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| 03:48.54 | Get_The_Fish | Nice, the git repo 13.7 branch shows the version as UNKOWN__and_probably_unsupported |
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| 05:10.52 | purf | Hello all |
| 05:15.15 | purf | Looking for a good SMS gateway for +011/US/Canada thought I might ask on here... 100k/day <50 CPS |
| 05:18.56 | purf | figured that the good folks at Digium might still have a presence here |
| 05:25.35 | [TK]D-Fender | * is not an SMS platform really |
| 05:25.57 | [TK]D-Fender | There are few limited techs that sorta work depending |
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| 05:32.29 | djiboutiii | What does it generally mean when you turn on RTP debug and see no packets sent or received? |
| 05:33.23 | djiboutiii | There's a lot on google for one-way audio issues. Not much on two-way. The NAT config seems fine to me. I'm lost. |
| 05:33.51 | purf | [TK]D-Fender: can you drop a name of one of those limited techs? |
| 05:34.07 | WIMPy | No audio is just two one-way audio issues at the same tieme :-) |
| 05:34.16 | djiboutiii | lol |
| 05:34.31 | djiboutiii | I suppose that's true |
| 05:34.36 | WIMPy | Seriousely. |
| 05:35.04 | WIMPy | It means you have twice the chance to get to one-way-audio at least. |
| 05:36.43 | WIMPy | After that you have to fix the right side, otherwise you're back where you are now. |
| 05:37.08 | djiboutiii | I'll keep that in mind. So this is a brand new vanilla install |
| 05:37.18 | [TK]D-Fender | some devices through chan_dongle, Some SIP carriers use SIP message to carry them. |
| 05:37.19 | djiboutiii | I'm able to forward a call to my cell phone |
| 05:37.29 | djiboutiii | but asterisk shows no RTP packets at all, leaving or receiving |
| 05:37.31 | [TK]D-Fender | Some E1 using a dAHDI insterface |
| 05:38.21 | WIMPy | For bulk SMS you'd most probably use http or https. |
| 05:41.33 | purf | Hrm. Thanks, I guess. Was hoping that someone on #asterisk knew what they were talking about. |
| 05:42.05 | [TK]D-Fender | We do |
| 05:42.10 | [TK]D-Fender | * is not an SMS platform |
| 05:42.25 | [TK]D-Fender | It can get it over various techs depending on your provider, etc. |
| 05:42.35 | [TK]D-Fender | but * does nthing special to helkp you with this |
| 05:43.16 | WIMPy | definitely not a sensible tool for the job. |
| 05:43.19 | [TK]D-Fender | Qty per day isn't really a factor that is relevant in any way I can see yet |
| 05:46.26 | purf | Thanks for your time. |
| 05:47.12 | WIMPy | thinks we gave the ``wrong'' answers... |
| 05:47.51 | purf | nono, I agree. Asterisk is not an SMS platform. |
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| 05:52.41 | purf | but "*" can establish a SIP session and afterward negotiate (and re-negotiate) various streams I'm working on integrating SMS into the mix, was wondering if there was a provider that could cut down my workload. Nevermind |
| 05:53.27 | purf | apologizes if he gave the wrong idea |
| 05:53.57 | WIMPy | Probably. But you didn't tell much about what you need. |
| 05:54.06 | [TK]D-Fender | sip-wise it's not part of any other stream, it['s just a packet all it's own. |
| 05:56.29 | purf | SIP enables and announces the stream |
| 05:56.51 | WIMPy | # |
| 05:56.55 | WIMPy | oops |
| 05:57.00 | WIMPy | What stream? |
| 05:58.01 | purf | SMS |
| 05:58.38 | WIMPy | That's not a stream. Just a single message. |
| 05:58.51 | purf | i.e: media |
| 05:59.41 | WIMPy | Thats a rather streched definition of "media". |
| 06:00.07 | purf | Single message=stream=announced packets over SIP |
| 06:00.48 | WIMPy | That's somehow the opposite of a stream. |
| 06:01.01 | purf | Whyso? |
| 06:01.10 | [TK]D-Fender | 1 drop of water != flow |
| 06:01.23 | [TK]D-Fender | hello = goodbye. |
| 06:01.27 | [TK]D-Fender | beginning = end |
| 06:01.30 | purf | 200 OK |
| 06:01.40 | [TK]D-Fender | #hiheresamessagetheend |
| 06:01.48 | [TK]D-Fender | nothing more comes |
| 06:01.53 | [TK]D-Fender | not a stream. |
| 06:01.57 | [TK]D-Fender | just because you ack it |
| 06:02.14 | [TK]D-Fender | And you COULD ... simply not ACK it. But that would be rude |
| 06:03.20 | purf | Agreed |
| 06:04.29 | [TK]D-Fender | This is an all new, and even more poorly scaled version of playing "But a switch IS a router!" |
| 06:04.41 | [TK]D-Fender | #funtimes |
| 06:07.23 | purf | [TK]D-Fender: Yeah, you may be right |
| 06:08.13 | purf | But I'm forcing SMS into the announce payload |
| 06:08.20 | purf | ...or trying to |
| 06:08.23 | purf | anyway |
| 06:14.14 | purf | It works, awesome |
| 06:14.40 | purf | Now I just need to find a suitable carrier |
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| 06:15.22 | purf | [TK]D-Fender, WIMPy, where you from? |
| 06:16.14 | purf | I'm not asking A/S/L, just curious as to your country of origin |
| 06:16.25 | purf | feel free to ignore |
| 06:18.57 | WIMPy | de |
| 06:19.21 | purf | ? |
| 06:19.35 | [TK]D-Fender | ca |
| 06:20.12 | WIMPy | points at ISO 3166 |
| 06:21.08 | purf | Heh, awesome |
| 06:21.16 | purf | ca here |
| 06:22.00 | purf | Check out https://tools.ietf.org/html/rfc4240#section-3 |
| 06:22.48 | purf | I'll post my code once polished... Thanks for talking |
| 06:22.58 | purf | 'night |
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| 10:34.28 | rexwin_ | what does Suppliers in Switch mean? |
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| 12:25.03 | DanQuinney | Can anyone suggest any software to load test an Asterisk box? Specificly multiple sip calls |
| 12:25.38 | juned | You can use Sipp may be |
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| 12:35.45 | DanQuinney | thanks juned |
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| 13:31.29 | rl1 | why does it show "T.38 support : No" on the peer but in the global config it shows "yes"? |
| 13:31.49 | rl1 | what am i doing wrong here |
| 13:40.28 | rl1 | yeah found the source of the problem |
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| 13:46.30 | mcargile | Does chrome have problems connecting to Asterisk 11's wss Websockets? |
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| 15:22.05 | gavimobile | my outgoing calls are being shown as blocked. after calling my tisp, they say they see the peer/user name. what do i need to change to have the trunk clid to show, or to allow me to use this Set(CALLERID(num)=${OUTCID1}) |
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| 15:38.20 | [TK]D-Fender | Go use that. |
| 15:38.30 | [TK]D-Fender | And we can't tell what you've done wrong until you show us. |
| 15:38.34 | [TK]D-Fender | Always show us |
| 15:38.42 | [TK]D-Fender | PASTEBIN is your friend |
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| 15:39.45 | purf | Hey all |
| 15:50.24 | gavimobile | [TK]D-Fender: here is my output without pastebin |
| 15:50.25 | gavimobile | http://pastebin.com/T2pbi0rM |
| 15:50.33 | gavimobile | without sip debug* |
| 15:50.44 | [TK]D-Fender | SIP debug IS what we need to see |
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| 15:50.57 | [TK]D-Fender | How are we supposed to prove what's being sent/rec when we can't SEE it? |
| 15:51.46 | gavimobile | [TK]D-Fender: no problem |
| 15:54.06 | gavimobile | [TK]D-Fender: http://pastebin.com/iBQ8L5Ec there ya go |
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| 15:56.34 | [TK]D-Fender | Use a proper SSH client |
| 15:56.45 | [TK]D-Fender | Your piped output is littered with ANSI garbage |
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| 16:06.45 | purf | [TK]D-Fender: What's a proper SSH client in your opinion? Just wondering if putty is still the norm? |
| 16:07.12 | [TK]D-Fender | PuTTY is a very solid choice. I don't know many others personally. |
| 16:10.51 | purf | gavimobile: Try using PuTTY http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html |
| 16:25.53 | newtonr | purf, on windows? I use https://www.bitvise.com/ssh-client which is free for individual use |
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| 16:54.46 | mub | soooo the new vicibox 7 installer is broken as fuck. In order to get it to run you must first set your network information first (the script fails to do this), correct the permissions on /usr/src/astguiclient/conf/vicibox-tel.sh |
| 16:55.01 | mub | And then after the install you have to chroot into the root and set the root password |
| 16:55.19 | mub | Just a FYI to anyone out there |
| 16:55.52 | purf | WTF is vicibox? |
| 16:59.08 | mub | purf: vicidial |
| 17:00.52 | purf | ah, gotcha |
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| 17:18.17 | cusco | hi folks |
| 17:18.49 | cusco | We've rebooted a asterisk server, and a PRI card seems its not working.. not sure what debug I produce |
| 17:19.22 | cusco | we have two cards, if I connect telco to the first card it works |
| 17:19.33 | cusco | if I connect it to the second it doesn't.. |
| 17:19.40 | cusco | dahdi_scan shows red alarm |
| 17:21.05 | cusco | here is output of dahdi_scan http://paste.debian.net/378373/ |
| 17:21.15 | cusco | spans 1,2 and 5 should be up |
| 17:21.29 | cusco | but 5 is not working |
| 17:21.58 | newtonr | cusco, I'd call Digium tech support and have them take a look |
| 17:22.35 | cusco | :/ |
| 17:22.37 | cusco | card is old |
| 17:22.52 | cusco | also here is system.conf from dadhdi: http://paste.debian.net/hidden/e56e046e/ |
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| 17:23.50 | cusco | I was hopping to get some way of learning more about my problem |
| 17:23.59 | cusco | isn't there a way I can debug? |
| 17:26.57 | newtonr | Ah yeah you have an older one. I haven't worked with the cards in a while, but IIRC "pri set debug on span <N>" |
| 17:27.06 | cusco | yes I can do that |
| 17:27.09 | cusco | did actually |
| 17:27.17 | cusco | only shows SENDING SAMBME or something |
| 17:27.20 | cusco | let me check |
| 17:28.03 | cusco | http://paste.debian.net/hidden/71ea4521/ |
| 17:28.06 | cresl1n | if you only see sending sabme, that means the other end either isn't responding or we're not seeing a response from them |
| 17:28.35 | cusco | but I connected the other end to another pri card, and it works |
| 17:29.14 | cusco | cresl1n: also before you joined: here is dahdi_scan http://paste.debian.net/378373/ ; and here is system.conf from dahdi: http://paste.debian.net/hidden/e56e046e/ |
| 17:29.20 | cusco | spans 1,2 and 5 should be up |
| 17:29.26 | cresl1n | are they green? |
| 17:29.29 | cusco | red |
| 17:29.34 | cresl1n | well. |
| 17:29.39 | cresl1n | fix red first :-) |
| 17:29.49 | cresl1n | Layer 1 problems are always where you should start |
| 17:29.56 | cusco | but whatever telco I connect to the second card (spans 5-8) |
| 17:29.58 | cusco | its red |
| 17:30.24 | cusco | its actually: RED/FLA |
| 17:30.29 | cusco | its actually: RED/LFA |
| 17:30.34 | cresl1n | yeah |
| 17:30.38 | cresl1n | L1 problems |
| 17:30.44 | cresl1n | are your framing and signaling right for the line? |
| 17:30.49 | cresl1n | usually timing won't do that |
| 17:30.59 | cusco | so. I rebooted asterisk server |
| 17:31.05 | cresl1n | also, is your cable legitimate? |
| 17:31.08 | cusco | and then PRI line on this second card stopped working |
| 17:31.21 | cusco | yes it was all working before, and I changed the cvable to the other pri card and it works |
| 17:31.29 | cresl1n | cool |
| 17:31.32 | cresl1n | that's a good start |
| 17:31.43 | cusco | but as we have 2 different telcos I would like to keep each telco on its own card |
| 17:31.47 | cresl1n | yeah |
| 17:31.51 | cusco | as we had problems with a single card for 2 telcos in the past |
| 17:31.57 | cusco | timing sources and what not |
| 17:32.31 | cusco | so cable and telco equipment is good as it started working once I connected it to the first PRI card |
| 17:32.41 | cusco | I'm wondering whats wrong with this 2nd PRI card |
| 17:32.41 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 17:33.46 | cusco | so.. any way to check card is working properly? I have a loop, I can plug it in.. |
| 17:33.53 | newtonr | when you restarted the machine was it physically moved around any? might be worth reseating the card - though I doubt that is it. |
| 17:34.21 | *** join/#asterisk jhester (jhester@nat/digium/x-ywtwzrrxcvokfeeb) |
| 17:34.26 | cusco | not fisically moved arround. by reseating you mean take it off clean contacts and plug it back in? |
| 17:35.10 | cusco | we just had a energy cut to fix the generator... so we powered everything down gracefully |
| 17:35.41 | cusco | ok I'm going to plug the loop just to check if it is detected as such |
| 17:36.53 | newtonr | cusco, http://kb.digium.com/articles/Configuration/How-do-I-run-a-loopback-test-on-a-Digium-E1-T1-card |
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| 17:37.10 | newtonr | there is the loopback test guide |
| 17:39.55 | cusco | er.. I just came to the server room, only have cli.. |
| 17:40.02 | cusco | so I connected the loop and still nothing |
| 17:40.06 | cusco | I should have some messages |
| 17:40.20 | cusco | asterisk stating that both ends trying to be master |
| 17:40.23 | cusco | or such |
| 17:40.28 | cusco | weird |
| 17:47.08 | newtonr | did you follow the guide? the configuration has to be specific for the loopback test |
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| 18:35.50 | cusco | ok regarding the /dev/dahdi/<num> |
| 18:35.59 | cusco | we don't have that |
| 18:42.06 | *** join/#asterisk onebree (d1bf0df7@gateway/web/cgi-irc/kiwiirc.com/ip.209.191.13.247) |
| 18:42.11 | onebree | hello, all |
| 18:42.54 | onebree | I ran an rtp and pjsip debug on an extension using webrtc. I was asked to find what cipher is being used, although IDK where to look in the log. |
| 18:44.21 | onebree | Here is the log: https://gist.github.com/onebree/4ce44fb9f8032c5c5869 |
| 18:44.59 | onebree | All I can tell is that there is a fingerprint in sha-256, and sipml5 uses the md5 algorithm (which I also found in the sipml5 code) |
| 18:46.29 | *** join/#asterisk catphish (~J@unaffiliated/catphish) |
| 18:48.20 | catphish | i got asked an odd question today: if someone dials a number on a sip handset, is it possible for the device that receives the SIP INVITE to issue a reinvite back to the UA with a new "To" header containing the name of the person they just dialed? would any phone display this? |
| 18:49.14 | [TK]D-Fender | CONNECTEDLINE <- |
| 18:49.17 | [TK]D-Fender | * can already do this |
| 18:49.23 | catphish | (so that my phone tells me the name of the person i just dialed) |
| 18:49.39 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONNECTEDLINE |
| 18:50.00 | catphish | oh that's very cool |
| 18:50.18 | [TK]D-Fender | User dials #, call hits dialplan, you do lookup, you push the name back to the phoe and carry on to do something with it |
| 18:50.52 | catphish | that's exactly what i'm after, do you know if the REST API can manipulate it? |
| 18:51.17 | [TK]D-Fender | it's a function.... |
| 18:51.24 | [TK]D-Fender | You can call it whenever you'd call any function |
| 18:51.30 | [TK]D-Fender | I don't know REST specifically |
| 18:52.38 | catphish | thanks, i'll look into it |
| 18:53.29 | onebree | [TK]D-Fender: any advice for my question? |
| 18:54.27 | [TK]D-Fender | [Feb 2 04:16:03] WARNING[18686]: res_config_mysql.c:1246 require_mysql: Possibly unsupported column type 'text' on column 'path' |
| 18:54.29 | [TK]D-Fender | [Feb 2 04:16:03] ERROR[18686]: res_pjsip_registrar.c:504 rx_task: Unable to bind contact 'sips:286@127.0.0.1:33994;transport=WSS;rtcweb-breaker=yes' to AOR '286' |
| 18:54.32 | [TK]D-Fender | That looks like a DB error |
| 18:54.45 | [TK]D-Fender | Straight up fail there |
| 18:55.04 | [TK]D-Fender | And a pile of clear dialplan errors at the top |
| 18:55.15 | catphish | "Possibly unsupported column type" is generally not fatal, as long as the column type is sensible (like text) |
| 18:55.23 | [TK]D-Fender | For which you didn't have basic verbose turned up (which there is never a reason not to be on verbose 5 for) |
| 18:55.48 | [TK]D-Fender | With that 2nd message following it I'd bet it IS kind fatal to this situation |
| 18:55.59 | catphish | "Unable to bind contact" looks fatal though, i dont know why |
| 18:56.07 | onebree | For the second error you listed, I do not have the contact listed in the database, rather it is being set by sipml5 when I connect |
| 18:56.39 | [TK]D-Fender | and FAILING to set it |
| 18:57.21 | onebree | My question was, rather, how do I find the cipher/crypto/etc in the log? |
| 19:13.50 | purf | Hey all |
| 19:14.08 | purf | onebree: did you reach a solution? |
| 19:14.54 | onebree | A solution for what, exactly? |
| 19:20.26 | *** join/#asterisk theron (~theron@2620:10d:c091:200::b:149c) |
| 19:21.05 | Kunsi | is there any solution to bridge an asterisk MeetMe to a mumble server (e. g. make voice from one side available to other side) |
| 19:21.08 | Kunsi | ? |
| 19:21.35 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 19:21.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 19:21.45 | purf | onebree: "I was asked to find what cipher is being used, although IDK where to look in the log." |
| 19:22.47 | onebree | purf: I am using firefox, and there is no mention of "cipher" or "crytpo" in the log file. I am performing another log right now to double check. I was advised to see if google chrome provided that header/somethign though |
| 19:25.08 | [TK]D-Fender | Kunsi: What protocols does it speak? |
| 19:25.50 | Kunsi | mumble speaks ⦠mumble ð |
| 19:26.02 | Kunsi | google finds https://github.com/slomkowski/mumsi |
| 19:26.07 | Kunsi | i'll give it a try |
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| 19:39.04 | *** join/#asterisk waj (~waj@ec2-54-164-240-122.compute-1.amazonaws.com) |
| 19:44.02 | waj | Seems like in 13.6, pjsip stops re-registering an endpoint if another (unrelated) endpoint is misconfigured, for example with a bad "match" value. |
| 19:44.32 | waj | I tested with the same config in 13.1 and it works fine (even after complaining about the wrong settings for the second endpoint) |
| 19:45.00 | waj | Is this a known issue? I could not find an open issue about this, but I can reproduce it very consistently. |
| 19:45.51 | onebree | waj: In my opinion, I would open an issue and explain the steps to reproduce. Worse thing that happens is they close your issue as a duplicate. |
| 19:46.12 | rmudgett | I expect it is a duplicate. |
| 19:46.28 | [TK]D-Fender | I would UPGRADE... then try to reproducer |
| 19:47.54 | waj | Ok, I will report. My first reaction was that it must be me, because I'm just learning pjsip after too many years of chan_sip :) |
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| 20:27.04 | *** join/#asterisk ZWay (4acb69c2@gateway/web/freenode/ip.74.203.105.194) |
| 20:28.01 | ZWay | How do you get the connecting channel's IP address using ARI? I've tried retrieving CHANNEL(recvip) and several other variables from the CHANNEL function, but all of them return "message": "Unable to read provided function" |
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| 20:45.15 | newtonr | ZWay, that is odd - what kind of channel? |
| 20:55.23 | ZWay | stasis channel |
| 20:55.31 | ZWay | from a softphone |
| 20:57.56 | catphish | does ARI allow you to call arbitrary functions on a channel then? |
| 20:58.25 | catphish | it looks like you can if the function acts like a variable? |
| 20:59.37 | mjordan | yes |
| 20:59.48 | ZWay | yeah, you can using GET /channels/{channelId}/variable |
| 21:00.29 | catphish | that's handy, i've been stupidly doing Stasis("myapp", function(), function(), function()) to get values into ARI |
| 21:00.41 | catphish | thats probably quicker, but very messy |
| 21:01.13 | catphish | and i couldn't find a way to set them, just read the docs and looks like its quite easy, thanks |
| 21:01.53 | ZWay | sure, now if I could only get the variable that I need, lol |
| 21:02.06 | ZWay | I can't seem to find a way to get the IP of the connected channel |
| 21:02.59 | catphish | you want the source IP that initiated the channel? |
| 21:03.09 | ZWay | yes |
| 21:03.29 | catphish | would it be in the from header? |
| 21:03.36 | ZWay | should be |
| 21:03.37 | catphish | not very nat-reliable, but maybe useful |
| 21:04.01 | catphish | so PJSIP_HEADER(read,from) |
| 21:04.07 | catphish | or SIP_HEADER(something) |
| 21:04.26 | ZWay | that gives me the server ip...hrm |
| 21:04.43 | ZWay | gives me the dial-string and some tag value |
| 21:04.54 | catphish | ah, this is kind of up the the client to set that when it makes the call |
| 21:05.09 | catphish | not sure if theres a variable with the actual endpoint IP or not |
| 21:05.26 | ZWay | documentation shows I should be able to call CHANNEL(uri) but it's not working |
| 21:17.04 | mjordan | ZWay: chan_sip or pjsip? |
| 21:17.12 | ZWay | pjsip |
| 21:17.24 | mjordan | ZWay: it should be CHANNEL(pjsip,remote_addr) |
| 21:17.41 | ZWay | awesome...that works perfect |
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| 21:26.39 | mub | e /var/lib/asterisk/sounds/en/{you-sound-cute.wav, yes-dear.wav, yes-dear2.wav} |
| 21:26.44 | mub | oops |
| 21:26.52 | catphish | lol |
| 21:27.06 | onebree | lmao, that vmade my day |
| 21:27.10 | mub | ... sometimes I like to listen to the extra asterisk sounds |
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| 21:47.32 | newtonr | ZWay, btw there is an #asterisk-ari channel for ARI discussion and questions |
| 21:47.51 | ZWay | awesome, thanks newtonr |
| 21:48.01 | newtonr | np |
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| 23:25.05 | badbit | Anyone in the UK having probelems? |
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| 23:26.10 | badbit | problems* |
| 23:26.19 | newtonr | badbit, That is pretty vague :D Probably want to be more specific |
| 23:26.46 | badbit | Sorry, anyone having problems with the SS7 network in the UK? |
| 23:28.27 | badbit | So so so many rel 17's. |
| 23:28.52 | badbit | I don't imagine that many people being on the phone right now. |
| 23:32.13 | badbit | Okay, I see now. Should have probably searched first. |
| 23:34.08 | badbit | Is this sothing to do with the EE merge? |
| 23:35.05 | badbit | Telephonica not happy about the LTE takeover so will just purchase them all? |
| 23:35.56 | badbit | Toys, pram? |
| 23:39.52 | badbit | Busy busy. |
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| 23:41.49 | *** join/#asterisk subhi (~subhi@146.214.112.87.dyn.plus.net) |
| 23:43.17 | subhi | Hi All, |
| 23:43.37 | subhi | I am trying to sort out a dialplan whereas if the extension is not available it puts the call through to voicemail |
| 23:43.41 | subhi | can't get it working |
| 23:44.26 | subhi | I have an extension I am using on my mobile phone over 3G/4G sometimes I don't have coverage |
| 23:47.40 | subhi | part of my dialplan is here: http://pastebin.com/Y8jiEk78 |
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