| 00:41.19 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:41.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 08:12.50 | jkroon | hi all, with logger.conf: full => ...,verbose,... doesn't actually log the verbose stuff to full log, which I need to debug a problem we're having. |
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| 08:13.31 | jkroon | did the syntax or something change? Does this now explicitly get filtered? |
| 08:18.03 | BarthezZ | kee |
| 08:22.40 | jkroon | ok, so the syntax is now ...,verbose(<level>),... no longer simple ,verbose,. |
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| 11:40.12 | zamba | meetme has been replaced with confbridge? |
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| 11:57.49 | youjelly | hi guys, I needed some help with cdr_mysql module |
| 11:59.19 | youjelly | like right now, its just creating 1 entry for each call |
| 11:59.53 | youjelly | I have this IVR, where the user connects and then they dial a number based on which dtmf they pressed, I want to track the duration of that call, after they pressed dtmf |
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| 12:08.54 | flymol0 | hi all |
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| 12:14.03 | zamba | is there a ppa or something available for asterisk 11 in ubuntu? |
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| 12:35.18 | puzzled | zamba: both meetme and confbridge are still available in asterisk-11 so you can choose. confbridge has better performance but is more work to setup than meetme |
| 12:35.55 | youjelly | anyone help with cdr_mysql? |
| 12:37.04 | puzzled | youjelly: never ask to ask :-) just state your problem with sufficient detail. use a pastebin if you need to show more than 3 lines of text |
| 12:37.40 | youjelly | <youjelly> hi guys, I needed some help with cdr_mysql module |
| 12:37.41 | youjelly | <youjelly> like right now, its just creating 1 entry for each call |
| 12:37.41 | youjelly | <youjelly> I have this IVR, where the user connects and then they dial a number based on which dtmf they pressed, I want to track the duration of that call, after they pressed dtmf |
| 12:39.32 | [TK]D-Fender | youjelly: "core show application ResetCDR" <- |
| 12:39.50 | youjelly | thanks TK |
| 12:41.31 | youjelly | so I'd reset before dialing? |
| 12:42.27 | *** join/#asterisk FlashDel (~benedict@static-87-79-94-28.netcologne.de) |
| 12:42.35 | [TK]D-Fender | youjelly: Yes |
| 12:45.46 | FlashDel | hi folks! i got a asterisk server connected via router to a fritzbox6360 (the fritzbox got 9 numbers configured) and everything is working fine so far, but i happens a few times that i cannot call in/out because of this error message: http://pastebin.com/Ji8A1XYa |
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| 12:49.12 | FlashDel | can somebody give me a hint? i am using asterisk 1.8.1 and here is my sip.conf http://pastebin.com/BRSHxKLB |
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| 13:09.45 | [TK]D-Fender | FlashDel: Your message showed a failure for Asterisk registering and we don't see any of those in the config |
| 13:10.05 | [TK]D-Fender | FlashDel: You also aren't showing any other tests for the DNS on that box |
| 13:10.42 | [TK]D-Fender | FlashDel: if your fritzbox is on a fixed IP on your local LAN then you should jsut make a hosts entry for it |
| 13:11.22 | [TK]D-Fender | FlashDel: And update your * because you are WAY behind and it represents a large security risk. |
| 13:14.59 | FlashDel | [TK]D-Fender, i will update the box next week :-) My problem is, that it shows dns errors, but i made an entry in the /etc/hosts file and i can ping the fritzbox and also the asterisk box works for a while, then these errors occur and after several minutes, they dissapear |
| 13:17.02 | [TK]D-Fender | FlashDel: It's also showing a timeout which means it DID find an IP.. and is failing to get an ANSWER |
| 13:17.14 | [TK]D-Fender | FlashDel: And I don't see you looking at the actual attempts in there. |
| 13:19.14 | FlashDel | [TK]D-Fender, mhh so my fritzbox is the problem? |
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| 13:21.40 | [TK]D-Fender | FlashDel: Hard to say... you aren;t looking at your actual registration attempts |
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| 13:47.23 | FlashDel | can i force asterisk to NOT reregister my sip peers? |
| 13:52.05 | [TK]D-Fender | Make the place it registers to not set it to expire |
| 13:52.18 | [TK]D-Fender | Which is a bad idea |
| 13:54.14 | FlashDel | its just for today, i got a routing problem (which eventually causes the errors above) and tommorow i will replace the router |
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| 14:17.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:54.32 | Sythius | dont know if this is an asterisk question but i want to use headsets for voip talk with the asterisk server, could i use any headset and what is the minimum spec for it? |
| 14:55.34 | WIMPy | Your question is rather vague. Do you plan to connect a headset to the server? |
| 14:56.52 | Sythius | sorry i will be using softphone called XLite on a laptop. |
| 14:58.17 | WIMPy | As VOIP usually uses a 8KHz samplerate, it makes sense to use a sound adapter that supports that samplerate. |
| 14:58.48 | WIMPy | Unfortunately no vendor will tell you. You have to try it out yourself or find someone who did so already. |
| 14:59.10 | WIMPy | Many headsets that are sold for VOIP use don't. |
| 15:01.02 | BeachBall | will i hit problems putting asterisk on the new ubuntu 14.04? |
| 15:02.03 | [TK]D-Fender | BeachBall: You... probably... others... maybe not :) |
| 15:02.19 | BeachBall | hmmm |
| 15:02.21 | BeachBall | i c |
| 15:02.55 | Sythius | WIMPy: ok thanks |
| 15:03.09 | [TK]D-Fender | BeachBall: Starting by wondering if Asterisk will fail on a distro is a bad start. Satisfy its dependencies and * doesn't care what name was underneath. |
| 15:05.17 | file | Asterisk is fine on 14.04, that's what I dev on |
| 15:08.08 | BeachBall | kk |
| 15:08.14 | BeachBall | :} |
| 15:08.33 | BeachBall | moving my digium card from 1 system to another, copying all the files in asterisk folder... should work? |
| 15:08.41 | BeachBall | right |
| 15:08.41 | BeachBall | ? |
| 15:09.25 | *** join/#asterisk CeBe (~CeBe@port-92-206-38-121.dynamic.qsc.de) |
| 15:09.48 | newtonr | BeachBall, you probably need the dahdi stuff in /etc/ as well |
| 15:10.10 | BeachBall | right |
| 15:10.33 | WIMPy | It should work. If it's a PCIe card it most probaly will. With PCI you never know. |
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| 15:14.39 | BeachBall | this is so exciting |
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| 15:17.43 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:19.48 | navaismo | good morning malcolmd, is there a chance to have a sticky post in the digium forum about webrtc troubleshooting? |
| 15:22.27 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 15:26.35 | BeachBall | asterisk support webrtc? |
| 15:26.43 | BeachBall | is there how-tos? |
| 15:27.09 | navaismo | is that a joke? |
| 15:27.26 | BeachBall | >:( |
| 15:27.28 | BeachBall | no |
| 15:27.41 | malcolmd | navaismo: sure. if you write something up, i'll be happy to sticky it. just let me know. :) |
| 15:28.02 | navaismo | malcolmd, ok i have some post about that... |
| 15:28.11 | malcolmd | navaismo: cool |
| 15:28.17 | navaismo | BeachBall, yes is supported check the asterisk wiki |
| 15:28.25 | MaliutaLap | does not want to know about navaismo and anything "sticky" ;) |
| 15:28.35 | navaismo | AHAHAHAHA LMAO |
| 15:29.11 | MaliutaLap | navaismo: I'll be here all week. Try the beef. ;) |
| 15:29.42 | navaismo | we have beef in the channel? |
| 15:30.07 | MaliutaLap | navaismo: there's lots of meat around |
| 15:30.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 15:30.34 | navaismo | eeewwww |
| 15:30.35 | MaliutaLap | navaismo: look - fresh meat ;) |
| 15:30.59 | navaismo | i imagine the worst |
| 15:31.18 | MaliutaLap | you mean wurst? |
| 15:31.41 | navaismo | vomit |
| 15:31.46 | navaismo | NEIN |
| 15:31.49 | MaliutaLap | trust me to make a meat joke into a sausage joke :) |
| 15:32.13 | MaliutaLap | navaismo: Juh ... du hast das wurst! |
| 15:32.25 | MaliutaLap | a/Juh/Jah/ |
| 15:32.37 | MaliutaLap | wow can't type ... must need sleep |
| 15:32.49 | navaismo | what is your TZ |
| 15:32.58 | MaliutaLap | UTC+10 |
| 15:33.03 | MaliutaLap | it's 01:30 |
| 15:33.12 | navaismo | :S |
| 15:34.17 | MaliutaLap | hmm, should we teach infobot about meat and wurst? |
| 15:34.23 | navaismo | no no |
| 15:34.32 | navaismo | too much "gayness" |
| 15:34.55 | MaliutaLap | navaismo: your text says "no, no" - but you're eyes say "yes, yes" ;P |
| 15:35.04 | navaismo | You cant see me |
| 15:35.18 | MaliutaLap | navaismo: wave into the webcam ;P |
| 15:35.28 | [TK]D-Fender | Oktoberfest: a competition to find the best wurst. |
| 15:35.44 | navaismo | for food sound great |
| 15:35.54 | navaismo | but for MaliutaLap's context NEIN |
| 15:36.10 | MaliutaLap | I didn't provide context |
| 15:36.15 | navaismo | ¬¬ |
| 15:36.50 | MaliutaLap | navaismo: if you thought it was something dirty it's your mind that's messy, not mine ;P |
| 15:37.12 | navaismo | ¬¬' |
| 15:39.11 | MaliutaLap | navaismo: I'm pretty sure "wurst" doesn't match in any of my dial plan contexts :) |
| 15:39.21 | MaliutaLap | navaismo: nor does "meat" |
| 15:39.43 | *** join/#asterisk ApPrOaCh (~sgadgil@199.255.40.83) |
| 15:40.53 | navaismo | no more word from my side |
| 15:40.59 | navaismo | i feel abused |
| 15:41.23 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:42.15 | MaliutaLap | navaismo: we aim to please ;) |
| 15:42.53 | navaismo | :'( stop it it hurts |
| 15:46.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:46.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 15:51.18 | ApPrOaCh | Not sure if some can help me to get a pointer for asterisk Manager API interface with PHP/PERL code |
| 15:52.23 | marceloamorim | guys, I'm wondering if I can configurate the asterisk to detect the callerid before the signalling, I'm from Brazil, and the callerid comming before the signalling |
| 15:56.18 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:56.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 15:59.11 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:59.24 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 15:59.33 | navaismo | ApPrOaCh, ~ask |
| 15:59.40 | navaismo | ~ask |
| 15:59.40 | 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. |
| 16:00.33 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 16:01.15 | ApPrOaCh | Apologize for that, sure will take care in future |
| 16:01.30 | navaismo | marceloamorim, https://issues.asterisk.org/jira/browse/DAHLIN-4 |
| 16:02.14 | marceloamorim | navaismo: I found this too http://pastebin.com/RaRzkYhQ |
| 16:02.22 | marceloamorim | it seems like a driver |
| 16:03.43 | marceloamorim | ohh, nice navaismo |
| 16:03.45 | navaismo | a patch for chan_dahdi but chech the version seems like used for asterisk 1.6 |
| 16:04.19 | marceloamorim | do you know how to apply those patchs? |
| 16:04.26 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:04.29 | *** join/#asterisk litn (~blice@alrig.ht) |
| 16:05.03 | litn | hey guys, having a music on hold issue... I'm using sln files for music on hold and asterisk shows no warnings or anything in the logs. It says it starts the hold music, ends the hold music, normal log, but the phone is just silent |
| 16:05.07 | litn | any ideas on what I can check next ? |
| 16:05.28 | navaismo | marceloamorim, what version are you using? |
| 16:06.00 | marceloamorim | asterisk 11.6-cert2 |
| 16:07.13 | navaismo | hmm those patches are for 1.6, its supposed that v > 1.6 must have it or I'm wring as usual? |
| 16:07.19 | navaismo | s/wring/wrong/ |
| 16:08.12 | navaismo | marceloamorim, to apply the patch you need to use the linux patch tool and run like patch -p0 < patchfilename |
| 16:09.15 | navaismo | ApPrOaCh, the intention of my message was for make you asking a specific question not to be an ass, sorry |
| 16:10.23 | ApPrOaCh | not an issue, I am new to the forum, so learning the way I can go along |
| 16:10.35 | marceloamorim | I'll check this command and try to found which file I need apply that diff |
| 16:10.45 | marceloamorim | find* |
| 16:11.20 | navaismo | marceloamorim, do it in a test machine not in production |
| 16:13.01 | navaismo | and marceloamorim did you already tried the others values for cidstart setting in your chan_dahdi.conf? |
| 16:13.39 | ApPrOaCh | Just to give background, I have installed asterisk and I came to know that we can interface with it via AMI and I see its having good potential, so wondering if some one may already has PHP/Perl code which interface with AMI, so trying to avoid reinventing the wheel |
| 16:13.51 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:13.59 | marceloamorim | ok, I'm working on machine to be an production machine, this is a feature, I don't intend to work too much days on that, I almost on the line to put on production |
| 16:15.12 | marceloamorim | I tried on my asterisk, but now I change the digium tdm800p for a khomp, and khomp was made on brazil, then works pretty well |
| 16:15.30 | navaismo | ApPrOaCh, http://www.voip-info.org/wiki/view/Asterisk+manager+API the see also section has some info |
| 16:15.46 | navaismo | for php many users use phpagi |
| 16:21.18 | marceloamorim | navaismo, anyway man, you opened my mind with those comments and url |
| 16:21.22 | marceloamorim | ty so much |
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| 16:28.34 | ApPrOaCh | thx I went through that link, I will search for key word phpagi, thx for pointer |
| 16:31.32 | [TK]D-Fender | ApPrOaCh: You were asking about perl. You should be searching for "perl asterisk AMI class" |
| 16:32.07 | ApPrOaCh | ah nice, sure will take a look for that class |
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| 17:03.04 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:03.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 17:11.19 | file | PEOPLE! |
| 17:11.53 | BeachBall | oh thats me |
| 17:11.55 | BeachBall | i'm people |
| 17:12.05 | file | hello. |
| 17:12.07 | BeachBall | Hi |
| 17:12.10 | BeachBall | can I help you? |
| 17:12.18 | file | potentially. |
| 17:12.21 | BeachBall | oh boy |
| 17:12.58 | file | but the questions is for everyone: is there something about Asterisk/in it that you'd like further explanation on in a blog post or something about VoIP in general? |
| 17:13.25 | BeachBall | YES! |
| 17:13.47 | BeachBall | I would like a 3 min video from each of the top dogs, about them selves |
| 17:13.56 | BeachBall | I think it would be nice to see them |
| 17:14.06 | BeachBall | :} |
| 17:14.08 | file | that's random. |
| 17:14.38 | file | I'm pondering a blog post |
| 17:14.55 | [TK]D-Fender | No, that's "top dogs". He didn't ask for "3 random employees" :) |
| 17:15.09 | WIMPy | There are open documentation issues. |
| 17:15.13 | file | [TK]D-Fender, I could still grab 3 random people... |
| 17:19.47 | file | [TK]D-Fender, you are now the chan_iax2 expert! congrats |
| 17:20.25 | wdoekes | file: rtp and rtcp and how it interacts with asterisk |
| 17:20.36 | file | wdoekes, ooh |
| 17:21.11 | file | wdoekes, nice one! |
| 17:21.14 | file | WIMPy, what's the issue numbers? |
| 17:23.11 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:23.47 | WIMPy | ASTERISK-23314 |
| 17:25.04 | WIMPy | But there are a few. |
| 17:25.27 | file | nods |
| 17:26.08 | file | nobody has gotten to it yet |
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| 17:33.53 | file | wdoekes, anything else Mr. Doekes? |
| 17:35.19 | drmessano | file: What about an up to date blog we can link explaining NAT+SIP+Asterisk using the NEWER NAT parameters in both the context of 11 and 12 |
| 17:35.29 | file | drmessano, added to list! |
| 17:37.37 | drmessano | What about a blog post address why Google Voice won't work anymore and how Google are a big bag of di... NAH. |
| 17:38.25 | file | tempting |
| 17:39.45 | WIMPy | Don't forget to save a copy with the Skype title then. |
| 17:40.48 | drmessano | I would love to see more small blog posts in general from the devs... little braggy things about big bugs that were squashed or "I rewrote this little bit of code and now ___ is 200x faster. This is so cool". |
| 17:41.09 | drmessano | I know you can "write code or write documentation/blog posts" but I kinda miss seeing those floating around |
| 17:41.21 | file | drmessano, noted! |
| 17:43.38 | drmessano | file, in general, us enthusiasts and fanboys want more to brag about with the project, in smaller ways than "Oh yeah, theres a new SIP stack, its really cool. Wonder what will be in there next year".. because you guys do SO much more than that and surely have lots of those moments like I described above. we just never hear about them |
| 17:44.07 | file | yup yup! |
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| 17:47.35 | navaismo | webrtc pro/cons, myths and realities, and troubleshooting |
| 17:47.49 | file | that would make me bitter |
| 17:47.56 | navaismo | you are bitter |
| 17:48.03 | litn | Hello... I am trying to get hold music to play.. I have tried every format etc., when I use a bad format I see it in the logs, but good format I just see that it's supposedly playing, but I don't hear anything. |
| 17:48.07 | litn | any ideas ? |
| 17:48.24 | [TK]D-Fender | litn: Show us. |
| 17:48.54 | litn | ah, I just discovered that it seems hold music does play internally from one ext to another |
| 17:48.59 | litn | only from the outside it isn't playing |
| 17:49.03 | litn | is that revealing at all? |
| 17:49.18 | [TK]D-Fender | Look at the call |
| 17:50.11 | navaismo | file what about a developing a new custom module for asterisk |
| 17:50.42 | WIMPy | Oh, wihile we are on that subject... |
| 17:51.06 | WIMPy | file: Can pjsip send out hold information? |
| 17:51.09 | litn | [TK]D-Fender: http://pastebin.com/Br88qXaw |
| 17:51.32 | file | WIMPy, the functionality does not currently exist but it would be extremely easy to add it |
| 17:52.06 | [TK]D-Fender | litn: We know nothing about your networking and we can't see what started the request for MoH, etc. You need to actually show us all the relevant bits |
| 17:52.16 | WIMPy | Ok, and now the bad question: What about outgoing overlap? |
| 17:52.36 | file | probably not THAT bad |
| 17:53.20 | litn | [TK]D-Fender: Ok hold on |
| 17:57.22 | litn | [TK]D-Fender: Here's the whole log, it's calling into a ring group which is supposed to play this hold music, |
| 17:57.25 | litn | http://pastebin.com/5EBDFA6M |
| 17:57.41 | litn | it does say that it is playing the hold music, but I don't hear anything (calling in from cellphone) |
| 18:00.03 | [TK]D-Fender | litn: how a working call. |
| 18:00.06 | [TK]D-Fender | show* |
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| 18:03.37 | WIMPy | file: Does the i extension work on pjsip? |
| 18:04.15 | file | 'i' is core functionality |
| 18:04.47 | WIMPy | But witch chan_sip it doesn't work. |
| 18:04.52 | WIMPy | with |
| 18:06.21 | file | specify what you mean by "work" |
| 18:06.28 | file | ie: scenario |
| 18:06.57 | WIMPy | It is never hit. Calls to invalid extensions never hit the dialplan. |
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| 18:07.38 | file | that's done on purpose as a 404 is sent back as a response instead |
| 18:08.23 | WIMPy | Would only make sense to me if no i extension was found. |
| 18:08.48 | file | the 'i' extension is used if you are in an IVR and someone enters an invalid choice |
| 18:09.16 | WIMPy | Other channels use it for the initial dial as well. |
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| 18:09.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:10.06 | file | what ones? |
| 18:11.13 | WIMPy | Doesn't dahdi? |
| 18:13.17 | litn | ok [TK]D-Fender |
| 18:13.19 | litn | http://pastebin.com/36Mw0F40 - WORKING moh, internal call to ring group. |
| 18:13.19 | litn | http://pastebin.com/rYMDY8k8 - Not working moh, external call to ring group. |
| 18:13.42 | litn | sorry it took so long, I made both the same moh class and stuff so that there was as little difference as possible for you to parse :) |
| 18:13.57 | [TK]D-Fender | [Apr 24 14:09:21] VERBOSE[17420][C-00000004] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/Flowroute-0000000a |
| 18:14.02 | [TK]D-Fender | One is NOT the same class |
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| 18:14.13 | litn | they are both default I thought |
| 18:14.14 | litn | hm |
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| 18:35.48 | litn | [TK]D-Fender: the confusion was that there is hold music for the incoming route and also the ring group and one was set different, I have it consistent now and still not working, pasting a new one |
| 18:37.05 | litn | http://pastebin.com/36Mw0F40 - WORKING moh, internal call to ring group. |
| 18:37.06 | litn | http://pastebin.com/ruudXae1 - Not working moh, external call to ring group |
| 18:41.58 | BeachBall | how to dial out from console? |
| 18:42.00 | BeachBall | for test |
| 18:45.33 | litn | [TK]D-Fender: seems about the same now, I see the same besides the original call in stuff |
| 18:48.09 | navaismo | BeachBall, channel originate or just originate |
| 18:48.16 | WIMPy | BeachBall: 'channel originate ...' or if you have it, 'console dial ...'. |
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| 18:55.36 | litn | ah [TK]D-Fender I just realized that no audio is going in/out from external, haha |
| 18:56.14 | BeachBall | i was successful |
| 18:57.21 | pabelanger | ~itsp-ca |
| 18:57.24 | pabelanger | ~itsp |
| 18:57.24 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:57.29 | pabelanger | ~itsplist-ca |
| 18:57.29 | infobot | rumour has it, itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 18:58.52 | pabelanger | ~itsplist-us |
| 18:58.53 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 21:01.55 | litn | hmm.. I am trying to get feature codes to work. When I put a phone on DND by dialing the feature code, the extension goes on dnd inside asterisk but the phone doesn't show it |
| 21:02.19 | litn | also if I press the dnd button on my phone it doesn't show in asterisk, it shows as "away" instead of dnd |
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| 21:08.06 | newtonr | litn, define " goes on dnd inside asterisk" |
| 21:12.24 | litn | newtonr: sorry, it shows a red circle on the extension in the flash operator panel |
| 21:14.30 | litn | so that's if I do it via feature code, which does not turn the dnd light on on the phone but does turn it on in flash operator |
| 21:14.48 | litn | and then on the other hand if I hit dnd on the phone the dnd light turns on, but the ext is unaffected in fop |
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| 21:17.09 | newtonr | litn, the flash operator panel is not Asterisk. It is code written on top of Asterisk and other technologies. Are you using a GUI to administrate Asterisk, maybe FreePBX ? |
| 21:17.31 | litn | maybe :( |
| 21:18.01 | litn | so- feature codes and interaction with the phones in this way, that's all on top of asterisk and not in asterisk? |
| 21:18.43 | newtonr | Are you using a web page to administrate Asterisk that looks something like these screen shots: http://tinyurl.com/k7bpn93 ? |
| 21:19.46 | litn | yes |
| 21:20.10 | newtonr | litn, It is a mix of both, you have to get real specific to know for sure. If you are not administrating Asterisk at a ground level then it could be a lot of work for you to find out. Most people in this room don't use a GUI like you are using, so they can't help you out as well. |
| 21:20.25 | newtonr | litn, head over to #freepbx and they may be able to help you better there. |
| 21:20.32 | litn | ok will do, thanks! |
| 21:21.01 | lvlinux | flash op panel probably uses ami to "ask" asterisk stuff |
| 21:23.04 | lvlinux | who knows lol... |
| 21:23.35 | newtonr | it is all smoke and magnets |
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| 21:25.54 | litn | I'm thinking maybe it's in the phone itself |
| 21:26.01 | litn | like maybe I can configure it to send a certain feature code? |
| 21:26.03 | litn | if dnd is pressed |
| 21:26.42 | lvlinux | litn: yes i was about to mention that I think it depends on the phone. on some of them I think that the DND button just shuts off the ringer and makes it not respond to SIP invites or something |
| 21:27.51 | lvlinux | if you can configure the button to do a speeddial you can do it in your dialplan, but I have no clue how you would do that with FreePBX. |
| 21:28.02 | lvlinux | More info here: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/usingCustomDeviceStates.html |
| 21:28.37 | navaismo | litn, DND from Freepbx only is visible by freepbx, DND on phone is set on the phone and it wil send you a busy back here when asterisk call it |
| 21:29.18 | litn | yeah, it returns busy, right |
| 21:31.45 | [TK]D-Fender | [17:25]litnif dnd is pressed <- This is PHONE SPECIFIC |
| 21:32.00 | [TK]D-Fender | the phone has to have that option |
| 21:34.01 | litn | the option to send something to the pbx? |
| 21:34.25 | [TK]D-Fender | Correct |
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| 22:14.26 | smkelly | hi |
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| 22:58.59 | c|oneman | anyone use didLogic? |
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