| 00:06.06 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 00:06.33 | *** join/#asterisk skywayskase (~skywayska@104.129.152.117) |
| 00:16.58 | *** join/#asterisk matrix1233 (~matrix123@197.0.84.220) |
| 00:20.56 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:20.56 | *** 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 |
| 00:33.06 | *** join/#asterisk skywayskase (~skywayska@4.16.15.100) |
| 00:51.49 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-30-44.hsd1.nj.comcast.net) |
| 00:53.56 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 01:12.33 | *** join/#asterisk TazzNZ (~pieter@202.89.40.241) |
| 01:12.39 | TazzNZ | good day all |
| 01:14.00 | TazzNZ | I am currently pulling out all of my remaining hair over T38. I have 2 asterisk boxes and I would like to send a fax between them, but I keep on failing T38 and I can't figure out why. On the sending server it keep on failing with : Failed to initialize UDPTL, declining image stream |
| 01:15.03 | TazzNZ | the receiving server says channel 'SIP/fax01-00000003' refused to negotiate T.38 |
| 01:15.29 | TazzNZ | receiving faxes over t38 works :( |
| 01:24.01 | *** join/#asterisk scgm11_ (~scgm11@r186-52-185-4.dialup.adsl.anteldata.net.uy) |
| 01:27.09 | *** join/#asterisk scgm11__ (~scgm11@r186-52-185-4.dialup.adsl.anteldata.net.uy) |
| 01:30.54 | *** join/#asterisk skywayskase (~skywayska@4.16.15.100) |
| 01:32.30 | *** join/#asterisk scgm11_ (~scgm11@190.64.210.154) |
| 01:35.06 | *** join/#asterisk matrix1233 (~matrix123@41.226.244.234) |
| 01:42.16 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 01:42.17 | xochilpili | hi all |
| 01:43.24 | xochilpili | im having some issues, with a register, i have a created a "trunk" and it registered, but when i try to set a register=> for a inbound routes, then i cant connect, but in asterisk there is no error |
| 01:44.34 | TazzNZ | you created a trunk and it registered, but you can't set register for inbound ?!? |
| 01:44.52 | TazzNZ | you should only register to your ITSP |
| 01:45.37 | xochilpili | but i have read that create a trunk, in sip.conf, i can make outgoing calls, but for incomming calls i need to create a register=>, am i wrong? |
| 01:45.55 | xochilpili | TazzNZ, ^ |
| 01:47.28 | TazzNZ | register is only used to tell the ITSP where you are if you don't have a static IP |
| 01:47.51 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 01:48.05 | TazzNZ | that sounded better in my head |
| 01:48.23 | TazzNZ | you need to register to get incoming calls if the ITSP doesn't have a static IP for you |
| 01:50.21 | xochilpili | thank you a lot! |
| 01:50.36 | xochilpili | i have read that ..., of course i was wrong |
| 01:50.42 | xochilpili | thank you TazzNZ :D |
| 01:51.02 | xochilpili | im off |
| 01:51.04 | TazzNZ | np |
| 01:51.09 | xochilpili | :D |
| 02:00.55 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 02:09.22 | *** join/#asterisk scgm11_ (~scgm11@r186-52-185-4.dialup.adsl.anteldata.net.uy) |
| 02:12.42 | *** join/#asterisk matrix1233 (~matrix123@41.226.244.234) |
| 02:20.17 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 02:23.50 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-30-44.hsd1.in.comcast.net) |
| 02:50.12 | *** join/#asterisk matrix1233 (~matrix123@41.226.244.234) |
| 03:24.36 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 03:57.58 | TazzNZ | OMG - seems I fixed it - it turns out that it *appears* as if the peer settings are ignored if t38 is disabled in the general section of sip.conf |
| 04:00.11 | drmessano | Well yeah |
| 04:05.38 | *** join/#asterisk matrix1233 (~matrix123@41.226.244.234) |
| 04:08.14 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 04:09.54 | Samot | Odd. |
| 04:10.05 | Samot | t38 is a general and a per peer setting |
| 04:10.12 | Samot | One doesn't need the other. |
| 04:12.32 | TazzNZ | Samot: that is what I thought too - but it seems like you have to have it on in general and then disable per peer |
| 04:12.37 | TazzNZ | which kinda makes sense |
| 04:13.03 | Samot | Well... |
| 04:13.18 | Samot | The receiving side is what says "Hey use t38" |
| 04:13.34 | TazzNZ | yeah - that worked |
| 04:13.47 | TazzNZ | the issue was the sending side would then go - stuff off, t38 is not enabled |
| 04:16.46 | TazzNZ | anyways - I am off - later all |
| 04:19.41 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 04:49.53 | *** join/#asterisk a|3x (~a|3x@mail.maximum.guru) |
| 04:49.55 | a|3x | hi |
| 05:48.59 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:54.05 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 05:56.35 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 06:06.03 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 06:08.56 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 06:25.27 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 06:34.36 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 06:48.10 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 07:01.13 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:11.19 | *** join/#asterisk bl3nto (~bl3nto@78.134.210.254) |
| 07:17.31 | *** join/#asterisk mmlj4 (~mmlj4@146.sub-174-218-6.myvzw.com) |
| 07:21.23 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:42.16 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 07:46.51 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 07:52.29 | *** join/#asterisk skywayskase (~skywayska@4.16.15.100) |
| 07:59.48 | *** join/#asterisk jkroon (~jkroon@165.16.204.37) |
| 08:21.38 | *** join/#asterisk DanB (~DanB@clt-195.192.205.145.ip-anschluss.net) |
| 08:36.19 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 08:38.52 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-30-44.hsd1.in.comcast.net) |
| 08:53.37 | *** join/#asterisk sotoz (~sotoz@095-097-255-066.static.chello.nl) |
| 08:54.30 | sotoz | Hello, how can I trigger asterisk (with pjsip) to send a 183 after answering a call? I tried using rpid_immediate=yes but it doesn't work :/ |
| 09:03.44 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 09:06.20 | *** join/#asterisk mmlj4 (~mmlj4@146.sub-174-218-6.myvzw.com) |
| 09:08.56 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 09:09.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:13.15 | *** join/#asterisk chocotov (~mike@p5B24F78E.dip0.t-ipconnect.de) |
| 09:14.18 | *** join/#asterisk navaismo (d5a5b553@gateway/web/freenode/ip.213.165.181.83) |
| 09:15.23 | sotoz | I found out. I just do a Progress() before the Answer(). I'm not sure why, but this works. |
| 09:16.51 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 09:20.04 | *** part/#asterisk chocotov (~mike@p5B24F78E.dip0.t-ipconnect.de) |
| 09:20.11 | *** join/#asterisk chocotov (~mike@p5B24F78E.dip0.t-ipconnect.de) |
| 09:24.36 | chocotov | hi, just registered here at freenode specially for #asterisk |
| 09:27.30 | chocotov | sometimes in our asterisk messages log, this error comes up: |
| 09:27.36 | chocotov | chan_sip.c: Rejecting secure audio stream without encryption details: audio 25354 RTP/SAVP 106 9 112 8 0 96 |
| 09:29.38 | chocotov | this happens a couple of times a week. In all cases these calls originate from an operator who will then retry to make the call. |
| 09:30.12 | chocotov | The second try of the operator always succeeds. |
| 09:31.28 | chocotov | we have a continous tcpdump running, so i had a chance to compare the INVITE between the failed call with the error and the succeeded |
| 09:32.55 | chocotov | there are only small difference between them, like the call-id and the crypto key |
| 09:34.35 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 09:34.48 | TandyUK | sure its not using AVP the second time? |
| 09:37.46 | *** join/#asterisk sekil (~sekil@cable-89-216-222-83.dynamic.sbb.rs) |
| 09:38.49 | chocotov | the SDP in both requests is identitical, it has SAVP in both |
| 09:39.42 | chocotov | identical apart from the ports and crypto key that is |
| 09:41.24 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 09:54.00 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 09:55.00 | navaismo | can you show peer configuration and sip debug? |
| 09:55.41 | chocotov | i have the suspicion that in these cases, that the crypto attribute of the SDP didn't make it to Asterisk, even though i can see it the pcap file |
| 09:58.20 | chocotov | hold on, the first case with debug turned on just happend |
| 10:18.55 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 10:42.27 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:52.39 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 10:57.22 | chocotov | i've got a debug log file and a pcap here: http://bama.ddns.net/~mike/Rejecting_secure_audio_stream_without_encryption_details |
| 10:58.03 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 10:59.20 | chocotov | there's two calls in there, the first rejected: chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pKFDVv2SoojBTVOOumoyrQRine30Qrdt5cWPnUoz... UNSUPPORTED OR FAILED. |
| 10:59.45 | chocotov | the second try was succesfull |
| 11:00.15 | chocotov | chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pKFDVv2SoojBTVOOumoyrcRECq1F0LtbFP8p0fWK... OK. |
| 11:34.41 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 11:53.58 | *** join/#asterisk skywayskase (~skywayska@4.16.15.100) |
| 11:54.58 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 12:11.19 | chocotov | just did a diff between the debug output of both calls. The succeeded call has these two lines right before the Processing media level (audio) OK: |
| 12:11.30 | chocotov | [Jun 6 11:38:53] DEBUG[2636][C-00000133] res_srtp.c: Adding new policy for SSRC 1340389802 |
| 12:11.33 | chocotov | [Jun 6 11:38:53] DEBUG[2636][C-00000133] sdp_srtp.c: SRTP policy activated |
| 12:11.56 | chocotov | the failed call doesn't have those lines .. |
| 12:12.03 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 12:12.34 | Samot | Show the whole call, snippets won't help. |
| 12:15.23 | chocotov | http://bama.ddns.net/~mike/Rejecting_secure_audio_stream_without_encryption_details/Rejecting_secure_audio_debug.txt |
| 12:18.19 | Samot | Well..a core debug is not helpful. |
| 12:18.25 | Samot | sip set debug on |
| 12:18.31 | Samot | core set debug 0 |
| 12:18.36 | Samot | core set verbose 10 |
| 12:18.39 | Samot | make the call again |
| 12:20.30 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 12:34.56 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:36.43 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:51.38 | Samot | Alright then. |
| 12:51.55 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:56.53 | chocotov | sorry, i can't reproduce the failed calls. They happen by chance. |
| 12:58.18 | Samot | Are these phone to phone or phone to trunk? |
| 13:00.21 | *** join/#asterisk scgm11_ (~scgm11@r186-50-180-154.dialup.adsl.anteldata.net.uy) |
| 13:02.21 | chocotov | phone to trunk i guess (softphone sends invite to asterisk, asterisk makes the call via SIP provider) |
| 13:07.22 | Samot | And it only works some of the time? |
| 13:07.47 | *** join/#asterisk dacod (~dacod@187.103.104.42) |
| 13:12.33 | chocotov | it works most of the time, i'd say maybe the problem happens once every 70 calls or so |
| 13:12.56 | Samot | Then pay more attention to those calls. |
| 13:13.16 | Samot | Because it is probably related to those calls and/or how they are sent. |
| 13:13.19 | Samot | or handle. |
| 13:13.22 | Samot | or handled. |
| 13:13.34 | chocotov | yeah, i figured that :) |
| 13:21.25 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-judjpvhbjuytqyda) |
| 13:21.25 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:25.30 | chocotov | just took a better look on how often it happens, once every 600 calls is a better estimate |
| 13:27.12 | Samot | So basically almost a non-issue. |
| 13:27.42 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 13:29.27 | chocotov | that's true |
| 13:43.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:54.05 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:55.43 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:56.00 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:58.34 | *** join/#asterisk kharwell (kharwell@nat/digium/x-rgxyxefashghulmr) |
| 13:58.34 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:04.49 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 14:05.15 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 14:05.34 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 14:06.24 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 14:12.28 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
| 14:14.43 | *** join/#asterisk pawiecki (~pawiecki@89.238.53.32) |
| 14:35.44 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-ooeetrulruosjyai) |
| 14:37.20 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:42.41 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 14:43.02 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-mizgcjtxniwjvfrw) |
| 14:43.02 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:52.23 | *** join/#asterisk pawiecki__ (~pawiecki@router.dir.pl) |
| 14:59.35 | *** join/#asterisk TeslaTrav (~TeslaTrav@COX-70-169-207-190-static.coxinet.net) |
| 14:59.49 | *** part/#asterisk TeslaTrav (~TeslaTrav@COX-70-169-207-190-static.coxinet.net) |
| 14:59.59 | *** join/#asterisk TeslaTrav (~TeslaTrav@COX-70-169-207-190-static.coxinet.net) |
| 15:00.09 | TeslaTrav | Anyone had to go from Asterisk to Mitel? |
| 15:00.46 | [TK]D-Fender | ~polls |
| 15:00.47 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:02.08 | TeslaTrav | Not a poll... Not even looking for help... Was looking to commeserate and have conversation. |
| 15:03.58 | *** join/#asterisk pawiecki_ (~pawiecki@gw.mondigroup.com) |
| 15:05.39 | [TK]D-Fender | Best to start with some actually commentary about your own experiences |
| 15:05.46 | [TK]D-Fender | Just so it doesn't come off as fishing. |
| 15:07.26 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 15:08.58 | lorsungcu | im considering going from asterisk to mitel |
| 15:10.05 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-159.dialup.adsl.anteldata.net.uy) |
| 15:11.59 | [TK]D-Fender | I did too for just a second there. Actually less than a second. More like .00027 of a second. So not even half a second. We need more accurate clocks... |
| 15:19.26 | wdoekes | :D |
| 15:20.13 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 15:23.46 | pawiecki_ | [TK]D-Fender: your clock is quite accurate anyway, if you ware able to measure that |
| 15:47.04 | *** join/#asterisk skywayskase (~skywayska@104.129.152.117) |
| 15:50.12 | *** join/#asterisk skywayskase (~skywayska@104.129.152.117) |
| 15:57.44 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 16:08.26 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 16:21.54 | *** join/#asterisk scgm11__ (~scgm11@190.64.210.154) |
| 16:23.34 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 16:29.05 | *** join/#asterisk imcdona (~imcdona@2607:f0d8:20:1001:94fe:2a00:16b0:7ff2) |
| 16:36.21 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 16:39.50 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-159.dialup.adsl.anteldata.net.uy) |
| 16:40.40 | *** join/#asterisk miralin (~Thunderbi@91.237.94.3) |
| 16:50.58 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.52) |
| 16:53.18 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:57.35 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:57.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:57.57 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 16:58.25 | *** join/#asterisk miralin1 (~Thunderbi@91.237.94.8) |
| 17:17.10 | *** join/#asterisk Samael28 (~Samael28@51.15.57.125) |
| 17:24.00 | *** join/#asterisk scgm11_ (~scgm11@r186-52-239-223.dialup.adsl.anteldata.net.uy) |
| 17:37.52 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
| 17:41.49 | dan_j | Hi. Does Asterisk PJSIP support instant messaging when used with Counterpath Bria? |
| 17:43.55 | Samot | Wrong question. |
| 17:44.00 | Samot | Flip it. |
| 17:44.07 | Samot | SIP SIMPLE and XMPP support <-- That's what Bria supports. |
| 17:45.19 | dan_j | Ah. Ok. So flipped, does Asterisk also support SIP SIMPLE and XMPP or do I need something else for that? |
| 17:45.37 | Samot | Asterisk supports SIP SIMPLe. |
| 17:45.39 | Samot | Asterisk supports SIP SIMPLE. |
| 17:46.01 | Samot | XMPP requires an xmpp server like Prosody or Let's Chat |
| 17:46.17 | dan_j | Ok. Will have a play around with SIP SIMPLE and see how I get on. |
| 17:47.29 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_message_context |
| 17:50.32 | dan_j | What happens in the event that a message is sent to an endpoint that's offline? |
| 17:50.54 | [TK]D-Fender | * doesn't do followups |
| 17:50.58 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 17:52.48 | dan_j | Ok, thanks Samot and [TK]D-Fender |
| 17:54.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 17:54.58 | Samot | You would need a method to store offline messages and send them to the user when they came online |
| 17:55.54 | dan_j | Ok. I'll sort with online and then look at dealing with the offline problem. Although XMPP server may be a simpler option if it does store and forward out of the box. |
| 17:56.05 | scgm11_ | dan_j using dialplan SendMessage you have the status of the message |
| 17:56.12 | scgm11_ | FAILED or SUCCESS |
| 17:56.41 | scgm11_ | Im using sip simple for all messages peer to peer but with chan_sip |
| 17:57.11 | dan_j | Oh, I didn't realise I need to write a dialplan. I thought that endpoint_message_context was just the endpoint's context. |
| 17:57.31 | dan_j | Now that makes sense if I have to put it through a dialplan |
| 17:57.48 | scgm11_ | using dialplan allows you to store messages and manage permissions or anything you like |
| 17:58.15 | dan_j | Once stored, how do you get the messages sent once the endpoint comes online? |
| 17:58.27 | [TK]D-Fender | You send it... |
| 17:58.28 | [TK]D-Fender | again |
| 17:58.30 | [TK]D-Fender | via the dialplan |
| 17:58.38 | [TK]D-Fender | just like the samples show you for your first attempt |
| 17:58.49 | [TK]D-Fender | You are the one who has to queue them up and retry |
| 17:59.15 | dan_j | I've not seen any samples. Where are they? |
| 17:59.58 | scgm11_ | sorry if I highjack your response [TK]D-Fender |
| 18:00.42 | [TK]D-Fender | dan_j, look for "sendmessage" |
| 18:00.48 | [TK]D-Fender | Wiki has this all |
| 18:03.00 | dan_j | I dont see much for 'sendmessage' in the wiki. Maybe 'senttext', but no sample dialplan |
| 18:04.08 | [TK]D-Fender | or "function message" |
| 18:04.41 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 18:07.25 | dan_j | Ok. Got a little more but no dialplan. My question is, since a dialplan needs to be executed (either by a SIP INVITE or SIP SIMPLE inbound message), how can I execute the dialplan when an endpoint registers but does nothing else? |
| 18:07.49 | scgm11_ | <PROTECTED> |
| 18:08.38 | scgm11_ | example: |
| 18:08.42 | scgm11_ | for chan_sip |
| 18:08.45 | scgm11_ | you have |
| 18:08.45 | scgm11_ | accept_outofcall_message=yes |
| 18:08.46 | scgm11_ | outofcall_message_context=messages |
| 18:08.52 | scgm11_ | so in context message |
| 18:09.03 | scgm11_ | you can use _XXXX |
| 18:09.20 | scgm11_ | and get the destination from EXTEN |
| 18:09.43 | Samot | This is all in the wiki |
| 18:09.46 | [TK]D-Fender | dan_j> Ok. Got a little more but no dialplan. My question is, since a dialplan needs to be executed (either by a SIP INVITE or SIP SIMPLE inbound message), how can I execute the dialplan when an endpoint registers but does nothing else? <- go find something that announces the registration and originate a local channel |
| 18:09.53 | [TK]D-Fender | as I said this is all dialplan and up to you |
| 18:10.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:10.31 | [TK]D-Fender | Go look for an event reporting registration or perform some other form of status check showing a change in state |
| 18:12.11 | dan_j | Would this require the use of an AMI client to monitor the state of an endpoint and originate if it comes online? Or is there a simpler wait to do it? |
| 18:12.41 | dan_j | Samot: I couldnt find it in the wiki |
| 18:14.36 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
| 18:15.18 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_MESSAGE |
| 18:15.35 | [TK]D-Fender | tehre are other pages with more complete samples |
| 18:15.37 | [TK]D-Fender | go look |
| 18:15.40 | [TK]D-Fender | drill |
| 18:28.41 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 18:28.45 | *** join/#asterisk netman (~netman@185.94.249.77) |
| 18:32.44 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 18:32.46 | xochilpili | hi all, |
| 18:37.25 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 18:48.30 | *** join/#asterisk phix (~threat@203.63.124.73) |
| 18:52.05 | *** join/#asterisk skywayskase (~skywayska@104.129.152.117) |
| 18:54.50 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 19:04.10 | xochilpili | yesterday i was able to send and receive calls from my ITSP, now i just got this: Received response: "Forbidden" from '"operadora" <sip:2286906800@189.194.93.46>;tag=as48c6db3b' "operadora" is the name of my local extension, and 189.194.93.46 is the ipaddr that my ITSP gave me |
| 19:04.34 | xochilpili | i tried to debug what's going on, but i cant figure it out |
| 19:06.13 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 19:06.14 | xochilpili | there's a authentication user, a user id and a password, with a 5301 port but in the trunk, i have set all this values, i can register, in sip show peers there's out-route/head_telephone serverip OK |
| 19:06.50 | xochilpili | but when i try to make a call, i just got Forbidden, does any one have a clue or something that helps? |
| 19:07.19 | drmessano | ?ssh |
| 19:07.29 | drmessano | :( |
| 19:07.44 | drmessano | SIP debug of a call would help |
| 19:08.31 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 19:11.26 | xochilpili | i have test with a zoiper, and in it's debug, it make the call using phone@proxyip < but in asterisk puts 2286906800@189.194.93.46 << which is my localip |
| 19:11.27 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 19:11.33 | xochilpili | where can i change this? |
| 19:13.04 | xochilpili | drmessano, im doing test with zoiper and his debug, i only have this connection wired, i will paste the debug of a sip call, just a sec |
| 19:13.05 | xochilpili | pls |
| 19:14.01 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 19:18.41 | *** join/#asterisk jkroon (~jkroon@165.16.204.168) |
| 19:20.53 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
| 19:29.37 | drmessano | xochilpili: ??? |
| 19:31.01 | xochilpili | drmessano, i solve it, i just restart network and magically works, i dont know what's going on |
| 19:31.40 | drmessano | Wouldnt have hurt to come back and say that |
| 19:36.40 | *** join/#asterisk n3ob_ (~n3ob@pool-108-36-197-170.phlapa.fios.verizon.net) |
| 19:50.01 | *** join/#asterisk klow (sid213056@gateway/web/irccloud.com/x-ppdsyaiuvmnlfexc) |
| 19:50.08 | klow | Hello, I have a phone that sends only IPv6 contact info in SDP because it in on T-Mobile network. T-Mobile uses NAT64 , I already have nat=yes set in that chan_sip extension but the RTP still is not working. Is there a better way to force Asterisk to use whatever the *apparent* IPv4 address is rather than getting confused at his IPv6 address in the SDP? I'm not operating on IPv6 at all on my side yet |
| 19:51.55 | klow | In other words I want asterisk/chan_sip to say "what IP is this guy connecting from" not what he says in the SDP .. i thought that was what nat=yes did though .. |
| 20:02.49 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 20:15.00 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-pjqmuaivwmyjwnog) |
| 20:39.29 | salviadud | klow, pastebin your sip.conf please |
| 20:40.18 | salviadud | to be precise, just the [general] section |
| 20:40.22 | salviadud | no passwords |
| 20:40.24 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 20:50.47 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 20:59.45 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 21:04.38 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:12.38 | klow | https://pastebin.com/u0hgYA0w |
| 21:22.54 | salviadud | klow, that phone you mention that is behind nat64, you have that registered as friend, user or peer? |
| 21:25.59 | klow | it is type=friend nat=force_rport,comedia |
| 21:26.24 | *** join/#asterisk jab416171 (~jab416171@c-24-10-151-191.hsd1.ut.comcast.net) |
| 21:28.57 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 21:29.35 | salviadud | what does sip set debug on say? |
| 21:29.47 | salviadud | Does it try to register? |
| 21:36.48 | klow | it registers fine, it just fails to get audio on a call because asterisk doesnt know how to connect o the IPv6 address sent by the phone in the SP |
| 21:36.51 | klow | SDP |
| 21:37.10 | klow | the ISP does a fine job of NATing the IPv6 phone to the IPv4 asterisk server |
| 21:37.28 | klow | but asterisk doesnt know how to deal with the SDP because it contains only IPv6 contact info |
| 21:37.44 | klow | If I used the term nat64 wrong i apologize |
| 21:38.00 | klow | i just mean somehow the IPv6 phone is connecting fine to the IPv4 asterisk through some NAT mechanism of the phone ISP |
| 21:42.33 | salviadud | Interesting network issue |
| 21:44.34 | salviadud | And, is there a way to make asterisk get an ipv6 address? |
| 21:48.47 | [TK]D-Fender | "get"? |
| 21:49.06 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 21:49.07 | Samot | Yes. |
| 21:49.12 | Samot | You order a block from your ISP. |
| 21:49.31 | Samot | Then you assign an IP to the Asterisk server.. |
| 21:49.50 | Samot | Then follow the Chan_SIP or Chan_PJSIP configs for listening on ipv6. |
| 21:52.08 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 21:52.48 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 22:03.36 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
| 22:03.59 | klow | Can chan_sip listen on both IPv4 and IPv6 ? |
| 22:15.23 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 22:54.16 | *** join/#asterisk mmlj4 (~mmlj4@182.sub-174-218-143.myvzw.com) |
| 22:57.26 | [TK]D-Fender | yes |
| 23:07.10 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 23:22.13 | *** join/#asterisk tuxd00d (~tuxd00d@wsip-184-179-14-87.ph.ph.cox.net) |
| 23:24.10 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 23:30.35 | *** join/#asterisk matrix1233 (~matrix123@41.226.247.24) |
| 23:30.43 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 23:32.30 | *** part/#asterisk kharwell (kharwell@nat/digium/x-rgxyxefashghulmr) |
| 23:56.01 | drmessano | klow: Have you tried setting your various bindaddr to 0.0.0.0? |
| 23:56.07 | *** join/#asterisk vstemen (~vince@ip68-227-102-113.ok.ok.cox.net) |
| 23:56.24 | drmessano | I dont recall if the default is [::] |
| 23:56.35 | drmessano | But 0.0.0.0 should effectively disable IPv6 |
| 23:56.40 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |