| 00:01.40 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 00:02.25 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
| 00:06.01 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
| 00:06.24 | *** join/#asterisk x5eb (~seb@seb-hpws2.w1.tele.crt1.net) |
| 00:06.42 | *** join/#asterisk hfp_ (~hfp@MTRLPQ0736W-LP130-02-1279513144.dsl.bell.ca) |
| 00:06.55 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 00:08.12 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:10.39 | *** join/#asterisk yokel_ (~yokel@unaffiliated/contempt) |
| 00:12.54 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 00:20.07 | *** join/#asterisk dlynes (~dlynes@ip099.216.145.98.nas.net) |
| 00:31.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:34.50 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-mbhcibzcatbampun) |
| 00:38.11 | *** join/#asterisk jeffspeff (~Jeff@173.218.60.159) |
| 00:39.02 | *** join/#asterisk voxter (~voxter@irc.voxter.net) |
| 00:40.35 | jeffspeff | i'm looking for softphone suggestions. video, IM, presence, outlook and/or active directory integration are required. opensource or free would be nice. |
| 00:41.30 | *** join/#asterisk malachi_constant (~toor@unaffiliated/malachiconstant/x-837457) |
| 00:41.54 | malachi_constant | Hi guys. I'm trying to use a Rhino channel bank to provide upstream phone lines to asterisk, and I'm having a hell of a time. |
| 00:47.03 | jeffspeff | malachi_constant, i'm not very familiar with the pstn integration, but did you load the drivers for the bank? i think you'll also need dahdi compiled and installed as well. |
| 00:50.26 | malachi_constant | jeffspeff: I don't believe there are any. I have a Sangoma T1 interface card in the PBX. |
| 00:50.39 | malachi_constant | Drivers are compiled and installed and so is DAHDI. |
| 00:53.46 | malachi_constant | When I try to dial, though, I get this: http://paste.ubuntu.com/8201645/ |
| 00:56.23 | jeffspeff | you may get a more descriptive error if you enable dahdi dugging |
| 00:56.44 | jeffspeff | i know for sip it's "sip set debug on" or something like that |
| 00:56.57 | [TK]D-Fender | FC-01-PBX-DESKTOP*CLI> originate DAHDI/1/1 extension [10digitphonenumber]@analog-lines |
| 00:57.11 | [TK]D-Fender | show us your registered channels |
| 00:57.48 | [TK]D-Fender | Actually .... |
| 00:57.50 | [TK]D-Fender | -- Executing [10digitphonenumber@analog-lines:1] Dial("DAHDI/1-1", "DAHDI/0/g1/{EXTEN}") in new stack |
| 00:57.54 | [TK]D-Fender | This is NOT valid |
| 00:58.17 | [TK]D-Fender | DAHDI/channel_or_group_identifier/numbertodial |
| 00:58.55 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 00:59.25 | [TK]D-Fender | <PROTECTED> |
| 01:06.45 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 01:07.41 | malachi_constant | [TK]D-Fender: Oooh, let me see. |
| 01:09.52 | *** join/#asterisk rock1979 (~rock@46.252.100.15) |
| 01:10.02 | rock1979 | hello |
| 01:10.23 | rock1979 | how can i force asterisk 11.5.0 to reject t38 invites |
| 01:10.25 | malachi_constant | [TK]D-Fender: Equipment on the other end is a Rhino 24-port channel bank, signalling is kewlstart. |
| 01:10.45 | rock1979 | Fender, nice to see you here |
| 01:10.57 | jeffspeff | rock1979, are you using sip? |
| 01:11.04 | rock1979 | yes |
| 01:11.05 | [TK]D-Fender | rock1979: Don't enable pasthrough, don't enable gateway mode |
| 01:11.17 | rock1979 | yes i did noth of them |
| 01:11.30 | jeffspeff | is t.38 enabled on the sip peers? |
| 01:11.51 | rock1979 | no |
| 01:11.52 | rock1979 | t38pt_udptl=no |
| 01:11.59 | rock1979 | setvar=FAXOPT(gateway)=no |
| 01:12.17 | rock1979 | but still asterisk is sending Sip 200 ok with g711 in sdp |
| 01:12.33 | rock1979 | it should send sip488 |
| 01:12.46 | rock1979 | so the remonte side fallback to g711 right ? |
| 01:13.06 | rock1979 | i also set verbose to 10 but nothing in the logs |
| 01:13.11 | jeffspeff | what about the faxdetect option? |
| 01:13.32 | [TK]D-Fender | verbose is NOT "sip debug" |
| 01:13.39 | [TK]D-Fender | and you haven;t shown us the problem\ |
| 01:13.44 | rock1979 | calls are coming from sip and sent to telco with dahdi |
| 01:13.56 | [TK]D-Fender | Since you'v been coming in here for days you should have learned by now : arrive with the debug in-hand |
| 01:14.02 | [TK]D-Fender | SHOW us the problem. |
| 01:14.25 | malachi_constant | [TK]D-Fender: Holy shit I think that was it. Thank you! |
| 01:14.26 | [TK]D-Fender | the configs AND the matching debug |
| 01:15.02 | rock1979 | i set " core set verbose 10 " |
| 01:15.17 | [TK]D-Fender | that is NOT SIP DEBUG |
| 01:15.50 | jeffspeff | rock1979, just type "core set debug on" in the cli |
| 01:15.56 | rock1979 | i set "sip set debug on ip XXX.XXX.XXX.XXX" |
| 01:15.56 | [TK]D-Fender | ... |
| 01:16.09 | rock1979 | i also got it with tcpdump |
| 01:16.33 | rock1979 | sip set debug on ip XXX.XXX.XXX.XXX it will be fine right ? |
| 01:20.54 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 01:23.52 | malachi_constant | rock1979: No. |
| 01:24.09 | rock1979 | sorry ? |
| 01:24.37 | malachi_constant | [TK]D-Fender: http://paste.ubuntu.com/8201844/ |
| 01:25.25 | malachi_constant | Any clues sir? |
| 01:28.56 | rock1979 | <[TK]D-Fender>: in asterisk cli i executed "sip set debug on ip XXX" i got the debug in log |
| 01:29.37 | rock1979 | will this be fine for you ? |
| 01:32.36 | malachi_constant | rock1979: core set debug 9 sip |
| 01:32.36 | *** join/#asterisk dms1 (~dms1@2601:0:5480:e4::1008) |
| 01:32.40 | malachi_constant | Or something like that. |
| 01:36.48 | *** join/#asterisk malachi_constant (~root@unaffiliated/malachiconstant/x-837457) |
| 01:44.22 | rock1979 | here is the debug |
| 01:44.24 | rock1979 | https://www.sharefest.me/d0cbb7927ab9798984261b782769ff7d |
| 01:47.03 | rock1979 | i am also sending sip.conf for the peer |
| 01:47.56 | rock1979 | https://www.sharefest.me/48c58b9f816388d1838a64d6a0b605b9 this is sip.conf |
| 01:48.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:50.39 | rock1979 | Fender are you there ? |
| 01:54.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:54.49 | rock1979 | i will be glad for help |
| 01:55.10 | rock1979 | really dont know how to solve this issue |
| 01:58.28 | rock1979 | i set t38pt_udptl=no and setvar=FAXOPT(gateway)=no but still asterisk is not sending SIP 488 to t38 invite |
| 02:00.44 | *** join/#asterisk JC_SoCal_ (sid1590@gateway/web/irccloud.com/x-qlfzedqibcmvjfjd) |
| 02:02.34 | rock1979 | the interesting part is that althoug asterisk is sending sip 200 with g711 in sdp. i am capable to receive fax with t38 |
| 02:03.09 | JC_SoCal_ | not sure what i'm doing wrong, but i'm in the asterisk console and i want to dial a number to test, so i try > console dial 6195551234 -- and i get these << Console Received digit x of duration >> but my phone never rings |
| 02:03.28 | rock1979 | with wireshark i am viewing the t38 messages |
| 02:03.38 | rock1979 | and the fax is sent to the remote side |
| 02:10.29 | JC_SoCal_ | what i would like to do is from the cli, take phone number A and phone number B and connect them |
| 02:12.29 | rock1979 | asterisk is receiving t38 invite |
| 02:12.38 | rock1979 | and sending first sip 100 trying |
| 02:12.50 | rock1979 | then sip 200 ok with voice codecs in sdp |
| 02:13.25 | rock1979 | any help will be appreciated |
| 02:36.38 | *** join/#asterisk floren (~floren@unaffiliated/floren) |
| 02:48.16 | *** join/#asterisk mutable_malachi (~mmalachi@unaffiliated/malachiconstant/x-837457) |
| 03:10.10 | floren | incredibly quiet, where is everyone? ChannelZ are you awake? :) |
| 03:10.46 | floren | just wanted to say good night to everyone |
| 03:33.19 | mutable_malachi | Good night floren. :) |
| 03:48.49 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 04:32.52 | [TK]D-Fender | malachi_constant: -- Executing [mytendigitnumber@analog-lines:1] Dial("DAHDI/1-1", "DAHDI/g0/{EXTEN}") in new stack |
| 04:33.00 | [TK]D-Fender | malachi_constant: You're clearly making a broken variable reference |
| 04:33.28 | mutable_malachi | [TK]D-Fender: Dammit, you're right. |
| 04:34.17 | mutable_malachi | It's ${EXTEN}, right? I'll screw with it when I get back in the office. |
| 04:38.36 | [TK]D-Fender | that's what you appear to be trying to do. |
| 04:39.29 | mutable_malachi | Right, it is. |
| 04:40.00 | mutable_malachi | Would that be a valid construction with ${EXTEN}? |
| 04:40.52 | [TK]D-Fender | yes |
| 04:41.25 | mutable_malachi | Terrific. Will see what's up. Thank for your help! |
| 04:51.08 | *** join/#asterisk vedic (~V@183.82.105.37) |
| 04:56.41 | *** join/#asterisk r00f (~r00f@av.r00f.us) |
| 05:02.33 | ChannelZ | I wandered off.. but goodnight floren |
| 05:05.52 | *** join/#asterisk jploh (~textual@49.146.179.249) |
| 05:06.25 | vedic | When a recording is saved via Asterisk agi command in pcm16 (.wav file), does that contain some header ? |
| 05:37.19 | *** join/#asterisk rock1979 (~rock@46.252.100.15) |
| 05:37.36 | rock1979 | hello |
| 05:37.56 | rock1979 | this my 7. day trying to solve an issue with asterisk |
| 05:38.10 | rock1979 | does anyone can halp me |
| 05:47.18 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 05:48.33 | Penguin | Did you already state the problem, or is this the 7th day of asking to ask? |
| 05:51.44 | r00f | day 7. they stil do not suspect me having a problem. |
| 05:55.42 | rock1979 | asterisk 11.5.0 with sip to dahdi calls |
| 05:56.16 | rock1979 | when t38 invite is received asterisk is sending SIP 200 ok with g711u,g711a,g729 codecs in SDP |
| 05:56.34 | rock1979 | <PROTECTED> |
| 05:59.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 06:03.34 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 06:13.19 | r00f | do you set t38pt_udptl=no globally? maybe it is still enabled on that peer |
| 06:13.53 | r00f | look sip show peer on that peer, what does it say regaring t38 |
| 06:18.49 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:e0c3:ac1d:394d:f910) |
| 06:19.16 | rock1979 | i checked with sip show peer on specific peer |
| 06:21.08 | rock1979 | T.38 support : No |
| 06:22.36 | rock1979 | the interesting part is althought t38pt_udptl=no , asterisk is sending and receiving T38 messages with remote side and receives t38 fax succesfully |
| 06:22.54 | rock1979 | the only problem is wrong reply to T38 invite |
| 06:27.09 | rock1979 | any idea ? |
| 06:39.23 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:57.45 | *** join/#asterisk vedic (~V@183.82.105.37) |
| 06:58.14 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:00.28 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:01.27 | *** join/#asterisk RadJackson (~RadJackso@cpe-xd001622.cust.jaguar-network.net) |
| 07:01.53 | wdoekes | rock1979: I believe you. I saw the pcaps last week. |
| 07:01.57 | wdoekes | let me check |
| 07:02.43 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 07:03.28 | RadJackson | Hi , i am using Read CMD , with 5 sec timeout , no limit max digits. i've noticed that everytime i type a digit , it reloads the 5 sec countdown |
| 07:03.50 | RadJackson | any idea how to make it 5 sec total timeout after launching the Read CMD ? |
| 07:12.12 | *** join/#asterisk grzew (~gwieczore@atmsa.atm.com.pl) |
| 07:12.36 | grzew | Hello, is anyone online here and have time to help me? |
| 07:12.48 | ChannelZ | ~ask |
| 07:12.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:13.13 | kleszcz | grzew: ? |
| 07:13.36 | grzew | here is my question: http://forums.digium.com/viewtopic.php?f=1&t=91235&sid=8755ab3a6c3e69b1bf95cbd285e1c121 |
| 07:14.49 | ChannelZ | Are you saying DTMF doesn't work, or you just can't "see them"? |
| 07:15.01 | grzew | doesn't work |
| 07:15.20 | grzew | When I do Read() it gets nothing |
| 07:16.02 | ChannelZ | What DTMF mode are you using on whatever peer you are dialing? What is the far end device? |
| 07:16.25 | grzew | End device is mobile phone |
| 07:16.45 | grzew | dtmf rfc2833 |
| 07:17.26 | grzew | I was trying other modes like inband, auto ... |
| 07:18.16 | grzew | The same mobile phone with console dial and channel originate |
| 07:19.06 | ChannelZ | well you can set RTP debug on and see if you are getting any type 101 packets from your ITSP |
| 07:19.09 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 07:19.34 | grzew | I will do this test right now. Give me a sec. |
| 07:21.35 | RadJackson | Is it a bug that Read cmd timeout reloads everytime a digit is typed ? |
| 07:22.49 | grzew | I set: "rtp set debug on" but nothing catched |
| 07:23.24 | grzew | RadJackson: no, timeout is executed after x second. |
| 07:23.48 | grzew | RadJackson: It not react on any digit. |
| 07:24.40 | RadJackson | <PROTECTED> |
| 07:24.53 | RadJackson | i type 1 , i can wait 4 seconds and type 2 , wait 4 seconds and type 3 |
| 07:25.09 | RadJackson | so for 5s timeout , it is (4s * number of digits typed) |
| 07:25.14 | grzew | RadJackson: Ok, testing now. |
| 07:25.33 | RadJackson | exten => s,n,Read(WHATEVER,beep,,,1,5) |
| 07:26.10 | grzew | -- User entered nothing. |
| 07:26.15 | grzew | after 5 sec. |
| 07:26.41 | RadJackson | once you hear the beep , type a digit , and wait for 3 seconds , and type the second |
| 07:26.50 | RadJackson | it will not go timeout after 5 seconds |
| 07:27.08 | RadJackson | its 5 seconds after the user stops entering |
| 07:27.16 | RadJackson | I want 5 seconds after lauching the cmd Read |
| 07:27.17 | RadJackson | is it possible ? |
| 07:27.33 | grzew | I was entering, but it ended after 5 sec with "User entered nothing" |
| 07:27.45 | RadJackson | can you pastebin your code ? |
| 07:27.58 | grzew | dialplan? |
| 07:28.04 | RadJackson | yes |
| 07:28.06 | grzew | ok |
| 07:28.35 | *** join/#asterisk shadej (~bzegeju@213.55.73.146) |
| 07:28.59 | grzew | http://pastebin.com/xgPfmquH |
| 07:29.23 | RadJackson | thanks , going to check it now |
| 07:30.13 | grzew | And I am originating calls like this: channel originate SIP/trunkname/mobilenumber application Macro macroname |
| 07:30.17 | shadej | I am reading getting started in ARI (https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI) but I am stuck with wscat |
| 07:30.40 | shadej | wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" does not give any response |
| 07:33.01 | ChannelZ | It's an interdigit timer, not an overall one. |
| 07:33.49 | grzew | But I was pressing digids and get disconnected after 5 sec. |
| 07:34.17 | grzew | The problem is with receiving dtmf with originated calls |
| 07:34.28 | grzew | When I do console dial it works. |
| 07:34.50 | grzew | I have no idea what is the difference |
| 07:36.08 | RadJackson | grzew on your code , you mentioned 1 max digit |
| 07:36.10 | ChannelZ | your paste showed a max digits of 1. |
| 07:36.32 | RadJackson | for my application , i do not know what is the maximuim of digit the user can type , all i need , is stop reading after 5 sec |
| 07:36.34 | RadJackson | try this : |
| 07:36.53 | RadJackson | http://pastebin.com/K4ZXTJAg |
| 07:37.31 | grzew | ok, give me a sec. |
| 07:38.46 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:39.10 | ChannelZ | try ReadExten instead |
| 07:39.11 | grzew | hmm, now I am getting RTP packet 101 |
| 07:39.25 | grzew | ReadExten works in macro? |
| 07:40.15 | ChannelZ | actually that won't work either |
| 07:40.39 | grzew | Ok, now with your code every digit is adding 5 sec to timeout |
| 07:40.45 | RadJackson | exactly |
| 07:41.04 | grzew | So, why I can't read this digit? |
| 07:41.17 | RadJackson | I need to stop reading 5 seconds after lauching CMD. |
| 07:41.21 | grzew | Or I can't see this on console? |
| 07:41.23 | RadJackson | my application is a game |
| 07:41.32 | ChannelZ | actually maybe ReadExten will work for you.. trying again |
| 07:41.34 | RadJackson | I dont want the user to google the question and type the answer |
| 07:41.35 | RadJackson | ok |
| 07:42.04 | grzew | ChannelZ: I am trying now with ReadExten |
| 07:42.17 | RadJackson | ChanServ same parameters for Read ? |
| 07:42.22 | RadJackson | ChannelZ* |
| 07:43.26 | ChannelZ | nah it's still an interdigit timeout |
| 07:44.26 | ChannelZ | RadJackson: wouldn't they have to be typing some of the right digits to keep the timeout from timing out while googling? In which case they'd be answering wrong? |
| 07:44.50 | RadJackson | yeah , but for example if asking which year X person was born ,90% of cases it starts with 1 |
| 07:44.53 | shadej | wscat does not give any response in ubuntu , help |
| 07:45.07 | RadJackson | 50% the second digit is 9 |
| 07:46.03 | ChannelZ | shadej: did you install npm? |
| 07:46.13 | shadej | ChannelZ: yes |
| 07:46.44 | ChannelZ | oh nm I thought you meant you didn't have that command |
| 07:47.07 | shadej | do I need to install node too? |
| 07:47.09 | grzew | Strange, when I used ReadExten I got: ########NEXT i ############# after pressing 5 |
| 07:47.34 | RadJackson | grzew same , it reads the extensions , since what you typed is not controlled , so it is the i extension (i think) |
| 07:48.02 | grzew | RadJackson: shouldn't i get digit number? |
| 07:48.23 | RadJackson | with Read cmd yes. , ReadExten , doesnt look like it gets the typed digits |
| 07:48.59 | grzew | but why Read cmd gets nothing? |
| 07:49.24 | RadJackson | for me , it gets all the digits typed after 5sec of time out |
| 07:49.35 | RadJackson | ayway i guess ive got no choice , i even tried the variables timeout(digit) and timeout(response) , still not working |
| 07:52.00 | grzew | RadJackson: Is there any way to get user digit? |
| 07:54.10 | RadJackson | yes , Read |
| 07:54.25 | RadJackson | If i understood your question well |
| 07:54.38 | grzew | So I will test with Read again |
| 07:55.52 | ChannelZ | I need a hotdog |
| 07:55.58 | grzew | User entered '9' |
| 07:56.39 | grzew | Great :) It is working now, but not immediately - only after timeout. |
| 07:56.50 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 07:56.58 | ChannelZ | It would, unless you only told it 1 max digit. |
| 07:57.40 | shadej | ChannelZ: I reinstalled;it is working but now I am getting server 404 error |
| 07:57.49 | grzew | How would it look like with max only digit? Now it is : Read(callcode,beep,,,1,5) |
| 07:58.05 | RadJackson | add 1 as the third parameter |
| 07:58.09 | RadJackson | thid parameter is maxdigitq |
| 07:58.15 | RadJackson | third* |
| 07:59.00 | grzew | Great, thank You very much with resolving my issue. |
| 07:59.56 | grzew | If we meet I am going to bue a beer for RadJackson and ChannelZ |
| 08:00.07 | grzew | *buy |
| 08:03.36 | RadJackson | yw |
| 08:05.29 | ChannelZ | oh did something get fixed while I was off making hotdogs? |
| 08:05.46 | RadJackson | yes ,for sure not my problem |
| 08:05.56 | JerJer | mmmm miscellaneous meat product |
| 08:06.38 | ChannelZ | Mmmmhmmm! |
| 08:07.15 | grzew | ChannelZ: Yes - my problem was resolved, thx. |
| 08:08.21 | ChannelZ | So what was it.. last I remember I told you to turn on RTP debug to see what was going on |
| 08:09.17 | shadej | ChannelZ: where do I find ari.conf? |
| 08:09.27 | ChannelZ | in /etc/asterisk ? |
| 08:11.14 | shadej | ChannelZ: I cant find it but I created one. |
| 08:13.11 | grzew | ChannelZ: I believe it was problem with my Read function arguments. |
| 08:14.48 | ChannelZ | well yeah.. if you want the sample one it's in the source, in configs |
| 08:15.47 | shadej | ChannelZ: this is the content of my new ari.conf http://pastebin.com/yhzaAfZX |
| 08:15.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:15.59 | shadej | any error? |
| 08:16.00 | RadJackson | ChannelZ have you an idea of any workarounds ? :) i'm thinking about launching READ CMD , in the meanwhile, runing an AGI that will redirect the channel to the timeout extension (ChannelRedirect) after 5 seconds... doesnt look like a clean solution for me.... |
| 08:16.09 | ChannelZ | dunno I've not used ARI yet |
| 08:16.36 | shadej | ChannelZ: so where do I got the sample file for ari.conf? |
| 08:16.43 | ChannelZ | seems logical though |
| 08:16.58 | ChannelZ | I told you, it's in the asterisk source, configs dir.. ari.conf.sample |
| 08:17.11 | RadJackson | shadej https://github.com/mojolingo/asterisk/blob/master/configs/ari.conf.sample |
| 08:17.37 | shadej | RadJackson: thank you |
| 08:21.48 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 08:22.06 | shadej | RadJackson: I am getting this error. error: Error: unexpected server response (404) |
| 08:25.47 | *** join/#asterisk CeBe (~CeBe@port-92-206-51-1.dynamic.qsc.de) |
| 08:26.01 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-51-1.dynamic.qsc.de) |
| 08:31.32 | shadej | okay it seems like I dont have the ari module but my asterisk is Asterisk 1.8.13.1~dfsg-3ubuntu3 |
| 08:35.01 | ChannelZ | oh wait.. you are trying to do websockets? mmyeah that ain't gonna work in 1.8 |
| 08:35.18 | ChannelZ | (or ARI) |
| 08:35.38 | ChannelZ | ARI is new for 12. |
| 08:37.23 | *** join/#asterisk areski (~areski@80.174.128.10.dyn.user.ono.com) |
| 08:37.34 | shadej | ChannelZ: I installed my asterisk tomorrow, do you mean it is not latest? |
| 08:37.44 | shadej | I mean yesterday |
| 08:37.46 | shadej | hah |
| 08:39.12 | ChannelZ | Yeah, 1.8 is kind of old. There's been 10, 11, and now 12. But lots of distros have old packages |
| 08:39.32 | RadJackson | ChannelZ i'm trying to launch an AGI script async , whats the right syntax ? |
| 08:39.37 | eirirs | ChannelZ: And 13.0 |
| 08:39.50 | RadJackson | i've tried AGI(redirector.php:async) and AGI(async:redirector.php) ; not working |
| 08:40.01 | ChannelZ | 13 is beta |
| 08:45.54 | ChannelZ | Rad I don't think it works like that, async AGI is basically using Manager. |
| 08:46.09 | RadJackson | ok , any idea how to launch an agi script async? |
| 08:46.23 | RadJackson | i'm thinking about lauching a bash script that will run the AGI on background |
| 08:46.27 | RadJackson | > /dev/null 2>&1 < /dev/null & |
| 08:51.31 | shadej | is it possible to upgrade asterisk1.8 to 12 using terminal commands? |
| 08:51.55 | ChannelZ | just build from the source. it's not that hard. |
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| 08:57.23 | *** join/#asterisk shadej (~bzegeju@213.55.73.146) |
| 09:00.52 | shadej | is asterisk 1.8 another name for asterisk 11 |
| 09:01.11 | ChannelZ | no |
| 09:06.25 | shadej | ChannelZ: ok, is this the tarball for asterisk 12? http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.12.0.tar.gz |
| 09:07.51 | ChannelZ | no that's 11 |
| 09:08.13 | ChannelZ | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12-current.tar.gz |
| 09:08.27 | ChannelZ | which is 12.5.0 |
| 09:16.45 | wdoekes | rock1979: 28.log:13:12 < rock1979> i didnt tried with latest version * 11 vers'on |
| 09:17.47 | wdoekes | if we ask you if you tried the latest version, it's generally not just a question, but a suggestion |
| 09:18.25 | wdoekes | your version is more than a year old |
| 09:18.46 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 09:20.52 | shadej | ChannelZ: but I cant see the reason that I cant get the latest asterisk , when I issue: apt-get install asterisk? |
| 09:24.07 | ChannelZ | because the repository you're using either doesn't have it or it's called something else |
| 09:26.09 | ChannelZ | Probably the former though. Most wouldn't include 12 because it's a feature release |
| 09:32.14 | wdoekes | rock1979: but.. I can confirm that asterisk will happily forward a T38 request from a device that has t38pt_udptl set to No, to a device that has it set to Yes |
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| 10:44.44 | rock1979 | wdoekes: t38pt_udptl is set to No for the sip peer |
| 10:45.05 | rock1979 | and the brighed side is dahdi |
| 10:45.52 | rock1979 | i couldnt manage to send SIP 488 to a T.38 re-invite |
| 10:49.30 | casdude | I have a weird problem where, very occasionally, when someone dials outbound the calls goes through fine, but the Asterisk CLI only shows 5 digits having been dialled. Does anyone know why it wouldn't show the rest of the digits? |
| 10:52.08 | casdude | There's nothing abnormal about the numbers being dialled either, very standard numbers |
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| 11:12.14 | wdoekes | rock1979: ignoresdpversion=yes -- try that |
| 11:13.04 | wdoekes | and collect the DEBUG log surrounding the T38 reINVITE if that doesn't help |
| 11:13.42 | wdoekes | rock1979: and you keep repeating yourself while giving us little to no extra info |
| 11:21.50 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 11:24.48 | wdoekes | (and I retract my "asterisk will happily forward" statement, that was user error on my part) |
| 11:29.25 | *** join/#asterisk hahainternet (~yeshello@87.98.174.109) |
| 11:30.57 | hahainternet | I've connected a legacy PBX to Asterisk via a Digium PRI card. The PBX dials out but always seems to aim for the 's' extension. It appears that it is either dialling no actual target number or there's a misconfiguration that's stopping asterisk from reading them |
| 11:31.06 | hahainternet | anyone come across this before? any recommendations for areas to investigate? |
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| 12:16.21 | shadej | I finally got my asterisk12 running, but the /etc/asterisk dicrectory is empty,is that normal? |
| 12:18.49 | kleszcz | make samples |
| 12:19.29 | kleszcz | after make install |
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| 12:36.13 | shadej | kleszcz: thank u, I got the sample confs now. I was having some sip config. in my old asterisk but now it seems like they are not working, I am trying to call using one of my sip clients but there is no any log on the asterisk CLI. |
| 12:43.38 | shadej | how do I test asterisk 12 if it is working? |
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| 12:48.28 | rock1979 | wdoekes: thanks for the valuable information (ignoresdpversion=yes) solved the issue |
| 12:48.59 | wdoekes | rock1979: |
| 12:48.59 | wdoekes | <PROTECTED> |
| 12:48.59 | wdoekes | <PROTECTED> |
| 12:49.00 | wdoekes | <PROTECTED> |
| 12:49.36 | wdoekes | if you had looked at the debug logs, you would have seen that the SDP is ignored becuase the session version isn't increased |
| 12:50.11 | wdoekes | i.e. your ITSP is doing it wrong |
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| 12:54.29 | [TK]D-Fender | shadej: Use it. |
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| 12:57.22 | rock1979 | great thanks wdoekes |
| 12:58.12 | shadej | [TK]D-Fender: sure that is the answer but it was not working for me, now I restarted it it is working. |
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| 13:33.48 | casdude_ | hi |
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| 13:52.20 | JC_SoCal | is there a way to start a recording in a call file? |
| 14:00.02 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:00.48 | [TK]D-Fender | You're in the dialplan just like any other call.... |
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| 14:17.15 | shadej | hi all, following this tutorial(https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI) . but I am stuck with this error any suggestion:https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI |
| 14:17.20 | shadej | oh sorry |
| 14:17.34 | shadej | this error:error: Error: unexpected server response (404) |
| 14:17.39 | shadej | after issuing the wscat |
| 14:24.34 | JC_SoCal | [TK]D-Fender: cool thx =) i made a macro |
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| 14:36.47 | shadej | can anyone tell me how to test ARI in asterisk 12? |
| 14:37.04 | shadej | I keep getting page not error(404) |
| 14:44.23 | Apteryx | Hello :) |
| 14:45.06 | *** join/#asterisk ggayan (~ggayan@190.215.47.74) |
| 14:45.22 | Apteryx | I'm trying to test if my Asterisk server is reachable outside my LAN, and the only two networks I have is my WLAN and my job VPN. |
| 14:47.43 | Apteryx | But I'm not sure if this is confusing Asterisk in a different way than when I am truly connected to my job's network and trying to reach it. |
| 14:49.46 | Apteryx | I have DirectMedia=No and NAT=force_rport,comedia |
| 14:50.29 | Apteryx | But I'm still getting one way audio (the SIP client on the VPN can send audio but cannot receive it) |
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| 15:19.37 | Stefan27 | How can i query the state of a channel in asterisk CLI? I have two bridges [SIP/A, Local/X@Y;2] and [SIP/B,LocalX@Y;1] and audio is indeed flowing fine both directions from A to B. BUT asterisk sends AMI events where LocalX@Y;2 has ChannelStateDesc: Ringing which is unexpected... LocalX@Y;1 seems to have state Up as well as the two sip channels |
| 15:20.45 | WIMPy | 'core show channels [...]' is what you get. |
| 15:22.42 | Stefan27 | thx, it confirms what I said. Should it even be possible for two legs of the same local channel to have the combination "Ringing" and "Up" as states? |
| 15:23.17 | WIMPy | I wouldn't think so. |
| 15:30.42 | Stefan27 | ok ill look into it and get back with more info |
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| 16:20.11 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 16:20.41 | dan_j | Hi. Is it possible to trigger an audio playback using the AMI, to a specific channel? |
| 16:31.00 | WIMPy | You can originate a chanspy. |
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| 16:41.41 | dan_j | chanspy? |
| 16:41.49 | dan_j | I'm not following. |
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| 16:42.59 | WIMPy | YOu originate a call from a chanspy with whisper/barge enabled on the channel you want to inject audio to a playback. |
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| 16:46.46 | dan_j | I think I understand. I can using the AMI Originate action, to create a new channel that executes the chanspy app on a specific channel, and then starts a playback. But doesnt that require me to still start a playback somehow? |
| 16:47.42 | dan_j | If I use AMI Originate to start the Playback application, do both channels hear the recording? |
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| 16:51.56 | WIMPy | The playback wil be the 2nd leg of the originate. |
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| 16:52.22 | WIMPy | See the ChanSpy options. You can select which channel(s) to play to. |
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| 16:58.56 | dan_j | I'm still not following 100%. In the originate action, channel would be a dialplan that plays a sound, and application would be chanspy pointing to the channel that i want to whisper to? |
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| 17:00.56 | dan_j | Any idea why no one made a PlaySound AMI action? There is a PlayDTMF one. |
| 17:01.54 | file | because you can't safely play sound to a channel unless you do the above |
| 17:02.47 | dan_j | Ok. Noted. |
| 17:08.56 | WIMPy | A shortcut would still be quite nice. |
| 17:09.21 | dan_j | I was going to say that. Wasn't sure if i'd missed something. |
| 17:12.06 | hahainternet | ok solved our problem, mitel device had deferred dialling on so it picked up the line and didn't dial immediately |
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| 17:33.32 | malachi_constant | [TK]D-Fender: IT WORKS! |
| 17:33.40 | malachi_constant | THANK YOU! |
| 17:34.07 | malachi_constant | [TK]D-Fender: I can't believe I was making such a stupid mistake. |
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| 19:03.10 | dan_j | WIMPy: I've managed to trigger a playback using AMI but can only get it to play to both channels by using the B option. Otherwise, theres no sound played. |
| 19:03.17 | *** join/#asterisk pramsky (~pramod@c-67-185-160-144.hsd1.wa.comcast.net) |
| 19:03.27 | dan_j | Any ideas what I could be doing wrong? |
| 19:04.59 | WIMPy | w/W don't work? |
| 19:07.23 | dan_j | Ah. W did it. Weird the wiki suggests that W simply stops listening, rather than allows talking. |
| 19:08.49 | *** join/#asterisk asteriskbizua (4d5b8405@gateway/web/freenode/ip.77.91.132.5) |
| 19:09.06 | WIMPy | use 'core show application ...' next time :-) |
| 19:09.36 | dan_j | Actually, its stopped working. One moment while i work out what i changed. |
| 19:10.15 | asteriskbizua | somebody using Goip 4 + sms forward feature ? |
| 19:10.18 | WIMPy | . o O ( Isn't the wiki generated from the same data? ) |
| 19:11.16 | dan_j | I think i mis-read it. |
| 19:11.22 | dan_j | Not sure why w didnt work though. |
| 19:12.39 | asteriskbizua | Who can help me with Goip 4 + sms forward feature ? |
| 19:13.06 | WIMPy | Possibly because the other audio stream isn't propperly discarded. |
| 19:13.09 | dan_j | Ok, its working now. I had to disable the beep on spy |
| 19:13.39 | dan_j | The beep was playing and delaying the playback by a few seconds. |
| 19:14.08 | WIMPy | That makes sense. |
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| 19:22.23 | quenode | Hi is there anyway to improve SIP call on <300ms ping latency. Europe - Australia. Using maybe another SIP provider option ? |
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| 19:48.26 | dan_j | WIMPy: thanks for the help. works perfectly. |
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| 20:12.09 | posixninja | is it still possible to connect to google voice through asterisk? I heard they made some changes recently |
| 20:12.41 | posixninja | I've read a few tutorials saying pjsip was needed now, but they didn't really talk about how to setup pjsip stuff |
| 20:14.44 | navaismo | posixninja: GV is dead, but many persons claims to have running. tm1000 told me that depends on your account setting if you use hangouts then nope if you use XMPP maybe it will work. Officialy the support is dead, and you need chan_motif to work on that |
| 20:15.33 | posixninja | damn, ok. also curious if anyone knows about wifi calling which is being pushed out from most carriers. would it be possible to make asterisk connect to say... t-mobile trunk (if you know username/password) |
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| 20:16.49 | posixninja | or alter the carrier files on the device to make it connect to your asterisk server |
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| 21:13.02 | dan_j | Is anyone finding that some call recordings are not working properly? I would say about 50% of my calls are corrupt. |
| 21:14.03 | ChannelZ | corrupt how |
| 21:14.17 | dan_j | They won't play back. |
| 21:14.38 | dan_j | I've made about 50 test calls just now. All calls execute mixmonitor at the start. |
| 21:15.16 | dan_j | and the h extension executes stopmixmonitor and moves the files out of spool to a different folder. |
| 21:15.46 | dan_j | about 20 didnt work and won't play back |
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| 21:27.49 | ChannelZ | well nevermind |
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| 21:34.47 | Apteryx | Hello! Could someone clarify what the "bind" option is for when defining transport mechanisms? |
| 21:38.29 | file | it is the IP address and optionally port to bind the transport to. |
| 21:45.39 | Apteryx | file: What does it does exactly? How is it different from an ACL? |
| 21:46.09 | file | an ACL determines whether traffic is accepted or denied from sources |
| 21:46.27 | file | binding is what the transport is listening for traffic on |
| 21:47.35 | Apteryx | file: Ok. That's clearer. Thanks. What's the advantage of defining it? 0.0.0.0 will listen on any adress? |
| 21:50.25 | file | might want to limit the scope |
| 21:51.45 | Apteryx | file: Ok. More restrictive == more security I guess, although I don't quite see how in this case. |
| 21:52.28 | file | if you have two interfaces and you only bind to one, then any traffic on the other won't be received |
| 21:54.28 | Apteryx | Ah, multiple interfaces. I see! Thanks again. |
| 21:54.59 | ChannelZ | the bind address is what IP/interface asterisk will listen for traffic on. It's the first 'physical connection' between asterisk and the network. |
| 21:56.33 | Apteryx | Is 0.0.0.0 a 'catch-all' or is it just defined as such in the examples to remind me to set it to a meaningful value? |
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| 22:18.25 | iceyp | Hey guys, got a weird problem where asterisk if it sees BUSY it continually tries to redial for a while... everywhere i've looked online there are instructions on how to ad this as a feature but not how to disable it :/ |
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| 22:24.31 | CeBe1 | iceyp: how would you add it? |
| 22:24.59 | CeBe1 | most likely that somehow covers how to disable it... |
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| 22:29.09 | iceyp | nah doesnt seem to be the same, asterisk shows Dial("SIP/sip02-00002f83", "SIP/vibesip01/093090361,60") in the log then goes Got SIP response 486 "Busy Here" then Everyone is busy/congested at this time (1:1/0/0) then repeats executing a dial over and over with the same messages |
| 22:29.48 | iceyp | seems there is mention of Auto fallthrough |
| 22:30.03 | iceyp | that may be the key, let me google that |
| 22:34.37 | Apteryx | I'm in the process of migrating to chan_pjsip, and would like to know which modules I need to load for a minimal example to work. I tried res_pjsip.so and chan_pjsip.so but this does not seem to suffice. |
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| 23:14.36 | Mango45 | I want to make a direct SIP call to some SIP URI. The other side seems to have a five minute session timer, and my connections expire after 90 seconds, so the invite is blocked by the firewall. |
| 23:15.03 | Mango45 | What is the best way to correct this, other than increasing the 90 seconds? |
| 23:24.14 | ChannelZ | turn on keepalive? |
| 23:25.49 | ChannelZ | although I don't entirely understand your question, which side is blocking? |
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