| 00:00.04 | WIMPy | indeed |
| 00:00.14 | WIMPy | What about the CPU? |
| 00:00.17 | ChannelZ | Though I don't think albania-asterisk ever said what codec he was using |
| 00:00.33 | albania-asterisk | yes i said :) |
| 00:00.42 | ChannelZ | I just missed it then |
| 00:00.53 | WIMPy | AOL |
| 00:01.36 | albania-asterisk | most of the calls are p2p , but small part around 5% are transcoding from g729 to g711 or vice versa, on p2p around 60% is g711 and rest g729 |
| 00:01.40 | ChannelZ | But in any case is something in the chain maybe just not able to keep up with the amount of packets it's having to analyize to route? A cheap-ass router or something |
| 00:02.12 | WIMPy | Quite possible. |
| 00:02.30 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:02.48 | albania-asterisk | you didn't tell me , how many concurrent calls should i think to have one asterisk box for production |
| 00:02.50 | albania-asterisk | on max |
| 00:03.14 | ChannelZ | I'm not sure there really is a number |
| 00:03.28 | WIMPy | No. Far too many factors. |
| 00:03.31 | ChannelZ | And I don't think this is an Asterisk problem in this case anyway. |
| 00:03.43 | albania-asterisk | for asterisk 1.4 or 1.8 |
| 00:04.41 | ChannelZ | If the interface is reporting packets disappearing that's upstream of Asterisk. UNLESS it happens to be the one not processing fast enough causing the buffers to overrun |
| 00:05.00 | ChannelZ | Is the loadav particularly high or anything? |
| 00:05.02 | WIMPy | But if you need a lot of g.729 transcoding, a hardware transcoder card may be cheaper. |
| 00:05.21 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 00:05.33 | albania-asterisk | during 1300-1500 concurrent calls , loadav is 10-18 |
| 00:06.04 | ChannelZ | shit and I just closed the pastebin window... How many cores? |
| 00:06.17 | albania-asterisk | 16 |
| 00:06.50 | WIMPy | Looks like scratching on the limit there. |
| 00:06.52 | ChannelZ | So it's working its tail off |
| 00:07.14 | WIMPy | If that's due to CPU time, that's an issue. |
| 00:07.26 | ChannelZ | yah |
| 00:07.57 | WIMPy | Take a look at how much idle time you have under heavy load. |
| 00:08.16 | WIMPy | If there isn't any left, you ran out of CPU power. |
| 00:08.41 | albania-asterisk | actually CPU i have seen max 480% |
| 00:09.19 | albania-asterisk | but this is spread through all cores, meaning 30% per CPU lets say |
| 00:10.06 | WIMPy | <gavimobile> ChannelZ: your answer is logical, what about my sip provider? |
| 00:10.11 | WIMPy | <gavimobile> my itsp* |
| 00:10.13 | WIMPy | <gavimobile> could they be the problem which causes my pbx to attemplt several |
| 00:10.18 | WIMPy | F*** |
| 00:10.30 | WIMPy | Where do you get that number from? |
| 00:10.34 | ChannelZ | paste diarrhea? |
| 00:10.47 | albania-asterisk | :) |
| 00:10.52 | WIMPy | Yes |
| 00:11.03 | ChannelZ | heh ok |
| 00:11.16 | WIMPy | It sometimes happes that I hit the mouse in a way that pastes the window back to itself :-( |
| 00:11.18 | albania-asterisk | never heard of paste diarrhea , seems fitting |
| 00:12.02 | WIMPy | Quite annoying. |
| 00:12.04 | albania-asterisk | so loadaverage is on the limit ? |
| 00:12.26 | WIMPy | Possibly. |
| 00:12.37 | WIMPy | Depends on the causes. |
| 00:12.48 | albania-asterisk | but i amsure that there is a lot of idle on cpus |
| 00:12.56 | albania-asterisk | during peak |
| 00:12.57 | WIMPy | That's why I suggested watching the idle time instead. |
| 00:13.51 | albania-asterisk | i should see tomorow and let you know |
| 00:13.56 | WIMPy | What about SMP capable reception? |
| 00:14.05 | WIMPy | What kernel do you have? |
| 00:14.06 | albania-asterisk | what is that ? |
| 00:14.41 | WIMPy | Not that long ago the network receive routines in the kernel got SMP enabled. |
| 00:15.06 | WIMPy | Can't remember if it was on the change to 3.0 or when. |
| 00:15.11 | albania-asterisk | 2.6.18-308.1.1.el5 |
| 00:15.25 | WIMPy | Get a new kernel. |
| 00:15.39 | WIMPy | You want that feature. |
| 00:15.53 | WIMPy | Or rather I think you may need it. |
| 00:16.53 | WIMPy | Unless the cause for your issue is outside of the server. |
| 00:17.32 | albania-asterisk | i see, |
| 00:18.06 | albania-asterisk | that fact that i dont have icmp packet loss or tcp errors is it telling me that the problem is not outside the server ? |
| 00:19.08 | albania-asterisk | if the switches/router gives low priority to udp yes, but there is not such thing |
| 00:19.17 | albania-asterisk | they are transparent |
| 00:19.17 | WIMPy | The number of overruns doesn;t seem to be the cause to me, but the udp errors you see seem strange to me. |
| 00:19.31 | WIMPy | But I have no clue what they mean :-( |
| 00:20.07 | albania-asterisk | i will upgrade server to new kernel |
| 00:20.29 | albania-asterisk | and tell tomorrow about idle time |
| 00:22.09 | albania-asterisk | thank you |
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| 00:49.47 | volga629 | http://fpaste.org/d39D/ |
| 00:50.13 | volga629 | this is what I see quality of video just bad |
| 00:51.27 | volga629 | me hard to tell what is not right |
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| 02:10.48 | nickgaw | Hi, Are the asus netbooks ok for a small home asterisk setup? |
| 02:11.10 | WIMPy | Easy |
| 02:11.56 | nickgaw | do any of the asterisk versions support wireless or just wired connections? |
| 02:12.24 | WIMPy | What does Asterisk have to do with your connection? |
| 02:13.04 | nickgaw | the netbook supports both wired and wireless network connections what I am trying to find out is what is best for a small home asterisk setup? |
| 02:13.39 | WIMPy | Wifi is not the best choice for realtime streams. |
| 02:14.39 | nickgaw | but can it work just to get the asterisk system installed over the network then I could always transfer it to the wired network when it is completely setup? |
| 02:15.12 | WIMPy | Sure |
| 02:15.29 | WIMPy | How could it make a difference what network connection you use to install? |
| 02:15.49 | nickgaw | well speed I thought was better on wired? |
| 02:16.17 | WIMPy | That's only a question of your patience. |
| 02:16.47 | nickgaw | but for the phone system working itself it is best to use wired networking? |
| 02:17.17 | WIMPy | Yes |
| 02:18.16 | nickgaw | with centos how do you configure wireless networking? I will have access to the system using ssh so I can have root access before I go and do the asterisk installation portion? |
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| 02:19.17 | WIMPy | You have to ask that to the RedHat guys. |
| 02:20.03 | unicron | my wifi is solid, i have a 1W dongle with a big yagi antenna |
| 02:20.17 | nickgaw | do they make adapters for normal phones that allow them to become sip phones over the wireless network or do they all need to be wired? |
| 02:20.22 | WIMPy | And no neighbours? |
| 02:20.33 | volga629 | http://fpaste.org/d39D/ |
| 02:20.51 | unicron | my antenna can see four neighbors' networks |
| 02:20.59 | volga629 | this debug for res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short |
| 02:21.05 | WIMPy | nickgaw: You can use Asterisk plus additional hardware to do that, yes. |
| 02:22.45 | nickgaw | would stores like bestbuy sell this type of stuff or any sip phones? |
| 02:37.03 | nickgaw | my asus netbook has no DVD burner so how can I make a complete backup of the windows 7 installation on it before installing any version of linux? |
| 02:37.24 | WIMPy | Network? |
| 02:38.02 | nickgaw | well then how can I restore the backup with just linux loadded? |
| 02:38.28 | WIMPy | Derpends on how you do the backup. |
| 02:39.38 | nickgaw | do they make a program in windows where I can have direct access to another computer's DVD burner threw usb? |
| 02:39.58 | WIMPy | Ask Microsoft. |
| 02:41.46 | nickgaw | is there a version of asterisk that runs under windows? |
| 02:42.36 | nickgaw | or is it best to backup windows and then just reinstall linux with asterisk? |
| 02:43.02 | WIMPy | There has been one. |
| 02:43.10 | WIMPy | I have no idea if it is maintained. |
| 02:43.33 | nickgaw | would using virtualbox on a netbook work? |
| 02:44.36 | WIMPy | Should be ok without hardware interfaces. |
| 02:45.22 | nickgaw | so devices don't exist that I attatch to the hardware phones to convert them into wireless sip phones? |
| 02:46.33 | WIMPy | Directely to wireless? Not sure. |
| 02:47.02 | WIMPy | I haven't seen such a thing. But I never looked for one, either. |
| 02:47.07 | nickgaw | what type of devices were you thinking of? |
| 02:47.41 | WIMPy | ? |
| 02:48.10 | nickgaw | phone hardware devices what types of devices would you normally use? |
| 02:48.54 | WIMPy | I prefer real phones. |
| 02:49.16 | nickgaw | yes me to do you mean sip phones that connect using a network cable? |
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| 02:50.14 | WIMPy | I use one of them, but the rest are old fashioned ISDN phones. |
| 02:50.42 | nickgaw | so the normal land line phones won't work? |
| 02:51.10 | WIMPy | It's just a question of the right interface/adaptor. |
| 02:52.56 | nickgaw | so it is possible what I am trying to do? |
| 02:53.21 | WIMPy | What *exactely* do you want to do? |
| 02:54.11 | nickgaw | convert land phones to sip phones but make them connect using wifi to the wireless network so I can place them all around the house and not have long wires going everywhere? |
| 02:54.38 | WIMPy | Short answer: yes |
| 02:55.21 | WIMPy | But as mentioned: I don't know Wifi ATAs, so you might need bit an ATA and some Wifi router for each phone. |
| 02:55.52 | WIMPy | Not sure how mich sense it makes to carry two devices plus two PSUs around with a telephone. |
| 02:56.04 | WIMPy | There are Wifi phones. |
| 02:56.35 | WIMPy | But you will usually be much better of using DECT/CAT-iq phones. |
| 02:56.40 | WIMPy | +f |
| 02:56.44 | nickgaw | ok wifi phones might work how much are they in the USA and are they easy to setup? |
| 02:57.30 | WIMPy | I don;t know your definition of easy, but it's about the same ans any SIP phone. |
| 02:57.45 | WIMPy | -n |
| 02:57.56 | nickgaw | do they also have network ports on them or are they only wifi? |
| 02:58.16 | WIMPy | Wifi phones are wifo phones. |
| 02:58.57 | WIMPy | Some desktop phones that usually connect via ethernet have support for USB wifi donges as an alternative. |
| 02:59.58 | nickgaw | so the phone has a menu where you configure the wifi settings and other settings? |
| 03:00.09 | WIMPy | yes |
| 03:00.19 | WIMPy | Or you use a web interface. |
| 03:00.54 | nickgaw | how much are the wifi phones? |
| 03:01.05 | WIMPy | No idea |
| 03:01.28 | nickgaw | would contacting an asterisk technition be any help? |
| 03:02.28 | WIMPy | I'd hope so. Otherwise they would be pretty pointless. |
| 03:03.12 | kikohnl | These may not be the best prices but I have used the company before, great service. http://www.voipsupply.com/voip-phones/wi-fi |
| 03:04.46 | nickgaw | what is the latest version of asterisk that supports google voice with an add on module? |
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| 03:10.06 | volga629 | res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short what will the cause for this message |
| 03:11.40 | WIMPy | Sounds like a broken packet to me. |
| 03:13.07 | volga629 | 2 hours can' find the cause I was previously point to problems on physical layer like cable speed, I checked everything and it not something on configuration not right |
| 03:13.52 | volga629 | ups sorry I mean it something with configuration |
| 03:14.39 | WIMPy | It can alwys be a software fault at any point. |
| 03:14.50 | volga629 | might on machine where vpn tunnel firewall ? |
| 03:15.05 | volga629 | not really |
| 03:15.07 | WIMPy | ANY point |
| 03:15.32 | volga629 | usually human factor always present |
| 03:17.26 | volga629 | you have match bigger experience in Voip that for sure |
| 03:17.31 | volga629 | me |
| 03:17.50 | WIMPy | I try not to care too much. |
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| 03:36.09 | volga629 | so what you think need check it to resolve the issue |
| 03:37.44 | WIMPy | I'd probably try to run wireshark and see if it moans about the packet contest. |
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| 03:41.28 | lkthomas | hey guys, other than dahdi_tool, what program can I use for non-interactive command check for dahdi status ? |
| 03:42.29 | WIMPy | Non-interactive? |
| 03:43.47 | lkthomas | dahdi_tool |
| 03:43.54 | lkthomas | it's interactive program |
| 03:45.15 | WIMPy | /proc/dahdi/... |
| 03:45.31 | WIMPy | Or use asterisk -rx "dahdi show ..." |
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| 03:52.23 | lkthomas | WIMPy: is that the same ? does dahdi_tool use procfs ? |
| 03:52.52 | WIMPy | I haven't looked at the internals of dahdi_tool. |
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| 03:55.41 | lkthomas | WIMPy: source code: http://www.xelatec.com/trac/pub/browser/xosipt/team/sh/d01/apps/ast_base/dahdi-2.2.1/tools/dahdi_tool.c |
| 03:57.31 | WIMPy | I guess that makes a "no". |
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| 04:10.01 | lkthomas | WIMPy: yes, seems it's using dahdi library instead |
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| 04:32.41 | fling | Hello! how can I increase sip registry refresh timeout? |
| 04:32.50 | fling | Also is not it a bad idea? |
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| 05:14.05 | ChannelZ | Are you refreshing due to dynamic IP? |
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| 05:41.41 | fling | ChannelZ: I'm refreshing due to stupid ISP :p |
| 05:41.52 | fling | ChannelZ: they have fixed most of the issues |
| 05:42.12 | fling | ChannelZ: but did not gave me any instructions on their service usage |
| 05:42.50 | fling | ChannelZ: they also said thas I need to increase the timeout to solve my issue |
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| 05:43.17 | fling | it is 45 seconds now, I need 20 minutes |
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| 05:47.44 | fling | ChannelZ: ip is not a dynamic one, it is 10.x.x.x |
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| 06:26.20 | fling | ChannelZ: registertimeout is not what I want? |
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| 07:03.06 | ChannelZ | well if they are forgetting your registration I guess. In any event look at defaultexpiry in sip.conf |
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| 07:46.38 | fling | ChannelZ: how to change for few hosts but not for all? |
| 07:56.21 | ChannelZ | see the ~expiry option in your register => line |
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| 08:10.40 | fling | ChannelZ: where? |
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| 08:12.01 | fling | register => …:…:…@…/…~(here?) |
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| 08:12.56 | ChannelZ | Open the example sip.conf and search for "register =>" and the format of the line is explained |
| 08:13.03 | ChannelZ | but basically yes |
| 08:14.57 | fling | ChannelZ: and if I set it to 1200 and I still see 45 in 'sip shor registry' then something wrong with my provider? |
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| 08:45.30 | oquidave | hello people |
| 08:46.29 | oquidave | i've been trying to install astaresk 1.8 on opensuse 12.1 this whole week to which i succeeded...but i've a few observations that still buffer me... |
| 08:47.15 | oquidave | when i start asaterisk using /usr/sbin/asterisk -gvvvc, and check for loaded modules using, module show, 0 modules are loaded!! |
| 08:48.31 | oquidave | however, when i start asterisk using, /etc/init.d/asterisk start it redirects to systemctl and just hangs untill i close the process and use /etc/init.d/asterisk restart |
| 08:49.08 | oquidave | then when i check for loaded modules, a long list of modules is rightfully loaded |
| 08:49.28 | oquidave | Question: why am i experiencing this behaviour? |
| 08:50.14 | oquidave | asterisk 1.8 is compiled and installed as root user togather with dahdi and libpri libraries on opensuse12.1 |
| 08:51.01 | oquidave | thanks |
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| 09:25.01 | *** part/#asterisk Serpent (Archer_Ser@unaffiliated/serpent) |
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| 09:44.23 | fling | I have an old phone connected to sip pata converter |
| 09:44.35 | fling | How can I use transfer using this phone? |
| 09:44.56 | WIMPy | See,the manual of your ata. |
| 09:45.00 | fling | oh |
| 09:45.20 | WIMPy | Or use features as in features.conf. |
| 09:45.24 | fling | WIMPy: are there some default numbers to dial for transfer? |
| 09:45.48 | WIMPy | Yes. See that file. |
| 09:45.57 | fling | thanks |
| 09:49.19 | fling | WIMPy: I do not see in the default file what to dial for a simple transfer |
| 09:51.33 | *** join/#asterisk hariom (~hariom@117.195.191.205) |
| 09:53.51 | hariom | I have asterisk server running on a remote server. I started asterisk using "asterisk" command without "-c" or "-r" options. When I login to the remote server terminal (i.e. Shell prompt. Not * CLI), and then make a phone call, I am able to listen prompt but when I am not logged into the terminal (i.e. exit from ssh), I am not able to connect via phone. Phone simply disconnects |
| 09:54.04 | hariom | How to fix this behavior? |
| 09:55.09 | hariom | Pls note that asterisk is running a normal user (not the root user). Does that mean the normal user needs to be logged in? |
| 09:55.46 | wdoekes | no it does not.. but it needs to be disowned before you can logout.. normally starting it without any options would do that (put it in daemon mode) |
| 09:56.05 | unicron | try running it like this: nohup asterisk >/dev/zero </dev/null & |
| 09:56.18 | unicron | er, did i get those backwards |
| 09:56.39 | wdoekes | er.. you don't want /dev/zero for either |
| 09:56.47 | unicron | or run it inside screen, then you can have your console too |
| 09:57.00 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:58.15 | wdoekes | perhaps hariom has nofork=yes in asterisk.conf? or some kind of wrapper script called asterisk? |
| 09:59.18 | hariom | unicron: I didn't get what you mean by run it inside screen? I just login to the remote server to check the log file time to time. I believe that asterisk with the command ("asterisk") without any options would run in the background. |
| 09:59.38 | wdoekes | hariom: what does "jobs" say after starting it? |
| 09:59.39 | hariom | wdoekes: Will check for nofork=yes. I am running an IVR via AGI |
| 09:59.43 | unicron | screen runs in the background and attaches your programs to it |
| 09:59.57 | unicron | so you can log back in and resume your screen and all your stuff will be running still |
| 10:00.17 | wdoekes | (but running screen shouldn't be needed, unless you want to) |
| 10:00.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:00.59 | fling | WIMPy: is *71 the default? |
| 10:02.26 | unicron | i'm a noob so i like to randomly interact with the CLI, that's why i like screen |
| 10:02.32 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 10:03.35 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 10:04.47 | *** join/#asterisk d00gster (~doughant@bba773204.alshamil.net.ae) |
| 10:05.11 | hariom | wdoekes: I have nofork=yes commented (i.e. disabled) |
| 10:05.29 | hariom | wdoekes: jobs doesn't give any output |
| 10:05.56 | *** join/#asterisk d00gster (~doughant@bba773204.alshamil.net.ae) |
| 10:08.25 | fling | WIMPy: it is not transferring |
| 10:08.35 | hariom | wdoekes: Any other suggestion? what is nofork=yes for? |
| 10:09.27 | WIMPy | fling: Then your dtmfmode may be wrong. Make sure the ATA and Asterisk are configured to the same way of transmitting DTMF. |
| 10:12.30 | hariom | When I exit from the remote server's shell prompt, I know that asterisk process is running as I see in "ps -e" when I ssh again to the server. But surprisingly, it doesn't handle any calls (call just drops no ring) when I exit from remote shell terminal. |
| 10:13.45 | hariom | If I keep myself logged in, asterisk handles the call. When I exit from the remoter server, asterisk drops the call. I am running asterisk with the command "asterisk" i.e. without any options |
| 10:14.15 | WIMPy | Is that some strange security "feature"? |
| 10:14.36 | *** join/#asterisk the_voice_ (~Ben@bzq-79-180-28-247.red.bezeqint.net) |
| 10:14.41 | unicron | maybe the server is killing your background processes when you logout |
| 10:14.51 | the_voice_ | anyone here know of any SMS enabled DID providers? |
| 10:15.25 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 10:16.00 | wdoekes | hariom: ok, if it's still running after you ssh again, it wasn't killed because of the lost shell (so fork/jobs is not an issue) |
| 10:16.11 | hariom | unicron: It doesn't kill the process as when I logs in, I can see asterisk process running (I can connect via "asterisk -r" and also ps -e says so) |
| 10:16.38 | wdoekes | in that case, I have absolutely no idea what your problem is :) |
| 10:16.41 | hariom | wdoekes: yea |
| 10:16.50 | wdoekes | increase debug level and verbosity and check that |
| 10:16.57 | wdoekes | (make sure it goes somewhere, see logger.conf) |
| 10:19.09 | hariom | wdoekes: Actually I have asterisk run at boot time as it is registered as a service to run at all run levels (I did make config) |
| 10:21.58 | fling | WIMPy: I can call from this phone, but numbers in menu are not working (and if I call from pstn dtmf are not working too) |
| 10:22.30 | fling | WIMPy: I have tried to change dtmf to info in pata-sip converter settings, still have this problem |
| 10:26.24 | WIMPy | The best choce is probably to use rfc2833 if available. |
| 10:26.33 | fling | WIMPy: ok |
| 10:26.42 | hariom | When asterisk running as non root user mode, should the user be a system user? |
| 10:27.12 | WIMPy | And if it's not working, there's probably a configuration mismatch between the ATA and Asterisk. |
| 10:27.22 | hariom | I am facing problem when I exit from the remote server. Asterisk stops handling the calls. But when I am logged into the server, it handles the calls fine |
| 10:27.29 | fling | WIMPy: ok, thanks, I will check it |
| 10:27.34 | WIMPy | What is a "system user"? |
| 10:27.42 | fling | WIMPy: the one from /etc/passwd |
| 10:27.53 | WIMPy | Oh, sure. |
| 10:28.03 | WIMPy | From where else? |
| 10:28.12 | fling | hehe |
| 10:28.47 | fling | or if we call about _system_ it is the user with uid<1000 |
| 10:28.55 | fling | s/call/talk |
| 10:29.29 | WIMPy | That's why I asked. |
| 10:29.30 | hariom | I don't think it is mandatory to remain logging into remote server for asterisk to handle calls |
| 10:29.50 | fling | hariom: umm? |
| 10:29.56 | WIMPy | hariom: Not something I have ever heard of before. |
| 10:30.33 | hariom | WIMPy: but here in my case, it is happening something like this. I am also clue less why is it happening like this |
| 10:31.22 | WIMPy | So am I. |
| 10:31.45 | fling | WIMPy: how can I simply test is my dtmf working? |
| 10:32.06 | WIMPy | hariom: Have you tried to start other demons that way? |
| 10:32.28 | WIMPy | fling: Call some extension that has a menu. e.g. VoiceMailMain. |
| 10:32.31 | hariom | WIMPy: other demons? which ones? |
| 10:32.41 | fling | right… |
| 10:33.27 | WIMPy | hariom: ftpd, httpd, anything. |
| 10:33.39 | fling | hariom: sshd :| |
| 10:33.46 | hariom | WIMPy: nope. I don't have any of these on my remote server |
| 10:35.20 | hariom | fling: sshd will run. The issue which I think could be is that sshd is installed as a root user where as asterisk is running as a normal user (non root user). By default, ftpd, httpd etc will also gets installed as root user ownership |
| 10:36.08 | WIMPy | They all would be as much root as Asterisk. |
| 10:36.15 | hariom | the question now is, Does running in non-root user mode forces asterisk to not handle calls when that user is not logged in? |
| 10:36.33 | WIMPy | They can run as root, but are usually configured to drop privileges. |
| 10:36.36 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 10:36.45 | hariom | WIMPy: I never tried running httpd, ftpd or sshd in non root user previleges |
| 10:36.57 | hariom | WIMPy: yea, once started |
| 10:37.21 | WIMPy | Usually, yes. |
| 10:37.50 | WIMPy | Not different from Asterisk. |
| 10:38.04 | hariom | Usually, yes ? for what |
| 10:38.08 | WIMPy | It has all 3 options as well. |
| 10:39.03 | fling | hariom: tell me what are you doing? :p |
| 10:39.31 | hariom | fling: I am facing problem when I exit from the remote server. Asterisk stops handling the calls. But when I am logged into the server, it handles the calls fine |
| 10:40.09 | fling | hariom: do you have root access? why are not you using initscript? |
| 10:41.34 | hariom | fling: I did "make config" . So I think when I restart, system uses initscript to start asterisk |
| 10:41.47 | fling | hariom: check it! |
| 10:41.58 | hariom | fling: checked it |
| 10:42.14 | fling | can you start and stop it with initscript? |
| 10:42.19 | WIMPy | Does it work after boot? |
| 10:42.29 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 10:42.31 | hariom | fling: when I reboot the system it starts * automatically |
| 10:42.37 | hariom | "make config" |
| 10:43.57 | fling | WIMPy: do I need to 'Enable Out-of-Band DTMF' ? |
| 10:44.07 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 10:44.27 | WIMPy | fling: You don;t strictly need it, but you want it, yes. |
| 10:45.07 | fling | WIMPy: only if I enable this option I can select rfc2833 or sipinfo |
| 10:45.20 | fling | so it is enabled and rfc2833 selected |
| 10:45.47 | WIMPy | good |
| 10:46.59 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 10:47.34 | the_voice_ | okay so I tai it no one here knows of any SMS enabled providers? Twilio, Plivo and Tropo all seem to have SMS's enabled on their numbers and I am trying to figure out how they did it |
| 10:48.25 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:48.39 | *** join/#asterisk cneb3000 (~cneb3000@b01977b3.bb.sky.com) |
| 10:49.44 | fling | WIMPy: it works now, I can walk in menu |
| 10:51.39 | fling | in CALL FEATURE CODE Call Transfer is set to *71 |
| 10:51.52 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 10:52.21 | fling | but when I press *71 ata sends these tones to asterisk and asterisk sends tones to the second caller |
| 10:52.29 | fling | instead of transfer |
| 10:53.07 | WIMPy | You need to enable transfers in the Dial application. See option t and T. |
| 10:53.33 | WIMPy | is AFK again |
| 10:53.54 | ectospasm | fling: "core show application Dial" << look for both T and t |
| 10:54.33 | wdoekes | hariom: stop repeating the question and start reading verbose/debug logs |
| 10:55.10 | hariom | wdoekes: nothing gets into it when I am logged out. |
| 10:55.41 | hariom | wdoekes: you doesn't know doesn't mean others also won't know the issue. I tried what you mentioned |
| 10:56.18 | wdoekes | your issue is seriously odd.. and spamming the channel is always uncalled for |
| 10:57.00 | wdoekes | pb the log before and after you logout.. if *nothing* happens, I doubt that your verbosity/debug is high enough |
| 10:57.06 | wdoekes | or asterisk is in SIGSTOP mode |
| 10:57.08 | fling | ectospasm: thanks |
| 10:57.12 | *** join/#asterisk grohne (~grohne@vortex.stw-bonn.de) |
| 10:57.17 | wdoekes | ps fax | grep T |
| 10:57.48 | *** join/#asterisk barbosa2 (~juliano.b@177.43.91.206) |
| 10:57.54 | hariom | wdoekes: 1358 pts/0 S+ 0:00 \_ grep --color=auto T |
| 10:58.04 | wdoekes | ok, so asterisk is running |
| 10:58.05 | hariom | nothing found. I have fax disabled |
| 10:58.15 | hariom | sorry! |
| 10:58.32 | hariom | :) |
| 10:58.48 | grohne | hi. if I need more features in libpri. to which mailinglist would I send patches? |
| 10:58.49 | hariom | I guess I should take a coffee |
| 10:59.42 | grohne | is that on topic on asterisk-dev@? |
| 11:00.24 | wdoekes | never send patches to the mailing list |
| 11:00.29 | wdoekes | attach them to a bug report in jira |
| 11:00.38 | wdoekes | issues.asterisk.org/jira |
| 11:01.07 | wdoekes | you may have to wait a couple of days before getting your submitters license approved |
| 11:02.09 | wdoekes | and.. even better, create a reviewboard item in reviewboard.asterisk.org after creating the ticket in jira |
| 11:02.23 | wdoekes | it will spam asterisk-dev that there is something to evaluate |
| 11:03.11 | fling | ectospasm: will it work with default features.conf? |
| 11:03.30 | ectospasm | fling: it should |
| 11:04.04 | fling | ectospasm: do I even need to setup anything on ATA? or asterisk should do the Transfer? |
| 11:04.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:e5cf:6fe6:6d65:275f) |
| 11:04.27 | ectospasm | fling: I wouldn't think that setting up anything on the ATA is required, but YMMV... |
| 11:05.32 | fling | ectospasm: but I don't see anything about *71 in asterisk config |
| 11:06.52 | ectospasm | fling: you probably need to set either atxfer or blindxfer to *71 in features.conf |
| 11:07.53 | grohne | wdoekes: thanks for the pointers. |
| 11:08.24 | *** join/#asterisk nisam (~nisam@125.22.91.2) |
| 11:09.07 | fling | ectospasm: so *71 is not the default? what is the default for asterisk? or I should set something anyway? |
| 11:14.17 | ectospasm | fling: see the annotated features.conf (sample file)... blindxfer => #1, atxfer => *2, afaict |
| 11:14.40 | fling | ectospasm: oh! thanks! |
| 11:15.08 | ectospasm | fling: but you can redefine them to be anything |
| 11:15.27 | ectospasm | ...if you want to change it so it matches your provider's vertical service codes, go for it |
| 11:16.16 | fling | no, I will keep it |
| 11:27.28 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:27.40 | fling | so I transfer using #, it works fine |
| 11:27.45 | fling | ectospasm: thanks ^ |
| 11:31.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:31.37 | *** join/#asterisk adolfomaltez (~taro@190.62.203.66) |
| 11:38.08 | *** join/#asterisk StaRetji (~LittleAll@87.250.55.105) |
| 11:38.13 | StaRetji | Howdy folks |
| 11:41.06 | StaRetji | if I use in extensions.conf, commands in order 1,2,3 and then I do #include another.conf and #include yetanother.conf and then 4,5,6 and so on |
| 11:41.34 | StaRetji | will asterisk process this conf in that order? |
| 11:41.55 | StaRetji | 1,2,3 values from 2 confs and after finish parsing them continue to 4.5.6? |
| 11:44.08 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:46.22 | StaRetji | like 1.do this, 2.do that, 3. parse some.conf (there are like 100 lines), once finish, continue parsing extensions.conf? |
| 11:46.52 | StaRetji | forgive this poor noob if question is silly :) |
| 11:47.32 | StaRetji | I am just affraid if I do 1.2. and parse some.conf that remaining of extensions.conf will be ignored |
| 11:48.26 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 11:50.47 | *** join/#asterisk angryuser (~Angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:51.44 | angryuser | Hello, i have a two port E1 card, one in NT connected to old pbx, and one in TE mode, when i dial by console by TE port it works, when i dial by PBX, it try to dial by TE port, i am getting ISDN cause code 44 |
| 11:51.49 | angryuser | can someone help me out ? |
| 11:52.40 | angryuser | i dont know where to search here, i did the pri debug, there is nothing diffrent there ;( |
| 11:53.01 | *** join/#asterisk sbszulu (~Sibusiso_@41.55.136.178) |
| 11:53.28 | bulkorok | Cause No. 44 - requested circuit/channel not available. |
| 11:53.29 | bulkorok | This cause is returned when the circuit or channel indicated by the requesting entity cannot be provided by the other side of the interface. |
| 11:54.55 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 11:54.58 | angryuser | bulkorok: well, so ? |
| 11:55.38 | angryuser | bulkorok: when i dial from console on this port it dials fine, i get this code only when my pbx dials from NT to TE port |
| 11:56.02 | bulkorok | 'core set verbose 5' in * CL |
| 11:56.07 | bulkorok | I |
| 11:57.14 | StaRetji | is my logic regarding #include okay folks? |
| 11:57.15 | StaRetji | thz |
| 12:02.07 | cneb3000 | Don't suppose anyone has any recommendation for UK sip carriers? |
| 12:02.27 | cneb3000 | currently making about 400k minutes of Nation calls, about 200k minutes of mobile calls. |
| 12:02.34 | cneb3000 | nation = national |
| 12:03.38 | nisam | hi |
| 12:04.49 | nisam | bulkorok:one help |
| 12:05.03 | bulkorok | lets see.. |
| 12:06.53 | nisam | help on bluetooth: now i am able install using chan_mobile module. But i can not see /etc/asterisk/mobile.conf. |
| 12:08.30 | nisam | wat will be the issue..? |
| 12:08.43 | bulkorok | lokk at ~asterisk-sources~/configs/chan_mobile.conf.sample |
| 12:09.09 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 12:09.41 | bulkorok | at least /etc/asterisk/chan_mobile.conf |
| 12:09.58 | bulkorok | if you issued 'make samples' |
| 12:10.01 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:10.45 | nisam | i issued make samples |
| 12:11.31 | nisam | chan_mobile.conf is available |
| 12:11.36 | bulkorok | :) |
| 12:13.36 | *** join/#asterisk StaRetji (~LittleAll@87.250.55.105) |
| 12:14.00 | WIMPy | angryuser: That might be an internal thing. Are the channels set up correctly in chan_dahdi.conf? |
| 12:14.24 | angryuser | How come the WIMPy yes i can call on any |
| 12:14.32 | angryuser | WIMPy yes i can call on any |
| 12:14.43 | angryuser | Inbound calls works fine by the way |
| 12:14.49 | nisam | both are same..? |
| 12:15.02 | angryuser | The come from TE > to NT port no problem, pbx anwers |
| 12:15.46 | bulkorok | nisam: chan_mobile.conf is for chan_mobile... i don't know if mobile.conf exists... |
| 12:16.26 | WIMPy | angryuser: And you have a full PRI? |
| 12:16.27 | angryuser | WIMPy: system.conf http://pastebin.ca/2206236 |
| 12:16.39 | angryuser | WIMPy: yes 30 channels |
| 12:16.57 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 12:18.26 | angryuser | WIMPy: my dialplan, there is no more simple |
| 12:18.27 | angryuser | http://pastebin.ca/2206237 |
| 12:19.31 | WIMPy | You might not want that extension under default, but that's a different topic. |
| 12:19.41 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 12:19.43 | angryuser | WIMPy: it is for console testing |
| 12:19.57 | WIMPy | The interesting bit of configuration is chan_dahdi.conf. |
| 12:20.07 | angryuser | WIMPy: when i dial like "console dial 12316546" it works |
| 12:20.09 | WIMPy | You can use an explicit context there as well. |
| 12:20.24 | WIMPy | But some debug output might be better. |
| 12:20.47 | *** join/#asterisk patrickximenes (~hexa@187.68.71.4) |
| 12:20.50 | WIMPy | There must be some diference, obviousely. |
| 12:21.00 | angryuser | WIMPy: http://pastebin.ca/2206239 |
| 12:22.28 | angryuser | WIMPy: yea, its been hours i am trying some different setups, even compared the Pri trace from console dial, and PBX dial, i dont see the difference |
| 12:23.49 | WIMPy | switchtype=national? Didn't you say E1? |
| 12:24.09 | WIMPy | Well, it's obviousely E1. |
| 12:24.48 | angryuser | WIMPy: yea its national |
| 12:25.04 | angryuser | WIMPy: so the pri trace from console http://pastebin.ca/2206240 |
| 12:25.11 | angryuser | (works) |
| 12:25.15 | WIMPy | Interesting. |
| 12:25.19 | WIMPy | Very interesting. |
| 12:26.37 | angryuser | WIMPy: pri trace from the PBX to provider http://pastebin.ca/2206243 which is not working |
| 12:26.52 | WIMPy | Why (or since when) is the hex dump formatted that way? |
| 12:26.59 | angryuser | Note how just after SETUP message i got cause 44 |
| 12:27.13 | angryuser | (on the second example) |
| 12:28.15 | angryuser | WIMPy: nationa or euroisdn changes nothing, as pridilaplan whatever options |
| 12:28.38 | angryuser | I can call in, PBX answers, he can not call out |
| 12:28.50 | WIMPy | pridialplan sets the type of number. |
| 12:29.38 | WIMPy | Span 2 is the NT port to the PBX? |
| 12:29.53 | angryuser | WIMPy: the only strange stuff i've found its when i call IN (works) i got PRI Span: 1 !! Unknown IE 36 (cs6) |
| 12:30.06 | angryuser | But the call is ok |
| 12:30.16 | *** join/#asterisk danfromuk (~IceChat77@2.27.28.56) |
| 12:30.19 | angryuser | Well, any ideas ? :) |
| 12:30.29 | *** part/#asterisk patrickximenes (~hexa@187.68.71.4) |
| 12:33.02 | WIMPy | The only wrong thing I see so far is in the working example. |
| 12:34.02 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:35.08 | angryuser | WIMPy: tell me |
| 12:35.37 | leifmadsen | I'm building an example that uses the dialplan_exec option in the ConfBridge() dtmf triggered menus. Anyone have an example they can think of that would be a good demonstration of this function? |
| 12:35.59 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 12:36.23 | leifmadsen | If you don't know what it does: allows a participant of the conference bridge to leave the conference temporarily to execute some dialplan and then rejoin the conference |
| 12:36.50 | WIMPy | Ooops |
| 12:37.00 | WIMPy | Look at the direction. |
| 12:37.22 | WIMPy | It's Asterisk that disconnects the call immediately after trying to place it. |
| 12:38.00 | leifmadsen | actually just thought of a good example I think |
| 12:38.03 | leifmadsen | thanks anyways : |
| 12:38.04 | leifmadsen | :D |
| 12:38.05 | WIMPy | Wait. How many calls are there? |
| 12:38.14 | angryuser | WIMPy: how come he does that ? |
| 12:38.42 | angryuser | WIMPy: do i need to indicate progress ? |
| 12:39.19 | WIMPy | Why are the Asterisk debug outputs so horrible to read? |
| 12:39.34 | WIMPy | And why do they have so many different formats? |
| 12:41.05 | WIMPy | Yes, Asterisk tears down the call before the remote can acknowledge it. |
| 12:41.17 | WIMPy | Why it does so, I have absolutely no idea. |
| 12:42.09 | leifmadsen | WIMPy: because it is an organic piece of software with many participants over a large period of time, and no one has spent the time to make the output consistent. |
| 12:42.16 | leifmadsen | This is pretty much the answer to any of those types of questions. |
| 12:42.45 | leifmadsen | same issue with the console commands until 4 of us at an AstriDevCon went through to make them consistent (for better or worse) |
| 12:42.59 | StaRetji | folks, how to block calls to certain dnid (mobile phone number)? thx |
| 12:43.04 | *** join/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 12:43.05 | WIMPy | Unfortunately that wastes so much time. |
| 12:43.07 | DelphiWorld | hey all |
| 12:43.21 | DelphiWorld | anyone know of any good widget that's Click to cable capable? |
| 12:43.27 | leifmadsen | StaRetji: just add a dialplan match for those numbers and have it playback a message stating that numbers not allowed to be called |
| 12:43.41 | StaRetji | is this okay? exten => xxxxxxxxxxxx,1,Hangup() |
| 12:43.44 | WIMPy | angryuser: I suspect there must be somethign going wrong between Asterisk and the PBX. Give us a trace from that port. |
| 12:43.49 | StaRetji | where xxxxx is mobile phone number |
| 12:43.51 | StaRetji | that leifmadsenq |
| 12:43.53 | leifmadsen | StaRetji: if it's a real number, yes |
| 12:44.13 | leifmadsen | specific number will get matched first over a pattern match, so that is valid |
| 12:44.22 | StaRetji | I get pbx.c:8312 add_priority: Unable to register extension 'xxxxxxxxxxxx', priority 1 in 'default', already in use |
| 12:44.34 | leifmadsen | then it's already been definied |
| 12:44.37 | StaRetji | where xxxxx is that mobile number |
| 12:44.39 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 12:44.41 | StaRetji | i added xxx |
| 12:44.44 | StaRetji | and 00xxxxx |
| 12:44.53 | StaRetji | thinking that maybe it wont match |
| 12:44.54 | leifmadsen | are you adding pattern matches, or real number? |
| 12:44.59 | StaRetji | real number |
| 12:45.03 | StaRetji | I added number |
| 12:45.05 | leifmadsen | then you've already definied it |
| 12:45.07 | StaRetji | and 00samenumber |
| 12:45.10 | leifmadsen | defined it* |
| 12:45.15 | leifmadsen | it can't be defined twice |
| 12:45.18 | leifmadsen | or you get that error |
| 12:45.21 | StaRetji | so, number and 00number is identical? |
| 12:45.25 | [TK]D-Fender | No |
| 12:45.25 | leifmadsen | no |
| 12:45.38 | leifmadsen | look at your dialplan -- you've got a typo or something defined twice |
| 12:45.40 | StaRetji | oh, I understand you!! I have number twice! |
| 12:45.43 | StaRetji | thx! |
| 12:45.45 | leifmadsen | yes..... |
| 12:46.05 | [TK]D-Fender | StaRetji, Which is exactly what that errors says |
| 12:47.13 | StaRetji | understood o7 |
| 12:47.14 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 12:47.20 | StaRetji | thx folks |
| 12:47.47 | StaRetji | but do I have to do 00number and number |
| 12:47.55 | StaRetji | or I can just do number? |
| 12:48.03 | StaRetji | cause I have 123456 and 00123456 |
| 12:48.18 | StaRetji | becasue with both 123456 and 00123456 you can make a call |
| 12:48.22 | [TK]D-Fender | StaRetji, Do want 2 different patterns or not? This is YOUR system. What do you need to dial? |
| 12:48.27 | StaRetji | so I thought I should be blocking both |
| 12:48.33 | leifmadsen | you need both if you want to block both |
| 12:48.37 | [TK]D-Fender | StaRetji, Blocking? |
| 12:48.42 | StaRetji | yep |
| 12:48.49 | StaRetji | roger that |
| 12:48.56 | [TK]D-Fender | StaRetji, Do what you want... it's your dialplan.... |
| 12:49.34 | StaRetji | will do, thx ;) |
| 12:49.35 | angryuser | WIMPy: sure one moment |
| 12:53.14 | angryuser | WIMPy: here it is http://pastebin.ca/2206253 |
| 12:54.09 | *** part/#asterisk deo (~dnepangue@203.177.214.75) |
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| 12:56.29 | angryuser | WIMPy: i am not that good in pri traces ;' |
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| 13:00.11 | ulogic | With sippeers in a realtime database, the CLI command "sip show users" doesn't show anybody until they register. Is that the way it is supposed to work?? |
| 13:00.23 | [TK]D-Fender | yes |
| 13:01.24 | ulogic | Also after a sip reload, "sip show users" doesn't show anybody. |
| 13:01.31 | leifmadsen | ulogic: yes |
| 13:02.00 | leifmadsen | that's how it works -- you might want to enable rtcache in sip.conf so it keeps the peers in memory after a module reload |
| 13:02.24 | ulogic | I'll give that a try right now |
| 13:04.48 | WIMPy | angryuser: It's your PBX that rejects the channel. |
| 13:05.03 | WIMPy | It it configured for a fractional PRI? |
| 13:05.21 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:05.31 | WIMPy | Or does it hace a configuration that only allows certain directions on certain channels? |
| 13:06.01 | angryuser | WIMPy: can you explain please ? |
| 13:06.24 | WIMPy | No. The PBXs manual might. |
| 13:06.26 | angryuser | WIMPy: the thing is if i connect the provider directly it works ;( |
| 13:06.48 | angryuser | WIMPy: i mean where do you see that it rejects ? |
| 13:06.58 | WIMPy | I can only see that it is disconnection the call with "requested channel not available". |
| 13:07.41 | WIMPy | From line 122. |
| 13:07.56 | ulogic | sip.conf.sample only shows rtcachefriends which I set to yes and did a sip reload |
| 13:08.10 | WIMPy | The disconnects goes from the PBX to Asterisk. |
| 13:08.26 | WIMPy | -s |
| 13:08.42 | ulogic | I force a phone to reregister, and it shows up in "sip show users", but after a sip reload, it still disappears |
| 13:09.06 | ulogic | Do I have to totally unload chan_sip.so and reload it that way? |
| 13:10.53 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 13:10.59 | xoveruk | hi |
| 13:11.14 | xoveruk | please, how do I place a call to a mobile from 'console dial'? |
| 13:11.14 | ulogic | Actually, I just force an unload and load of chan_sip.so, but doing a sip reload still clears the users list |
| 13:11.43 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 13:12.45 | ulogic | xoveruk: have you tried console dial mobile-phone-number@your-outbound-context ? |
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| 13:15.09 | *** join/#asterisk lminiero (~lminiero@2001:760:3403:fffc:2c0:9fff:fec3:5b07) |
| 13:15.30 | lminiero | hi there |
| 13:16.09 | lminiero | anybody around? |
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| 13:16.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:17.58 | xoveruk | ulogic: with the full dialplan@context? |
| 13:21.27 | xoveruk | ulogic: I get console hangup after it dials the number |
| 13:21.43 | xoveruk | I am using 9 for outside lines, must I use 9no@context? |
| 13:27.24 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
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| 13:28.18 | DelphiWorld | hi pabelanger |
| 13:28.36 | pabelanger | indeed |
| 13:28.50 | DelphiWorld | pabelanger: do you know any good C2C Service? |
| 13:29.05 | pabelanger | C2C? |
| 13:29.41 | DelphiWorld | pabelanger: Click to call |
| 13:29.56 | pabelanger | Ah. No, don't use it |
| 13:30.32 | DelphiWorld | np |
| 13:30.34 | cneb3000 | DelphiWorld: You mean something that highlights numbers in web pages and lets you click them and dial automatically? |
| 13:30.43 | cneb3000 | I think firefox uses to have an add on that did that? |
| 13:30.47 | grohne | uhm. the license agreement on jira is very restrictive. I does not permit uploading a modified patch. Do I really need to upload a diff to a diff in this case? |
| 13:30.59 | DelphiWorld | cneb3000: yes but built-in web phone, and i want only one number to be dialed that's a sip uro |
| 13:31.04 | DelphiWorld | s/uro/uri/ |
| 13:31.14 | grohne | s/I does/It does/ |
| 13:32.30 | DelphiWorld | cneb3000: http://www.phono.com but i'm a js less person |
| 13:32.32 | Katty | hello my asterisk does not work at all how to fix plz |
| 13:32.47 | cneb3000 | DelpiWorld: I was juist going to suggest that :) |
| 13:32.56 | grohne | Katty: you will have to provide more details to get any help at all. |
| 13:33.03 | DelphiWorld | Katty: rm -Rf * ;-) |
| 13:33.08 | DelphiWorld | joke |
| 13:33.16 | cneb3000 | Katty: del c:\system32 |
| 13:33.31 | DelphiWorld | cneb3000: format c:/q |
| 13:33.34 | DelphiWorld | shutdown -s |
| 13:33.52 | grohne | is anyone actually using libpri trunk? it seems heavily modified. |
| 13:34.25 | cneb3000 | DelphiWorld: I cant remember its name, but I found a flash soft phone once |
| 13:34.34 | cneb3000 | maybe that can get you started on a googling journey |
| 13:34.41 | DelphiWorld | cneb3000: red5? |
| 13:34.48 | cneb3000 | ahh yes! thats it |
| 13:34.52 | cneb3000 | long time ago though |
| 13:34.52 | DelphiWorld | cneb3000: you do JS? |
| 13:35.09 | cneb3000 | hmm, i can read it, but never really written in it |
| 13:35.59 | cneb3000 | just reading the documentation for phono though |
| 13:36.20 | cneb3000 | do you have an API key already? |
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| 13:38.50 | DelphiWorld | cneb3000: i do |
| 13:39.00 | cneb3000 | DelphiWorld: because there's an example on http://phono.com/docs that looks like it'll work |
| 13:39.15 | DelphiWorld | cneb3000: it's not easy to cleanup :-) |
| 13:39.28 | cneb3000 | DelphiWorld: Just change phono.phone.dial("985-655-2500", { to whatever number you want to dial :p |
| 13:39.49 | DelphiWorld | :P |
| 13:40.03 | cneb3000 | without testing it.. i think that will do! |
| 13:44.15 | *** join/#asterisk d00gster (~doughant@bba773204.alshamil.net.ae) |
| 13:46.07 | robl^ | What's the cleanest way to implement automated announcements? i.e. I want to dial an extension from a phone, asterisk will then call another extension (Overhead paging device) and play a pre-recorded message |
| 13:48.47 | *** part/#asterisk nisam (~nisam@125.22.91.2) |
| 13:50.02 | [TK]D-Fender | robl^, Record file, originate the playback |
| 13:50.39 | [TK]D-Fender | Well .. actually yuo didn't mention having to record it in that step.... so jsut originate the playback |
| 13:51.10 | robl^ | right. just a series of standard announcement pre-recorded. |
| 13:53.12 | robl^ | ohh! I missed originate() that does make it a lot easier. thanks [TK]D-Fender. I was thinking weird hacks with call files to originate |
| 14:00.21 | robl^ | [TK]D-Fender: PERFECT! Thanks! |
| 14:00.34 | [TK]D-Fender | robl^, You're welcome. |
| 14:01.37 | robl^ | I really need to update things. I still think in terms of 1.4 era. |
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| 14:23.09 | grohne | wdoekes: so I attached an updated patch to PRI-106. |
| 14:23.34 | grohne | wdoekes: how about more intrusive changes to pri. is the trunk version to be released any time soon? |
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| 14:36.52 | wdoekes | trunk become 11 very recently, so no |
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| 14:38.23 | wdoekes | nm, that's asterisk. I know nothing about libpri |
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| 14:43.34 | grohne | wdoekes: sorry, I got disconnected. can you repeat? |
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| 14:46.14 | kuku | I'm trying to do chanspy with just using the sip extension ( without the channel ) it works 80% of the time, the other 10% it doesn't and the other 10% it connects to an entirely different extension... any advice ? |
| 14:46.34 | WIMPy | grohne: The issue is of a much more general nature and can probably be found in many to most implementations. |
| 14:46.55 | WIMPy | It affects most IEs including caller IDs. |
| 14:47.06 | grohne | WIMPy: indeed. |
| 14:47.19 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:48.47 | [TK]D-Fender | kuku, "sip extension" is not a meaningful term. Show us debug of the actual failure |
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| 14:51.23 | wdoekes | grohne: that I know nothing about the release cycle of libpri |
| 14:52.06 | nisam | hi |
| 14:52.23 | nisam | which is the best sip trunk provider in india |
| 14:52.55 | grohne | WIMPy: so I have some patches that attach lengths to party_numbers |
| 14:53.19 | grohne | WIMPy: i just wonder whether it makes any sense to publish them, since they target 1.4.12, but this seems mostly like a stable release |
| 14:53.25 | *** join/#asterisk jrod (jrod@2600:3c00::f03c:91ff:fe93:649a) |
| 14:53.43 | WIMPy | Ah, nice. |
| 14:53.53 | grohne | just should I base them on 1.4.12 or trunk? |
| 14:54.23 | grohne | in any case it *will* severly break api |
| 14:54.26 | WIMPy | GI guess you're better off to do them against trunk. |
| 14:54.43 | WIMPy | But if it's not too much work, you could just do both. |
| 14:55.02 | grohne | it is basically duplicating the work, since the diff stable..trunk is huge |
| 14:55.42 | grohne | trunk looks like a rewrite. |
| 14:55.50 | WIMPy | An API change will certainly not go in to anything but trunk. |
| 14:56.01 | grohne | i certainly agree |
| 14:56.05 | WIMPy | I haven't looked that deep. |
| 14:56.22 | WIMPy | In fact I try to avoid libpri now. |
| 14:56.22 | grohne | this is why I retained api in my patch to PRI-106 |
| 14:57.05 | grohne | so is there something else providing similar functionality? |
| 14:57.36 | WIMPy | mISDNuser |
| 14:58.44 | grohne | thanks. I'll have a look |
| 14:58.48 | WIMPy | And yes, I'd be ery astounded if the same bug didn't exist there, |
| 14:58.53 | WIMPy | very |
| 15:00.09 | grohne | can you point me to the component implementing q931 in misdnuser? |
| 15:00.24 | AkkerKid | Good day everyone! If I wanted to build a script where you could dial the last two numbers of an extension that had the same first 3 numbers of your own and it would ring the full 5 digit extensions for you, how would I do it? |
| 15:01.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:02.07 | WIMPy | grohne: lib/layer3 |
| 15:02.13 | grohne | thanks. |
| 15:02.16 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:02.59 | [TK]D-Fender | AkkerKid, give us an example number.... |
| 15:03.24 | AkkerKid | I'm at exten 12345 and I want to dial 12399 by only hitt digits 99 |
| 15:03.42 | grohne | WIMPy: so at least mi_decode_calling_nr preserves length. which is more than libpri does. |
| 15:04.01 | [TK]D-Fender | AkkerKid, then make and exten to match 2 digits and shove the 123 in your Dial() |
| 15:04.14 | jeffspeff | yep, that would work |
| 15:04.39 | grohne | WIMPy: wait. it doesn't. it just copies everything necessary. :-( |
| 15:04.56 | carrar | exten => _XX/300XX,1,Dial(SIP/300${EXTEN},100) |
| 15:04.58 | AkkerKid | the hard part for me is setting it up so using the same script, someone with ext 12445 and dial 12499 aswell with only 99 |
| 15:05.04 | carrar | something like that |
| 15:05.27 | [TK]D-Fender | AkkerKid, You can't do both patterns in one. |
| 15:05.50 | carrar | exten => _XX/_300XX,1,Dial(SIP/300${EXTEN},100) |
| 15:05.57 | [TK]D-Fender | carrar, And close ... as abstractions go.... |
| 15:06.01 | carrar | heh |
| 15:06.36 | carrar | AkkerKid, just write a simple AGI |
| 15:06.37 | AkkerKid | there's no way to get the first three digits of the source extensions and the two digits dialed to create the destination extension? |
| 15:07.20 | AkkerKid | The idea is that a branch office can call their local extensions withour dial the full 5 digits |
| 15:07.28 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-uvyevcbvzvtryuai) |
| 15:07.33 | carrar | or just a do a if on comapir of the first 3 digits |
| 15:07.35 | WIMPy | grohne: I guess that's what you will find almost everywhere :-( |
| 15:07.40 | *** part/#asterisk jrod (jrod@2600:3c00::f03c:91ff:fe93:649a) |
| 15:07.43 | *** part/#asterisk nisam (~nisam@125.22.91.2) |
| 15:07.56 | carrar | of both the caller id and exten |
| 15:08.32 | carrar | them pop those 3 diigts into a var |
| 15:08.52 | carrar | send to same 2 digit extension with the var in front of it |
| 15:08.58 | carrar | BING |
| 15:08.59 | carrar | BAMG |
| 15:09.01 | grohne | WIMPy: hmm. ok. anyway I probably cannot use misdn with this hardware. |
| 15:09.12 | carrar | OPPAN GANGNAM STYLE !!!!!!!!! |
| 15:09.16 | [TK]D-Fender | AkkerKid, First not call your calling device an "extension". If you configured the CALLERID of the SIP DEVICE to match some arbitrary extension (extens = line in extensions.conf) then you can use that. |
| 15:09.39 | WIMPy | grohne: What hardware? |
| 15:09.42 | [TK]D-Fender | AkkerKid, So if the callerid of the calling phone holds it, then take that instead of a fixed prefix. |
| 15:09.59 | WIMPy | And yes, it has many limitations, like DSS1 only. |
| 15:10.08 | grohne | WIMPy: mostly junghans bris and pris |
| 15:10.35 | WIMPy | grohne: Sure they work with mISDN. Anything HFC based does. |
| 15:10.42 | WIMPy | Including USB dongles. |
| 15:11.07 | niluje | file: I'm answering here to avoid pollution on the dev channel |
| 15:11.13 | WIMPy | The real limitation ATM are not in misdn, however, but in LCR. |
| 15:11.15 | file | niluje, thanks |
| 15:11.26 | file | niluje, that was the mailing list post that mjordan was referring to |
| 15:11.30 | niluje | actually, the functionality I need to implement is this one: https://issues.asterisk.org/jira/browse/ASTERISK-19571 |
| 15:11.40 | grohne | hm. ok. on the other hand I am already working with dahdi and libpri. and misidn doesn't fix the "bugs" I am currently working on. switching seems more like a risk than a gain |
| 15:11.57 | AkkerKid | Alright, thanks all. I've got to figure how to get the source's channel variables out of freepbx parsable... or something... |
| 15:12.03 | niluje | it works fine, the problem is there are memory leaks introduced with the patch - I did not constat them, I've been told so |
| 15:12.32 | niluje | the patch seemed to be complicated for what it does, so I thought there would be a simpler way to play a sound |
| 15:12.32 | mjordan | there's a lot more than just memory leaks that that patch will introduce |
| 15:12.40 | [TK]D-Fender | AkkerKid, I just gave you something very specific to consider. IS it the same number you are looking for? |
| 15:12.43 | mjordan | race conditions galore |
| 15:12.51 | niluje | mjordan: right, I read the comments ;p |
| 15:13.06 | grohne | WIMPy: thanks for your insight |
| 15:13.35 | AkkerKid | [TK]D-Fender: I'm looking into what FreePBX is setting for that. Thanks! |
| 15:13.51 | [TK]D-Fender | AkkerKid, You shouldn't have to look. This is something you should immediately know. |
| 15:13.59 | AkkerKid | My second problem is that I'm programing under Elastix/FreePBX |
| 15:16.07 | [TK]D-Fender | AkkerKid, And you keep adding extra facts that aren't answering the question.... |
| 15:16.44 | WIMPy | grohne: Possible. Sure. I don't know what your needs are, so I cant give a recommendation anyway. |
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| 15:18.49 | AkkerKid | [TK]D-Fender: You're losing me. Am I going the wrong direction? |
| 15:19.04 | [TK]D-Fender | AkkerKid, What did I ask you? |
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| 15:20.21 | AkkerKid | Honestly, I'm not sure. |
| 15:21.58 | AkkerKid | but the direction I'm going is this exten => _#xx,1,Dial(${something}${EXTEN:-2,2}) |
| 15:22.17 | AkkerKid | without the opund once i'm out of testing of course. |
| 15:22.30 | AkkerKid | pound* |
| 15:23.24 | AkkerKid | the something is somehow populated with the first three digits of the extensions associated with the caller's device |
| 15:24.44 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:26.29 | [TK]D-Fender | AkkerKid, I asked you if the SIP DEVICE that was placing the call had a CALLER ID that matched the "extension" that was related to it. |
| 15:26.48 | AkkerKid | I believe it does not. |
| 15:28.39 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-81-32-194.hsv.bellsouth.net) |
| 15:31.37 | [TK]D-Fender | AkkerKid, So when you dial from that phone the number the person you call is seeing is NOT the same as the 5 digit "extension" they would dial from their phone to call it back? |
| 15:32.16 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:32.28 | AkkerKid | I suppose internally, it does but externally it doesn't so it should work for my purposes. |
| 15:32.49 | [TK]D-Fender | AkkerKid, You are now being even more vague about some implication of intern VS external. |
| 15:32.51 | AkkerKid | this is what i've got now... |
| 15:32.53 | AkkerKid | Dial(${CHANNEL:4:2}${EXTEN:-2,2}) |
| 15:33.12 | [TK]D-Fender | AkkerKid, I asked a very simple question.... |
| 15:33.14 | AkkerKid | sorry, Dial(${CHANNEL:4:3}${EXTEN:-2,2}) |
| 15:33.17 | [TK]D-Fender | there is no "maybe about it. |
| 15:33.30 | [TK]D-Fender | we are talking about a SIP device registered to a single server. |
| 15:33.35 | [TK]D-Fender | I asked about the CALLING phone. |
| 15:34.05 | AkkerKid | then as far as we're concerned, yes. |
| 15:34.26 | [TK]D-Fender | [11:09] <[TK]D-Fender> AkkerKid, First not call your calling device an "extension". If you configured the CALLERID of the SIP DEVICE to match some arbitrary extension (extens = line in extensions.conf) then you |
| 15:34.43 | [TK]D-Fender | [11:09] <[TK]D-Fender> AkkerKid, So if the callerid of the calling phone holds it, then take that instead of a fixed prefix. |
| 15:35.43 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 15:36.19 | AkkerKid | understood. |
| 15:36.56 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:39.18 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
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| 15:56.21 | AkkerKid | I've got it working. Thanks for your help [TK]D-Fender! exten => _#xx,1,Goto(from-internal,${CALLERID(number):0:-2}${EXTEN:-2},1) |
| 15:59.15 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
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| 16:22.56 | ChannelZ | I think CALLERID(number) is deprecated, use CALLERID(num) |
| 16:26.48 | *** join/#asterisk sbszulu (~Sibusiso_@197.173.185.150) |
| 16:26.50 | carrar | so sad |
| 16:32.25 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zheezhuviejvqccu) |
| 16:36.24 | *** join/#asterisk sbszulu (~Sibusiso_@197.173.185.150) |
| 16:49.45 | leifmadsen | CALLERID(number) and (num) are the same thing and I've never heard of either being deprecated |
| 17:00.50 | ChannelZ | my bad then, I thought it was going/had gone away |
| 17:01.44 | ChannelZ | it's certainly not listed in the function help. But maybe I'm thinking of the old channel variables or something |
| 17:02.18 | *** join/#asterisk jsjc (~Adium@27.Red-2-137-30.dynamicIP.rima-tde.net) |
| 17:03.05 | jeffspeff | what's a good way to secure a sip based * with lots of remote users that work from their homes that have dynamic IP addresses? |
| 17:05.55 | *** join/#asterisk j4m3s__ (~j4m3s_@adsl-98-81-85-124.hsv.bellsouth.net) |
| 17:08.20 | [TK]D-Fender | Strong passwords, and a firewall strategy (fail2ban, port knocking, other login), or VPN, etc. |
| 17:11.22 | *** join/#asterisk jsjc (~Adium@138.Red-83-56-193.dynamicIP.rima-tde.net) |
| 17:15.15 | jeffspeff | using strong passwords, firewall has to be open for SIP because i never know what network or IP a legitimate user will be registering from... the Polycom phones (IP330, 335, and 340) don't support VPN like the Cisco SPA's do. |
| 17:17.04 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:17.04 | *** part/#asterisk sbszulu (~Sibusiso_@197.173.185.150) |
| 17:17.08 | [TK]D-Fender | jeffspeff, There is a question about starting open and then CLOSING OFF attackers |
| 17:18.14 | jeffspeff | [TK]D-Fender, what do you mean by "closing off" ? |
| 17:19.02 | jeffspeff | I monitor the system as much as possible and block the IP's of networks I see that are spamming AND not located in the US. but that only limits it to attacks from outside the US |
| 17:19.29 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 17:25.05 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:30.29 | *** join/#asterisk j4m3s__ (~j4m3s_@adsl-98-81-34-31.hsv.bellsouth.net) |
| 17:31.54 | Qwell | j4m3s_: ohai |
| 17:33.25 | *** join/#asterisk j4m3s___ (~j4m3s_@adsl-98-81-90-238.hsv.bellsouth.net) |
| 17:41.55 | [TK]D-Fender | jeffspeff, You should already have been Googling "fail2ban" and have come up with that answer..... |
| 17:43.48 | jeffspeff | [TK]D-Fender, but how accurate is that? |
| 17:44.16 | [TK]D-Fender | jeffspeff, How accurate is what? |
| 17:45.51 | jeffspeff | i don't see how it can explicity secure the system... it'd be good for catching brute force hacks and others of the like |
| 17:46.36 | [TK]D-Fender | jeffspeff, And what would qualify as "explicit"? |
| 17:46.52 | jeffspeff | idealy i would like to set a host ip for each sip user, but as they have dynamic IP's i can't. |
| 17:47.10 | [TK]D-Fender | jeffspeff, what that's one way ... of NOT answering the question.... |
| 17:47.33 | [TK]D-Fender | well* |
| 17:49.00 | jeffspeff | explicit as in only these group of IP's are allowed to access the server, or IP 192.168.1.1 is the only one allowed to connect to extension/sip user 1234 |
| 17:51.09 | [TK]D-Fender | jeffspeff, Yes and you've already shot down being able to just enter those flat-out |
| 17:51.17 | [TK]D-Fender | So it isn't viable. So this ISN'T an option. |
| 17:51.43 | [TK]D-Fender | So what process can you possibly imagine that can be explicit when they can connect from ANYWHERE? |
| 17:51.49 | jeffspeff | right, so I'm trying to explore other alternatives to provide equivelant protection |
| 17:51.59 | [TK]D-Fender | You can't HAVE equivalent |
| 17:52.10 | [TK]D-Fender | The situations are not the same. |
| 17:52.33 | [TK]D-Fender | I want to be as safe from falling while in a plane as I am when I'm sitting on the ground. |
| 17:52.36 | [TK]D-Fender | NOT POSSIBLE. |
| 17:53.48 | [TK]D-Fender | So you have to do something else |
| 18:03.17 | *** join/#asterisk sustav (~vpp@nat/digium/x-vwjwuahuwiduvtzz) |
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| 19:08.59 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:12.17 | *** join/#asterisk morfin (~morfin@morfin.telenet.ru) |
| 19:12.20 | morfin | hello |
| 19:12.52 | morfin | is that possible to move CDR data from MySQL to PostgreSQL without losing data? |
| 19:15.09 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:15.12 | Qwell | morfin: sure |
| 19:16.03 | morfin | i need that because i don't think storing that in MySQL is good idea actually |
| 19:17.39 | Qwell | What's wrong with MySQL? |
| 19:18.42 | drmessano | The Internet + Oracle + Slashdot |
| 19:28.41 | morfin | actually nothing |
| 19:29.02 | morfin | but i have projects on PostgreSQL that should use CDR |
| 19:29.31 | morfin | and i think it would be much better if i'll have data on PostgreSQL |
| 19:30.43 | robl^ | if one insists on MySQL, at least check out MariaDB |
| 19:30.48 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:30.48 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:33.03 | TechSmurf | bashes his head |
| 19:33.27 | TechSmurf | My *now box has decided not to play ringing() when I call it |
| 19:33.32 | TechSmurf | wtf? |
| 19:35.10 | *** join/#asterisk d00gster (~doughant@173.195.1.109) |
| 19:38.22 | TechSmurf | it did it once after a reboot... now it's stopped again |
| 19:40.05 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 19:40.13 | TechSmurf | if I tell it to play MOH instead of ringing it works |
| 19:44.31 | [TK]D-Fender | TechSmurf, Where's the debug? |
| 19:45.40 | TechSmurf | I'm fried after all the shit I've tried to fix today. Please treat me like a noob |
| 19:46.10 | TechSmurf | (sans usual noob insults, preferrably) |
| 19:47.28 | [TK]D-Fender | TechSmurf, Where's the debug? <- |
| 19:47.32 | TechSmurf | What debug? |
| 19:47.42 | [TK]D-Fender | Show us us evidence of the problem. |
| 19:47.43 | TechSmurf | I don't know how to get what you call a debug |
| 19:47.45 | [TK]D-Fender | Show us the cal. |
| 19:47.52 | TechSmurf | I don't know what you even mean |
| 19:47.52 | [TK]D-Fender | Show us configs.Tell use what version of *. |
| 19:48.02 | TechSmurf | 2.0.2 i believe |
| 19:48.09 | [TK]D-Fender | That is not an Asterisk version # |
| 19:48.18 | TechSmurf | sry. as I said, fried |
| 19:48.44 | [TK]D-Fender | Yuo appear to be beyond any semblance of a clue. Take a break an come back when you're capable of processing things... |
| 19:48.52 | TechSmurf | I can't |
| 19:48.53 | [TK]D-Fender | You'll burn us out along with you otherwise |
| 19:49.07 | TechSmurf | I have to fix this stupid thing. but ok, I'll just do it myself |
| 19:52.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:52.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:06.35 | TechSmurf | laughs |
| 20:06.42 | TechSmurf | (insanely) |
| 20:07.28 | TechSmurf | destination == extension, no ring tones. ring group, no ring tones. terminate call -> play ringtones until caller hangs up, I get ring tones. |
| 20:07.35 | Sprocks | if I use a DNS for "host=" in my sip.conf will that cause problems? |
| 20:07.47 | TechSmurf | Sprocks: so long as dns works, no |
| 20:10.14 | Sprocks | ok so as long as I keep it pointing to the ip of the machine in question I should have no problems |
| 20:10.28 | TechSmurf | as far as I've experienced. |
| 20:10.56 | TechSmurf | if the machine's ip won't be changing, though, I'd consider it one more thing to possibly go wrong |
| 20:12.17 | TechSmurf | [TK]D-Fender: what levels of verbose and debug do you expect would best highlight such a problem? |
| 20:13.06 | [TK]D-Fender | TechSmurf, Everything maxxed clearly. Details of what is actually going on. We know nothing at all right now. |
| 20:13.22 | TechSmurf | ok. max 7? |
| 20:16.07 | TechSmurf | shakes his head |
| 20:17.29 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 20:17.53 | TechSmurf | it didn't like 'core set debug maxxed' |
| 20:18.03 | AkkerKid | LOL |
| 20:18.34 | TechSmurf | I was having a hard time coming up with a more polite way to get my point across. |
| 20:19.01 | TechSmurf | I shouldn't have to go google wtf maximum debug is when he could have "clearly" said a fucking arabic numeral. |
| 20:19.19 | *** join/#asterisk jkroon (~jkroon@dsl-244-21-47.telkomadsl.co.za) |
| 20:19.34 | AkkerKid | D-Fender had to deal with me earlier. He's not in a great mood. |
| 20:19.57 | TechSmurf | heh |
| 20:20.32 | TechSmurf | beats Qwell with a deadly tuna. |
| 20:20.49 | jkroon | hi guys, when using call origination it seems the translation paths end up being sub-optimal (asterisk 1.8.12.1 and 1.8.15.1 confirmed). in particular even though both SIP channels of the resulting bridge uses g729 asterisk ends up using 2 g729 encoders and decoders ... this doesn't seem right. |
| 20:21.21 | jkroon | does anyone perhaps know why this happens and possibly how to avoid it? (ie, get it to function in pass-through mode) |
| 20:21.36 | AkkerKid | jk: codec from digium? |
| 20:21.43 | jkroon | yes |
| 20:21.47 | AkkerKid | are you recording anything? |
| 20:21.52 | AkkerKid | mixmon? |
| 20:22.09 | [TK]D-Fender | That would do it |
| 20:22.12 | jkroon | AkkerKid, with and without recording it happens. with recording it's two encoders and four decoders, using monitor |
| 20:22.19 | jkroon | without recording it's 2/2 |
| 20:22.50 | jkroon | in both cases it should be 0/0 if i'm not mistaken |
| 20:23.00 | AkkerKid | This issue has surpassed my Level of tech support. Allow me to forward you to level 2 service. Please hold. |
| 20:23.10 | jkroon | rofl |
| 20:23.26 | jkroon | [TK]D-Fender, any bets? |
| 20:23.34 | jkroon | pointers at how to debug? |
| 20:23.39 | AkkerKid | if you're recording, asterisk will have to translate both directions of audio into your recording format |
| 20:24.21 | *** join/#asterisk niluje (~niluje@82.227.67.242) |
| 20:24.38 | jkroon | AkkerKid, agreed, but I'd expect it to use 2 decoders and 1 encoder with mixmonitor, not 2 encoders and 4 decoders. |
| 20:25.22 | TechSmurf | What is the log supposed to say when it starts playing the default ringing tone back to the caller? |
| 20:25.24 | jkroon | with monitor (not mixmonitor) i expect to see 0/0 since it just copies the raw streams to files (permitting the codecs are the same as the channels) |
| 20:25.33 | [TK]D-Fender | What is monitor without recording? |
| 20:25.46 | AkkerKid | {channel} is ringing ? |
| 20:26.04 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 20:26.29 | [TK]D-Fender | jkroon, Incorrect. |
| 20:26.41 | jkroon | [TK]D-Fender, there are two mechanisms for recording, Monitor() and MixMonitor(), MixMonitor mixes in real-time and is generally considered better from the most documentation. |
| 20:26.45 | jkroon | [TK]D-Fender, please explain ? |
| 20:26.49 | [TK]D-Fender | jkroon, Monitor ercords in a FORMAT. the file it says could be in something OTHER than the call. |
| 20:27.19 | [TK]D-Fender | jkroon, Have an end-to-end G.729 call? TFB when when you save as **wav** |
| 20:27.23 | jkroon | [TK]D-Fender, that's why I said permitting the ... nm, codecs vs format, ok, permitting the FORMAT is the same as the channels. |
| 20:27.37 | [TK]D-Fender | jkroon, Not the channels.. the RECORDING FILE FORMAT |
| 20:28.04 | jkroon | no, end-to-end G.729, with MixMonitor to G.729 => 2 decoders, 1 encoder, same, but with Monitor usually gets met 0/0. |
| 20:28.22 | [TK]D-Fender | Mix monitor = TRANSCODE 100% of the time |
| 20:28.41 | jkroon | [TK]D-Fender, ok, but I see 2/2 even without either of Monitor or MixMonitor. |
| 20:28.42 | [TK]D-Fender | Because mixing requies decodeing , mix, re-encode |
| 20:28.50 | [TK]D-Fender | jkroon, Show us the calls. |
| 20:29.08 | jkroon | verbosity? |
| 20:29.12 | jkroon | 4 sufficient? |
| 20:29.16 | [TK]D-Fender | collect that up, I'm heading home. Back in 20ish |
| 20:30.00 | TechSmurf | stabs d-fender |
| 20:31.40 | AkkerKid | a little murder never killed anyone... |
| 20:32.55 | jkroon | rofl |
| 20:33.22 | TechSmurf | reaches critical mass and obliterates freenode entirely. |
| 20:33.27 | drmessano | Funny that everyone wants his help, but they talk about him behind his back and complain about his methods |
| 20:33.43 | TechSmurf | actually, I'm pretty much done with his help for tday |
| 20:33.48 | TechSmurf | he's as burned out as I am... |
| 20:33.55 | TechSmurf | we probably would kill each other. |
| 20:34.45 | TechSmurf | puts the keyboard down and goes to find a few minutes of zen before he goes back to googling. |
| 20:35.43 | *** join/#asterisk sustav_ (~vpp@nat/digium/x-whpcrbhgtdtthmrh) |
| 20:36.44 | AkkerKid | I can't understand being grumpy. If I spend a bunch of time trying to wring information out of inquisitive people without being paid, i'd be grumpy too. |
| 20:36.57 | AkkerKid | i can understand* |
| 20:37.54 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:39.15 | jkroon | drmessano, not funny at all actually. |
| 20:40.38 | drmessano | jkroon, indeed |
| 20:42.04 | jkroon | ok, i've got a theory but it's wack. |
| 20:42.49 | jkroon | my call origination uses Local/ext@aanswer so as to manipulate things a bit before I issue Dial(), now it seems that the Local/ channel uses "ulaw", so at that stage it transcodes .., |
| 20:43.33 | jkroon | now it dials out to another SIP/ or IAX2/ channel, which ends up being g729, as soon as it gets bridged the Local/ channel gets optimized away, but the transcoding remains ... ?!? |
| 20:44.57 | *** join/#asterisk tompaw (~tompaw@93.159.144.165) |
| 20:46.22 | TechSmurf | AkkerKid: I actively opped for efnet #linuxhelp for years. grumpy just happens :) |
| 20:47.38 | TechSmurf | should go ban some people.. that always made him feel better |
| 20:48.16 | jkroon | AkkerKid, imagine never being left alone ... as in ever. whenever you're online, people want your knowledge. half the time being ungrateful for that help, being demanding as hell and insisting that you owe them support. i'd get a little more than just grumpy. |
| 20:48.46 | TechSmurf | jkroon: It's actually quite easy to ignore. |
| 20:49.13 | drmessano | But he doesn't |
| 20:49.20 | jkroon | probably, but still |
| 20:49.33 | TechSmurf | I mean, you have to *really* ignore it tho |
| 20:49.46 | TechSmurf | because once you're invested in fixing someone's problem... |
| 20:49.59 | jkroon | jip. |
| 20:50.17 | TechSmurf | heaven help that guy you throw ideas at for 20 minutes who quits irc without saying whether it worked or a thanks |
| 20:50.36 | jkroon | and it's not always possible, and then people really go like "but wtf?!? you said you'd help" |
| 20:50.41 | jkroon | ooh yea, that's annoying. |
| 20:51.10 | TechSmurf | I'm the bastard crafting a shitlist entry in response to that one. |
| 20:51.19 | sruffell | …let's not forget that on the mailing lists as well….questions …answers…..then nothing... |
| 20:51.23 | TechSmurf | come back. ask another question. see what happens |
| 20:51.24 | drmessano | He helps everyone, regardless of how helpless and useless they are. That's the real issue. If you can't be slightly empowered to follow directions and generally act like a useless twit, you earn being labeled a useless twit. If you hang in there for two hours helping the useless twit, you earn being able to apply the label. |
| 20:51.35 | TechSmurf | sruffell: I couldn't do mailing list support. |
| 20:51.41 | TechSmurf | I'd go batty |
| 20:51.49 | sruffell | btw, who are we talking about (can't follow IRC) |
| 20:51.51 | sruffell | ? |
| 20:52.12 | TechSmurf | fender.. but it seems to have digressed to all of our support gripes |
| 20:52.21 | sruffell | heh |
| 20:52.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.31 | jkroon | timing ... |
| 20:52.33 | drmessano | SHHHH.. he's back |
| 20:52.35 | TechSmurf | right? |
| 20:52.43 | TechSmurf | wanders back into his time-out corner |
| 20:52.53 | jkroon | wb [TK]D-Fender |
| 20:52.55 | sruffell | I like to make sure mailing list threads are accurate since they get picked up by google. I figure trying to help people on the mailing lists has a good chance of preventing the next question from ever being asked. |
| 20:53.00 | jkroon | almost ready with a pastebin for you |
| 20:53.12 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:53.23 | sruffell | but…there is only a really small subset of things that I can help people with anyway…so I'm not going batty. |
| 20:53.32 | jkroon | sruffell, that's a good aim :) |
| 20:53.34 | TechSmurf | I think I'll just nuke my pbx next time I get physical access |
| 20:53.47 | jkroon | no no no |
| 20:53.51 | TechSmurf | dumbest issue I've ever tried to chase down |
| 20:54.01 | jkroon | no nukes ... we probably want the city around it still :p |
| 20:54.14 | TechSmurf | it's ok |
| 20:54.17 | jkroon | TechSmurf, what're you stuck on? (can't guarantee help) |
| 20:54.19 | AkkerKid | social selection |
| 20:54.19 | TechSmurf | The box is in reno |
| 20:54.20 | drmessano | The more data we have indexed the less people use it. The problem is that there's so much data out there, and people hold their time in such a high regard, they refuse to follow a link that may not necessarily apply to their issue. |
| 20:54.29 | TechSmurf | I'll just drive it down to the NTS first. |
| 20:54.37 | TechSmurf | nobody will notice |
| 20:54.53 | *** join/#asterisk Tom_123 (~b0ot002@cpe-69-207-81-7.rochester.res.rr.com) |
| 20:55.12 | Tom_123 | I don't really have a good concept of DID in the voip world |
| 20:55.19 | TechSmurf | jkroon: It stopped playing ring tones back to callers |
| 20:55.24 | Tom_123 | wouldn't all voip by default really be supporting DID? |
| 20:55.25 | drmessano | You tell someone to google for some specific term, and they can't be bothered filtering search results.. They want YOU to be their Lucky button. |
| 20:55.33 | TechSmurf | Tom_123: yes |
| 20:55.34 | jkroon | TechSmurf, channel tech? |
| 20:55.47 | WIMPy | Tom_123: I'd say the opposite. |
| 20:56.01 | WIMPy | But that may (as usual) depend on the definition used. |
| 20:56.11 | jkroon | i had some interesting stuff with 180 Ringing vs 183 Progress on SIP ... |
| 20:56.34 | TechSmurf | hrm |
| 20:56.35 | sruffell | drmessano: heh..yeah….and those people I have to just ignore *unless* they are polluting the Google cache with just plain wrong informatin. |
| 20:56.42 | TechSmurf | there's something I hadn't pondered |
| 20:56.52 | TechSmurf | is the ring coming from my pbx or the remote pbx |
| 20:57.15 | jkroon | well, enjoy the sniffing :) |
| 20:57.31 | TechSmurf | oh wait, I did ponder that |
| 20:57.48 | TechSmurf | I called from my cell, still no ring tone |
| 20:58.00 | TechSmurf | all the handsets in the office ring just fine |
| 20:58.09 | TechSmurf | but the caller gets dead air till it's answered |
| 20:58.25 | anonymouz666 | sruffell: do you recommend to always use irqbalance? |
| 20:58.33 | TechSmurf | if I switch the route destination to "play ringtones until caller hangs up", it does that. |
| 20:58.51 | TechSmurf | if I switch the announce from ring to any MOH, that works too |
| 20:59.41 | TechSmurf | and I can't see a single bloody thing in the logs |
| 20:59.47 | WIMPy | anonymouz666: In a hopeless situation I found out that dahdi can work better with only one CPU. |
| 20:59.58 | TechSmurf | all it talks about is the ringing the sip handsets |
| 21:00.18 | anonymouz666 | what do you mean by "hopeless situation"? |
| 21:00.38 | WIMPy | Far too little CPU for SWEC. |
| 21:00.41 | TechSmurf | jkroon: and after rebooting the pbx, it works correctly once. |
| 21:01.47 | jkroon | TechSmurf, now that sounds like one of those nuke-worthy issues. |
| 21:01.52 | TechSmurf | right? |
| 21:01.53 | anonymouz666 | heh... I always with lots of CPUs |
| 21:02.01 | anonymouz666 | I am |
| 21:02.13 | anonymouz666 | at least 4 |
| 21:02.26 | WIMPy | Yes, but I go a little further with only one instead of two. |
| 21:02.28 | TechSmurf | not my ideal answer, but chasing down this snafu is like running a marathon for a diet coke. |
| 21:02.28 | jkroon | WIMPy, too little cpu for swec? |
| 21:02.42 | WIMPy | yes |
| 21:02.44 | jkroon | ok, so internally SIP to SIP everything is OK? |
| 21:02.52 | TechSmurf | yep |
| 21:03.05 | jkroon | but when you come in from some external source no ringing on the "external" side? |
| 21:03.13 | TechSmurf | yep |
| 21:03.39 | jkroon | hmm, for interests sake, what happens if you put a Progress() somewhere on the incoming path forcing the local asterisk to generate ringing? |
| 21:03.44 | TechSmurf | externally via iax trunk no issue either |
| 21:03.57 | jkroon | so only from dahdi? |
| 21:04.14 | TechSmurf | third party sip provider |
| 21:04.22 | jkroon | is indications.conf set up correctly? |
| 21:04.41 | TechSmurf | afaik. I've never touched it and everything works with our old sip provider |
| 21:04.47 | jkroon | ok, try forcing local indications ... perhaps they don't deal correctly with receiving a simple 180 ... |
| 21:05.15 | TechSmurf | would the "play ring tones until caller hangs up" black hole do that itself? |
| 21:05.45 | [TK]D-Fender | sees we are still firmly in MaybeLand |
| 21:05.56 | jkroon | [TK]D-Fender, always :) |
| 21:06.10 | jkroon | TechSmurf, try generating it yourself by using Progress(). |
| 21:06.24 | [TK]D-Fender | A notable suburb of GuessVille |
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| 21:10.19 | jkroon | [TK]D-Fender, http://pastebin.com/Pw00r6z1 - two examples |
| 21:10.54 | jkroon | first is without any recording, SIP and IAX/2 channels, both g729, so should just function in pass-through, but ends up using 2/2 indicating that it's transcoding. |
| 21:11.52 | jkroon | my theory currently is that because I originate via a Local channel initially for some reason it decides to transcode via ulaw |
| 21:12.14 | jkroon | any suggestions to get me a little further along? |
| 21:13.33 | [TK]D-Fender | jkroon: first call had NO monitoring in it that I could see |
| 21:13.51 | [TK]D-Fender | jkroon: And I dont see SIP DEBUG for the first leg of the call. |
| 21:14.25 | jkroon | [TK]D-Fender, that's the point ... no monitoring - why is it using up codecs? |
| 21:14.31 | jkroon | adds the sip debug |
| 21:16.03 | jkroon | http://pastebin.com/qjX56mA1 |
| 21:16.36 | jkroon | wonders why pastebin's of * dialplan code always seems to trigger a captcha ... |
| 21:17.48 | jkroon | hmm, is that snom even allowed to respond directly with 180 ringing instead of 100 trying? |
| 21:19.31 | wdoekes | 100 trying is not mandatory |
| 21:19.53 | jkroon | wdoekes, ok, so i learn something ... thanks. |
| 21:19.59 | wdoekes | 200 directly is ok too, as long as it's quick enough |
| 21:20.28 | jkroon | very interesting ... so any response to an INVITE is really OK as long as it's quick enough then I'm guessing? |
| 21:20.59 | wdoekes | the 100 is there so the uac doesn't have to guess whether the packet is lost |
| 21:21.09 | jkroon | makes sense |
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| 21:25.09 | [TK]D-Fender | jkroon: Ok, I don't see anything in there that should require transcoding..... what version of * is this? |
| 21:26.51 | jkroon | 1.8.15.0, tried 1.8.15.1 and 1.8.12.1 as well (other machines, same dialplan) |
| 21:27.55 | [TK]D-Fender | jkroon: I'm wondering if the Local channel in the middle is involved in some way. Do a test with a direct dial between them |
| 21:28.10 | jkroon | seen some odd behaviour with * "losing" translation paths too off recently, requiring a restart to get things working again ... may or may not be related, haven't had time to debug that either |
| 21:29.04 | jkroon | give me a few minutes, will need to reconfigure the phone to auto-answer ... |
| 21:34.53 | TechSmurf | hrm |
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| 21:37.50 | jkroon | [TK]D-Fender, ok, originate SIP/102 application Dial IAX2/ulsvoip-uls/0845158255: http://pastebin.com/TDeQK0dv - works as expected. |
| 21:38.10 | [TK]D-Fender | jkroon: No, no originate.. direct calls |
| 21:38.41 | jkroon | ok, don't have physical access to the phones, but they work as expected, no transcoding. |
| 21:39.02 | TechSmurf | well, it's definitely sending 180 back to the third party provider |
| 21:39.08 | jkroon | this only happens when I originate the calls via the Local/ channels. |
| 21:39.21 | jkroon | TechSmurf, i'm telling you, try using Progress to generate the ringing locally. |
| 21:39.55 | [TK]D-Fender | jkroon: Try using the /n suffix on those Local channel dials |
| 21:40.07 | jkroon | ok, one second. |
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| 21:48.03 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 21:48.09 | jkroon | [TK]D-Fender, as per request: http://pastebin.com/fKJ9zZH2 |
| 21:50.47 | jkroon | same result really as far as I can tell, not sure what you were hoping to learn? |
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| 21:52.08 | jkroon | except in this case it would actually make sense to use 2 encoders+decoders if the Local channel runs on something other than g729. |
| 21:52.10 | [TK]D-Fender | Wondering if the way Local's bridge back on answer is interfering with it and forcing it through slin, etc |
| 21:52.33 | [TK]D-Fender | because the Local layer is really the only differnce from the dirct one |
| 21:53.27 | TechSmurf | jkroon: Unfortunately I'm doing an excessive amount of googling to try to turn that into english |
| 21:54.01 | jkroon | ok, i'm reasonably sure that you're correct, in that if I originate SIP/102 directly into from-phone I don't see the behaviour, and the only logical conclusion that I can make is that somehow just having Local/ involved is causing the transcode to happen. |
| 21:54.23 | jkroon | perhaps not the "answer" portion, but something to do with the Local/ channel |
| 21:55.01 | [TK]D-Fender | jkroon: Nothing else stands out from there..... |
| 21:55.23 | jkroon | would it help at all to know what ${CHANNEL(nativeaudioformat)} is on the Local/ channel when performing the call routing? ie, just before the Dial() onto the IAX/2 channel? |
| 21:56.05 | jkroon | [TK]D-Fender, ok, how do I confirm/go about testing your theory? |
| 21:58.47 | TechSmurf | jkroon: progressinband=yes/prematuremedia=no fixed it |
| 21:59.17 | jkroon | TechSmurf, interesting. |
| 21:59.28 | jkroon | what does the SIP signalling look like after setting that? |
| 21:59.37 | jkroon | still sending 180s? or 183s? |
| 22:00.08 | jkroon | [TK]D-Fender, just before Dial() I added a NoOP, with the following info: audioreadformat=slin,audiowriteformat=slin,audionativeformat=slin |
| 22:00.40 | jkroon | so IMHO that pretty much confirms it for me ... Local/ is forcing it via slin ... how to prevent that now becomes the appropriate question? |
| 22:01.21 | TechSmurf | 100->180->183 |
| 22:02.15 | jkroon | TechSmurf, Progress() probably would have achieved the same, basically as the call comes in you just call Progress() on that channel, which will immediately send a 183 to the calling channel, and from there asterisk will (if I understand it correctly) signal "inband" |
| 22:02.52 | TechSmurf | yes, but I have no idea wtf you mean, and that's as close as google got me ;) |
| 22:03.24 | jkroon | exten => ???,n,Progress() ... _somewhere_ in your dialplan that will get executed on the calls :) |
| 22:04.26 | jkroon | [TK]D-Fender, is there any way to inform to the Local channel what it's "native" format should be? |
| 22:05.43 | jkroon | from what I can tell, if I do a Dial(Local/foo) somewhere in my dialplan it inherits from the channel executing the Dial(), but obviously with originate there will be only the two Local/foo-1;2 channels and slin is the obvious choice there, but they don't switch to whatever other formats are on either end ... ? |
| 22:06.21 | jkroon | so in my originate case I'd expect the NoOP() above to report g729 since that call leg has already been bridged to a g729 channel ? |
| 22:06.27 | jkroon | or am I missing something? |
| 22:11.56 | jkroon | [TK]D-Fender, ok, local_new initializes those formats, and slin seems as sane a default as any from app_originate, from whence AST_FORMAT_SLINEAR is hard-coded. |
| 22:12.10 | [TK]D-Fender | jkroon: Not that I can see..... This was just a suspect on my side now confirmed... but I dont know of a reason for it... |
| 22:13.25 | jkroon | ok, let's assume i'd like to do something about it ... specifically, when ;1 gets bridged, I'd like to see if it's possible to switch from slin to whatever format ;1 is bridged to (which is obviously possible since ;2 isn't connected yet), how would I go about doing that? |
| 22:14.06 | jkroon | is there some hook in chan_local that I can react on? |
| 22:14.50 | [TK]D-Fender | jkroon: this is a rather in-depth question you should really take to -dev. It has definitely left the scope of what I can assist you with |
| 22:15.23 | jkroon | [TK]D-Fender, hehe, that answer most certainly surprises me. it was just said a short whilst back that you are pretty knowledgeable. |
| 22:15.30 | jkroon | i thank you for sharing that knowledge. |
| 22:17.14 | jkroon | really thanks! you've been a great help. much appreciated. |
| 22:17.20 | [TK]D-Fender | jkroon: I am on the usage side, but you're right into the source at this point .... I've never coded for * directly |
| 22:17.41 | jkroon | you do work for digium right? |
| 22:17.56 | jkroon | or am i off the mark? |
| 22:18.26 | [TK]D-Fender | jkroon: At the very least I'm good for making sure we look at every layer that is presented to us... it higlighted the layer that is indeed responsible ... now how to change that to something usefull.... well thats for others to pick up for you |
| 22:18.52 | [TK]D-Fender | jkroon: No affiliation with any telephony related company |
| 22:19.00 | jkroon | hehe, ok, let's see if I can phrase this nicely to -dev. |
| 22:19.00 | [TK]D-Fender | jkroon: I'm just here" |
| 22:19.17 | jkroon | [TK]D-Fender, rofl, that's very kind of you. |
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| 22:30.33 | carrar | points to the bed in the corner of the room with the guitar case on it |
| 22:30.39 | carrar | thats where TK sleeps |
| 22:31.35 | carrar | Thats his katana next to it, advise you not to touch it |
| 22:31.59 | carrar | Where there is one katana, there is always another!! |
| 22:35.48 | [TK]D-Fender | carrar: I've got 3... |
| 22:36.01 | zenmaster | Hi, I was intrested in setting up filters on call lists. Is there any good examples on this? |
| 22:36.02 | carrar | see! |
| 22:36.11 | carrar | IPTABLES :) |
| 22:36.24 | carrar | block all UDP |
| 22:36.33 | zenmaster | haha. |
| 22:37.18 | [TK]D-Fender | But only 1 filter. If you put 2 the coffee can't pass through and you'll block up the whole system |
| 22:37.43 | carrar | espresso! |
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