| 00:00.28 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 00:03.48 | [TK]D-Fender | You're entitled to your opinion. Consider the possiblity that your perception isn't perhaps average, and then that it is not the message I'm conveying |
| 00:03.52 | [TK]D-Fender | Do with it what you will. |
| 00:04.39 | [TK]D-Fender | but add something constructive instead of just a claim of my intent. |
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| 01:00.32 | UncleKiwi | hey there, what does this mean |
| 01:00.36 | UncleKiwi | main-00000028 requested media update control 26, passing it to SIP/1023-00000027 |
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| 06:55.52 | imihaylov | Hello! Does someone know how to block caller which number starts with + (plus) |
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| 07:12.57 | imihaylov | Hello! Does someone know how to block caller which number starts with + (plus) ? |
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| 07:23.14 | ChannelZ | Look at ${CALLERID(num):0:1} to see if it's a + and behave accordingly |
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| 07:27.54 | ChannelZ | like maybe GotoIf($[${CALLERID(num):0:1}=+]?hanguponthem,1) and then have an extension called 'hanguponthem' that does something interesting |
| 07:32.20 | ChannelZ | actually come to think of it the + will probably confuse the parser... you probably have to quote it.. |
| 07:32.44 | ChannelZ | GotoIf($["${CALLERID(num):0:1}"="+"]?hanguponthem,1) |
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| 07:48.05 | *** join/#asterisk HouseMD (~HouseMD@mesi-yar.ru) |
| 07:48.33 | HouseMD | ÐобÑÑй денÑ! ÐÑоÑÑ Ð¿Ð¾Ð¼Ð¾Ñи по Asterisk+e1 |
| 07:49.40 | HouseMD | плаÑа Wildcard TE220 (4th Gen), ÑÑаÑÑÑ ÐÐ |
| 07:50.53 | HouseMD | пÑи звонке пиÑÐµÑ Ð¾ÑибкÑ: Unable to create channel of type 'DAHDI' (cause 34 Circuit/channel congestion) |
| 07:51.02 | HouseMD | Everyone is busy/congested at this time (1:0/1/0) |
| 07:52.31 | HouseMD | пÑокидÑваем звонок Ñак: Dial(DAHDI/g2/${EXTEN},60,TK); |
| 07:52.41 | HouseMD | на вÑоÑой Aster |
| 07:58.14 | HouseMD | in english ? |
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| 12:15.13 | pawiecki | Hello, I want to collect CLI output from asterisk on a remote server to analyze problem with calls. Problem is i don't have screen or tmux installed there (can't install anything). I'm not familiar with asterisk logging. How can i collect CLI output for a fixed time, like 3 days? |
| 12:16.10 | [TK]D-Fender | pawiecki: logger.conf <--- |
| 12:16.32 | [TK]D-Fender | Ther is no normal way to capture CLI itself, but you can log the parts yuo care about |
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| 12:37.20 | pawiecki | [TK]D-Fender: thx, got it working now. |
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| 13:33.24 | Rico | hi |
| 13:33.33 | Rico | I'm still having a problem with asterisk moh |
| 13:33.45 | Rico | first time the caller is put on hold works fine |
| 13:34.06 | Rico | seconds time, caller does not hear anything (no rtp stream sent from asterisk to caller) |
| 13:35.32 | Rico | don't know where to look for |
| 13:35.52 | Rico | I've got the same problem with to "local" phones (without nat) |
| 13:35.55 | robmal | The second time he isn't put on hold. |
| 13:36.54 | robmal | He can hum some tune if he's bored. |
| 13:37.17 | Rico | RobertLaptop: http://pastebin.com/YjqzCeBw |
| 13:38.12 | robmal | This is from the 'no moh' call? |
| 13:38.39 | Rico | yes |
| 13:40.57 | Rico | robmal: I think that's strange that line 17 comes just after line 16 |
| 13:42.30 | robmal | No, it has to bridge moh with the sip channel, but why it stops playing 10s later is weird. |
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| 14:13.59 | leftleg | hello all |
| 14:14.51 | file | hi |
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| 14:23.12 | leftleg | i have 3 asterisk boxes sip trunked. my main pbx has a sip provider. how could i allow/route the other two asterisk boxes to make calls to the pstn using the sip provider on the main pbx? |
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| 14:23.59 | leftleg | i tried google searching, but i'm not sure what i'm searching for.... or the correct words to search for. |
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| 14:24.29 | WIMPy | In that case you might want to start with the book. |
| 14:24.34 | WIMPy | ~book |
| 14:24.34 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:24.39 | WIMPy | or |
| 14:24.41 | WIMPy | ~primer |
| 14:24.41 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 14:24.57 | leftleg | i will try that, thank you |
| 14:26.10 | WIMPy | And if you get some basic understanding, the question hopefully just answes itself :-) |
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| 15:00.42 | polysics | hello! |
| 15:01.04 | polysics | question for the devs: does `_.` catch `h` now? |
| 15:01.13 | polysics | now as in 12+ |
| 15:01.53 | WIMPy | It always did. |
| 15:02.29 | WIMPy | That's why you shouldn't use it. |
| 15:05.55 | polysics | hmm, there is some slightly different behavior, but maybe I am seeing things |
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| 15:52.47 | CptBurger | Has anyone else had trouble detecting DTMF from Sprint mobile phones? |
| 15:53.08 | CptBurger | Seems to only happend when its a Sprint mobile. |
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| 16:06.15 | AndyML | greetings all. I have a ConfBridge user reporting that idle conference bridge users are hearing conf-muted.gsm after 30 minutes in the bridge, and are not able to use *1 to unmute themselves. This is Asterisk 11.16.0 |
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| 16:38.29 | *** join/#asterisk litn (~blice@alrig.ht) |
| 16:38.52 | litn | hello. I use the AMI to change the music on hold class of certain channels, basically depending on what kind of customer they are in our database |
| 16:39.08 | litn | if I change it immediately when I get the new channel event, it works, but if I wait a couple seconds or it takes too long, it does not work |
| 16:39.36 | litn | I check the channel variables in the asterisk cli and it did change them, but it seems that if the call is already connected and going then asterisk doesn't re-read them when I put them on hold |
| 16:41.48 | mjordan | litn: the music on hold class is checked on the channel when MoH is first started on the channel. If something sets it after that, it won't be re-read. It is, however, definitely checked when the MoH app is started on the channel; if you're seeing something else, than I'm not sure how you're producing that behaviour. |
| 16:46.35 | litn | mjordan: so I put in a sleep for 3 seconds before I issue the Setvar command |
| 16:46.40 | litn | and when I do that, the moh does not change |
| 16:46.45 | litn | if I take out the sleep, then it does change |
| 16:46.53 | litn | in both cases the variables look correct in the console |
| 16:48.23 | litn | and in the full log it shows it playing the default moh in the sleep one and in the one where I set it as soon as the channel is created it's correct |
| 16:48.26 | litn | any ideas? |
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| 16:54.05 | mjordan | nope. Frankly, looking at the code, the StartMusicOnHold application evaluates the MoH on the channel. If you're experiencing something different, than I'm not sure what you're doing or why your system would have a different effect. |
| 16:54.44 | mjordan | there is nothing in Asterisk that at some point in time says, "don't evaluate the MoH property on the channel" |
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| 16:56.00 | litn | you had said before that maybe if moh is called before I set it- having looked at the code just now, does it read moh class each time? |
| 16:56.05 | litn | so it should work either way? |
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| 17:06.21 | Geek-Linux | Hi: My asterisk SIP box is behind the firewall. during the call when user hangsup BYE doesnt receive on the box. and the call goes to infinity. thanks in advance. |
| 17:09.01 | [TK]D-Fender | Show the actual call |
| 17:09.14 | [TK]D-Fender | Descriptions don't prove what's happening. |
| 17:21.48 | litn | mjordan: well, how do you think I should move forward debugging this? could it be my version of asterisk? |
| 17:29.19 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 17:40.01 | CptBurger | litn: when you say sleep, do you mean wait? |
| 17:45.42 | *** join/#asterisk Bouldr (47fb5705@gateway/web/freenode/ip.71.251.87.5) |
| 17:48.46 | litn | CptBurger: I mean, the program waits, yeah |
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| 17:48.52 | litn | it's written in python which uses time.sleep |
| 17:51.50 | Bouldr | I'm having trouble with inbound calls on only one of our extensions (remote location) - whenever they try to answer calls I get a busy signal, and their phone acts as if the call is already hung up. Here's a sip debug log: http://pastebin.com/pMfmvx9W |
| 17:51.58 | Bouldr | Any ideas? |
| 17:52.28 | Bouldr | I also have a sip debug log of a similar extension where the call is actually able to be answered, if applicable |
| 17:57.07 | CptBurger | litn: your time.sleep sounds like the problem. I'm not an asterisk dev and they may be able to speak more to this but it sounds like you put the thread to sleep and Asterisk goes ahead and reads the channel variable for MOH. Asterisk is multi-threaded and doesn't wait until the sleep is over to read the variable so thats why it doesnt work. You come back from sleep after Asterisk has started |
| 17:57.07 | CptBurger | the MOH? |
| 17:57.28 | *** part/#asterisk mjordan (mjordan@nat/digium/x-pdjmehsfsojjweqc) |
| 17:58.02 | CptBurger | litn: or the sleep is messing with Asterisk ability to read the channel. Is there an alternative to sleep? |
| 17:59.42 | litn | CptBurger: Oh sorry- I should have clarified I guess, I am just using the sleep to debug the issue, |
| 18:00.04 | litn | the actual blocking event that occurs is I'm looking up the customer in the database, to see which hold music I should play, and this takes a second |
| 18:00.20 | litn | if I take out the database call and instead just put a condition, it plays instantly, without the lag of the db lookup |
| 18:00.37 | litn | so what I did was I put a sleep(2) first to make sure it wasn't the db connection itself but rather just the amount of time it was taking to do the lookup |
| 18:00.40 | litn | does that make sense? |
| 18:00.57 | litn | so I was just confirming that if I wait a moment before setting the mohclass variable, it does not work, but if I set it instantly as I get the newchannel event it does work. |
| 18:03.33 | litn | CptBurger: the other part that maybe wasn't clear to you is that I am doing this over AMI with a program that is running on another server |
| 18:03.40 | litn | so the sleep() on the program should not affect asterisk |
| 18:03.49 | litn | besides sending the setvar command for moh later |
| 18:07.38 | CptBurger | ltin: ok, interesting. Sounds like an AMI problem... |
| 18:10.39 | CptBurger | Bouldr: [2015-06-08 13:10:21] WARNING[18161][C-0000001f]: chan_sip.c:23081 handle_response_invite: Received response: "200 OK" from '800' without SDP |
| 18:11.40 | CptBurger | Bouldr: thats where the remote side gets hung up. |
| 18:12.20 | CptBurger | Bouldr: I've had terrible experiences with Grandstream devices. I've tried them several times and always get burned. |
| 18:12.52 | Bouldr | CptBurger: You are correct... I was just going back and forth and that definitely is it - any ideas? All of our locations use the same ones, and outbound calls work fine |
| 18:14.13 | CptBurger | bouldr: Post a successful call sip debug and compare the 200 OK responses maybe? |
| 18:14.28 | CptBurger | A successful call to another Grandstream |
| 18:14.32 | Bouldr | http://pastebin.com/VVhMbq1E |
| 18:14.50 | Bouldr | This is a successful call to another location using the same grandstream dp715 |
| 18:16.36 | CptBurger | bouldr: but now your old past is gone... |
| 18:16.39 | CptBurger | paste |
| 18:17.18 | litn | CptBurger: I'm not sure it's a problem with AMI. I'm changing the mohclass variable, right- when I check the variables on those channels in the asterisk console, they have been changed. But the moh is still playing default. |
| 18:17.40 | litn | just not sure where to even begin debugging this now. |
| 18:18.04 | Bouldr | CptBurger: Oi, Sorry about that - I had it expire in an hour. Here ya go: http://pastebin.com/ZFqtQqXP |
| 18:18.11 | CptBurger | litn: hrm.. how low can you sleep before it works? 1 second? |
| 18:18.26 | litn | CptBurger: good question, let me check |
| 18:20.17 | *** part/#asterisk vaskozl (~vaskozl@unaffiliated/einherjar) |
| 18:22.03 | CptBurger | Bouldr: the 200 OK response is malformed on the non-working one. It reports the content length to be 2450, which is incorrect, and it is also missing the first line v=0. The Firmware on the two is different. The working one reports 1.0.0.8 and the non-working one reports 1.0.0.23. I'll bet updating the firmware to 1.0.0.8 will fix it. |
| 18:22.35 | *** join/#asterisk exuberocity (~exuberoci@cpe-174-101-43-157.columbus.res.rr.com) |
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| 18:22.53 | Bouldr | CptBurger: Thanks, I'll give that a shot now |
| 18:24.36 | CptBurger | litn: If 1s doesn't work, try .01 or .0001. I'm curious if it is the sleep that is doing this. |
| 18:25.01 | CptBurger | litn: go as low as you can |
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| 18:27.00 | litn | CptBurger: anything 0.04 or greater plays default music |
| 18:27.11 | litn | 0.03, 0.02, 0.01 play the hold music I set |
| 18:30.52 | litn | CptBurger: in the asterisk log, when the call is first made, it runs through the dialplan from the conf and applies them, it looks like, |
| 18:31.13 | litn | [2015-06-08 14:25:54] VERBOSE[12114][C-00003c94] pbx.c: -- Executing [@from-internal:2] Set("SIP/442-0000b3c3", "MOHCLASS=default") in new stack |
| 18:31.42 | litn | so I'm wondering if maybe if I don't set moh before this happens, it plays default? |
| 18:32.46 | litn | doesn't make sense, you would think the opposite would happen, that if I set mohclass before this, this would overwrite it |
| 18:33.44 | CptBurger | litn: so you have 0.04 seconds for you DB lookup. Problem solved! :D |
| 18:33.58 | litn | lol |
| 18:34.30 | litn | it happens again a bit later down, that it sets setmusic, |
| 18:34.32 | litn | [2015-06-08 14:25:54] VERBOSE[12114][C-00003c94] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/442-0000b3c3", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack |
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| 18:38.18 | Bouldr | CptBurger: Just updated firmware to 1.0.0.33 - same error "[2015-06-08 14:36:13] WARNING[18536][C-00000035]: chan_sip.c:23081 handle_response_invite: Received response: "200 OK" from '800' without SDP" |
| 18:48.54 | Bouldr | CptBurger: I just noticed, the one with the higher firmware was the one that wasn't working |
| 18:49.34 | litn | Bouldr: see if you can find the exact firmware of the one that is working just to rule out the variable |
| 18:53.51 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 19:03.31 | Bouldr | CptBurger: It appears the manufacturer has their firmware dating messed up, 1.0.0.8 is actually one of the first versions - and 1.0.0.33 is the latest. |
| 19:04.26 | Bouldr | CptBurger: Anyhow - what is generating the incorrect 200 repsonse? the phone itself? |
| 19:16.09 | CptBurger | Bouldr: yes, that SIP response comes from the phone. |
| 19:16.39 | CptBurger | Any chance you can downgrade the phone to test it? |
| 19:17.11 | Bouldr | CptBurger: unfortunately I can't find the old firmware anywhere - the 1.0.0.8 version |
| 19:18.38 | CptBurger | You could also update a phone from 1.0.0.8 to 1.0.0.33 but you risk breaking that phone... Oh! Can you back up the firmware on one with 1.0.0.8 and use that backup on the other? |
| 19:19.13 | *** join/#asterisk ketas (~ketas@226-209-191-90.dyn.estpak.ee) |
| 19:24.05 | CptBurger | Are future Editions of Asterisk: The Definitive Guide going to be made freely available online or is the 4th Ed the end of that? |
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| 19:32.52 | [TK]D-Fender | CptBurger: We're only psychic on Tuesdays, sorry.... |
| 19:33.20 | Bouldr | CptBurger: any idea how I could snag the firmware off the device? |
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| 20:13.32 | wanda_ | hullo, 17th june VoIP open source fan organize a meeting with Mumble to speak about Asterisk et VoIP open source http://barcamp.org/w/page/97007235/BarCampMumble-VoIP-OpenSource |
| 20:13.59 | wanda_ | https://twitter.com/v_dagrain/status/606483428402429952 the news in english :D |
| 20:14.24 | wanda_ | http://www.asterisk-france.org/content/75-MAJ-avec-compte-rendu in french - thanks to share - message over :D |
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| 20:24.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 20:30.13 | Bouldr | CptBurger: Any ideas or should I just chalk it up as a phone issue and buy a new one? |
| 20:31.41 | CptBurger | Bouldr: sorry i was away, in the web interface would be the way to find and backup the existing firmware. I'm not familiar with that exact device. |
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| 20:32.56 | Bouldr | CptBurger: I had checked, the only option available was backup the config settings - and really no luck searching the internet |
| 20:33.10 | Bouldr | For the old firmware, anyway |
| 20:34.26 | CptBurger | Bouldr: yeah.. idk, you can look around and verify that all your working phones are on 1.0.0.8. If you find working phones on the same .23 or .33 firmware, then its not the case. But if it is, then maybe you start a support request through Grandstream. |
| 20:35.33 | CptBurger | But otherwise it may be a defective phone? Its hard to beleive that this error is due to a defective phone, it seems so much like software but hey, computers are magic! |
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| 20:55.28 | Bouldr | CptBurger: Ok, confirmed not the firmware. Found another working phone that has 1.0.0.23 - the same version the broken one had originally |
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| 21:02.57 | CptBurger | bouldr: it really seems like a defective phone then, verify by swapping it with a know working phone. |
| 21:03.55 | CptBurger | Bouldr: If that doesn't do it, then something is messing with your packets and thats weird. |
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| 21:46.09 | karlfife | Does anybody know if the Digium 1TE235F also uses the wct4xxp driver like the TE220? |
| 21:46.40 | karlfife | IOW, can they be used interchangably with identical configuration? |
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| 21:47.20 | karlfife | It appears the TE220 is EOL, and the 1TE235F replaces it. |
| 21:47.30 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 21:53.58 | malcolmd | negative, it uses a new driver |
| 21:54.18 | malcolmd | the configuration is the same, iirc, but you can't use the older wct4xxp driver w/ the te235 |
| 22:00.31 | *** part/#asterisk mjordan (mjordan@nat/digium/x-kvupgktmmpgpfenl) |
| 22:01.11 | karlfife | Got it. Thanks Malcom |
| 22:01.31 | karlfife | for that, I could just call out wct4xxp AND the new driver in /etc/dahdi/modules |
| 22:01.34 | karlfife | no? |
| 22:01.59 | karlfife | Any reason to prefer the new card? |
| 22:02.03 | malcolmd | yes, i think that'd be fine. if the other card isn't present, any system.conf lines for dahdi wouldn't get applied to a non-present card |
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| 22:05.38 | malcolmd | new card's low profile, so that's nice. new card's also currently manufactured, so that's nice too. old card's taken out of rotation, so if we ever had to replace a te220 during warranty, it could get replaced by a te235 anyway. if you're buying new, buying the newer card makes more sense, as you wouldn't have to worry about that. as far as performance is concerned, i dunno, i was off of cards (i'm phones) when they came into being. s |
| 22:08.05 | malcolmd | new cards also do easier field upgrading of firmware, so that's handy as well |
| 22:09.24 | file | malcolmd is now known as "Bringer of dialtone in phones" |
| 22:09.37 | malcolmd | ohm, ohm, ohm |
| 22:12.50 | karlfife | :-) |
| 22:13.36 | karlfife | Great info I appreciate it. |
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| 23:15.44 | deraps | is it possible to do a custom(or is a prepackaged other language) for the VM prompts? (specifically the "The person at extension" and "when done, press pound or hang up") TIA! |