| 00:05.38 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
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| 01:09.18 | viraptor | how can I debug a problem with res_config_mysql? I get the "Binding iaxpeers to mysql/asterisk/iax_friends" message, but the db entries are not displayed on the peer list :/ |
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| 02:37.29 | saxa | hi, can anybody point me where to begin looking to reduce the delay I have when I call from a SIP hardphone to my DAHDI outbound line ? |
| 02:38.15 | saxa | i can dial a number , but then everything is mutted for a long time , before i can hear any thing |
| 02:38.37 | saxa | the called party usually drops off as he doesnt hear anything also |
| 02:38.46 | saxa | but the call is done |
| 02:38.54 | saxa | any ideas ? |
| 02:42.25 | saxa | oh I use a grandstream gxp285 sip phone |
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| 03:12.40 | Kobaz | grandstream isn't known for being high quality |
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| 04:52.24 | tonsofpcs | anyone know of a decent J2ME SIP 'phone' client? or a decent, fully functional blackberry client? |
| 05:10.19 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
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| 06:45.24 | Dovid | is there any difference for T.38 on 1.6.2.X vs .1.8.X? |
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| 07:14.53 | BlackBishop | ancan I escape the , in ,Macro(mysql, INSERT INTO `logs` (`username`,`numar`) VALUES ('${CALLERID(num)}','${digito}')) so it sees the whole thing as ${ARG1} ? |
| 07:17.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:26.43 | BlackBishop | damn comma :| |
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| 07:30.03 | *** join/#asterisk astro__ (astro@silvia.drft.net) |
| 07:45.23 | benngard | any knows if MEMBERINTERFACE is broken in 1.8.2.3? |
| 07:47.16 | BlackBishop | I can't escape the damn commmaaaa... |
| 08:17.47 | ChannelZ | eh? |
| 08:21.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
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| 08:25.02 | p3nguin | He was being eaten alive by commas. |
| 08:26.01 | kaldemar | BlackBishop: no. use Gosub instead. |
| 08:36.07 | *** join/#asterisk Daejeo (~chatzilla@S010600265af0e1ad.cg.shawcable.net) |
| 08:36.27 | Daejeo | yum find anyon awake* |
| 08:37.30 | Dovid | lol |
| 08:37.32 | Dovid | nope. all dead here |
| 08:38.40 | p3nguin | It was likely the typo that caused a lack of relevant results. |
| 08:39.08 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:43.30 | Daejeo | Dovid: awakw |
| 08:43.39 | Daejeo | thanks yum |
| 08:43.47 | Daejeo | :) |
| 08:44.49 | BlackBishop | kaldemar: I'll just create 2 macros ... I only needed an insert and an update |
| 08:46.10 | p3nguin | Macros are deprecated in favor of Gosub() and Return(). |
| 08:47.01 | p3nguin | Although I find some places where I cannot figure out how to do some things using GoSub/Return and Macro does it just fine. |
| 09:01.49 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 09:03.26 | benngard | any1 knows how MEMBERINTERFACE is supposed to work? |
| 09:04.12 | benngard | either i am a complete fool or it is broken |
| 09:04.59 | p3nguin | What does core show function MEMBERINTERFACE say about it? |
| 09:05.33 | benngard | it is not a function it is a variable |
| 09:05.56 | benngard | should be set by Queue function |
| 09:06.16 | p3nguin | oh |
| 09:06.19 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 09:06.28 | p3nguin | What was the problem with it? |
| 09:06.34 | benngard | Queue sets for example QUEUENAME coreectly |
| 09:06.44 | benngard | MEMBERINTERFACE is empty |
| 09:07.49 | *** join/#asterisk SillyWalker (debian-tor@gateway/tor-sasl/sillywalker) |
| 09:07.55 | SillyWalker | hey ppl |
| 09:07.59 | benngard | hi |
| 09:08.35 | SillyWalker | just set up some asterisk/hylafax servers :p |
| 09:08.49 | SillyWalker | Both are awesome |
| 09:10.53 | SillyWalker | Well we are in need of someone to donate some SIP trunks for a good cause |
| 09:12.31 | SillyWalker | Anyone interested please check here http://typewith.me/oplibya-fax |
| 09:12.37 | SillyWalker | thanks! |
| 09:13.24 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 09:13.40 | p3nguin | I've got an Asterisk trunk for you. |
| 09:14.42 | SillyWalker | nice! We are in desperate need right now |
| 09:14.51 | SillyWalker | PM me for details pls |
| 09:14.51 | p3nguin | http://imagebin.org/138853 |
| 09:15.30 | SillyWalker | lolz |
| 09:15.40 | SillyWalker | fucking asterisk humor.... |
| 09:15.54 | SillyWalker | awww man you trolled me hard.. |
| 09:16.03 | p3nguin | Yep, because I cannot stand when people say "SIP trunk." |
| 09:16.51 | SillyWalker | Well excuse my french! |
| 09:17.07 | p3nguin | Il y a plus d'une façon de peler un chat. |
| 09:18.04 | SillyWalker | well any suggestions as to getting those faxes into libya? |
| 09:22.50 | astro__ | why fax libya |
| 09:22.58 | astro__ | they are busy rioting |
| 09:24.42 | *** join/#asterisk donttrustem (~Trustem@80.30.140.114) |
| 09:25.29 | donttrustem | hi guy's I have a client that cannot make outgoing calls but I can call them from my mobile but not my telefonica landline? |
| 09:25.35 | donttrustem | any ideas? |
| 09:26.48 | astro__ | what do the logs say |
| 09:27.08 | donttrustem | that the problem I don't have access to the logs :( |
| 09:27.15 | donttrustem | jst asking what it could be |
| 09:28.57 | p3nguin | Do you know anything about their configuration? |
| 09:30.30 | donttrustem | Well .... they use a Sysmaster switch not asterisk but all these system work very much the same. I think it a call routing issue |
| 09:33.14 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 09:36.44 | astro__ | anyone here run an fxo card with vsphere? |
| 09:36.59 | *** part/#asterisk SillyWalker (debian-tor@gateway/tor-sasl/sillywalker) |
| 09:37.00 | p3nguin | That's possible? |
| 09:37.22 | astro__ | i was told it is, just not good for high volume |
| 09:38.36 | p3nguin | I'm not sure how it would work at all. ESX server would have to either have some type of way to emulate the hardware or transparently pass it through to the guest. |
| 09:39.02 | astro__ | i spent the last few days consolidating all my junk servers at home to vsphere to save power |
| 09:39.21 | astro__ | at this old pbx would be nice to ditch |
| 09:39.26 | p3nguin | What kind of server hardware specs do you have? |
| 09:39.54 | astro__ | its just a whitebox asus board w/ core i7 quadcore w/ 20gig of ram |
| 09:40.11 | p3nguin | not bad |
| 09:40.38 | astro__ | seems to be doing the job for a home machine |
| 09:46.44 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 09:51.13 | titter | Sorry all for the join/parts ... Windows 7 and firewire is complete garbage. |
| 09:52.11 | p3nguin | I had to put you on ignore for JOINS/QUITS months ago. |
| 09:52.30 | titter | Ya I fixed that issue |
| 09:52.38 | titter | Silly dd-wrt |
| 09:53.56 | titter | Trying to get my Tascam working with Win 7 and it's just giving me fits. Granted I had it working, but random sync drops doesn't make for nice recordings. So I am screwing with different drivers. |
| 09:53.56 | p3nguin | If you know you have network problems, and you obviously know, you really shouldn't put IRC channels on autojoin. |
| 09:55.02 | titter | No no, I didn't realize since I wasn't home for that long period of time. 99% of the time I am just rdp'd into my home desktop. That is where Pidgin runs from. However traveling, I figured the hotel internet or my android tethering was to blame. |
| 09:55.54 | titter | Once I got home and someone complained about it I looked into it, dd-wrt was rebooting once an hour for a different purpose. |
| 09:56.12 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 09:56.15 | sigius | d |
| 09:58.18 | benngard | any1 knows a way to get the devicename (the called device of a not answered queue call? |
| 09:59.26 | titter | sleep time. |
| 09:59.46 | benngard | i thought MEMBERINTERFACE would do the trick but that variable is empty |
| 09:59.47 | sigius | I am capturing an asterisk phonecall with MixMonitor but dtmf tones are not heard in the capture (i think they are sent as rtp events (rfc2833)). How can I capture these as well, using MixMonitor or otherwise ? |
| 10:00.47 | p3nguin | You're probably never going to hear the full DTMF tones in a recording. Asterisk seems to always grab them. |
| 10:05.22 | sigius | p3nguin, well I can hear them when I am using my softphone (qutecom), I do not hear them when using SIPp. Dont know for a fact but i guess the first uses inband and the second uses rfc2833 |
| 10:05.46 | *** join/#asterisk geoffmcc (~geoffmcc@cpe-76-180-197-68.buffalo.res.rr.com) |
| 10:05.56 | sigius | hear them 'in the recording' that is |
| 10:15.20 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:21.38 | astro__ | say if i have pstn coming into a 110 block in a house which spreads off to all the analog phones, can i run a wire off the 110 block to my fxo and have asterisk answer |
| 10:22.13 | astro__ | or will all the analog phones ring |
| 10:23.57 | p3nguin | If you connect the house wiring to your Asterisk FXO port, all the phones will ring, but yes you can have Asterisk answer. |
| 10:24.36 | astro__ | so if i dont want them to ring im gonna have to put them all on fxs and make extensions ? |
| 10:25.14 | p3nguin | If you connect to Asterisk, it will make Asterisk act as another telephone on the wiring. |
| 10:25.57 | p3nguin | Ideally, you would have only Asterisk connected to the line wiring and use IP phones to talk to Asterisk. |
| 10:26.25 | astro__ | i want to stay totaly pstn for half the phones and use ip for the other half |
| 10:27.12 | astro__ | basicaly just have asterisk act as an ivr on the line |
| 10:27.17 | p3nguin | By adding an FXO to Asterisk, you make it just like any other phone. a special phone, but just another phone nevertheless. |
| 10:27.49 | astro__ | yea i know i was hopeing their was some sort of magic to make my situation work ;-) |
| 10:33.09 | astro__ | i think i have a few old sipura 2000s kicking around i can use |
| 10:51.35 | benngard | when u trasfer a call, can u play "normal ring-tones" instead of moh? |
| 10:52.55 | benngard | a calls b, when b press # and new extension a hears moh, but i would lilke toplay normal ring tones instead |
| 11:01.18 | ChannelZ | attended transfers put them on hold because it doesn't make sense for the other person to just hear ringing while you're talking to the person you're transferring them to before you actually complete the transfer |
| 11:03.34 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 11:03.51 | benngard | yes, but my boss wants it that way so i have to figure out a way to do it |
| 11:03.59 | benngard | i dont know why, but he hates moh |
| 11:04.09 | ChannelZ | so make moh silent |
| 11:04.15 | ChannelZ | or a ringing noise |
| 11:04.34 | benngard | i was actually thinkg of that |
| 11:05.27 | benngard | in features.conf i found, parkedmusicclass=default guess i can record ring tones i a file a replace default |
| 11:06.18 | ChannelZ | it seems dumb though to have a person hear ringing for 15 seconds while you're talking to the transfer destination saying "hey, I have this irate customer on the phone, he's being a total d-bag, he says his TV catches fire every time he turns it on...." |
| 11:06.45 | ChannelZ | no that's not what that does |
| 11:08.58 | ChannelZ | just don't configure music on hold and they will hear nothing. If your schizophrenic boss doesn't like music on hold then what does he expect people to hear when they get put on actual hold? ringing there too? |
| 11:09.55 | benngard | yes :( |
| 11:10.52 | ChannelZ | what kind of company is this? |
| 11:15.27 | benngard | i am working on it-department of a swedish company named Input Interiör, www.inputinterior.se |
| 11:15.39 | astro__ | quit |
| 11:17.07 | WIMPy | input interior = kitch furniture and output interior = bathroom furniture? |
| 11:17.40 | *** join/#asterisk guifort (~guifort@86.67.148.134) |
| 11:17.47 | guifort | Hi All |
| 11:20.46 | p3nguin | benngard: I'd tell the boss that it's a silly idea and explain why. What if it takes you like 2 or 3 minutes before you can complete the transfer? I'd be pretty upset if I had to listen to ringing for 3 minutes -- nice tunes would keep me calm. |
| 11:23.06 | p3nguin | benngard: Also, you don't have to use music in your music on hold... you can use a prerecorded voice message talking about products and such. |
| 11:24.01 | *** join/#asterisk admin0 (~admin0@cm180.omega140.maxonline.com.sg) |
| 11:28.59 | benngard | p3nguin: maybe i can get him to "buy" a prerecorded message |
| 11:29.19 | p3nguin | benngard: Any reason you wouldn't want to record it yourself? |
| 11:29.57 | p3nguin | WHAT THE FUCK |
| 11:30.01 | benngard | p3nguin: sorry, wrong word "buy" |
| 11:30.10 | p3nguin | <PROTECTED> |
| 11:30.47 | p3nguin | wondered why the system was /slow/ for a while |
| 11:31.25 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 11:31.33 | p3nguin | benngard: I don't understand what you are meaning. Why do you need to buy it? Why can't you record it yourself? |
| 11:32.24 | benngard | p3nguin: i was trying to say that my boss could accept a prerecorded message |
| 11:32.32 | p3nguin | It used up all its RAM and half its swap. |
| 11:32.48 | WIMPy | What is "it"? |
| 11:34.03 | p3nguin | the system, computer, PC, machine, box |
| 11:34.41 | p3nguin | I don't think I've ever run a system with THAT high of a load average before. I think the highest I ever had was around 350. |
| 11:45.24 | astro__ | i have never seen a load average like that before |
| 11:45.56 | p3nguin | Yeah, I have some runaway processes for an unknown reason. |
| 11:46.45 | *** part/#asterisk toresbe (toresbe@simula.gunkies.org) |
| 11:47.18 | p3nguin | A cron job apparently didn't exit each time it ran, and just kept starting a new instance. Rinse. Repeat 4000 times. |
| 11:48.13 | p3nguin | This isn't a new system nor is the cron task new. The only thing I have done to that computer is replace the hard drive about 11 days ago. |
| 11:48.41 | p3nguin | That was done via cp -ax ... ... so that wouldn't have changed the behavior of the script. |
| 11:49.59 | p3nguin | I'm surprised that old P3 Coppermine holds up to a load like that. |
| 11:52.06 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:54.16 | astro__ | i had to lug 25 p3s down 2 flights of stares last week to a dumpster |
| 11:54.44 | astro__ | old cases had so much more weight to them |
| 11:55.11 | p3nguin | They don't make anything like they used to. |
| 11:55.55 | astro__ | its gonna be nice when the new ones hit the dumpster it will be so much easyer |
| 11:55.56 | *** join/#asterisk bn-7bc (~bjarne-im@m90-137-139-161.cust.tele2.no) |
| 11:55.57 | p3nguin | SIGKILL isn't removing these processes for me, so I'm probably going to have to reboot. |
| 11:56.14 | p3nguin | I am down to 1079 processes from 4158. |
| 11:58.20 | p3nguin | load average is still 981.16, and that's just not good enough. Phone calls are clear again, finally! |
| 11:59.35 | p3nguin | It was so bad that autodial was canceled when I'd dial a number and then lift the handset. Never saw that before. |
| 12:06.50 | astro__ | why cant you just reboot? |
| 12:11.21 | p3nguin | I can, and I probably will. I was just hoping to resolve it without a reboot. |
| 12:12.07 | WIMPy | What are those procs doing that you can't get rid of them? |
| 12:12.29 | p3nguin | It's a whole mess of "mount -l" processes. |
| 12:13.24 | p3nguin | I don't see any reason for them to be stuck, nor should they ignore KILL. |
| 12:14.26 | WIMPy | Stuff stuck in I/O can be hard to get rid of. |
| 12:17.29 | p3nguin | I just ran mount -l from my prompt and it stuck, too. I wish I knew what caused this to start happening. |
| 12:18.31 | WIMPy | Whatever you try to mount seems to be unavailable in an way that it has hopes that it might become available again. |
| 12:21.44 | p3nguin | mount -l should simply be showing the mounted file systems with labels. |
| 12:21.53 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 12:22.41 | WIMPy | Ah, that was the only option. Well then it seems unable to read the label of some FS. |
| 12:23.06 | p3nguin | Now I'm curious what the heck would be wrong with that volume. |
| 12:23.09 | WIMPy | Sounds dangerous. |
| 12:24.14 | WIMPy | Do you have anything mounted that's been removed? |
| 12:26.31 | p3nguin | It's possible that the volume causing this hang up could have been removed. I can't go check it physically right now. |
| 12:27.45 | WIMPy | Maybe you can umount -f it. |
| 12:28.51 | p3nguin | The mtab does indicate that it's mounted. |
| 12:29.28 | p3nguin | But /proc/mounts does not. |
| 12:30.33 | WIMPy | I'd remove it manually from mtab then. |
| 12:31.19 | WIMPy | Won't help you to get rid of the stuck mounts but should prevent it from happening again. |
| 12:31.38 | astro__ | 0a:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 12:31.43 | astro__ | shitty |
| 12:32.04 | astro__ | junk drawer you have failed me |
| 12:32.19 | WIMPy | ??? |
| 12:33.35 | WIMPy | Weren't that thos standard HFC things that are sold like high end stuff? |
| 12:33.47 | p3nguin | I removed the line from mtab, then ran mount -l again and it DIDN'T hang! |
| 12:34.14 | astro__ | i thought it was a real digium fx0 but i think its a knockoff |
| 12:34.40 | WIMPy | Oh, an analog thing? |
| 12:35.00 | p3nguin | I guess I just have to reboot it since I can't kill off the stuck mount processes. |
| 12:35.09 | astro__ | kernel: NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 |
| 12:35.15 | p3nguin | Unless you have any other idea before I reboot it. |
| 12:36.14 | WIMPy | Not much. I have tried killing stuck processes by sending them a SIGSEGV. But I doubt that will work. |
| 12:36.28 | p3nguin | It's worth a shot. |
| 12:36.50 | WIMPy | But they shouldn't cause much harm siitting there doing nothing. |
| 12:37.18 | p3nguin | It's keeping the load average at 982. |
| 12:37.35 | WIMPy | That doesn't mean much. |
| 12:42.31 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 12:42.42 | p3nguin | The system is going down for reboot NOW! |
| 12:43.06 | p3nguin | That's not something I typically do. Ever. |
| 12:43.37 | *** join/#asterisk sourcode (~code@ppp-58-8-229-221.revip2.asianet.co.th) |
| 12:43.43 | WIMPy | was just about to suggest that you might be able to make it forget about the device it's waiting for, |
| 12:44.04 | p3nguin | What were you thinking I could try? |
| 12:44.11 | WIMPy | What kind of device was it? |
| 12:44.26 | p3nguin | USB mass storage |
| 12:45.00 | WIMPy | The it might be possible to forcefully unload usb-storage. |
| 12:45.15 | WIMPy | Or remove the SCSI ID. |
| 12:45.44 | p3nguin | I would have tried it, even though I bet it wouldn't have helped in this case. |
| 12:45.59 | WIMPy | Maybe not. |
| 12:46.47 | WIMPy | But the problem with removes USb storage usually doesn't happen any more, |
| 12:49.15 | WIMPy | d |
| 12:51.32 | p3nguin | It's still connected. I can mount it after the reboot. |
| 12:52.39 | WIMPy | Maybe it was shortly disconnected and then reconnected? |
| 12:52.57 | p3nguin | It's weird, though... I'm trying to delete some vim swap files that were left behind, and it says "rm: cannot remove `/mnt/etc/asterisk/.sccp.conf.swp': Read-only file system", but mount shows rw for that file system. |
| 12:53.39 | p3nguin | If it got write protected, that could have maybe caused the issue. |
| 12:54.08 | WIMPy | it has hardware WP? |
| 12:54.30 | p3nguin | yes, a switch on the device. |
| 12:54.40 | p3nguin | It must be switched because I can't edit any files. |
| 12:55.02 | WIMPy | I guess that could be an explanation. |
| 12:55.05 | p3nguin | Either that or it got damaged and is locked. |
| 12:55.57 | p3nguin | The reason I changed the hard drive was because the storm and power problems did a number on the hard drive. It could have damaged the USB drive too. |
| 12:56.09 | WIMPy | If you set it to remount-ro on errors, I'm not sure mtab would reflect the change. But you should see some message then. |
| 12:56.17 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 12:56.50 | WIMPy | BTW: I should set that on the new drive... |
| 12:57.06 | p3nguin | I'm not familar with "remount-ro on errors". |
| 12:57.31 | WIMPy | tune2fs -e remount-ro |
| 12:57.42 | WIMPy | Can also be used with mkfs. |
| 12:57.45 | p3nguin | /dev/sdb1 /mnt vfat ro,relatime,fmask=0022,dmask=0022,codepage=cp437,iocharset=iso8859-1,shortname=mixed,errors=remount-ro 0 0 |
| 12:57.57 | p3nguin | It is mounted that way. |
| 12:58.21 | WIMPy | Ok, so that would make it RO as soon as it hits a FS error. |
| 12:58.33 | WIMPy | Time to umount and fsck. |
| 12:58.35 | p3nguin | and mtab doesn't reflect it. |
| 12:59.29 | WIMPy | It'd need som user space support to change mtab. But /prc/mounts should get changed. |
| 13:00.16 | WIMPy | Why am I missing letters all the time. That's annoying. |
| 13:01.03 | p3nguin | My keyboard has a couple that it likes to skip. I usually see it and go back to correct before sending, but one will sometimes slip through. |
| 13:01.19 | p3nguin | I know r does it, and maybe t. |
| 13:01.46 | WIMPy | Not sure if it's the keyboard or me. |
| 13:02.04 | p3nguin | I know mine is the keyboard. |
| 13:02.15 | p3nguin | At least 98% keyboard. |
| 13:02.36 | p3nguin | 1) Truncate first to 0 bytes |
| 13:02.37 | p3nguin | 2) Truncate second to 0 bytes |
| 13:02.45 | p3nguin | I guess it wants me to pick. |
| 13:04.51 | p3nguin | Since I'm not good as guessing games, I'm restarting fsck.vfat with the -a option. |
| 13:05.58 | WIMPy | ... and hope it's better at guessing? |
| 13:06.23 | p3nguin | exactly |
| 13:06.43 | p3nguin | Why are these IronKey USB keys so bloody expensive? |
| 13:08.18 | p3nguin | Okay, fsck finished. The result is... |
| 13:08.24 | p3nguin | drumroll please |
| 13:08.27 | WIMPy | Why do all keyboards with exta function keys some of the usual keys in non-standard places? |
| 13:08.32 | Tozz_ | *drumroll* |
| 13:08.33 | p3nguin | Root directory is full. |
| 13:08.37 | WIMPy | It asks anyway? |
| 13:09.00 | p3nguin | I guess it can't do any work with a full volume. |
| 13:09.04 | WIMPy | Don't you just love msdos? |
| 13:10.04 | p3nguin | I knew I needed to replace the device for lack of space anyway. I guess I could format the replacement with an actual file system. |
| 13:10.31 | p3nguin | Is XFS safe for USB sticks? |
| 13:10.44 | WIMPy | Should be good enough if you just move the stuff to a subdir. |
| 13:10.45 | p3nguin | Or should I stick to ext3? |
| 13:11.26 | WIMPy | wouldn't use anything with a journal for flash storage because of the extra writes. |
| 13:11.51 | p3nguin | Since I use it for backup of asterisk, I really don't need to make it easily readable by those other OSs. |
| 13:12.13 | p3nguin | So XFS and EXT3 are no good? EXT2 is your suggestions? |
| 13:12.23 | WIMPy | I'd use ext2, yes. |
| 13:12.40 | p3nguin | I think that's a good idea. |
| 13:12.59 | WIMPy | And if you also read it, with -o noatime,nodiratime. |
| 13:14.41 | p3nguin | It's not supposed to get much activity. It's supposed to stay plugged in all the time, but it should get mounted, write everything related to the PBX onto it, then umount, once per day. |
| 13:16.04 | p3nguin | It's supposed to serve as a quick restore solution in the event of a mistake, to make it easier and faster to recover from it than going to the permanent backups for traditional restoration. |
| 13:17.54 | WIMPy | If you umount anyway, a journal would be pretty pointless. |
| 13:18.39 | WIMPy | You could remount RW/RO instead of mount/umount. |
| 13:21.32 | p3nguin | Speaking of USB flash memory, I recently bought this cool little USB stick made by Verbatim... it's the tiniest USB flash I have ever seen. I migrated my ESXi install to it from a regular hard drive to try to save power and reduce heat. |
| 13:22.18 | p3nguin | http://www.newegg.com/Product/Product.aspx?Item=N82E16820235039 |
| 13:22.45 | sigius | I am capturing an asterisk phonecall with MixMonitor but dtmf tones are not heard in the capture (i think they are sent as rtp events (rfc2833)). How can I capture these as well, using MixMonitor or otherwise ? |
| 13:23.18 | WIMPy | That's not small. There are these mini-SD cards with USB at the other end. |
| 13:23.38 | WIMPy | sigius: I think that's not possible. |
| 13:24.03 | p3nguin | I've seen the mini SD cards, but the adapters I have seen are pretty big in comparison to the mini SD. |
| 13:24.27 | *** join/#asterisk philfine__ (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 13:24.45 | WIMPy | Nono. the card itself has the SD connector at one end and USB at the other. |
| 13:25.15 | WIMPy | So 1/3 of the card disappears in the USB socket. |
| 13:25.28 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 13:25.58 | p3nguin | These Verbatim sticks are only 30.2mm x 12.7mm x 1.5mm. |
| 13:26.13 | WIMPy | If they continue that way you will need pliers to remove tha cards from the socket. |
| 13:26.21 | p3nguin | for sure |
| 13:26.41 | p3nguin | That's kind of funny. |
| 13:27.04 | WIMPy | Especially when you produce some short circuits in doing so. |
| 13:27.17 | p3nguin | Maybe they can provide tweezers in the package for the flash memory. |
| 13:29.13 | WIMPy | http://www.amazon.com/SanDisk-Ultra-Plus-2GB-Card/dp/B000EWI8IK |
| 13:29.33 | WIMPy | These were quite cool already, but I think I've seen smaller ones. |
| 13:30.19 | p3nguin | I should figure out what I need to do to replace this heavy-weight apache with lighttpd or nginx. |
| 13:30.26 | philfine__ | Hello everyone |
| 13:31.42 | philfine__ | Is there a way to create something as a function is a dialplan ? |
| 13:32.07 | WIMPy | http://www.dealextreme.com/p/world-s-smallest-microsd-transflash-usb-2-0-mini-card-reader-11294 |
| 13:32.19 | WIMPy | There's a nice little card reader. |
| 13:32.32 | philfine__ | Currently I was setting a context that would match to anything _. in case some other context Goto to it |
| 13:32.38 | WIMPy | philfine__: You mean IN the dialplan? |
| 13:32.42 | philfine__ | Yes |
| 13:32.57 | philfine__ | I have to outbound peers |
| 13:33.08 | WIMPy | No, you need to use variables. |
| 13:33.19 | philfine__ | I want to create a context for each performing several tasks such as logging, etc |
| 13:33.20 | p3nguin | You should almost never use _. as your pattern. |
| 13:33.39 | philfine__ | And the other context / matches would Goto to those contexts |
| 13:33.59 | philfine__ | I have no configuration on those context |
| 13:34.03 | p3nguin | Contexts are not matched by patterns; contexts are assigned per peer. |
| 13:34.04 | philfine__ | Only Gotos go there |
| 13:34.08 | sigius | WIMPy, ok , too bad, thanks |
| 13:34.26 | p3nguin | The peer match (or lack of) determines the context where the call is sent. |
| 13:34.39 | p3nguin | Once the call is in the context, then it has to match an extension. |
| 13:35.04 | p3nguin | _. is a terrible extension to match. You should almost ever use that pattern. |
| 13:35.23 | philfine__ | It is just that I do not want to repeat all the Verbose, LOG, perform the call in every single rule that needs to redirect a call out |
| 13:35.57 | WIMPy | uses _. or _! a lot. |
| 13:35.58 | p3nguin | Yeah? You could use _X. as long as every call will have at least two numbers. |
| 13:36.18 | WIMPy | And starts with a digit. |
| 13:36.19 | *** join/#asterisk viraptor (~viraptor@distillery.viraptor.info) |
| 13:36.40 | viraptor | hi all |
| 13:36.45 | p3nguin | wimpy: When a person doesn't even know the difference between a context and an extension, he probably shouldn't be using _. at all. |
| 13:36.46 | viraptor | how can I debug a problem with res_config_mysql? I get the "Binding iaxpeers to mysql/asterisk/iax_friends" message, but the db entries are not displayed on the peer list :/ |
| 13:37.09 | philfine__ | I do know the difference |
| 13:37.27 | p3nguin | Just a couple minutes ago you were calling _. a context. |
| 13:37.56 | philfine__ | If I did that was a mistkae |
| 13:37.59 | philfine__ | :D |
| 13:38.12 | WIMPy | Or maybe it's just a language thing? |
| 13:38.27 | philfine__ | I have to context outboud1 outbound2 in my dialplan |
| 13:38.27 | p3nguin | very likely |
| 13:38.57 | p3nguin | You know I only speak and read English. |
| 13:39.21 | philfine__ | sorry, I am native in Portuguese |
| 13:39.41 | philfine__ | So there might be some language things :) |
| 13:40.13 | philfine__ | Can I post my dialplan somewhere and you comment on it ? |
| 13:40.19 | p3nguin | Yes. Continue with your case please. |
| 13:40.20 | WIMPy | ~pb |
| 13:40.20 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:40.42 | p3nguin | pastebin.com is usually good enough. |
| 13:41.41 | philfine__ | Inside outbound1 I was setting a extension matching ._ to and performing logging, Dial though right channel. |
| 13:41.41 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 13:41.46 | philfine__ | Same in outbound2 |
| 13:42.59 | philfine__ | And in some context lets say "national-call" I do a GotoIfTime and depending on the time of the day I redirect to one of this contexts |
| 13:43.23 | philfine__ | either outbound1, outbound2 |
| 13:43.39 | philfine__ | Anything really bad until now ? |
| 13:44.14 | philfine__ | neither outbound2 and outbound1 are defined as any device contexts |
| 13:44.22 | WIMPy | no |
| 13:44.57 | p3nguin | What do you think would be lighter, lighttpd or nginx? I'm probably just going to be showing asterisk stats (cdr) and serve a few basic files. |
| 13:45.15 | WIMPy | No idea. |
| 13:45.49 | p3nguin | I may just stick to apache since it works. |
| 13:46.14 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 13:46.15 | p3nguin | It's only using 20M memory. |
| 13:48.48 | philfine__ | Well I post it on pastebin to simplify things |
| 13:49.06 | p3nguin | It would make it much easier for us to see it. |
| 13:50.33 | p3nguin | Actually, I forgot I had something I needed to do, so I probably won't get a chance to look at it. |
| 13:51.04 | philfine__ | http://pastie.org/1585666 |
| 13:51.31 | philfine__ | My problem is in from-internal context last extension |
| 13:51.37 | philfine__ | Everything matches there |
| 13:52.05 | WIMPy | That's what you told it to do. |
| 13:52.59 | philfine__ | But I also include many other specific extensions |
| 13:53.07 | philfine__ | I mean more specifi |
| 13:53.34 | WIMPy | Included contexts are always checked after all extensions. |
| 13:53.42 | philfine__ | Ok |
| 13:54.13 | WIMPy | So you need to put that matchall into another context you can include as well. |
| 13:54.20 | philfine__ | Ok, should I create other context to make them at same level ;-) |
| 13:54.30 | philfine__ | Ok |
| 13:54.33 | philfine__ | Understaood |
| 13:54.35 | philfine__ | Thanks |
| 13:56.10 | philfine__ | Worked great :) |
| 14:00.19 | *** join/#asterisk Andrew_M_ (6274247f@gateway/web/freenode/ip.98.116.36.127) |
| 14:01.50 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:03.38 | Andrew_M_ | Q: How do I get Google Contacts to my Polycom, so I can dial the numbers? I have Asterisk 1.8.2. |
| 14:03.42 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:05.09 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 14:05.37 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 14:14.36 | benngard | i know i asked earlier today but se if any that has joined know: is ${MEMBERINTERFACE} broken in 1.8.2.3? |
| 14:20.20 | Jasnejac | tbenngard: here's a change in the queue behaviour in the 1.8.3 release candidate. don't know if that affects it |
| 14:21.44 | benngard | ${QUEUENAME} works but ${MEMBERINTERFACE} is alay empty :( |
| 14:21.52 | benngard | always* |
| 14:23.21 | benngard | i found https://issues.asterisk.org/view.php?id=15813 exactly the same problem i got with 1.8.2.3 |
| 14:24.20 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 14:33.28 | *** join/#asterisk geoffmcc (~geoffmcc@cpe-76-180-197-68.buffalo.res.rr.com) |
| 14:34.48 | geoffmcc | using opensource g729 codec. voicemail is real choppy. trying to get to be near perfect but no luck. is this cause it is opensource version of codec or would this be a config problem, any ideas? |
| 14:38.00 | leifmadsen | benngard: use DumpChan() to see what channel variables are set. I've not really heard of MEMBERINTERFACE before |
| 14:38.31 | Andrew_M_ | geoffmcc: How is the network between the phone and the voicemail? |
| 14:38.39 | leifmadsen | geoffmcc: g729 is a patent incumbered codec, thus there is no opensource version. What you're using is typically considered pirated software. |
| 14:39.01 | leifmadsen | just spend the $10 and get something that is far more likely to work |
| 14:39.19 | leifmadsen | unless of course the amount of time you spend trying to get it to work is worth less than $10 |
| 14:43.09 | geoffmcc | @leifmadsen: i did not realise it was pirated. they made it sound opensource. just fiddling really and prob wont ever use so not worth the $10. was just bugging me that it was choppy |
| 14:43.09 | geoffmcc | makes me sound cheap, but money is just tight.. thank u |
| 14:43.53 | leifmadsen | there are many other codecs you can use which use less bandwidth such as g726, gsm, etc. |
| 14:43.56 | leifmadsen | ilbc, speex |
| 14:44.57 | geoffmcc | @leifmadse: i tried gsm i believe but was still kinda choppy. will try g726 |
| 14:45.09 | Andrew_M_ | leifmadsen: Isn't the $10 if you are transcoding g729? |
| 14:45.16 | geoffmcc | @leifmadse: sure i can tell im a newb. first real sit down with it |
| 14:45.22 | leifmadsen | Andrew_M_: then it's likely a network or CPU issue |
| 14:46.02 | leifmadsen | geoffmcc: I wasn't trying to imply anything -- I'm just saying if you need codecs that require less bandwidth there are many other options. It's highly unlikely it's the codec that is introducing the chop, but rather network conditions are far more likely. |
| 14:46.15 | leifmadsen | Andrew_M_: true story -- non-transcoding requires no license |
| 14:46.31 | leifmadsen | goes back to deploying his new Asterisk GUI on this production box, w00t |
| 14:46.59 | Andrew_M_ | leifmadsen: I am trying to remember what Jared thought me. |
| 14:47.27 | Andrew_M_ | ...tought me |
| 14:48.22 | geoffmcc | @leifmadsen if i use ulaw sounds perfect but my outbound audio is choppy. so this led me to believe was codec |
| 14:49.17 | geoffmcc | but i know next to nothing so if u say probably network then probably is |
| 14:50.09 | Andrew_M_ | geoffmcc: Could be a latency or bandwidth issue. |
| 14:50.20 | leifmadsen | Andrew_M_: taught I think :) |
| 14:50.49 | leifmadsen | geoffmcc: sounds more like bandwidth or jitter to me. Enable a jitterbuffer on the other side and see if that helps. |
| 14:50.59 | Andrew_M_ | I need a spell-checker and thesaurus in here! |
| 14:53.32 | geoffmcc | prob a stupid ? but if in debug says Reliably Transmitting (NAT) would def not be a nat issue right |
| 14:55.29 | benngard | leifmadsen: i added DumpChan to the dialplan, i dont se MEMBERINTERFACE there, but when i dig into the source code it is there, dont understand |
| 14:55.35 | Andrew_M_ | To my knowledge NAT issues cause no audio in SIP. Not voice quality issues. |
| 14:56.45 | geoffmcc | Andrew_M_: k, thank u. time to read up on jitterbuffers |
| 14:57.56 | benngard | but i found a fun variable: ~HASH~SIP_CAUSE~SIP/0317998972-00000093~=SIP 180 Ringing :) |
| 15:00.04 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 15:01.37 | leifmadsen | benngard: I think you're using it wrong |
| 15:01.49 | leifmadsen | benngard: show the full dialplan in a pastebin of how you're using it |
| 15:02.04 | benngard | oki sec |
| 15:03.50 | benngard | leifmadsen: http://pastebin.com/hVf6kKsS |
| 15:05.38 | leifmadsen | benngard: ya that won't work -- MEMBERINTERFACE would be set on the channel the Queue() is calling, not in the dialplan after the call falls out of the queue |
| 15:05.53 | leifmadsen | you're not using Queue() right |
| 15:06.01 | leifmadsen | check the queue chapter at... |
| 15:06.03 | leifmadsen | ~newbook |
| 15:06.03 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 15:06.53 | geoffmcc | @leifmadsen: google search on jitter buffer led me to explaination and example setup. using what this user posted has it way better than b4 so once i figure out how to tweak it i should be good. Thank u |
| 15:08.03 | *** join/#asterisk imox1234 (~imox1234@p4FC5C137.dip0.t-ipconnect.de) |
| 15:08.15 | leifmadsen | geoffmcc: np, you'll be adding latency, but it sounds like your outbound network connection isn't quite able to handle VoIP calls (either that, or turn off your torrents) |
| 15:11.24 | geoffmcc | @leifmadsen: that being said, if my problem is with voicemail only and not inbound calls, should i not use jitterbuffer and take the hit on chopy msgs (while checking threw IPKall trunk) |
| 15:11.54 | *** join/#asterisk imox1234 (~imox1234@p4FC5C137.dip0.t-ipconnect.de) |
| 15:11.55 | geoffmcc | also have 2mbps up, what is suggested min? |
| 15:12.54 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 15:17.05 | Jasnejac | 2 mbps is plenty |
| 15:17.40 | Jasnejac | can run 20+ consurrent ulaw or alaw calls on 2 mbps although that might be pushing it a little |
| 15:19.33 | *** join/#asterisk sourcode (~code@ppp-58-8-229-221.revip2.asianet.co.th) |
| 15:28.51 | leifmadsen | geoffmcc: if it's voicemail only it could be disk I/O for some reason |
| 15:28.55 | leifmadsen | shrugs |
| 15:31.26 | geoffmcc | @leifmadsen: for whatever reason it seems to be working fine, even now after removing jitter buffer settings... i dont get it. Think i need to take a break for a while |
| 15:33.16 | leifmadsen | I doubt it's anything you changed in asterisk |
| 15:33.29 | leifmadsen | runs off to play some Forza 3 or something |
| 15:37.26 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
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| 16:36.49 | *** join/#asterisk philfine_ (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 16:36.53 | philfine_ | How do I do pattern matching with caller id of incomming call ? |
| 16:38.18 | WIMPy | exten/cid |
| 16:38.30 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 16:39.01 | philfine_ | Cool :) |
| 16:39.26 | philfine_ | you mean: exten => s/CID ? |
| 16:39.36 | philfine_ | For the general case |
| 16:40.13 | WIMPy | yes |
| 16:40.37 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 16:50.36 | Andrew_M_ | Q: How do I get Google Contacts to my Polycom, so I can dial the numbers? I have Asterisk 1.8.2. |
| 16:57.01 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 17:09.59 | saxa | Kobaz: ok, but i already have it, so do i throw it away ? |
| 17:23.59 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 17:25.23 | viraptor | can anyone help out with IAX realtime? |
| 17:29.17 | saxa | Kobaz: anyway, do you have an idea on how can i solve my problem with the delay ? |
| 17:38.29 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 17:39.04 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 17:51.57 | *** join/#asterisk Dr-Linux (~Dr-Linux@182.177.195.252) |
| 17:52.35 | Dr-Linux | is 1.6.2.x stable and recommended to upgrade from 1.6.1.x ? |
| 17:54.13 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 17:55.21 | pabelanger | Dr-Linux: Yes, 1.6.1 is in security mode. |
| 18:00.55 | *** join/#asterisk philfine_ (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 18:02.48 | philfine_ | Lets say I want to be able to perform the following: Be able to call the asterisk from pstn line and if I do hangup in the first 2sec of ringing, asterisk will call me back and redirect me to voice mail. If I do not hangup start ringing internal phone |
| 18:02.54 | philfine_ | How can I do something like that ? |
| 18:06.48 | pabelanger | philfine_: Answer(), Hangup(), Wait(5), Dial(previous ANI), Answer(), Voicemail() |
| 18:06.57 | pabelanger | can all be done via the dialplan |
| 18:07.44 | philfine_ | That first Answer I do not want to do that |
| 18:07.55 | philfine_ | Since otherwise I pay a call from my cellphone |
| 18:07.59 | pabelanger | Progress() |
| 18:08.03 | philfine_ | I want the sip provider to call my phone |
| 18:09.57 | philfine_ | Ok, My mistake |
| 18:10.46 | philfine_ | Thats no what I want is it ? |
| 18:11.02 | pabelanger | Yes? |
| 18:11.16 | philfine_ | Answer will immediatelly asnwer the call right ? |
| 18:11.42 | pabelanger | Infact you don't even need Answer() or Progress(), just setup an incoming extension that will read the incoming ANI on the line, if it matches your cell phones, Hangup() then start the callback procedure |
| 18:12.56 | philfine_ | Then it doesn't allow me to make the calls from my cell |
| 18:13.11 | philfine_ | What I want is asterisk to detect that I did hangup |
| 18:13.22 | philfine_ | hangup my cellphone |
| 18:13.30 | philfine_ | And not the system to hangup me |
| 18:13.33 | *** join/#asterisk imox1234 (~imox1234@p4FC5C137.dip0.t-ipconnect.de) |
| 18:13.40 | pabelanger | philfine_: Asterisk cannot detect a hangup if you don't answer the line |
| 18:13.55 | philfine_ | I don't see why not |
| 18:14.25 | philfine_ | It detects that I am calling, why it doens't detect that I hangup the call before any action on its side |
| 18:15.31 | pabelanger | Then use the exten => h extension |
| 18:31.00 | *** join/#asterisk ManxPower (~ewieling@user-24-214-153-32.knology.net) |
| 18:31.16 | *** part/#asterisk ManxPower (~ewieling@user-24-214-153-32.knology.net) |
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| 19:16.16 | tash | anyone in here have much luck with Cepstral TTS voices? As in ... making them sound more realistic and less robotic? |
| 19:16.52 | tash | I really like ATT Natural Voices, but it is too expensive. Neospeech is also really good, but it too is a little on the steep side. I like Cepstral's pricing, but the quality seems to be lacking. |
| 19:18.28 | *** part/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 19:21.23 | *** part/#asterisk Plazma (~Plazma@freenode/staff/plazma) |
| 19:24.22 | ChannelZ | tashHouseWork_bb: You answered your own question. You get what you pay for |
| 19:32.35 | *** join/#asterisk fab5freddy (~bigd0gg@bas2-montrealak-1096582482.dsl.bell.ca) |
| 19:34.30 | fab5freddy | i installed asterisknow.. everything looks good in the control panel. i added extensions. twinkle registers with the asterisk server. but when i try to dial the extensions i created it says number is not in service. |
| 19:34.44 | fab5freddy | any ideas? |
| 19:36.41 | *** join/#asterisk Daejeo (~chatzilla@S010600265af0e1ad.cg.shawcable.net) |
| 19:38.03 | Daejeo | yum FIND *girl -- sexy |
| 19:38.32 | Daejeo | yum FIND *girl -- sexy --irc |
| 19:39.14 | Daejeo | something wrong with repos |
| 19:39.20 | Daejeo | :( |
| 19:39.40 | nestAr | hhehe |
| 19:40.20 | fab5freddy | daejeo: are you familiar with asterisknow? |
| 19:43.01 | Daejeo | fab5freddy: go ahead and ask what you want to know. do not ask to ask |
| 19:43.41 | Daejeo | there are so many asterisknow head on irc |
| 19:44.21 | fab5freddy | daejeo: i setup asterisknow. added extensions. installed twinkle. twinkle registers with th asterisk server. but when i try to dial the extensions created the asterisk voice answers saying the number is out of service. |
| 19:45.09 | nestAr | i am absolutely baffled by this IP650 |
| 19:45.15 | nestAr | i can not get it to ring |
| 19:47.51 | ChannelZ | can you be more vague? |
| 19:48.23 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 19:50.28 | *** join/#asterisk fross (~fross@68-112-140-187.dhcp.stcd.mn.charter.com) |
| 19:51.05 | nestAr | when you call it, it doesn't make any noise. |
| 19:51.35 | nestAr | all my 450's work fine.. just this 650 is being a thorn in my side. |
| 19:51.44 | nestAr | when i turn the ring volume up, it makes noise.. |
| 19:51.56 | nestAr | but when a call comes in, nada.. shows up on the LCD, but no tones |
| 19:53.31 | *** join/#asterisk mach-bar (~mach-bar@xdsl-89-0-43-129.netcologne.de) |
| 19:55.04 | *** join/#asterisk mach-bar (~mach-bar@xdsl-89-0-43-129.netcologne.de) |
| 19:58.46 | fab5freddy | what would be my steps to troubleshoot why an extension would be out of service when i can connect to the server |
| 19:59.01 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 20:00.14 | ChannelZ | nestAr: sounds like a setting probably, have you done a factory reset on it to wipe its brain? |
| 20:00.50 | ChannelZ | fab5freddy: the extension isn't properly associated with the device? but that's a FreePBX question |
| 20:00.57 | p3nguin | daejeo: Did you have a question? |
| 20:01.33 | nestAr | ChannelZ: yeah.. a couple times.. i'm on the right track.. i just did it again with out the directory or the overrides.. |
| 20:01.43 | nestAr | now to re-enable things one at a time. |
| 20:02.07 | Daejeo | p3nguin: nope |
| 20:02.16 | p3nguin | fab5freddy: I would suggest that the context is wrong or the peer isn't matching your peer entry correctly. You can troubleshoot it by going to the asterisk cli and running "core set verbose 4" and then make a call. |
| 20:02.59 | *** join/#asterisk rusty2011 (~dfrdfn2@184.46.83.50) |
| 20:14.00 | nestAr | hrmm... it's something in my directory file |
| 20:17.19 | nestAr | might be that <rt>1</rt> |
| 20:17.21 | nestAr | duuuuur |
| 20:21.27 | viraptor | why do I get messages like this sometimes? "reg_source_db: IAX/Registry astdb host:port invalid - '212.11.78.225:65181'" -> I don't see anything invalid about it |
| 20:23.19 | nestAr | Not only did that solve the problem, it also solved the other issue i had that the call wasn't being seized on off-hook. |
| 20:23.48 | nestAr | i guess maybe polycom assumes that if you have the ring set to silent for a contact, you might not want to accidently answer it on off-hook. |
| 20:23.51 | nestAr | makes sense to me. |
| 20:23.59 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:24.52 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 20:42.39 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
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| 20:44.42 | tashHouseWork_bb | ChannelZ: so I'm afraid that I'm left with crappy TTS or fork over the dough? |
| 20:44.55 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.6.75) |
| 20:44.58 | DelphiWorld | hello guys |
| 20:45.07 | DelphiWorld | my asterisk is unable to make out any iax2 call right now |
| 20:45.11 | DelphiWorld | from me to provider |
| 20:45.13 | DelphiWorld | how do i check? |
| 20:45.24 | DelphiWorld | just error unknowne |
| 20:46.53 | fenrus | start debugging |
| 20:48.18 | DelphiWorld | fenrus: the provider is not replying |
| 20:48.31 | DelphiWorld | fenrus: evean the demo digium extension 500 isn't replying to me |
| 20:49.41 | DelphiWorld | if i do iax2 reload |
| 20:49.43 | DelphiWorld | i got: [Feb 20 20:49:43] NOTICE[3655]: chan_iax2.c:12039 iax2_poke_peer: Still have a c |
| 20:49.43 | DelphiWorld | allno... |
| 20:52.56 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:54.08 | DelphiWorld | strange that i don't have any call! |
| 20:54.14 | DelphiWorld | just loaded asterisk |
| 20:54.21 | DelphiWorld | if i reload iax2 i got [Feb 20 20:54:07] NOTICE[4310]: chan_iax2.c:12039 iax2_poke_peer: Still have a callno... |
| 21:05.24 | *** join/#asterisk manji (~manjiki@adsl-142-168.adsl.ntua.gr) |
| 21:14.00 | DelphiWorld | any help? |
| 21:20.10 | p3nguin | delphiworld: I have no idea what that error means, but I see it all the time and it doesn't cause any problem, nor does it appear to be indicative of any problem. |
| 21:20.39 | Andrew_M_ | Q: How do I get Google Contacts to my Polycom, so I can dial the numbers? I have Asterisk 1.8.2. |
| 21:20.47 | DelphiWorld | p3nguin: but my iax2 connection failing evean to the default digium sample server |
| 21:21.15 | p3nguin | delphiworld: I believe you when you say it is failing, but it's not because of that message. |
| 21:21.25 | DelphiWorld | p3nguin: i agree :D |
| 21:21.28 | DelphiWorld | p3nguin: but how do i know? |
| 21:21.38 | DelphiWorld | p3nguin: how do i use tcpdump to debug in port 5060? |
| 21:21.50 | p3nguin | delphiworld: IAX2 doesn't run on port 5060. |
| 21:22.04 | DelphiWorld | p3nguin: sory 4569 |
| 21:22.19 | p3nguin | delphiworld: Start with the iax debug on asterisk cli. |
| 21:22.37 | DelphiWorld | p3nguin: i see no reply |
| 21:22.58 | DelphiWorld | p3nguin: iax2 set debug on |
| 21:22.59 | p3nguin | delphiworld: If you see nothing, then your phone is not reaching asterisk. |
| 21:23.07 | DelphiWorld | p3nguin: or iax2 set debug peer peername on |
| 21:23.18 | p3nguin | delphiworld: Is your phone a sip phone and the peer is iax2? |
| 21:23.23 | DelphiWorld | p3nguin: so no, my phone is reaching it and hanging up |
| 21:23.34 | p3nguin | delphiworld: core set verbose 4 |
| 21:23.35 | DelphiWorld | p3nguin: yes phone is gxp1200 and the peer is iax2 |
| 21:23.52 | DelphiWorld | p3nguin: ok is 4 |
| 21:23.53 | Andrew_M_ | DelphiWorld: do you have Asterisk -vvvr (verbosity 3) |
| 21:24.02 | p3nguin | delphiworld: Now make a call. |
| 21:24.03 | Andrew_M_ | Never mind |
| 21:24.04 | DelphiWorld | Andrew_M_: yes now is 4 |
| 21:24.26 | p3nguin | delphiworld: Then pastebin everything from the time you turned on debug until the time you hangup the handset. |
| 21:25.02 | DelphiWorld | p3nguin: ok |
| 21:25.47 | Andrew_M_ | DelphiWorld: have you tried "iax2 show peers" |
| 21:26.19 | DelphiWorld | Andrew_M_: all peers down :( |
| 21:26.23 | DelphiWorld | p3nguin: http://www.dpaste.de/xse5/ |
| 21:26.29 | DelphiWorld | p3nguin: and no iax debugging :( |
| 21:26.49 | p3nguin | Unable to creat |
| 21:26.50 | p3nguin | e channel of type 'IAX2' |
| 21:27.08 | p3nguin | delphiworld: module show like iax |
| 21:27.46 | DelphiWorld | p3nguin: module showing one loaded, chan_iax2.so |
| 21:27.57 | DelphiWorld | p3nguin: PM you the debugging of iax2 register to the remote peer |
| 21:28.32 | p3nguin | I charge $100 per hour for private support. |
| 21:29.59 | DelphiWorld | p3nguin: while pastebin it then. |
| 21:30.11 | DelphiWorld | p3nguin: i didn't want to pastebin it here just because of the username |
| 21:30.19 | DelphiWorld | Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ |
| 21:30.19 | DelphiWorld | Timestamp: 00007ms SCall: 00903 DCall: 00000 [78.129.153.20:4569] |
| 21:30.19 | DelphiWorld | USERNAME : xxxxxxxx |
| 21:30.33 | DelphiWorld | and no reply from the carrier. |
| 21:31.15 | p3nguin | delphiworld: Are you saying that IAX2 is working, but there is no "connection" between your asterisk and that peer? |
| 21:31.27 | DelphiWorld | p3nguin: yes, a local peer is working |
| 21:31.35 | DelphiWorld | p3nguin: a iaxcomm softphone is working |
| 21:32.42 | p3nguin | delphiworld: What does iax2 show registry say about the registration of your asterisk to that peer? |
| 21:33.01 | DelphiWorld | p3nguin: to the working or not working one? |
| 21:33.15 | p3nguin | delphiworld: I'll let you decide. |
| 21:33.35 | DelphiWorld | p3nguin: :) |
| 21:33.54 | DelphiWorld | p3nguin: the not working one is a provider while the working one is a softphone |
| 21:33.58 | p3nguin | delphiworld: Your statement makes me think you have a second ITSP and you use IAX2 to it. |
| 21:34.01 | DelphiWorld | here's registration to the provider: |
| 21:34.02 | DelphiWorld | 78.129.153.20:4569 N 114516 <Unregistered> 60 Request |
| 21:34.03 | DelphiWorld | Sent |
| 21:34.30 | p3nguin | delphiworld: iax2 show registry shows YOUR asterisk's registration to OTHER peers. What does that have to do with your softphone? |
| 21:34.39 | DelphiWorld | p3nguin: :) |
| 21:35.08 | DelphiWorld | p3nguin: i didn't know this then |
| 21:35.26 | p3nguin | delphiworld: Now you know. |
| 21:35.43 | DelphiWorld | p3nguin: thank you so much for the info :P |
| 21:36.14 | p3nguin | delphiworld: So it shows that the request is sent. Are you sure that you configured the peer entry correctly? |
| 21:36.30 | DelphiWorld | p3nguin: while pastebin it, ok? |
| 21:36.34 | p3nguin | delphiworld: okay |
| 21:38.06 | DelphiWorld | p3nguin: http://dpaste.de/rX5w/ |
| 21:38.47 | p3nguin | delphiworld: Do you want to see my config for voipms? |
| 21:39.03 | DelphiWorld | p3nguin: yes if you don't mind? |
| 21:41.51 | p3nguin | delphiworld: http://pastebin.com/3h7sG9b5 |
| 21:42.52 | DelphiWorld | p3nguin: while try it |
| 21:43.05 | p3nguin | delphiworld: If I were you, I would copy mine and paste it into your system without changing anything except your username and secret. That should successfully register you to the chicago PoP. |
| 21:43.17 | p3nguin | delphiworld: If that works, then change your host to london. |
| 21:43.48 | DelphiWorld | p3nguin: i hop... let try :) |
| 21:43.57 | p3nguin | delphiworld: Or I could just change mine to london and see how it works. |
| 21:44.06 | DelphiWorld | p3nguin: lol try! |
| 21:44.54 | DelphiWorld | p3nguin: so same :( |
| 21:44.59 | p3nguin | delphiworld: I almost forgot... you can also delete that setvar I use in mine. |
| 21:45.09 | DelphiWorld | p3nguin: i removed it allready :-) |
| 21:46.07 | p3nguin | 64.120.22.242:4569 <Unregistered> Request Sent |
| 21:46.19 | p3nguin | 64.120.22.242:4569 <Unregistered> Registered |
| 21:46.25 | p3nguin | No problem. |
| 21:46.33 | DelphiWorld | p3nguin: so you can't register? |
| 21:46.34 | p3nguin | delphiworld: Now I'll make a call via london. |
| 21:47.51 | p3nguin | delphiworld: It is registered. I didn't copy the entire line and I reused the first line I wrote to you. It actually doesn't say <Unregistered> it shows my IP address. |
| 21:48.04 | DelphiWorld | p3nguin: mine still not |
| 21:49.35 | DelphiWorld | ohh p3nguin! |
| 21:49.40 | p3nguin | delphiworld: I guess you have other problems that you aren't telling me about. |
| 21:49.42 | DelphiWorld | p3nguin: i evean removed my older one |
| 21:49.49 | DelphiWorld | but is still looking for london.voip.ms |
| 21:50.23 | DelphiWorld | p3nguin: i don't have london pops now but is still looking for it :( |
| 21:50.57 | DelphiWorld | feeling mad right now |
| 21:52.17 | DelphiWorld | still not registered. |
| 21:52.18 | p3nguin | delphiworld: Care to try SIP instead of IAX2? |
| 21:52.28 | DelphiWorld | ok p3nguin, a question :-) |
| 21:52.35 | DelphiWorld | p3nguin: why i'm using iax and no sip? |
| 21:53.10 | p3nguin | delphiworld: Why are you using IAX2 and not SIP? |
| 21:53.53 | DelphiWorld | p3nguin: ip access list extended 100 deny udp any any eq 5060 log deny udp any any eq 5060 log |
| 21:53.55 | DelphiWorld | p3nguin: :) |
| 21:53.58 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:54.45 | p3nguin | delphiworld: You use IAX2 because you entered a firewall rule to block SIP? Wouldn't it be just as easy to remove the rule as it is to configure IAX2? |
| 21:55.06 | DelphiWorld | p3nguin: lol, not me, if was me i wouldn't do it! is my crazy ISP :P |
| 21:58.14 | DelphiWorld | p3nguin: my ISP block sip :( |
| 22:02.28 | *** join/#asterisk linuxplatform (~fedora@188.148.251.129) |
| 22:11.32 | delphiWorld | anyone use tshark to capture traffic? |
| 22:17.49 | *** join/#asterisk Andrew_M_ (6274247f@gateway/web/freenode/ip.98.116.36.127) |
| 22:18.44 | Andrew_M_ | Q: How do I dial Google Contacts on my Polycom? |
| 22:20.05 | DelphiWorld | Andrew_M_: me too i love to import it through my gxv3140 :) |
| 22:20.51 | DelphiWorld | Andrew_M_: the idea is to assign each one a speed dial number and put it in your address book :P |
| 22:21.01 | Andrew_M_ | I would like it to be "live" if possible. I save from my Android, then dial it from my Polycom. |
| 22:21.38 | DelphiWorld | Andrew_M_: but how doe polycom know it |
| 22:21.41 | Andrew_M_ | I am sure I can import, it, format it, upload it, but how often is that practical? |
| 22:22.07 | Andrew_M_ | DelphiWorld: That is part of my Q. |
| 22:22.25 | DelphiWorld | Andrew_M_: and that's part of my A :P |
| 22:23.23 | Andrew_M_ | I would like it to be automatic, either periodical, like once a day, or with some push technology. |
| 22:24.06 | Andrew_M_ | How about being able to save from Polycom, then use it in Android. |
| 22:24.36 | DelphiWorld | Andrew_M_: i don't have a polycom so no idea :-) |
| 22:25.09 | *** part/#asterisk tashHouseWork_bb (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 22:25.14 | Andrew_M_ | DelphiWorld: Any Asterisk phone, Polycom or other. |
| 22:25.29 | DelphiWorld | Andrew_M_: and the issue is how doe polycom know that's a contact to be called and not a email address? |
| 22:26.10 | Andrew_M_ | DelphiWorld: My first question would be how to link the Contacts with the Directory. |
| 22:26.43 | DelphiWorld | Andrew_M_: ldap? |
| 22:27.33 | Andrew_M_ | DelphiWorld: There is software available for ldap, Outlook, Google, but they run on Window$. I would like it on the phone instead. |
| 22:27.54 | DelphiWorld | Andrew_M_: polycom support ldap isn't it? |
| 22:28.06 | Andrew_M_ | DelphiWorld: yes |
| 22:28.16 | DelphiWorld | Andrew_M_: so use it? |
| 22:28.37 | Andrew_M_ | How do I get Google Contacts to ldap? |
| 22:29.55 | DelphiWorld | Andrew_M_: try to sync it through some tools in a linux box and sync the polycom there |
| 22:31.01 | Andrew_M_ | DelphiWorld: Thanks for the great idea. I have to read up on ldap admin. |
| 22:31.11 | DelphiWorld | Andrew_M_: :P |
| 22:32.53 | DelphiWorld | Andrew_M_: a simple idea: |
| 22:33.12 | DelphiWorld | Andrew_M_: use outlook or thinderbird and import your contact from gmail, and push it back to a ldap server |
| 22:33.14 | Andrew_M_ | DelphiWorld: Simple are the best ones. |
| 22:33.38 | Andrew_M_ | DelphiWorld: I already have it in thunderbird. |
| 22:33.54 | DelphiWorld | Andrew_M_: not sure if thinderbird support ldap |
| 22:34.02 | DelphiWorld | Andrew_M_: Evolution support it too |
| 22:34.25 | DelphiWorld | should go to sleep |
| 22:34.28 | Andrew_M_ | DelphiWorld: Something will support it. Evolution is too buggy for my taste. |
| 22:34.30 | DelphiWorld | good night Andrew_M_ |
| 22:34.43 | Andrew_M_ | DelphiWorld: Good night. Thanks again! |
| 22:34.53 | DelphiWorld | Andrew_M_: np! |
| 22:34.55 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.6.75) |
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