samot | You need to be a little more detailed. |
| 22:09.52 | ruben23 | Samot: i have an asterisk server, then i have a plivo system that recieved calls, which i can route to my asterisk, the problem how do i route calls to my asterisk just like a local SIP transfer so it will not pass to any VoIp line just plane SIP transfer |
| 22:11.29 | Samot | You need a SIP peer between them just like anything else. |
| 22:13.57 | ruben23 | Samot: both system i need to add sip peers.? |
| 22:14.23 | Samot | ruben23: It's two endpoints. |
| 22:14.40 | Samot | You need something so they know each side. This is no different than Phone -> PBX |
| 22:15.09 | Samot | Or PBX -> Provider |
| 22:15.59 | ruben23 | Samot: can you share how to guide how to set that up somehow |
| 22:16.21 | Samot | No I've never used a Plivo system. |
| 22:16.45 | ruben23 | i mean a guide like how to make Sip peers between two VoIP system |
| 22:17.41 | Samot | What two systems? I just Google Plivo and they look like some sort of provider/vendor. |
| 22:19.39 | ruben23 | i mean asterisk and plivo, yes correct, the setup might be like peering two asterisk server |
| 22:19.52 | Samot | Did you look at their guides? |
| 22:21.00 | *** join/#asterisk chris349 (~office@104-12-70-21.lightspeed.miamfl.sbcglobal.net) |
| 22:21.32 | chris349 | I reboot 1 digium phone and now its trying to contact some digiumcloud.net server |
| 22:21.49 | chris349 | How do I stop this and make the phone contact my local server? |
| 22:22.58 | chris349 | I try to reset phone to factory defaults but it doesnt clear this setting, I believe for Digium Configuraiton Server |
| 22:28.20 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:37.30 | *** join/#asterisk Oatmeal (~Suzeanne@2600:8802:1500:74d:c42e:ccf6:4f39:587) |
| 22:44.31 | *** join/#asterisk NightMonkey_ (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 22:44.45 | Reinhilde | Oatmeal: did you know that you're poisonous to diabetics |
| 23:32.13 | *** join/#asterisk mindthelion (~techquila@121-75-249-73.dyn.vf.net.nz) |
| 23:36.33 | *** join/#asterisk bimbam2103 (577bc518@gateway/web/freenode/ip.87.123.197.24) |
| 23:42.09 | *** join/#asterisk Typhon (~Typhon@dslb-088-068-202-219.088.068.pools.vodafone-ip.de) |