| 00:00.44 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 00:03.29 | LemensTS | . |
| 00:03.51 | *** join/#asterisk jcoffi1 (n=jcoffi@208.87.0.146) |
| 00:07.12 | NMR_1122 | to get calls in from my voip provider, do I need to forward a specific port to asterisk? |
| 00:08.17 | NovceGuru | damn broadvoice sucks |
| 00:08.19 | NMR_1122 | (using IAX) |
| 00:09.42 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 00:09.57 | hi365 | anyone have experience with a linksys 3201? |
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| 00:12.35 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 00:12.42 | pagec | is there some way to use the command line to turn off features in `make menuconfig` for asterisk (i want to turn off the RAS appliction, it isn't compiling and I don't care aobut it)? |
| 00:18.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 00:32.45 | isamar | hi folks.. |
| 00:34.03 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:34.36 | s14ck | i have this error http://pastebin.com/m6443e5f2 |
| 00:35.02 | s14ck | i can't recive call from e1 trunk |
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| 00:44.21 | s14ck | the error say me what the extension is not found in the contex |
| 00:44.42 | isamar | <PROTECTED> |
| 00:45.01 | pagec | yeah, what is the full phone number being dialed? xxx-xxx-1011? |
| 00:45.33 | pagec | maybe you have only the last 4 digits being sent to you, and you need to cut that down on your pbx or have the CO send you all 12 |
| 00:45.41 | pagec | *10 |
| 00:46.34 | s14ck | but i put DNIS max in 4 |
| 00:47.23 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-61.dyn.centurytel.net) |
| 00:48.02 | pagec | well i am not sure how you configured it in the dial plan, but i don't think you are picking up the 4 digits |
| 00:48.17 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-133-195.dsl.stlsmo.sbcglobal.net) |
| 00:48.28 | s14ck | pagec: how can I do it? |
| 00:49.41 | pagec | you need 1011 as an extension in the context asterick looks into from the hardware recieving the call, you might want to check the web for setup instructions |
| 00:57.43 | NMR_1122 | Is there a way to record myself for voice menu use using the phone? Or does it have to be recorded in a separate audio application and then copy the file over to the asterisk server? |
| 00:58.07 | *** join/#asterisk Woody4286 (i=Woody214@2001:470:5:3c9:7057:f912:7cfc:df4b) |
| 01:02.38 | pagec | check the Record function |
| 01:02.57 | pagec | you can dial an extension and run it when that extention picks up |
| 01:03.42 | NMR_1122 | And it'l put a permanent sound file somewhere that I can pass to playback()? |
| 01:03.48 | pagec | something like http://pastebin.com/m1b8eec06 |
| 01:04.23 | pagec | you can change it out of tmp if you want it permanently |
| 01:04.38 | pagec | if you are testing out greeting i'd put it in temp and then copy it when you have a good one |
| 01:04.54 | NMR_1122 | does temp get auto-erased? |
| 01:05.04 | pagec | at boot it should |
| 01:05.31 | NMR_1122 | so I should probably just create a special folder for "working" recordings |
| 01:05.40 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 01:05.52 | NMR_1122 | Otherwise Ill get it just right and the power will go out! |
| 01:05.55 | pagec | yeah, but they can grow huge very quickly, one reason to use tmp |
| 01:06.13 | NMR_1122 | oh.... |
| 01:06.22 | NMR_1122 | like how big? |
| 01:06.22 | pagec | lol, you can boot of a CD should that happen, but really, who reboots a linux machine, i'd trust tmp for this |
| 01:07.38 | NMR_1122 | ok, thanks. :) |
| 01:07.38 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 01:10.51 | NMR_1122 | Is it possible to control the way extensions ring? |
| 01:13.03 | NovceGuru | ringtone? |
| 01:13.41 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:14.02 | jplank | if I use fxs_ls in zapata.conf, why would it keep using kewlstart? |
| 01:14.19 | NMR_1122 | yeah... right now with Bellsouth, the phone rings "normal" when called via a local number and a "double ring" when someone call the 800 number. |
| 01:26.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:28.02 | jeff_phillips | NMR_1122: you would have to change your alert info |
| 01:28.36 | jeff_phillips | and it depends on the phones you're using. |
| 01:28.49 | NMR_1122 | Hi Jeff |
| 01:29.03 | jeff_phillips | hi |
| 01:29.16 | NMR_1122 | right now im using a SPA-2102 |
| 01:29.23 | NMR_1122 | to test with |
| 01:29.36 | NMR_1122 | we haven't gotten any ip phones yet |
| 01:29.39 | jeff_phillips | that one should be pretty straight forward |
| 01:29.56 | jeff_phillips | I have an audiocodes MP-124 and it's a pain in the neck |
| 01:30.51 | NMR_1122 | isnt there a command to send from asterisk to tell it which ring to use? |
| 01:32.44 | jeff_phillips | Yes it's your Alert-Info header |
| 01:33.14 | NMR_1122 | where does that go? in Dial()? |
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| 01:35.13 | jeff_phillips | well it depends on what distinguishes which ring you want to use |
| 01:36.03 | NMR_1122 | it should be based on the number the caller dials to reach us. |
| 01:37.56 | *** join/#asterisk mrbnet_ (n=mrbnet@c-75-73-142-28.hsd1.mn.comcast.net) |
| 01:39.40 | jeff_phillips | NMR_1122: I appologize I'm just starting with asterisk & am more familer with the freepbx, but in there you'd set it up in your incomming route to add the alert-info header |
| 01:40.06 | mrbnet_ | I setup asterisk on a debian system using apt. I cannot remember or find the config file I am supposed to change so asterisk loads on boot. Any ideas? |
| 01:41.22 | jeff_phillips | <PROTECTED> |
| 01:41.23 | jeff_phillips | <PROTECTED> |
| 01:42.00 | jeff_phillips | was how it used to work |
| 01:42.22 | jeff_phillips | now I think you do something like exten => s,1,SIPAddHeader(Alert-Info: something) |
| 01:44.13 | jeff_phillips | You would of course have the "something" match whatever is in the ring cadence options in the regional settings of your SIP device |
| 01:45.39 | jblack | mrbnet_: look at /etc/init.d |
| 01:46.03 | NMR_1122 | it looks like my device has 9 fields, labeled "Ring1 Cadence:" |
| 01:46.16 | NMR_1122 | with values like "60(.4/.2,.4/.2,.4/4)" |
| 01:46.50 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 01:47.05 | jeff_phillips | NMR_1122 That's where you can change how long each ring burst is and how long the pauses are between the bursts |
| 01:47.05 | NMR_1122 | Actually I have "Distinctive Ring Patterns" and "Distinctive Ring/CWT Pattern Names" |
| 01:47.52 | jeff_phillips | does it show what it is looking for in the alert-info? Usually something like bellcore-r1, r2, r3 or dr1 dr2 or something |
| 01:48.19 | NMR_1122 | Ah yes, that's in a different group |
| 01:48.30 | NMR_1122 | it says ring1 name |
| 01:48.51 | mrbnet_ | jblack: I have looked there. I thought there was a popup after I installed asterisk that said I needed to uncomment a line |
| 01:48.56 | NMR_1122 | so i assume ring1 name of "bellcore-r1" causes ring1 cadence to occur |
| 01:49.23 | jeff_phillips | Sounds right |
| 01:49.53 | jeff_phillips | if it says ring1's name is bellcore-r1, then you'd want to send "bellcore-r1" as your alert-info in the invite, and it should trigger the ring cadence for ring1 |
| 01:50.07 | NMR_1122 | So I want: exten => s,1,SIPAddHeader(Alert-Info:Bellcore-r1) |
| 01:50.20 | jeff_phillips | yeah that's what I'd try |
| 01:50.30 | NMR_1122 | ok, lets see... |
| 01:50.43 | jeff_phillips | i'm not sure if it's case sensitive |
| 01:51.10 | *** join/#asterisk iamfuzz (n=brian@c-24-126-234-225.hsd1.ga.comcast.net) |
| 01:52.11 | iamfuzz | Hi, I'm trying to setup a simple IAX conference room and have followed the guide here: http://www.voipplanet.com/backgrounders/article.php/3631256 |
| 01:52.34 | iamfuzz | however, I am unable to connect to the asterisk server with any client I've tried, and no messages appear in any logs or on the asterisk console |
| 01:52.37 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:53.03 | iamfuzz | I guess my question is, how do I connect to an asterisk server running on the local network - complete asterisk noob here |
| 01:53.18 | jeff_phillips | connect what? |
| 01:53.21 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 01:53.39 | iamfuzz | trying to call in with a soft client to a conference room |
| 01:54.04 | jeff_phillips | can you connect to make other types of calls? |
| 01:54.11 | iamfuzz | kiax to be specific. I've also tried another client on another box on the network, and can't connect. I get no error messages or anything - it just times out |
| 01:54.21 | iamfuzz | I can't connect squat :-) |
| 01:54.34 | NMR_1122 | It works Jeff. You're awesome. |
| 01:54.40 | jeff_phillips | i am? |
| 01:54.44 | jeff_phillips | nah... |
| 01:54.47 | NMR_1122 | apparently |
| 01:55.03 | iamfuzz | Running Ubuntu 8.04, downloaded asterisk, it's running. tried this tutorial for setup: http://www.voipplanet.com/backgrounders/article.php/3631256 |
| 01:55.09 | iamfuzz | can't connect to asterisk |
| 01:55.14 | jeff_phillips | iamfuzz: I dunno... firewall? |
| 01:55.40 | iamfuzz | guess it's possible, but I can't even connect on the same machine running the server |
| 01:55.42 | denon | hmm, so I moved an older * config (zap) to current/dahdi .. someone remind me what I'm forgetting.. |
| 01:55.44 | denon | channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 |
| 01:55.44 | denon | app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) |
| 01:56.09 | iamfuzz | IaxWrapper::event_unknown() Uknown message: Type=4 |
| 01:56.11 | denon | dahdi show channels does show all the channels |
| 01:56.16 | jeff_phillips | iamfuzz: i dunno, hard to say what the problem is when you don't have an error or something to point us in some direction |
| 01:56.22 | iamfuzz | that's all my client outputs as an error, over and over |
| 01:56.30 | denon | dahdi show status shows the circuit OK |
| 01:56.59 | denon | 's bangin head against wall |
| 01:57.13 | pagec | anyone recommend any good free windows sip clients? |
| 01:57.38 | jeff_phillips | pagec: I tried 20 different free windows sip clients and they all suck |
| 01:57.57 | iamfuzz | jeff_phillips, how bout a good free iax client? |
| 01:58.18 | AlexGC | iamfuzz: try zoiper |
| 01:58.20 | jeff_phillips | Kapanga was the best for what I wanted to do, but that was merely to keep the thing minimized and auto-answer calls to the soundcard output (one direction) as extensions in a PA paging group |
| 01:58.25 | iamfuzz | AlexGC, thx |
| 01:58.37 | jeff_phillips | all the other ones kept popping up and were quite obnoxious |
| 01:58.38 | denon | zoiper works fine, eyebeam is worth the money |
| 01:59.45 | jeff_phillips | iamfuzz: I dunno I was only looking for those two features (auto answer & don't get in my face). I tried a couple iax clients but as soon as I realized they lacked one or both of those two options, I immediately uninstalled them |
| 02:01.19 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:02.59 | iamfuzz | is there a good, simple way to do a demo call just to test that your server is up? |
| 02:03.07 | iamfuzz | I am confused out of my mind with this |
| 02:03.12 | iamfuzz | granted it's a small mind |
| 02:03.36 | NMR_1122 | I've got a strange bug going on now... for some reason, when I call in to asterisk from my cell phone, the extension here rings once, and then asterisk hangs up on me. the console says "everyone is busy/congested at this time" but the phones aren't in use! |
| 02:07.09 | pagec | i see eyebeam has xlite for free, does that work? |
| 02:07.32 | jaytee | yep |
| 02:07.44 | denon | no g729 |
| 02:08.28 | pagec | does g729 really make that much of a difference? and if you use asterisk for free, isn't that not included anyway? |
| 02:09.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 02:13.16 | pagec | anyway, i've just been using the free protocols and they seem to work fine |
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| 02:32.16 | jplank | what would cause asterisk not to disconnect an FXO port properly (mind you, it only happens with a local disconnect. when the far end disconnects, it works properly) |
| 02:34.29 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 02:35.40 | hardwire | jplank: far end being PSTN or SIP/IAX? |
| 02:36.28 | hardwire | sorry.. I misunderstood.. I getcha now |
| 02:36.37 | jplank | pstn |
| 02:37.00 | hardwire | so making calls out of your asterisk box to the PSTN.. when far side disconnects it kills the channel |
| 02:37.09 | jplank | yea |
| 02:37.19 | hardwire | what is your local device? |
| 02:37.28 | jplank | tdm2400 |
| 02:37.41 | hardwire | is it the console in asterisk? |
| 02:37.46 | hardwire | a sip phone on your asterisk system? |
| 02:37.47 | jplank | ohh |
| 02:37.50 | jplank | no a sip phone |
| 02:38.07 | jplank | I see the hangup inside the console |
| 02:38.11 | hardwire | what happens when you issue a soft-hangup from the console? |
| 02:38.24 | hardwire | does it wrap up the zap channel or does it stay open? |
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| 02:40.13 | NMR_1122 | Did you know that music on hold during dial doesn't work unless you use playback() before dial()? |
| 02:40.27 | ManxPower | NMR_1122: that is not true. |
| 02:41.10 | jplank | hardwire, thats whats weird |
| 02:41.16 | NMR_1122 | That seems to be the case here.... If i comment out playback, music on hold says its playing, but you hear no music |
| 02:41.18 | jplank | if I do a show channels, I don't see it open |
| 02:41.30 | jplank | but if I do a zap show channel 1 it shows it offhook |
| 02:41.45 | NMR_1122 | if i play a sound file first, you actually hear the music play |
| 02:42.03 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 02:42.15 | jplank | I'm confused, because if I ANI the number, the call completes |
| 02:43.46 | ManxPower | chances are the playback causes an answer to happen |
| 02:44.12 | NMR_1122 | in both cases I'm calling answer() first |
| 02:44.48 | hardwire | jplank: weird |
| 02:45.54 | jplank | inbound seems to work, just outbound is the problem |
| 02:47.11 | hardwire | jplank: tried kickstart vs loopstart? |
| 02:47.35 | hardwire | is busydetect or indications turned on in zapata.conf for that channel? |
| 02:47.42 | jplank | yea |
| 02:47.48 | jplank | well kewl not kick |
| 02:47.53 | jplank | the line is def loop though |
| 02:47.54 | hardwire | kewl |
| 02:47.54 | hardwire | sorry |
| 02:48.02 | jplank | its weird |
| 02:48.09 | hardwire | weird. |
| 02:48.18 | jplank | if I make an outbound call, it will work |
| 02:48.33 | jplank | then on the second outbound call I get a error message telling me I didn't dial all the digits |
| 02:48.35 | hardwire | if you ANI the number the call completes? explain that to me real quick |
| 02:48.50 | jplank | if I wait about 20 seconds then try another call, it works |
| 02:49.13 | hardwire | tried this with a regular phone? |
| 02:49.19 | jplank | but if I make a inbound call, while its not working, the inbound completes, and if I hangup (far end) then with outbound, it works |
| 02:49.20 | jplank | yes |
| 02:49.32 | jplank | I also had it rewired about 4 times already |
| 02:49.36 | hardwire | hah |
| 02:49.42 | hardwire | regular phone reacts properly? |
| 02:49.43 | jplank | right now the line is going straight from the VZ can to the 2400 |
| 02:49.45 | jplank | yea |
| 02:49.52 | hardwire | openvz? |
| 02:50.00 | jplank | VZ = verizon |
| 02:50.04 | hardwire | ah |
| 02:50.11 | hardwire | tried using redial on the phone? |
| 02:50.18 | hardwire | vs manually dialing? |
| 02:50.38 | jplank | both, yes |
| 02:50.47 | hardwire | tried extending dtmf in zapata.conf? |
| 02:51.01 | jplank | extending DTMF? I'm dialing with a w |
| 02:52.12 | hardwire | for recording? |
| 02:52.29 | jplank | for a pause |
| 02:52.35 | hardwire | ahha |
| 02:52.47 | hardwire | but that doesn't change how the dtmf tones are |
| 02:53.15 | jplank | I'm not sure what you mean by extending the DTMF |
| 02:53.56 | ManxPower | jplank: is the line an analog line? |
| 02:54.14 | jplank | yes |
| 02:54.17 | jplank | do you mean relaxdtmf=yes ? |
| 02:54.23 | hardwire | toneduration=100 |
| 02:54.25 | hardwire | try 200 |
| 02:54.33 | hardwire | in 1.4+ |
| 02:55.10 | hardwire | ManxPower: yeh.. sip in and fxo out to verizon pots. |
| 02:55.11 | jplank | trying it |
| 02:55.12 | ManxPower | jplank: you are STARTING to dial too fast. put a "w" at the beginning of the number. Also, I suggest toneduration=500 for testing, you can shrink the number later. |
| 02:55.32 | ManxPower | eg. Dial(Zap/G1/w${EXTEN}) |
| 02:55.49 | ManxPower | w = wait .5 second (for the telco to realize you went off hook and want to dial) |
| 02:55.54 | jplank | hmmm the toneduration has been giving me better results so far |
| 02:55.59 | denon | ManxPower: what am I missing/forgetting? : No translator path exists for channel type DAHDI |
| 02:57.03 | ManxPower | denon: I've never been any good at fixing translator path problems. |
| 02:57.19 | jplank | moving up the toneduration seems to be working better |
| 02:57.24 | denon | ManxPower: well .. I just moved from zap to dahdi .. |
| 02:57.25 | jplank | is that number in MS? |
| 02:57.28 | jplank | ms* |
| 02:57.28 | ManxPower | denon: check codec_dahdi.so or something like that |
| 02:57.32 | denon | was working fine with zap :) |
| 02:57.33 | ManxPower | jplank: yes, in MS. |
| 02:57.49 | ManxPower | denon: I still live in a Zap world. |
| 02:58.15 | denon | ManxPower: me too .. but thats gotta change |
| 02:58.52 | ManxPower | denon: not for me. My employer uses specific versions of Asterisk/Zaptel |
| 02:59.17 | denon | ah ic |
| 03:00.02 | denon | somethin is seriously hosed here |
| 03:00.17 | ManxPower | denon: and there are 1.4isms and 1.2isms everywhere in their dialplan the put on customer machines. |
| 03:00.36 | denon | well, I'm still pretty focused on 1.4 |
| 03:00.40 | tzafrir_laptop | this is not related to codec_dahdi normally |
| 03:00.49 | tzafrir_laptop | codec_dahdi and chan_dahdi are not related |
| 03:00.54 | denon | tzafrir_laptop: care to enlighten me? |
| 03:01.21 | ManxPower | whatabout lack of a zttranscode.ko? |
| 03:01.27 | tzafrir_laptop | a g729 call that ended up in a DAHDI/whatever channel? |
| 03:02.04 | tzafrir_laptop | codec_dahdi is for those with a transcoder card |
| 03:02.59 | denon | tzafrir_laptop: no, ulaw to dahdi |
| 03:03.07 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 03:03.24 | denon | getting two errors: |
| 03:03.25 | denon | channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 |
| 03:03.25 | denon | app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) |
| 03:04.04 | ManxPower | denon: do me that says chan_dahdi.so is not loaded. |
| 03:04.18 | tzafrir_laptop | denon: 256: g729 |
| 03:04.33 | tzafrir_laptop | 76: slin|ulaw|alaw |
| 03:04.47 | tzafrir_laptop | core show codecs |
| 03:05.25 | denon | sorry, you were right -- that client was g729, but on ulaw, I still get: |
| 03:05.26 | denon | app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 03:05.49 | denon | but core show channeltypes shows DAHDI |
| 03:05.51 | tzafrir_laptop | And this is not FXO, right? |
| 03:06.08 | denon | no, it is fxo |
| 03:06.33 | tzafrir_laptop | can you set it to ls rather than ks and see if this changes anything? |
| 03:06.37 | denon | fxs doesnt give me anything .. no message, no dialtone |
| 03:06.56 | denon | sure |
| 03:07.40 | denon | actually, dahdi_gen wanted me to use pri_cpe |
| 03:07.47 | denon | there anything to that? |
| 03:08.00 | ManxPower | do you have a PRI? |
| 03:08.04 | tzafrir_laptop | http://bugs.digium.com/view.php?id=14577 |
| 03:08.09 | denon | (it's cross-connected to an adtran 750) |
| 03:08.16 | tzafrir_laptop | ah, no FXO. ignore the above |
| 03:08.27 | denon | some FXO, some FXS |
| 03:08.48 | denon | on the adtran |
| 03:09.18 | tzafrir_laptop | if it is FXO (fxs_ks signalling) , then do look at the above report |
| 03:09.49 | tzafrir_laptop | still does not understand what that issue has not been fixed |
| 03:10.15 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:10.20 | tzafrir_laptop | Breaks FXO on Asterisk 1.6.x and has been open for quite some time |
| 03:10.30 | denon | well ,Ive always used fxs_ks / fxo_ks in the past |
| 03:10.34 | denon | this is 1.4.x |
| 03:10.43 | denon | 1.4.24 |
| 03:10.46 | tzafrir_laptop | what version of 1.4? |
| 03:10.59 | tzafrir_laptop | ah, ok. |
| 03:11.33 | denon | was running fine on zaptel 1.4.12 before |
| 03:11.40 | denon | decided I should do dahdi on the upgrade |
| 03:11.43 | denon | (dumb move :) |
| 03:11.54 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 03:12.08 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 03:12.17 | tzafrir_laptop | next move would be to get a PRI trace, I guess |
| 03:12.25 | tzafrir_laptop | err... no PRI here |
| 03:12.40 | denon | it's a t1 cross-connect to adtran 750 |
| 03:13.02 | denon | on a single t1 digium card (t100P or so) |
| 03:13.14 | tzafrir_laptop | hmm... what was the Dial string you used? |
| 03:14.01 | hardwire | jplank: all happy? |
| 03:14.03 | denon | well, it doesnt seein inbound either |
| 03:14.06 | jplank | not yet |
| 03:14.08 | jplank | still testing |
| 03:15.19 | denon | tzafrir_laptop: but for outbound, Dial(DAHDI/18/18005551212) or such |
| 03:15.39 | tzafrir_laptop | Channel 18 exists, right? |
| 03:15.44 | denon | yeah |
| 03:15.58 | tzafrir_laptop | What is the output of: dahdi show channel 18 |
| 03:18.01 | denon | bleh, just overwrote a config .. sec .. so actually, in this situation, should they be fxs_ks/fxo_ks? or is the gen right wanting to do pri_cpe? |
| 03:18.40 | hardwire | closer? |
| 03:18.47 | denon | chan_dahdi whines if I don't set it to pri_cpe |
| 03:19.38 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 03:20.03 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 03:21.12 | denon | tzafrir_laptop: but to answer your question, here's the channel 18 output: http://pastebin.ca/1369944 |
| 03:21.43 | jplank | even if I set toneduration to 1000 if I place two calls over the same trunk to close together I get the issue |
| 03:22.13 | tzafrir_laptop | signalling type: ISDN PRI |
| 03:22.16 | *** part/#asterisk califus (n=chatzill@210.212.160.101) |
| 03:22.23 | denon | tzafrir_laptop: yeah, thats what I was saying above |
| 03:22.42 | denon | it whines if I dont set it to pri_cpe |
| 03:22.53 | tzafrir_laptop | who whines? |
| 03:23.13 | tzafrir_laptop | chan_dahdi won't let you chang esignalling just like that |
| 03:23.34 | denon | chan_dahdi.so won't load |
| 03:23.47 | tzafrir_laptop | but you can change whatever you need on system.conf, run dahdi_cfg, and then in asterisk run: dahdi restart |
| 03:23.56 | denon | well .. it wont load if I dont set it to that, says I need to change it |
| 03:24.07 | tzafrir_laptop | (and that's nothing new, except that you can run 'dahdi restart instead of a full restart) |
| 03:24.18 | denon | yeah, Im doing all that |
| 03:24.29 | denon | well, I dont know what changes I'd need to make in system.conf, but i am in chan_dahdi.conf |
| 03:24.51 | denon | then I'm /etc/init.d/asterisk stop; /etc/init.d/dahdi restart; /etc/init.d/asterisk start |
| 03:24.51 | tzafrir_laptop | what is the output of cat: /proc/dahdi/* |
| 03:25.51 | denon | http://pastebin.ca/1369953 |
| 03:25.55 | denon | (ignore span 2) |
| 03:29.08 | jplank | hardwire: still no, pretty much the same thing |
| 03:29.41 | jplank | if I do a zap show channel, it shows it as offhook |
| 03:30.21 | jplank | is there any way to slow down the disconnect? |
| 03:30.23 | denon | tzafrir_laptop: that pastebin was for you, obviously |
| 03:31.19 | tzafrir_laptop | denon, so it is configured for PRI |
| 03:31.24 | tzafrir_laptop | (in system.conf) |
| 03:31.39 | tzafrir_laptop | edit system.conf and re-run dahdi_cfg |
| 03:33.22 | hardwire | jplank: when would that happen? |
| 03:34.20 | denon | tzafrir_laptop: there any harm in just copying zaptel.conf to system.conf? |
| 03:34.30 | denon | the zaptel.conf that was working for years prior, that is |
| 03:34.34 | jplank | when it hangs up the line |
| 03:34.48 | hardwire | how would you slow that down? |
| 03:34.58 | jplank | I'm just grasping at straws here |
| 03:35.06 | tzafrir_laptop | denon, no. but you should add echo canceller information |
| 03:35.08 | hardwire | I'm just wondering how odd that sounds. |
| 03:35.09 | hardwire | heh |
| 03:35.23 | jplank | why when I hang up the line zap show channel still shows it offhook |
| 03:35.35 | tzafrir_laptop | denon, basically: echocanceller=mg2,1-25 |
| 03:35.39 | tzafrir_laptop | or something similar |
| 03:35.43 | hardwire | do you have a sound card in your asterisk box jplank? |
| 03:35.44 | jameswf | jplank, foo or fxs |
| 03:35.52 | jplank | fxo |
| 03:35.58 | jplank | no soundcard hardwire |
| 03:36.05 | jameswf | isn't hook state fxs only |
| 03:36.11 | hardwire | no worries.. if you could test something for me. |
| 03:36.31 | tzafrir_laptop | it does have some meaning on FXO. Though strange. Not really sure |
| 03:36.36 | hardwire | modprobe snd-dummy |
| 03:36.47 | hardwire | modprobe snd-pcm-oss |
| 03:36.58 | jplank | no return on either |
| 03:37.00 | hardwire | then restart asterisk with the oss module |
| 03:37.07 | hardwire | if it's not already loaded |
| 03:37.22 | hardwire | then from the console try to 'dial number@outcontext' |
| 03:37.31 | hardwire | then issue a hangup.. and see if you still have issues. |
| 03:37.48 | denon | tzafrir_laptop: now I'm making headway .. heh, I was just getting mismatched and confused by the errors .. not realizing where it was getting that data |
| 03:38.37 | hardwire | at this point in life.. why aren't microsoft updates send via broad satellite multicast every few minutes to government issues reception devices? |
| 03:38.42 | hardwire | send/sent |
| 03:38.47 | hardwire | I mean srsly.. |
| 03:39.12 | jplank | hardwire I don't follow, what does loading the oss drivers have to do with an FXO issue |
| 03:39.15 | denon | tzafrir_laptop: you prefer mg2? |
| 03:39.29 | hardwire | jplank: maybe it's not just an fxo issue. |
| 03:39.34 | hardwire | process of elimination ftw. |
| 03:39.39 | jplank | inbound works noproblem |
| 03:39.40 | jameswf | I prefer pretzels |
| 03:39.45 | tzafrir_laptop | denon, of the built-in ones it's the best |
| 03:39.50 | tzafrir_laptop | I naturally prefer oslec |
| 03:39.51 | jplank | sip works in or out |
| 03:39.56 | hardwire | jplank: just working the process. |
| 03:40.07 | denon | tzafrir_laptop: I've got some hpec licenses for this box .. |
| 03:40.08 | hardwire | feel free to skip a few steps and scratch your head all night. |
| 03:40.12 | denon | but last time I used hpec it was worse .. so .. |
| 03:40.21 | tzafrir_laptop | denon, should also work very well |
| 03:40.26 | denon | I see dahdi still likes to ring all the fxs channels when it loads with adtran |
| 03:40.38 | jplank | load => chan_oss.so |
| 03:40.38 | tzafrir_laptop | (and in dahdi it will be easy to fall back to mg2 from hpec) |
| 03:40.39 | jplank | right? |
| 03:40.40 | denon | was kinda hoping that'd be fixed in the past 10 years or so :) |
| 03:40.42 | jameswf | sounds like an adtran bug |
| 03:40.58 | denon | jameswf: perhaps |
| 03:41.14 | jameswf | our channel banks dont ring all the channels :) |
| 03:41.30 | jplank | hardwire: your saying just add load => chan_oss.so to modules.conf and restart? |
| 03:42.06 | tzafrir_laptop | denon, on startup dahdi sends an "on-hook" to all channels |
| 03:42.08 | hardwire | if it's not already loaded |
| 03:42.18 | hardwire | like.. do you have "dial" as a valid command in the console? |
| 03:42.25 | jplank | no |
| 03:42.32 | hardwire | then add it and restart asterisk if you can. |
| 03:42.33 | denon | tzafrir_laptop: that makes adtran want to ring? |
| 03:42.35 | tzafrir_laptop | err... not dahdi. Asterisk's chan_dahdi |
| 03:42.44 | tzafrir_laptop | I don't know adtran |
| 03:42.58 | denon | 750s have done this for years .. |
| 03:43.08 | denon | always surprised me, since it was pretty common place back in the day (the 750) |
| 03:43.14 | denon | for obvious historical reasons with mark |
| 03:43.26 | jameswf | denon, a distaste for reality makes the adtran ring |
| 03:43.56 | denon | who'z yer dahdi |
| 03:44.01 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 03:44.04 | jameswf | if adtran cared you would call them and they would fix it |
| 03:44.13 | *** join/#asterisk trymi1 (n=please@dip5-237.bagan.net.mm) |
| 03:44.19 | jplank | I won't be able to use dial from the console, I don't have a soundcard |
| 03:44.29 | denon | sure, they could make it do sip over serial too .. question is do they want to |
| 03:44.34 | jameswf | jplank, originate |
| 03:44.42 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 03:44.56 | jameswf | denon new features is not the same as fixing basic function |
| 03:45.01 | hackeron | hey, I'm trying to dial out, my outgoing server is asterisk and I have freeswitch configured to dial out through asterisk - but I get: Call from '' to extension '918002255288' rejected because extension not found. -- any ideas? |
| 03:45.24 | denon | jameswf: well, I talked to them many many years ago .. and they basically said |
| 03:45.29 | denon | "what's asterisk?" |
| 03:45.34 | denon | of course, that's changed a little |
| 03:45.39 | jameswf | exactly they are to big for you |
| 03:45.46 | jplank | am I using originate wrong |
| 03:45.53 | jameswf | that is no way to run a company |
| 03:45.56 | denon | nah, adtran's always been great to work with |
| 03:45.56 | jplank | originate zap/1 number@context |
| 03:46.05 | denon | just that asterisk wasn't exactly on the radar yet |
| 03:46.44 | jameswf | denon you should try one of our channel banks they simply work and if something is broke we don't blame it on the software |
| 03:47.07 | denon | ahhh, a commercial .. |
| 03:47.12 | denon | I thought you sounded funny :) |
| 03:47.31 | denon | astribank? |
| 03:47.34 | jameswf | denon no commercial if something is broke it should be fixed |
| 03:48.01 | jameswf | anyone who is not willing to fix their bugs because they dont like your software is a joke |
| 03:48.01 | denon | which CBs? |
| 03:48.19 | jameswf | asterbanks are tzafrir_laptop |
| 03:48.22 | jplank | hardwire: I get the same thing when I use originate |
| 03:48.25 | jameswf | RHino |
| 03:48.49 | denon | ah, not used RHino I dont think.. |
| 03:49.07 | denon | mostly because I've been too lazy to eval em I guess heh |
| 03:50.26 | jplank | does someone have an * with an fxo card in front of them right now |
| 03:50.27 | [TK]D-Fender | Rhino CB's are prety decent IMO |
| 03:50.36 | [TK]D-Fender | If you really want to head that way of course... |
| 03:50.38 | denon | jplank: yes |
| 03:50.41 | jplank | maybe the problem I'm having isn't really a problem per se |
| 03:50.44 | jameswf | denon even if you dont come our way you should look at a companies willingness to address issues as a factor so if they dont find someone who will, hell go with an asterbank at that rate you know when something is broke tzafrir_laptop will probably loose sleep to make it right |
| 03:51.03 | jplank | can you call out that trunk, as soon as it starts ringing, hang up, and call out it again within a second or two |
| 03:51.13 | jplank | disconnecting on the * side |
| 03:51.17 | jplank | never picking up the far end |
| 03:51.30 | denon | jplank: what, just the delay before it gives the channel back? |
| 03:51.54 | jplank | oh your connected to a channel bank, I don't know if that will work the same |
| 03:52.10 | denon | jplank: nah, Ive got a pile of other PBX windows open |
| 03:52.15 | denon | with tdm cards |
| 03:52.20 | jameswf | downloading ubuntu-alpha I should really wait the 3 days for beta... |
| 03:52.32 | denon | but I cant do what you want, as I dont have a handset to one of em |
| 03:52.47 | jplank | anyone want to take a look at my zapata.conf and zaptel.conf, maybe I'm missing something |
| 03:52.50 | denon | hm, ftp.digium.com is gone? bummer |
| 03:53.11 | denon | jameswf: yeah, I'm sure you're right .. though, in the past, adtran has gotten firmware out to me to fix my issues .. |
| 03:53.14 | jameswf | denon downloads.digium... |
| 03:53.19 | denon | jameswf: I've not pestered them on this one |
| 03:53.52 | denon | oh that's right, http only now |
| 03:54.05 | *** part/#asterisk trymi1 (n=please@dip5-237.bagan.net.mm) |
| 03:55.09 | jameswf | jplank, pastebin |
| 03:55.55 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:56.12 | hardwire | jplank: ok. |
| 03:56.30 | hardwire | sorry for the wild goose chase.. I usually start with chan_oss before originating to a local app |
| 03:57.35 | jplank | hardwire: jameswf http://pastebin.com/m62360b6a |
| 03:57.54 | jplank | the only thing I didn't add was zapata-addtional.conf and thats because thats blank |
| 03:58.03 | hardwire | freepbx? |
| 03:58.30 | hardwire | just guessing.. I don't have a lot of experience with it |
| 03:58.43 | hardwire | I know a lot of solutions use the additional.conf's however |
| 03:58.44 | *** join/#asterisk tawker (n=ahuman@wikipedia/Tawker) |
| 03:58.59 | jplank | freepbx |
| 03:59.15 | jplank | but really, freepbx didn't touch zapata |
| 03:59.22 | *** join/#asterisk propellerhead (n=yogurt2u@200.43.87.56) |
| 03:59.34 | denon | I guess I should really give hpec another chance |
| 03:59.43 | jplank | and genzaptelconf generated zaptel.conf and zapata-auto.conf |
| 03:59.48 | [TK]D-Fender | What is the actual problem? |
| 04:00.19 | jameswf | comment out #include zapata-additional.conf see what happens |
| 04:00.21 | *** part/#asterisk tawker (n=ahuman@wikipedia/Tawker) |
| 04:02.38 | jplank | its blank, but ok |
| 04:03.20 | jplank | same thing |
| 04:03.30 | jplank | fender, I can't always make outbound calls |
| 04:03.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 04:03.44 | jplank | I sometimes get a message saying that I didn't dial all the digits |
| 04:03.49 | jplank | inbound works |
| 04:03.55 | jplank | I have a w in the dial plan |
| 04:04.02 | jplank | and I have the toneduration set to 100 |
| 04:04.11 | jplank | (tried even 1000 but same thing |
| 04:04.12 | jplank | ) |
| 04:06.10 | [TK]D-Fender | jplank: show me. |
| 04:06.59 | denon | hm, dahdi init.d script wants to run zaphpec enable .. nice. :) |
| 04:07.15 | jplank | how can I show you? its a recorded message I'm hearing on the line, console looks perfect |
| 04:07.19 | denon | (instead of dahdihpec_enable) |
| 04:07.39 | [TK]D-Fender | jplank: Show me the faled call. |
| 04:07.59 | jplank | sure, but FYI - the call doesn't look like it failed |
| 04:08.05 | denon | busts out ln -s |
| 04:10.26 | jplank | hmmm with toneduration at 300, everyone of my calls completed |
| 04:10.31 | jplank | but it was a 6 second connect time |
| 04:10.43 | jplank | so I'm sure it has to have something to do with that. |
| 04:11.05 | denon | hmm, asterisk detected a problem with my dahdi and will shutdown |
| 04:11.17 | denon | it's like installing windows on a free after rebate motherboard |
| 04:11.54 | NovceGuru | gets a bag of pcchips |
| 04:11.59 | denon | hehe |
| 04:12.12 | jplank | 250 worked, 5 1/2 isn't as bad |
| 04:17.34 | jplank | seems 200 with a w might be the magic number |
| 04:17.45 | jplank | having the customer test, seeing if its too long |
| 04:18.31 | [TK]D-Fender | w = 500ms |
| 04:19.29 | jplank | yea |
| 04:19.31 | jplank | .5 second |
| 04:19.55 | jplank | customer is testing right now, hopefully 4 seconds (more or less) isn't too long |
| 04:22.20 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 04:23.22 | denon | ah well, hpec still blows |
| 04:24.13 | denon | at least it's easier to disable in dahdi |
| 04:24.44 | *** join/#asterisk columbo (n=columbo@pool-173-51-16-137.lsanca.dsl-w.verizon.net) |
| 04:24.49 | *** part/#asterisk columbo (n=columbo@pool-173-51-16-137.lsanca.dsl-w.verizon.net) |
| 04:24.57 | *** join/#asterisk columbo (n=columbo@pool-173-51-16-137.lsanca.dsl-w.verizon.net) |
| 04:25.46 | *** join/#asterisk AndyCrawford (n=andy@dynamic-65-161-142-80.tvscable.com) |
| 04:31.08 | hardwire | [TK]D-Fender: so jplank is having issues with the remote side dealing with dtmf.. and longer tones seem to fix it.. but when he uses a regular phone and hits redial (3 second dial time or so) it works fine every time. |
| 04:31.17 | hardwire | Maybe a volume issue as well? possibly echo can? |
| 04:31.29 | hardwire | I doubt echo can.. but maybe theres something I'm missing there. |
| 04:31.53 | [TK]D-Fender | gains could certainly be a factor |
| 04:32.05 | hardwire | there *is* a disconnect tone too.. right? |
| 04:32.16 | hardwire | or is that just crazy talk. |
| 04:33.00 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 04:33.31 | hardwire | it's a loop.. so I guess it doesn't need a tone. |
| 04:34.24 | hardwire | http://pastebin.com/m62360b6a |
| 04:34.31 | hardwire | gai is 0 there.. and echo cancel is on |
| 04:34.34 | jplank | hardwire echo can is on (hardware) |
| 04:34.40 | hardwire | gain |
| 04:34.42 | jplank | fender I left the gains defaiult |
| 04:35.03 | hardwire | yar.. whats the tool to calibrate those? is it just ztmonitor and an echo test? |
| 04:35.29 | hardwire | hates dealing with pots. |
| 04:35.36 | hardwire | I wish you and your customers would never have to deal with them ever again. |
| 04:35.55 | hardwire | alas.. we need something that works when the zombies attack.. so it's still quite heavily used. |
| 04:36.49 | jplank | so do I |
| 04:36.57 | jplank | I just told that to my client |
| 04:37.08 | jplank | if he would of ordered SIP trunks from us, none of this would ever happen |
| 04:37.22 | hardwire | I bet he's scared of zombies. |
| 04:37.27 | hardwire | might be worth asking later on. |
| 04:37.48 | jplank | this is the only client we have that we put a PBX in for, but don't have their voice and or data |
| 04:38.01 | hardwire | I'm down with that |
| 04:38.12 | hardwire | it's kinda neat having crazy little scenarios like that. |
| 04:38.19 | hardwire | cause it adds contrast to other crap you have to deal with. |
| 04:38.27 | hardwire | "At least it's not a client x scenario" |
| 04:38.28 | jplank | yea |
| 04:38.43 | hardwire | and then everybody chuckles and goes for tea. |
| 04:39.03 | jplank | well I already told my tech to take the client out for drinks when they leave |
| 04:39.26 | hardwire | I forgot where you worked. |
| 04:39.29 | hardwire | I knew at one point. |
| 04:40.29 | hardwire | <- self employed masochist. |
| 04:42.00 | jplank | interglobe communications |
| 04:42.04 | jplank | don't look at our website |
| 04:42.08 | jplank | it sucks (for now) |
| 04:42.14 | jplank | its from like 7 years ago |
| 04:42.38 | jplank | we are waiting for the designer to finish it off |
| 04:43.08 | hardwire | haha |
| 04:43.16 | hardwire | sorry boss |
| 04:43.25 | *** part/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 04:43.53 | jplank | they are doing an awesome job though, we sent the logo to landsend to make us some new shirts, and they liked our new logo so much, they want to use it in their catalogs :P |
| 04:44.05 | hardwire | orly |
| 04:44.21 | hardwire | interglobe is a neat name too |
| 04:44.28 | hardwire | and it fits well with techie parts of catalogs |
| 04:44.54 | jplank | its funny, there's a company called interglobe in like india or something that runs calling card scams |
| 04:45.01 | jplank | we always here about it |
| 04:45.05 | jplank | esp at comptel |
| 04:45.09 | hardwire | doh |
| 04:46.30 | denon | well, looks like dahdi is stable .. |
| 04:46.36 | denon | I really shoulda done this on an offline system .. oh well |
| 04:47.11 | denon | I should move that box to all sip so I can stuff i into an esxi platform |
| 04:47.25 | denon | as much as I hate abstraction layers when it comes to voice |
| 04:57.04 | *** join/#asterisk cr4z3d (n=cr4z3d@unaffiliated/cr4z3d) |
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| 05:07.37 | denon | hm, when did asterisk console stop ignoring ^c? |
| 05:07.44 | denon | sighs |
| 05:16.07 | cr4z3d | i've got a cisco ip phone 7970. should i use the skinny driver or chan_sccp driver |
| 05:21.53 | *** join/#asterisk raulfragoso (n=raulfrag@189-68-132-231.dsl.telesp.net.br) |
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| 05:41.19 | Cuban0 | hello everyone i have a problem with DAHDI and x100p maybe somebody can help me, i have tried all by google and no results |
| 05:41.51 | Cuban0 | can someone help ??? |
| 05:42.08 | Cuban0 | this is log |
| 05:42.09 | Cuban0 | [root@elastix ~]# dahdi_hardware |
| 05:42.10 | Cuban0 | pci:0000:01:09.0 wcfxo- 1057:5608 Wildcard X100P |
| 05:42.20 | Cuban0 | so hardware is being recognized |
| 05:42.22 | Cuban0 | but later |
| 05:42.49 | Cuban0 | [root@elastix ~]# dahdi_cfg -v |
| 05:42.49 | Cuban0 | DAHDI Tools Version - 2.2.0-rc1 |
| 05:42.49 | Cuban0 | DAHDI Version: 2.2.0-rc1 |
| 05:42.49 | Cuban0 | Echo Canceller(s): |
| 05:42.49 | Cuban0 | Configuration |
| 05:42.49 | Cuban0 | ====================== |
| 05:42.51 | Cuban0 | 0 channels to configure. |
| 05:43.04 | Cuban0 | and only Timer DUMMY appears |
| 05:43.20 | mog | you have to configure it Cuban0 |
| 05:43.31 | mog | also please dont paste multiple lines into the channel |
| 05:43.36 | mog | use pastebin |
| 05:43.38 | Cuban0 | oops i´m sorry |
| 05:43.45 | Cuban0 | i´m new around here |
| 05:43.49 | mog | no worries |
| 05:43.56 | Cuban0 | listen bro |
| 05:43.59 | mog | you have a folder called /etc/dahdi/system.conf |
| 05:44.02 | Cuban0 | i have this problem in boot up |
| 05:44.07 | mog | you need to configure it for your card |
| 05:44.48 | Cuban0 | NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 |
| 05:44.49 | Cuban0 | Failed to initailize DAA, giving up... |
| 05:45.06 | Cuban0 | already patched wcfxo.c |
| 05:45.43 | Cuban0 | when i use dahdi_cfg the card does not appears, only Dummy |
| 05:46.03 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 05:46.24 | Cuban0 | mog can you please tell me the next step ? |
| 05:46.54 | mog | please read what i said |
| 05:47.11 | mog | your card wont be useable till its configured in system.conf |
| 05:47.18 | Cuban0 | ok i will nano it |
| 05:47.21 | Cuban0 | wait |
| 05:47.33 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 05:48.27 | Cuban0 | i have it , it says it was autogenerated and only has two variables inside defaultzone and loadzone |
| 05:48.50 | Cuban0 | what do i need to put inside ? |
| 05:49.34 | Cuban0 | it says "do not hand edit" |
| 05:51.23 | *** join/#asterisk PDani (n=pekdanie@89.133.156.227) |
| 05:53.24 | Cuban0 | it is generated by dahdi_genconf but it does not put the Wildcard parameters inside |
| 05:53.44 | Cuban0 | i´m stuck there :( |
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| 06:04.42 | Cuban0 | can somebody help ? |
| 06:05.54 | mog | fxsks=1 |
| 06:06.32 | mog | recommends reading the book or going to voipinfo, or paying one of the friendly consultants |
| 06:09.42 | Cuban0 | only that ? |
| 06:10.39 | mog | thats what you need to bring the card up, but then you also have to bring up card in asterisk which is a seperate issue |
| 06:15.59 | jplank | anyone know what this means, http://pastebin.com/m6d382526 |
| 06:18.19 | *** join/#asterisk PDani (n=pekdanie@89.133.156.227) |
| 06:19.18 | mog | your crashing |
| 06:19.56 | jplank | I got that ;) |
| 06:20.02 | jplank | I'm curious why |
| 06:20.28 | Cuban0 | after putting fxsks=1 now it says |
| 06:20.29 | Cuban0 | [root@elastix ~]# dahdi_cfg |
| 06:20.29 | Cuban0 | DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) |
| 06:20.37 | mog | modprobe wcfxo |
| 06:20.41 | mog | your card isnt loaded |
| 06:21.33 | jplank | I could make it crash |
| 06:21.35 | Cuban0 | modprobe does not returns nothing |
| 06:21.40 | jplank | by making a bunch of inbound calls |
| 06:21.53 | Cuban0 | i guess i must get rid of that first boot error |
| 06:22.20 | jplank | but why is the card crashing? |
| 06:22.35 | mog | thats what modprobe is supposed to return |
| 06:22.49 | mog | in gnu/linux if things work its normal for it to return nothing |
| 06:22.55 | mog | \you usually print on problems |
| 06:23.04 | Cuban0 | OKAY |
| 06:23.05 | Cuban0 | THEN |
| 06:23.12 | Cuban0 | sorry for caps |
| 06:23.26 | Cuban0 | why it does not detects it ? |
| 06:24.43 | Cuban0 | it does not looks like hardware problem |
| 06:24.43 | Cuban0 | because |
| 06:24.43 | Cuban0 | [root@elastix ~]# dahdi_hardware |
| 06:24.43 | Cuban0 | pci:0000:01:09.0 wcfxo- 1057:5608 Wildcard X100P |
| 06:26.05 | mog | your driver isnt loaded if the dadhi_cfg fails |
| 06:26.44 | Cuban0 | on bott up it says |
| 06:26.47 | Cuban0 | NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 |
| 06:26.47 | Cuban0 | Failed to initailize DAA, giving up... |
| 06:26.47 | Cuban0 | wcfxo: probe of 0000:01:09.0 failed with error -5 |
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| 07:02.24 | drmessano | 24 is branching out |
| 07:02.30 | drmessano | I saw an IP330 on 24 tonight |
| 07:03.55 | JT | polycom must've given more blowjobs to 24 producers |
| 07:04.00 | JT | than cisco |
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| 07:10.54 | jplank | tonight just isn't my night, first outgoing call problems, and when I get that fixed, this http://pastebin.com/m6d382526 when more then 3 or four calls come in at once |
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| 08:29.09 | *** join/#asterisk ultrav1olet (n=ultrav1o@92.255.134.224) |
| 08:32.58 | ultrav1olet | I cannot quite understand how I can implement the following things: we have three groups of employees in my organization, say super-users, sales managers and it department. I want let people from each group call every other person in the organization, and at the same I want to let super users call any numbers, and all sales calls should be recorded. How can I write a dialplan to make it all possible? |
| 08:33.36 | Octothorpe | From scratch, I have no idea. With FreePBX that wouldn't be too difficult to implement. |
| 08:33.59 | Octothorpe | I couldn't code a dialplan from scratch if you held a gun to my head. |
| 08:34.01 | Octothorpe | :) |
| 08:34.10 | ultrav1olet | :) |
| 08:34.34 | ultrav1olet | I do understand the extensions and dialplan, I just cannot see the whole picture |
| 08:36.00 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:37.35 | ultrav1olet | ok, is it possible to have an extension which is included by default for each and every user in iax.conf/sip.conf/users.conf? |
| 08:41.39 | kaldemar | ultrav1olet: make a context with all employees in it, and a contexts that allow any calls. include the context with all employees for the normal people and all the other too for the super-users. |
| 08:42.38 | kaldemar | simple and easy. |
| 08:44.01 | ultrav1olet | I'll try, thank you |
| 08:44.13 | kaldemar | if you want to go further, you can have all sorts of contexts with different kind of extensions and then different levels of rights class contexts that include different extension contexts. |
| 08:45.14 | *** join/#asterisk Badrobot- (i=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 08:45.26 | ultrav1olet | My problem is that I was thinking wrong - I created a nested structure of contextes and it no longer allows growing - I have undesired results from including one context into another one. So, I'll try your suggestion |
| 08:46.31 | *** join/#asterisk ew01f (n=angomg--@201.171.75.205.dsl.dyn.telnor.net) |
| 08:47.10 | kaldemar | drawing pictures is a good way to keep up with all the includes. :) |
| 08:47.25 | harryv | how do I figure out which framing/coding to use? and what are they acronyms for? |
| 08:48.00 | harryv | cas/ccs - ami/hdb3 |
| 08:48.06 | harryv | /crc4 |
| 08:48.13 | kaldemar | would be nice to have a script that would construct a dot file from a dialplan and then draw a graph with graphviz or something. |
| 08:48.46 | kaldemar | harryv: what country are you in? |
| 08:48.50 | harryv | denmark |
| 08:49.12 | kaldemar | what kind of interface do you have and what are you connecting it to? |
| 08:49.41 | harryv | digium TE121 -> nokia modem -> isdn30 |
| 08:51.01 | kaldemar | most likely you need ccs,hdb3 without crc4. |
| 08:51.27 | kaldemar | ccs means common channel signaling, i.e. you have a single separate channel that handles signaling for all the other channels. |
| 08:51.54 | kaldemar | cas would be channel associated signaling, so there would be no separate signaling channel (aka d-channel). |
| 08:53.29 | kaldemar | hdb3 coding is usually used in E1 PRI's, which you most likely have. |
| 08:53.44 | harryv | yep |
| 08:54.26 | kaldemar | then, define the 16th channel of a span as dchan and the rest as bchan. |
| 08:54.26 | harryv | atm I have the crc4 too.. |
| 08:54.52 | kaldemar | you may need it too. |
| 08:55.02 | harryv | will it do any harm? |
| 08:55.47 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 08:59.07 | kaldemar | depends on the other end |
| 08:59.44 | kaldemar | try it out. |
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| 09:18.53 | ultrav1olet | how can I implement this: an outsider calls our asterisk PBX and no matter if someone picked up or not an internal SIP telephone, a caller has three seconds to enter the number of another employee (without any announcement) - so that a caller could request any person without talking to a secretary |
| 09:24.54 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
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| 09:26.18 | *** join/#asterisk stmaher (n=stephen@mateus.province5.tv) |
| 09:26.57 | stmaher | Hi guys.. I have asterisk 1.6 and I believe its crashed twice. I cant see any core dump anywhere and the logfile doesnt show much.. |
| 09:27.24 | stmaher | COuld someone please advise me on what to do? possibly increse logging? |
| 09:29.49 | freh | stmaher: in the asterisk.conf file you can increase the debug level |
| 09:30.11 | freh | debug = x |
| 09:32.08 | freh | stmaher: Also in the asterisk.conf file you can set "dumpcore = yes" ... which dumps the core on a crash |
| 09:32.33 | stmaher | just saw that.. thank you :-).. I have no idea why its crashing.. will keep an eye.. |
| 09:32.35 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:32.44 | stmaher | If i do get logs and a coredump.. where do I send it to? |
| 09:33.16 | freh | I had some problems with 1.6 too. It crashed several times so now I'm using 1.4 |
| 09:33.40 | freh | I don't know, I'm just a regular user myself |
| 09:33.50 | stmaher | Interesting.. |
| 09:33.56 | stmaher | Thanks will take that onboard |
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| 10:13.23 | *** join/#asterisk scruz (n=scruz@196.216.253.116) |
| 10:13.30 | scruz | good day |
| 10:16.11 | *** join/#asterisk mort___ (n=mort@nsabfw1.nsab.se) |
| 10:23.21 | freh | So, actually, which is the latest stable release of asterisk? |
| 10:23.37 | freh | Is 1.6.0.6 stable? |
| 10:24.21 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:31.01 | ultrav1olet | it is |
| 10:31.44 | RypPn | tzafrir_laptop around? |
| 10:31.56 | tzafrir_laptop | RypPn, yup |
| 10:32.24 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:32.30 | tzafrir_laptop | freh, I think I'd actually try 1.6.0.7-rc1 (or is it 2?) |
| 10:32.30 | RypPn | hi, I dont remember if you remember me mentioning outbound issues with dahdi recently? |
| 10:32.50 | tzafrir_laptop | sure |
| 10:33.28 | RypPn | I've been doing some more digging trying to work out why http://bugs.digium.com/view.php?id=14577 doesn't work on 1.6.1.x and 1.6.2.x |
| 10:34.01 | RypPn | It would seem that the digits aren't send via the tdm card and the pstn provider times out giing their usual message |
| 10:34.09 | RypPn | giving* |
| 10:34.48 | RypPn | I'm hoping this is meaningful for someone with greater knowldge of the internals of how it all works :) |
| 10:35.14 | RypPn | digits=dialled digits |
| 10:35.36 | scruz | tzafrir_laptop: o/ |
| 10:36.19 | tzafrir_laptop | RypPn, what do you mean by "not sent"? |
| 10:36.31 | tzafrir_laptop | How do you see that? |
| 10:36.41 | RypPn | when I tail the full log, I see the digits on 1.6.0.7 |
| 10:37.05 | tzafrir_laptop | Asterisk does not ask dahdi to dial? |
| 10:37.30 | RypPn | yes, it asks it to dial, but the dialstring doesn't seem to be passed |
| 10:37.58 | tzafrir_laptop | I applied the same patch to the 1.6.1 (rc3) package I'm testing now |
| 10:38.03 | tzafrir_laptop | So far it look OK |
| 10:38.04 | scruz | tzafrir_laptop: o/ == "wave" |
| 10:38.34 | tzafrir_laptop | scruz, hi |
| 10:38.52 | tzafrir_laptop | it actually looked like some sort of a smily to me |
| 10:39.26 | *** join/#asterisk propellerhead (n=yogurt2u@200.43.87.56) |
| 10:39.33 | scruz | RypPn: is does chan_dahdi replace chan_ss7? |
| 10:39.39 | scruz | *-is |
| 10:39.52 | *** join/#asterisk smooth_penguin (n=smoove_@59.95.33.11) |
| 10:40.03 | RypPn | dahdi replaces zaptel |
| 10:40.05 | tzafrir_laptop | chan_dahdi (as of 1.6.0, IIRC) has SS7 support through libss7 |
| 10:40.18 | tzafrir_laptop | So in that respect it replaces chan_ss7 |
| 10:40.35 | scruz | so i just need to build libss7, right? |
| 10:40.51 | tzafrir_laptop | AFAIK chan_ss7 is still developed independently, but I'm quite unfamiliar with it |
| 10:41.12 | tzafrir_laptop | yes, you just need libss7 installed before building asterisk |
| 10:42.43 | scruz | ok, but i guess it would work on * 1.4 |
| 10:42.53 | scruz | as i'm using * 1.4 |
| 10:43.15 | scruz | likes freshmeat's new look. |
| 10:44.03 | tzafrir_laptop | doesn't |
| 10:44.19 | tzafrir_laptop | takes more hunting to see the same information |
| 10:44.34 | scruz | that's a fail. i noticed it too |
| 10:44.52 | scruz | the UI is more visually pleasing, is what i meant |
| 10:46.15 | scruz | ack, i'll need to use zaptel or upgrade *. guess i'll use zaptel |
| 10:46.50 | scruz | libss7 requires * 1.6+ |
| 10:47.03 | tzafrir_laptop | you can use dahdi with latest asterisk 1.4 |
| 10:47.24 | tzafrir_laptop | (not sure how chan_ss7 handles this, though) |
| 10:49.20 | kaldemar | stmaher: http://www.asterisk.org/doxygen/trunk/AstDebug.html |
| 10:50.02 | *** join/#asterisk destructure (n=de@67-23-12-32.static.slicehost.net) |
| 10:51.21 | scruz | i'll go the zaptel way because dahdi is relatively unused over here, and other people might need to work on the * install |
| 10:51.46 | scruz | and the book has no info on dahdi |
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| 11:00.04 | RypPn | tzafrir_laptop I think I've found the bit I'm referring to in my log if it helps? |
| 11:11.25 | *** join/#asterisk beherit (n=albert@203.153.6.16) |
| 11:12.07 | beherit | we are having problem navigating the banks IVR option, any idea where will i start troubleshooting. Is it correct that its a dtmf issue? |
| 11:12.14 | tzafrir_laptop | RypPn, can you be more specific? |
| 11:13.07 | Chainsaw | beherit: It might be. Are your DTMF signals not reaching their IVR at all or are they garbled? |
| 11:13.39 | RypPn | tzafrir_laptop sure, http://rafb.net/p/gRSOGx70.html not working , http://rafb.net/p/d69i8X20.html working |
| 11:14.25 | RypPn | 1st paste the string only feeds the 1st 0 on line 56, you can see in the 2nd paste the hole string is fed |
| 11:14.33 | RypPn | whole* |
| 11:15.10 | beherit | chainsaw, i think its just incomplete |
| 11:16.24 | Chainsaw | beherit: Try calling your cellphone and see whether holding down a key gets you a clear DTMF tone or not. |
| 11:17.10 | Chainsaw | beherit: Generally only uLaw & aLaw are able to support in-band DTMF (the other codecs will turn it into garble). |
| 11:17.58 | beherit | Chainsaw: we are using g729 codec, any idea how to resolve this issue? |
| 11:18.36 | Chainsaw | beherit: And you're using that outbound to the PSTN as well? |
| 11:19.25 | beherit | we don't have pstn |
| 11:20.00 | beherit | Chainsaw: its a small call center. with no PSTN just 2 voice provider. Both are using g729 codec |
| 11:20.40 | Chainsaw | beherit: Cellphone test still applies. |
| 11:20.44 | tzafrir_laptop | RypPn, have you noticed the digits get detected |
| 11:20.58 | tzafrir_laptop | this is a dialout? |
| 11:21.01 | tzafrir_laptop | What device? |
| 11:21.16 | RypPn | Its a sangoma A200D |
| 11:22.31 | RypPn | On the first paste? line 53? yeah I noticed that |
| 11:23.23 | RypPn | Strange why it stops after sending the first digit |
| 11:24.10 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:24.48 | Chainsaw | beherit: But I would like to quote this to you, from wikipedia: "Music or DTMF tones can only be transported reliably with this codec using the RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals as specified in RFC 2833" |
| 11:29.39 | tzafrir_laptop | RypPn, the detected digits could be related to echo. In both cases Asterisk did send the full number |
| 11:29.56 | tzafrir_laptop | And we see that in the bad case the dialing stopped in the middle |
| 11:30.48 | beherit | yes i am using rfc2833 in dtmfmode |
| 11:30.52 | tzafrir_laptop | Is it the same version of dahdi in both cases? Of the Sangoma drivers? |
| 11:31.11 | RypPn | tzafrir_laptop yes, dahdi is 2.1.0.4 and wanpipe is 3.3.16 |
| 11:33.09 | tzafrir_laptop | In that trace I didn't see any explicit disconnect. Rather it was DAHDI (the kernel) that signalled an early "dial complete" |
| 11:33.10 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 11:34.40 | RypPn | tzafrir_laptop If it helps to get a cleaner trace I can put 1.6.1.x or 1.6.2.x back on before the wife gets home :) |
| 11:39.22 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 11:39.56 | *** join/#asterisk smooth_penguin (n=smoove_@59.95.55.181) |
| 11:41.44 | RypPn | tzafrir_laptop you think this could be oslec-related in 1.6.2 ? |
| 11:42.05 | tzafrir_laptop | I still can't see how this is related |
| 11:42.31 | tzafrir_laptop | for some reason the dialing buffer got reset. I'm not sure how |
| 11:42.52 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:43.16 | RypPn | I can set up 1.6.1 again and give you ssh access if it helps, lemme know :) |
| 11:45.03 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 11:58.32 | harryv | I have the TE120 series guide, but it doesn't say: Where is the e1/t1 jumper on te121? |
| 11:58.46 | harryv | I can see two jumbers, labelled p3 and p8 |
| 11:59.28 | ultrav1olet | how can I make MixMonitor record only an actual conversation and not error codes like congestion, busy, etc? |
| 11:59.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:08.44 | *** join/#asterisk Ether_Man (i=Ether_Ma@h229n2fls310o1101.telia.com) |
| 12:08.53 | bobsaccamano | hi all..im trying to simulate international direct distance dialing for SIP Channels in asterisk...accordingly my dial plan looks like this: http://pastebin.com/m373dc176 |
| 12:09.16 | bobsaccamano | I am getting a 503 Server error..saying the HangupCause code=20 |
| 12:09.30 | bobsaccamano | any idea where im missing the plot |
| 12:09.30 | bobsaccamano | ? |
| 12:10.21 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 12:10.39 | harryv | found it. |
| 12:12.00 | *** part/#asterisk fish-bulb (n=cstewart@nat/digium/x-31143f1c9f5a014f) |
| 12:12.36 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-f2afe336fea09726) |
| 12:12.48 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:15.48 | harryv | w/ E1, will channel 16 always be d-chan? |
| 12:16.59 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.rubicom.hu) |
| 12:18.57 | tzafrir_laptop | yes |
| 12:20.19 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 12:20.23 | ultrav1olet | how can I make MixMonitor record only an actual conversation and not error codes like congestion, busy, etc? Or is it possible to erase the call if it wasn't a real call? |
| 12:26.15 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:27.01 | kaldemar | ultrav1olet: check DIALSTATUS and remove the files according to it. |
| 12:27.08 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 12:28.17 | kaldemar | bobsaccamano: you don't have a SIP peer 544444444444444 |
| 12:36.48 | ultrav1olet | OK, here's how my dialplan looks like: http://pastebin.ca/1370362 |
| 12:37.54 | ultrav1olet | two questions: I cannot access $(filename) in a Macro; how can I call system(/bin/rm) upon !receiving ANSWER? |
| 12:38.12 | ultrav1olet | ! = NOT |
| 12:39.13 | ultrav1olet | I don't want to add a command for every s-STATUS where STATUS != Answer |
| 12:40.01 | kaldemar | give ${filename} as an argument for the macro |
| 12:40.35 | ultrav1olet | good idea! what about the second question? how can I call system(/bin/rm) upon !receiving ANSWER? |
| 12:41.16 | kaldemar | if you goto to the dialstatus, you have to do something in every different status. |
| 12:41.31 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:42.34 | harryv | http://pastie.org/425325 - what's going on there? TE121 |
| 12:43.27 | harryv | but every 2nd zap restart gives http://pastie.org/425326 |
| 12:43.44 | ultrav1olet | kaldemar: that doesn't sound nice :( OK, I'll do that way |
| 12:44.19 | kaldemar | harryv: you have a signaling problem. yellow alarm. |
| 12:45.03 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 12:45.11 | coppice | your signaling is afraid |
| 12:45.34 | harryv | wtf. I shut down the computer, took the card out, sat it back in, it is recognized. now. trouble. :( |
| 12:47.17 | bobsaccamano | kaldemar, i do |
| 12:47.25 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 12:48.48 | ultrav1olet | How can I make this extension simpler? http://pastebin.ca/1370369 |
| 12:49.22 | ultrav1olet | Is there a way to make the first two commands of each extension a function (extension,macro,etc)? |
| 12:50.26 | zeeesh | we have cdrs in Master.csv... can we find rtp logs.... i need to know when both legs established. and when disconnected... is there any posibility... ? |
| 12:52.32 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 12:52.35 | kaldemar | ultrav1olet: make an extensions that matches all those three with only priorities 2 and 3. |
| 12:53.39 | kaldemar | -s |
| 12:53.39 | harryv | kaldemar: how would I diagnose this? there's connection to the network terminal |
| 12:55.39 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 12:56.35 | ultrav1olet | kaldemar: how can I do that? |
| 12:56.53 | ultrav1olet | do you mean regular expression? |
| 12:59.07 | kaldemar | ultrav1olet: no, a regular pattern |
| 13:00.11 | ultrav1olet | http://www.google.com/search?hl=en&q=asterisk+regular+pattern&btnG=Search nothing :( |
| 13:00.39 | kaldemar | a pattern just like the ones you have in that context. |
| 13:00.58 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.rubicom.hu) |
| 13:01.02 | ultrav1olet | kaldemar: my numbers are quite different - I cannot think I can come up with a pattern that matches all of them |
| 13:01.03 | kaldemar | but one that matches all those three |
| 13:01.34 | kaldemar | ultrav1olet: yes you can |
| 13:01.36 | ultrav1olet | _89XXXXXXXXX _8800XXXXXXX and _2XXXXXX |
| 13:02.03 | ultrav1olet | my head exploded :) |
| 13:02.13 | kaldemar | _XXXXXXX. would match all those |
| 13:02.46 | *** join/#asterisk trelane (i=trelane@funtoo/staff/trelane) |
| 13:03.01 | ultrav1olet | I don't get you - the last one has only 7 digits, the first two ones have 10 digits |
| 13:03.28 | kaldemar | . <- matches to any amount of anything |
| 13:03.29 | harryv | ultrav1olet: look at the . |
| 13:05.48 | ultrav1olet | ok :) |
| 13:05.54 | kaldemar | ahem. . is one or more, not any amount. |
| 13:06.14 | *** join/#asterisk Thiago_Lima (n=tslima@201-13-219-237.dial-up.telesp.net.br) |
| 13:06.23 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 13:07.46 | ultrav1olet | hm, none of System(rm filename) in http://pastebin.ca/1370387 works. MixMonitor creates file _after_ hangup thus, the file is not deleted :( |
| 13:08.45 | ultrav1olet | any ideas? |
| 13:10.23 | kaldemar | make a hangup extension in the context that triggered the macro |
| 13:11.13 | ultrav1olet | but it needs to run only based on DIALSTATUS - how can I do that? |
| 13:11.59 | *** join/#asterisk orly_owl (n=DavoDink@c122-108-50-15.sunsh1.vic.optusnet.com.au) |
| 13:12.29 | orly_owl | Is it possible to test an IP phone by setting up an asterisk server to connect it to? |
| 13:12.38 | kaldemar | gosub instead of macro would be handy for that, you could return to the previous context with app Return. |
| 13:13.20 | kaldemar | but since you use a macro, try to save the status to a channel variable. |
| 13:14.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:14.52 | *** join/#asterisk jeff_phillips (n=ceramics@66.112.49.13) |
| 13:15.39 | *** join/#asterisk destructure (n=de@67-23-12-32.static.slicehost.net) |
| 13:18.04 | *** join/#asterisk mort_gib (n=mjensen@83.36.63.16) |
| 13:18.20 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 13:18.28 | *** part/#asterisk orly_owl (n=DavoDink@c122-108-50-15.sunsh1.vic.optusnet.com.au) |
| 13:19.03 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:19.15 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 13:19.15 | ultrav1olet | I created a very cool hack :) |
| 13:19.26 | jeff_phillips | ? |
| 13:20.10 | ultrav1olet | I made a wrapper for a /bin/rm which is run in background (#! /bin/sh /bin/sleep 60; /bin/rm "$1" &>/dev/null) |
| 13:21.21 | ultrav1olet | then I ran System(/usr/local/bin/slowrm /var/spool/asterisk/monitor/${ARG2} &) |
| 13:21.45 | ultrav1olet | so now it asterisk deletes a file past hangup ;) |
| 13:21.50 | *** join/#asterisk ReD-MaN (i=rox-ur-s@216.75.172.220) |
| 13:24.06 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 13:24.45 | *** join/#asterisk path_ (n=path_@240-117-21-190.adsl.terra.cl) |
| 13:25.42 | tzafrir_laptop | ultrav1olet, not "$1" . "$@" |
| 13:26.07 | tzafrir_laptop | and use 'exec' to save an unnecessary process |
| 13:26.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:27.08 | ultrav1olet | tzafrir_laptop: I delete one file only, there's no need for "$@" :) |
| 13:27.48 | ultrav1olet | But will exec work? I spawn a process in background (note & at the end) |
| 13:28.44 | tzafrir_laptop | And don't underestimate the run-time of rm: http://lists.debian.org/debian-user/2009/03/msg01507.html |
| 13:28.48 | tzafrir_laptop | :-) |
| 13:29.31 | ultrav1olet | LOL |
| 13:29.42 | tzafrir_laptop | why do you run it in the background only after 60 seconds? |
| 13:29.57 | ultrav1olet | That man definitely needs a FS on top of RAM disk |
| 13:30.08 | tzafrir_laptop | The sleep should also be in the background |
| 13:30.25 | ultrav1olet | most normal people will hang up if there's no answer for sixty seconds |
| 13:30.47 | tzafrir_laptop | you should have a subshell in the background. That subshell will exec rm eventually |
| 13:30.52 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:30.53 | ultrav1olet | my whole System runs in a background, so sleep shouldn't run in a background |
| 13:32.27 | ultrav1olet | my only concern in http://pastebin.ca/1370387 is the following: can s-ANSWER become s-DIFFERENT_STATUS? |
| 13:32.48 | ultrav1olet | is it theoretically possible? |
| 13:33.02 | ThoMe | hello |
| 13:33.08 | ThoMe | is it posible to list all meetme rooms |
| 13:33.18 | ThoMe | also rooms what not active at the moment |
| 13:33.27 | ThoMe | example all in my meetme.conf |
| 13:35.07 | *** part/#asterisk ultrav1olet (n=ultrav1o@92.255.134.224) |
| 13:35.26 | kaldemar | ThoMe: sure, use meetme.conf to list them |
| 13:35.44 | kaldemar | where are you accessing the information? |
| 13:36.14 | ThoMe | manager |
| 13:38.59 | russellb | i think there is a manager action for that ... |
| 13:39.08 | ThoMe | russellb: hey. ok. thank you. |
| 13:39.16 | ThoMe | russellb: the command "MeetmeAdmin" what is it? |
| 13:39.20 | ThoMe | http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetmeAdmin |
| 13:39.25 | ThoMe | but i havent the command |
| 13:39.54 | russellb | ThoMe: MeetmeList is the manager action i was thinking of |
| 13:41.55 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 13:42.03 | ThoMe | russellb: No such command 'MeetmeList' (type 'help MeetmeList' for other possible commands) |
| 13:42.09 | ThoMe | fputs($socket, "Command: MeetmeList\r\n\r\n"); |
| 13:42.22 | ThoMe | and fputs($socket, "Command: meetme list $room\r\n\r\n"); list only the active rooms. |
| 13:42.25 | Ether_Man | What modules do I need for a pure SIP phone environment? It cant be just chan_sip.so can it? |
| 13:43.57 | [TK]D-Fender | Ether_Man: All related dialplana pps, MoH, Features, etc. Piles of stuff |
| 13:45.29 | *** join/#asterisk Sanjoy (n=Sanjoy@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 13:45.54 | kaldemar | looks like MeetmeList is only in 1.6. |
| 13:47.04 | Ether_Man | [TK]D-Fender, and how do I find out exactly which ones of those I need? :/ |
| 13:47.48 | [TK]D-Fender | Ether_Man: I doubt there is a guide anywhere. Try stuff and see where things fail. |
| 13:48.02 | [TK]D-Fender | Ether_Man: What is your goal? |
| 13:48.24 | Ether_Man | Single network, just internal SIP phone connections |
| 13:49.23 | kaldemar | asterisk has so many modules for basic functionality that IMO it is better to start with all and drop stuff when you know you don't need it. |
| 13:50.15 | [TK]D-Fender | Ether_Man: No, why are you trying to go ultra-minimalist on the basic modules? |
| 13:50.46 | Ether_Man | Because I dont like having anything loaded that Im not going to use |
| 13:51.53 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:51.56 | [TK]D-Fender | Ether_Man: Don't waste your time going nuts over this at the start. Typically you'll only want to NOLOAD specific modules you know you won't need like H.323, Skinny/SCCP, etc that could pose an actual risk |
| 13:52.33 | [TK]D-Fender | Ether_Man: Anything more tends to indicate you're either running on an extremely limited embedded environment or jsut another Gentoo Ricer :p |
| 13:52.36 | *** join/#asterisk copas2 (n=copas2@apps.zalaszam.hu) |
| 13:52.50 | copas2 | hi |
| 13:53.38 | copas2 | i'm looking for a support person regarding asterisk g.729 codec licenses |
| 13:54.03 | Ether_Man | [TK]D-Fender, no not really.. It means I actually care about not running useless things I have no reason to run if I dont have to... |
| 13:54.04 | [TK]D-Fender | copas2: Call up Digium support. You bought it, they support it |
| 13:54.12 | *** join/#asterisk chazz (n=chazz@173-24-217-85.client.mchsi.com) |
| 13:55.06 | Ether_Man | [TK]D-Fender, like, there's a reason Im running Arch instead of say Ubuntu or Fedora |
| 13:55.12 | copas2 | already emailed them but it's kinda urgent |
| 13:56.06 | [TK]D-Fender | copas2: then just state your questions and maybe someone with experience is around |
| 13:56.42 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:56.49 | copas2 | just want to split the newly acquired licenses to 6 different ones |
| 13:56.51 | [TK]D-Fender | Ether_Man: Ok, well there isn't any guide I'm aware of for this. feel free to start Googling or go by trial & error |
| 13:56.57 | jeff_phillips | is g.729 worth buying? I mean, is the sound quality comparable to make it worth the savings in bandwidth? |
| 13:57.13 | [TK]D-Fender | copas2: then you DEFINITELY need to contact Digium support |
| 13:57.37 | [TK]D-Fender | jeff_phillips: If you need the BW, then G.729 is the way to go. |
| 13:57.40 | copas2 | jeff_phillips: it was inevitable because of bandwidth situation |
| 13:57.44 | copas2 | right |
| 13:58.12 | copas2 | jeff_phillips: and the pstn gateway is accepting only G... codecs |
| 13:58.26 | jeff_phillips | Well, "need" and "can definately use" are two different things... I could certainly benefit from using less of the limited bandwidth we have, but I don't want to sacrafice sound quality |
| 13:59.07 | jeff_phillips | if it sounds about the same I might want to buy a license |
| 13:59.08 | coppice | G.729 is pretty good until you call a cellphone |
| 13:59.20 | jeff_phillips | oh, how does it sound with a cell phone? |
| 13:59.48 | copas2 | jeff_phillips: you'll have to make that compromise |
| 13:59.52 | coppice | you end up with two low bit rate codecs in series and it sounds pretty nasty |
| 14:00.04 | jeff_phillips | i see |
| 14:00.27 | jeff_phillips | well a lot of our SIP calls are to/from cell phones so that might not be such a good idea for me then |
| 14:01.13 | copas2 | Ether_Man: so i will have to wait for an answer by email |
| 14:01.52 | copas2 | jeff_phillips: use voip/cell gateways then |
| 14:02.32 | jeff_phillips | HMMM..... |
| 14:03.15 | jeff_phillips | nah, even with mobile to mobile it'd be cheaper *for me* to do it through SIP |
| 14:04.48 | jplank | cam someone help me debug a core dump, asterisk keeps crashing when theres 2 or 3 calls coming in over a 2400p with FXO modules, no errors in /var/log/asterisk/full |
| 14:06.16 | *** join/#asterisk sakajawebe (n=chazz@nat/digium/x-5eeed25c2e206425) |
| 14:06.33 | sakajawebe | copas2: have you already activated your G.729 license? |
| 14:07.00 | copas2 | sure :-) |
| 14:07.18 | sakajawebe | eh, then there won't be anything that can be done for splitting then |
| 14:08.08 | copas2 | why not, what makes the difference? |
| 14:08.24 | copas2 | there is one more chance to active them |
| 14:08.35 | sakajawebe | because once it has been activated, there is a license for the full amount of channels for that key |
| 14:09.02 | sakajawebe | the only way to split them is to change teh one key so that it doesn't have that many channels anymore, and then create new keys for the difference in channels |
| 14:09.02 | copas2 | then it has to be deleted and 6 new ones to be created |
| 14:09.10 | ThoMe | russellb: have my meetme-rooms now insert in my mysql-db. |
| 14:09.20 | ThoMe | russellb: now i can make a small query for list my all rooms. |
| 14:09.21 | ThoMe | :-) |
| 14:09.27 | sakajawebe | but see, deleting it doesn't stop that one activation license that has already happened |
| 14:09.28 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:09.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:09.32 | copas2 | sakajawebe: but i cannot change it, can i? |
| 14:09.52 | sakajawebe | cannot change what? the key? |
| 14:09.57 | copas2 | sakajawebe: yes |
| 14:10.24 | sakajawebe | you can't change it, but now you have the license file, so you can use it on that system |
| 14:10.25 | copas2 | sakajawebe: that's what digium support does |
| 14:10.53 | copas2 | sakajawebe: that system doesn't need it that's why it has to be split |
| 14:11.09 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:11.16 | sakajawebe | but now that system has it, and there isn't anyway to validate that it no longer does |
| 14:11.47 | copas2 | sakajawebe: since we are just reselling (and unfortunately installing) them, we didn't want to buy six different licenses for that |
| 14:12.06 | copas2 | sakajawebe: still hope that support can fix it |
| 14:12.38 | sakajawebe | I'm pretty sure once it has been activated there is nothing that can be done |
| 14:12.51 | copas2 | sakajawebe: how can you be so sure? |
| 14:13.26 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 14:14.02 | jplank | how could I debug a segmentation fault? |
| 14:14.14 | copas2 | sad thing is that it's written nowhere what to do in that case. i mean if you want to install that license on multiple system. it might be obvious for digium but it's not for end users |
| 14:14.19 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:14.35 | sakajawebe | it is against the license agreement to install a license on more than one system |
| 14:14.36 | copas2 | systems |
| 14:14.51 | copas2 | aha |
| 14:15.13 | copas2 | so it reveals itself only after installing |
| 14:15.22 | Nugget | the g729 codec used to be tied to a specific machine (via mac address) |
| 14:15.30 | copas2 | on that multiple page agreement :-) |
| 14:15.31 | Nugget | probably still is, but I dunno |
| 14:15.47 | sakajawebe | the agreement is avialable online in the documentation section |
| 14:15.54 | copas2 | Nugget: it's okay. i can abandon that license, it's not a problem |
| 14:17.06 | copas2 | sakajawebe: okay, okay. still hope that support can do something. if not, we have to return it and buy new ones. they cannot handle this in an agreement can they |
| 14:17.29 | copas2 | it's just a product |
| 14:17.53 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:19.31 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 14:20.20 | stmaher | \Hi guys.. Im trying to add people automatically to a conference.. but with auto answer on a polycom phone |
| 14:20.40 | stmaher | I tried the outbound call drop file option.. but need to specify a SIP origin.. |
| 14:20.56 | stmaher | I want that SIP origin to be an extenion in extensions.conf.. but its not working.. |
| 14:20.58 | stmaher | any suggestions? |
| 14:24.10 | keith4 | wtf? Skype SIP? |
| 14:24.20 | [TK]D-Fender | stmaher: Show us what you've done so far |
| 14:24.54 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.24.187) |
| 14:25.32 | VaGoNeTaS | hi |
| 14:25.44 | VaGoNeTaS | i got this error installing asterisk on a brand new machine, 1.4.24 |
| 14:26.01 | VaGoNeTaS | root@reportes:/usr/src/asterisk-1.4.24# make menuselect |
| 14:26.01 | VaGoNeTaS | ************************************************** |
| 14:26.01 | VaGoNeTaS | *** Install ncurses to use the menu interface! *** |
| 14:26.15 | VaGoNeTaS | but |
| 14:26.18 | VaGoNeTaS | root@reportes:/usr/src/asterisk-1.4.24# apt-get install libncurses-dev |
| 14:26.23 | VaGoNeTaS | is already the last version |
| 14:26.29 | *** join/#asterisk PDani (n=pekdanie@catv-89-133-156-227.catv.broadband.hu) |
| 14:26.35 | VaGoNeTaS | somebody knows what package is missing? |
| 14:26.40 | stmaher | here is my drop file |
| 14:26.41 | stmaher | Channel: SIP/3006 |
| 14:26.41 | stmaher | MaxRetries: 2 |
| 14:26.42 | stmaher | RetryTime: 60 |
| 14:26.42 | stmaher | Context: int-hardphones |
| 14:26.44 | stmaher | Extension: 2441 |
| 14:26.55 | stmaher | Channel: <- can you omit that? |
| 14:28.32 | BlargMaN00 | VaGoNeTaS: What distro and version are you using?? |
| 14:28.44 | anonymouz666 | stmaher: why you would omit that? |
| 14:28.50 | VaGoNeTaS | Ubuntu 8.04 |
| 14:28.55 | VaGoNeTaS | and is asterisk 1.4.24 |
| 14:29.18 | VaGoNeTaS | the machine is new, but i've just installed the libncurses-dev |
| 14:29.24 | VaGoNeTaS | and still not working |
| 14:29.32 | [TK]D-Fender | stmaher: No. You are telling * to call the "channel" and on answer dump them in the specified place in the dialplan. |
| 14:29.41 | BlargMaN00 | try 'apt-get install libncurses5 libncurses5-dev' and see if it installs... |
| 14:29.58 | copas2 | VaGoNeTaS: bc pciutils patch unifdef mysql-server libmysqlclient15-dev make gcc g++ libncurses5-dev apache2 php5 libxml2-dev libtiff4-dev lame php-pear php5-mysql php5-gd libssl-dev libcpan-mini-perl bison libaudiofile-dev curl sox php-db flex xsltproc unixodbc-dev mpt-status acpid libnewt-dev libsqlite0-dev libsqlite3-dev ntp git-core atftpd python-xml |
| 14:30.12 | VaGoNeTaS | libncurses5 ya está en su versión más reciente. |
| 14:30.19 | VaGoNeTaS | that is in spanish |
| 14:30.24 | VaGoNeTaS | but that means that is already in the last version |
| 14:31.03 | VaGoNeTaS | BlargMaN00 : its already installed both of it |
| 14:31.05 | VaGoNeTaS | both packages |
| 14:31.29 | VaGoNeTaS | it seems that another package is missing |
| 14:31.32 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:31.34 | VaGoNeTaS | besides of ncurses |
| 14:31.39 | BlargMaN00 | do this... 'dpkg -l | grep ncurses' and tell me what it says... |
| 14:31.42 | BlargMaN00 | ~p |
| 14:31.43 | jbot | methinks p is q and not q |
| 14:31.57 | stmaher | [TK]D-Fender Ok perhaps I should rephrase.. How would i join a load of people to a conference.. with auto pickup? |
| 14:32.40 | [TK]D-Fender | stmaher: autopickup usually requires you to add headers to instruct the phone to do so |
| 14:32.50 | [TK]D-Fender | stmaher: And multiple phone would be multiple call-files. |
| 14:33.55 | *** join/#asterisk jshriver (n=jshriver@72.240.39.37) |
| 14:33.58 | jshriver | greetings |
| 14:34.22 | jshriver | Is it possible in asterisk to require a user to type in a number (password) prior to making a PoTS call? |
| 14:34.32 | [TK]D-Fender | jshriver: Yes. |
| 14:34.34 | jshriver | how? |
| 14:34.49 | [TK]D-Fender | jshriver: this is YOUR dialplan, go prompt them yourself. "core show application read" |
| 14:35.05 | jshriver | main reason I ask, is that we want to keep track of who is making calls that are not business related, LD, or 800 numbers. |
| 14:35.38 | jshriver | ok, second question :) |
| 14:35.59 | jshriver | I keep getting cdr errors saying database is full. I see it's a sqlite db, but the file itself is only a couple megs dont understand how it can be full |
| 14:36.06 | jshriver | how can I clear it? |
| 14:36.45 | jshriver | no command core |
| 14:37.04 | [TK]D-Fender | jshriver: remove "core" as you appear to be on 1.2 or lower. |
| 14:37.13 | jshriver | aye that did it ty |
| 14:37.18 | jshriver | 1.2 :) |
| 14:37.41 | [TK]D-Fender | jshriver: And welcome to 2009 where 1.6.0 is current and 1.6.1 nearling release |
| 14:38.12 | jshriver | I'm new to * this was kind dumped on me and my boss doesnt like to upgrade |
| 14:38.21 | mort_gib | Just had a meeting with a prospective client, manager comes in stressing that he will not accept any extra payment for Voicemail :-) |
| 14:38.27 | jshriver | we're still using Mandriva 2006 for a server lol |
| 14:39.02 | jshriver | anyway I appreciate the leads, read looks nice. |
| 14:39.03 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:39.03 | mort_gib | Turns out they had an Avaya dude doing that trick with them, trying to sneak some EUR 1500 in under the radar He |
| 14:39.04 | [TK]D-Fender | jshriver: And I recognize that you've been around here for a few months now... |
| 14:39.19 | jshriver | yup off on. Learning just enough to do what is needed. |
| 14:40.05 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 14:40.59 | jshriver | Anyway appreciate the help. Off to resume work, hope you all have a good day. |
| 14:41.08 | *** part/#asterisk jshriver (n=jshriver@72.240.39.37) |
| 14:41.59 | *** join/#asterisk bbryant (n=bbryant8@68.208.65.34) |
| 14:43.35 | coppice | mort_gib: I find that hard to believe. when did Avaya charge that little for *anything*? |
| 14:44.18 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:44.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:45.12 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 14:45.54 | chazz | VaGoNeTaS: try running make distclean in the asterisk directory and starting over and see if that helps at all |
| 14:45.55 | [TK]D-Fender | coppice: Oh... he meant PER BOX :p |
| 14:45.57 | mort_gib | coppice: That was for a 5 user install |
| 14:46.11 | mort_gib | coppice: My client is 120 or something |
| 14:47.07 | VaGoNeTaS | ok, i've just fixed the ncurses problem |
| 14:47.09 | mort_gib | coppice: and then they asked if it was possible to do conference (three way calling) and how much extra that would cost B-) |
| 14:47.20 | VaGoNeTaS | now i need to where can i get this module: |
| 14:47.22 | VaGoNeTaS | cdr_adaptive_odbc |
| 14:48.53 | *** part/#asterisk NotJerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 14:48.59 | BlargMaN00 | VaGoNeTaS: 'apt-get install unixodbc' |
| 14:49.22 | harryv | how do I define which of my two te121 are used for what in /etc/zaptel.conf ? I want to swap their functionality. change span number, or? |
| 14:50.39 | VaGoNeTaS | is already installed but i cannot select that on the menuselect |
| 14:50.43 | VaGoNeTaS | it appears like XXX |
| 14:51.13 | BlargMaN00 | VaGoNeTaS: did you install it yourself, or was it pre-installed on the distro?? |
| 14:51.31 | VaGoNeTaS | i belive was pre-installed coz as i told you before this machine is brand new |
| 14:51.41 | VaGoNeTaS | i've just installed ubuntu server 8.04 hardy on it |
| 14:51.43 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 14:52.09 | VaGoNeTaS | what's your suggestion related to this? |
| 14:52.40 | path_ | VaGoNeTaS hahahah you're playing around asterisk |
| 14:52.42 | mort_gib | VaGoNeTaS: So you need to log to a database?? |
| 14:52.43 | BlargMaN00 | VaGoNeTaS: 'apt-get install libltdl3 libltdl3-dev' |
| 14:53.02 | BlargMaN00 | VaGoNeTaS: then you will need to rerun the ./configure script |
| 14:54.33 | VaGoNeTaS | what u mean im playing around with asterisk? |
| 14:55.15 | VaGoNeTaS | este qliao |
| 14:55.32 | VaGoNeTaS | what tha fuck r u doing here dude |
| 14:56.24 | path_ | having fun :-D |
| 14:56.37 | path_ | I thought you where a security expert and that nobody knows more than you do |
| 14:56.41 | path_ | s/where/were |
| 14:57.15 | harryv | as far as I can tell it's determined by the order of the module load. but, this is two identical cards, thus using the same module |
| 14:58.25 | *** join/#asterisk bbryant (n=bbryant8@68.208.65.34) |
| 14:58.34 | VaGoNeTaS | that was more that 10 years ago |
| 14:58.48 | VaGoNeTaS | u cant blaim me , i was 16 years old |
| 15:00.31 | jeff_phillips | haha, well if that were true we wouldn't have been able to 'blame' jacob in my class in high school for lieing about how he had a limo to take to foot ball practice |
| 15:02.13 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:02.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:02.41 | VaGoNeTaS | maybe |
| 15:03.16 | VaGoNeTaS | people grows up |
| 15:03.18 | VaGoNeTaS | i did |
| 15:03.21 | VaGoNeTaS | he didnt |
| 15:03.25 | copas2 | well, thanks, bye |
| 15:03.35 | copas2 | . |
| 15:03.50 | jeff_phillips | VaGoNeTaS: You sure you 'grows' up? lol |
| 15:04.39 | VaGoNeTaS | maybe my english is not the best in the world but im almost sure that u can understand the point |
| 15:05.23 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 15:06.46 | [TK]D-Fender | VaGoNeTaS: "Understanding" isn't jeff_phillips' strong suit ;) |
| 15:07.03 | [TK]D-Fender | zing! |
| 15:07.16 | VaGoNeTaS | maybe, thought you were here to help |
| 15:08.00 | [TK]D-Fender | VaGoNeTaS: Anyway, whats left for your install? |
| 15:09.35 | VaGoNeTaS | i've just installed dahdi-linux and dahdi-tools |
| 15:09.45 | VaGoNeTaS | did the make config on dahdi-tools so he made the scripts |
| 15:09.47 | VaGoNeTaS | init scripts |
| 15:10.20 | VaGoNeTaS | im on the 'make menuselect' option |
| 15:10.27 | VaGoNeTaS | chan_dahdi is XXX |
| 15:10.44 | VaGoNeTaS | and cdr_odbc XXX |
| 15:10.45 | [TK]D-Fender | VaGoNeTaS: Configure and initialze dahdhi first |
| 15:10.48 | jasonwoot | ZOMG, so I JUST located "all-your-base.gsm" in \var\lib\asterisk\sounds |
| 15:11.01 | [TK]D-Fender | VaGoNeTaS: and you'll have to do "./configure" after to ahve * pick it up |
| 15:11.08 | [TK]D-Fender | jasonwoot: lol |
| 15:11.18 | VaGoNeTaS | actually i've removed the asterisk-1.4.24 |
| 15:11.21 | VaGoNeTaS | and started all over |
| 15:11.38 | jasonwoot | oh my Gods, who do I fracking thank for that one? |
| 15:11.45 | VaGoNeTaS | root@reportes:/usr/src/dahdi-tools-2.1.0.2# /etc/init.d/dahdi stop |
| 15:11.45 | VaGoNeTaS | dahdi_cfg not executable |
| 15:12.07 | VaGoNeTaS | and |
| 15:12.07 | VaGoNeTaS | root@reportes:/usr/src/dahdi-tools-2.1.0.2# /etc/init.d/dahdi start |
| 15:12.07 | VaGoNeTaS | dahdi_cfg not executable |
| 15:12.49 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 15:12.51 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:15.40 | VaGoNeTaS | what seems to be the problem there? |
| 15:16.20 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 15:16.28 | tzafrir_laptop | VaGoNeTaS, ls -l /usr/sbin/dahdi_cfg |
| 15:16.28 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 15:16.34 | VaGoNeTaS | there is no file |
| 15:16.43 | tzafrir_laptop | maybe you didn't run 'make install' ? |
| 15:16.46 | VaGoNeTaS | i was looking for dahdi_cfg on /usr/sbin |
| 15:16.47 | VaGoNeTaS | i did |
| 15:16.50 | raden_work | anyone recommend a good softswitch ? |
| 15:16.58 | VaGoNeTaS | just a min |
| 15:18.18 | harryv | [Mar 24 16:18:07] WARNING[3812]: chan_zap.c:9294 pri_dchannel: PRI Error on span 0: We think we're the network, but they think they're the network, too. |
| 15:18.43 | VaGoNeTaS | still got this: |
| 15:18.43 | VaGoNeTaS | XXX 4. cdr_odbc |
| 15:18.48 | harryv | there's not even a span 0 !? I have span 1 and 2, and span 2 is signalling=pri_cpe |
| 15:20.18 | tzafrir_laptop | VaGoNeTaS, you need to re-run ./configure (in asterisk) |
| 15:21.06 | tzafrir_laptop | And if 'make install' did not install it: maybe it has failed in the middle? |
| 15:21.11 | VaGoNeTaS | i did tzafrir_laptop |
| 15:21.15 | tzafrir_laptop | Did it complete successfully? |
| 15:21.31 | jaytee | harryv, pastebin your zaptel.conf and zapata.conf files |
| 15:22.27 | VaGoNeTaS | i was missing |
| 15:22.35 | VaGoNeTaS | unixodbc-dev |
| 15:22.45 | VaGoNeTaS | hehehe, sorry |
| 15:23.00 | VaGoNeTaS | what i need is cdr_adaptive_odbc |
| 15:23.04 | raden_work | would it be possible to setup a VOIP service provider with asterisk ? |
| 15:23.21 | [TK]D-Fender | raden_work: Sure |
| 15:23.23 | *** join/#asterisk HuntsMan (n=hunts@88-69-246-201.adsl.terra.cl) |
| 15:23.39 | harryv | jaytee: http://sprunge.us/DICY & http://sprunge.us/jEjj |
| 15:23.48 | raden_work | [TK]D-Fender, I understand id need software for billing but anything else ? |
| 15:24.04 | [TK]D-Fender | raden_work: What do you think? |
| 15:24.29 | raden_work | [TK]D-Fender, what do you mean what do i think ? |
| 15:24.51 | [TK]D-Fender | raden_work: You tell ME what you think you need to become a provider. |
| 15:25.08 | raden_work | why are you answering my question with a question |
| 15:25.43 | raden_work | need some sort of soft switch , billing software and wholesale termination / origination service and alot of patients, tech support setup etc... |
| 15:25.48 | [TK]D-Fender | raden_work: Because if you can't come up with a basic list of the kind of things required then you would seem to be completely out of your element in trying to take on such a task. |
| 15:26.30 | [TK]D-Fender | raden_work: Sounds like you've got a basic list there. |
| 15:26.47 | raden_work | [TK]D-Fender, I'm just looking to get something together for like 5 people on one block run and test for a while then expand local phone company ripping us $36 a month for land line + 37 for DSL its nutz |
| 15:27.21 | [TK]D-Fender | raden_work: And what will you be offering them? |
| 15:27.22 | raden_work | not looking to be the next vonage just need to know the basics of what i might need and go from there looking for suggestions thats all |
| 15:27.47 | harryv | jaytee: span 1 is connected to another box in here, span 2 to the telco. (I'm forwarding some traffic from span 2 -> span 1) |
| 15:27.47 | [TK]D-Fender | raden_work: you need to process their calls and bill them. Thats what a service provider generally does. |
| 15:27.55 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:28.13 | raden_work | WELL my WIFI reches everyone at 24 mb with practically no latency soo internet and phone for $30 a month 3 cents a min outbound |
| 15:28.23 | jaytee | harryv, don't see anything wrong with the config files as far as timing and signalling |
| 15:28.26 | mort_gib | [TK]D-Fender: And slag them off when they have problems |
| 15:28.44 | stmaher | [TK]D-Fender is there a way to add exten => _877XXXX,1,SIPAddHeader(Alert-Info: Auto Answer) to an outbound call file? |
| 15:28.44 | mort_gib | [TK]D-Fender: And have "premium rate" support numbers |
| 15:29.01 | harryv | jaytee: I don't get why it says Error on span 0 - what is span 0 ? |
| 15:29.03 | jaytee | harryv, what "other box in here" ? |
| 15:29.11 | [TK]D-Fender | stmaher: You do not add DIALPLAN to a call file. Go reconsider what kind of CHANNEL you are calling. |
| 15:29.19 | harryv | jaytee: some legacy VoiceGuide thing. it's not even connected at this point. |
| 15:29.37 | stmaher | [TK]D-Fender im considering joining a phone to a conference.. |
| 15:29.45 | raden_work | [TK]D-Fender, you think what im offering seems unreasonable ? |
| 15:29.52 | [TK]D-Fender | mort_gib: Yes... the Ferrengi Rules Of Acquisition is required reading in business school... |
| 15:29.58 | stmaher | [TK]D-Fender but need to add that header in order for it to automatically pick up |
| 15:30.27 | mort_gib | [TK]D-Fender: LOL yes! |
| 15:30.27 | jaytee | harryv, don't know why it would report span 0 if you've got your spans correctly defined as 1 and 2 in zaptel.conf which you do and the signalling is correct as near as I can tell from looking at the zapata.conf |
| 15:30.39 | [TK]D-Fender | raden_work: And what are you using for connectivity? |
| 15:30.41 | *** join/#asterisk bartpbx (n=bartpbx@p5099e196.dip0.t-ipconnect.de) |
| 15:30.52 | bartpbx | hello |
| 15:31.04 | bartpbx | is anyone out there using dahadi with B410P Cards? |
| 15:31.11 | jaytee | harryv, what does it display on the CLI when you type pri show spans? |
| 15:31.21 | [TK]D-Fender | stmaher: As I said look at the channel type you are calling. SIP will not let you do more. |
| 15:31.26 | mort_gib | bartpbx: Yep, at least I'm trying to |
| 15:31.59 | harryv | jaytee: http://sprunge.us/UgPL |
| 15:32.01 | [TK]D-Fender | stmaher: Go read over the complete list of Asterisk Channel Types a few dozen times and see which one will let you use Dialplan apps prior to calling a device like that |
| 15:32.02 | raden_work | 768 / 6 MB DSL |
| 15:32.07 | *** join/#asterisk AndyCrawford (n=andy@dynamic-65-161-142-80.tvscable.com) |
| 15:32.08 | bartpbx | mort_gib: that does not sound very good |
| 15:32.11 | raden_work | 3 ms packet jitter 0 packet loss |
| 15:32.17 | bartpbx | what are your problems? |
| 15:32.25 | mort_gib | bartpbx: Well, (l)users |
| 15:32.36 | [TK]D-Fender | raden_work: 768... and what do you think you can survive in terms of calls & connectivity on that little link? |
| 15:32.55 | mort_gib | bartpbx: They gave me a 20 minute windows to replace an * server in a 45 user environment |
| 15:32.56 | [TK]D-Fender | raden_work: Esp as you are including internet services. |
| 15:33.07 | bartpbx | oh |
| 15:33.14 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:33.14 | raden_work | i have 2 lines one will be dedicated VOIP and one will be dedicated internet |
| 15:33.26 | jaytee | harryv, I don't know why it would report span 0 either with your configs. Strange |
| 15:33.28 | [TK]D-Fender | raden_work: Who is your provider? |
| 15:33.36 | raden_work | ATT |
| 15:33.42 | mort_gib | bartpbx: Have rescheduled, though, Install went fine as long as you use 1.6 |
| 15:33.47 | raden_work | main office like 4 blocks away fiber from there |
| 15:33.55 | harryv | I'll try and remove one of the TE121 cards. I suspect it is broken. |
| 15:34.04 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 15:34.05 | [TK]D-Fender | raden_work: Normal residential service? |
| 15:34.16 | raden_work | Business |
| 15:34.25 | raden_work | they said i can do whatever i want with the line |
| 15:34.43 | [TK]D-Fender | raden_work: take a VERY close look at your terms of service. Its still all but certain that you will be in violation for trying to resell over it. |
| 15:34.43 | raden_work | i pay for the bandwith and the line its mine to do what i please with they said |
| 15:34.57 | raden_work | [TK]D-Fender, what u recomend ? |
| 15:35.02 | [TK]D-Fender | raden_work: and "whatever i want with the line" generally does not include "reselling" |
| 15:35.05 | VaGoNeTaS | ok guys |
| 15:35.13 | raden_work | our t1 line we have says we cant resell over it |
| 15:35.15 | VaGoNeTaS | i got cdr_adaptive_odbc.c on the machine |
| 15:35.30 | VaGoNeTaS | now i need to add it to the "make menuselect list" |
| 15:35.58 | raden_work | [TK]D-Fender, i do appreciate the this is going to go wrong scenarios thats what im looking for :) |
| 15:36.38 | [TK]D-Fender | raden_work: Not knowing the one I just mentioned is like jumping out an airlock without a space-suit on.... |
| 15:36.46 | [TK]D-Fender | "oops" |
| 15:36.56 | jasonwoot | ah raden_work... I remember when I was young and idealistic |
| 15:37.02 | jasonwoot | before the first lawsuit |
| 15:37.05 | [TK]D-Fender | :p |
| 15:37.21 | raden_work | im not exactly young nor idealistic i own 3 businesses and there is no idealistic in me no more |
| 15:37.24 | coppice | [TK]D-Fender one day that will be the cutting edge of extreme sports |
| 15:37.30 | jaytee | harrv, the cards have a switch on them if you're using multiple cards so that when the module for the card loads it will always assign the correct card to the correct spans. make sure they aren't both set the same. |
| 15:37.32 | raden_work | jasonwoot, lawsuit for what ? |
| 15:37.36 | jaytee | lunchtime, brb |
| 15:37.42 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:37.47 | [TK]D-Fender | coppice: and I know a thing or two about cutting edges :) |
| 15:38.09 | harryv | jaytee: they're.. |
| 15:38.11 | *** join/#asterisk gultig (n=dbriggs@70.96.32.114) |
| 15:38.15 | harryv | this just started happening |
| 15:38.26 | harryv | could a broken card cause this? |
| 15:39.30 | bartpbx | is there any way to debug line events on a dahadi interface? |
| 15:40.10 | mort_gib | bartpbx: What's your problem?? |
| 15:42.34 | bartpbx | i have a system with 3 B410P cards |
| 15:43.03 | bartpbx | 1 for external ( telco) and 2 for internal ISDN devices |
| 15:43.39 | mort_gib | bartpbx: Yeah.... |
| 15:44.00 | mort_gib | So NT/TE mode, and you have power on the right cards |
| 15:44.35 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:44.44 | VaGoNeTaS | [*] 1. cdr_adaptive_odbc |
| 15:44.45 | VaGoNeTaS | hehehehe |
| 15:44.46 | VaGoNeTaS | nice |
| 15:45.29 | VaGoNeTaS | i've just downloaded cdr_adaptive.odbc.c to cdr/ inside of asterisk-1.4.24 |
| 15:45.44 | VaGoNeTaS | and added the following to menuselect-tree |
| 15:45.59 | VaGoNeTaS | <category name="MENUSELECT_CDR" displayname="Call Detail Recording" remove_on_change="cdr/modules.link"> |
| 15:45.59 | VaGoNeTaS | <member name="cdr_adaptive_odbc" displayname="Adaptive ODBC CDR backend" remove_on_change="cdr/cdr_adaptive_odbc.o cdr/cdr_adaptive_odbc.so"> |
| 15:46.00 | VaGoNeTaS | <PROTECTED> |
| 15:46.00 | VaGoNeTaS | </member> |
| 15:46.07 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:46.09 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.28 | raden_work | [TK]D-Fender, how would you go about things ? |
| 15:47.29 | bartpbx | yes, TE/NT but power issn't the problem |
| 15:47.53 | bartpbx | curenty i have the problem that all console dial comands end up in a Congestion with hangupcause 34 |
| 15:48.24 | mort_gib | I had the same issue in the brief period I had it connected |
| 15:48.24 | bartpbx | that sounds to me that no ISDN device is "listinging" to the number on the port / group i dial |
| 15:48.42 | bartpbx | yeah |
| 15:48.51 | [TK]D-Fender | raden_work: Get a connection I'm allowed to resell |
| 15:48.54 | mort_gib | Well I tried outgoing only co Dial(DAHDI/g1... |
| 15:49.05 | mort_gib | No joy, like not connected |
| 15:49.23 | bartpbx | ok |
| 15:49.25 | raden_work | [TK]D-Fender, but out T1 at our office says we cant resell it |
| 15:49.29 | bartpbx | so it might be a gneral issue |
| 15:49.32 | mort_gib | But I had green light in the cards, like no alarm |
| 15:49.34 | [TK]D-Fender | raden_work: then you're screwed |
| 15:49.40 | coppice | tzafrir_laptop: www.xorcom.com seems to not exist |
| 15:49.45 | bartpbx | no alarm on these two ports |
| 15:49.54 | bartpbx | what versions are you unsing |
| 15:50.01 | mort_gib | Dunno, didn't have time to find out to be honest |
| 15:50.15 | mort_gib | Latest, Asterisk 1.6.0X |
| 15:50.36 | mort_gib | dahdi show channels shows up in cli all SHOULD be good |
| 15:50.38 | *** join/#asterisk infernix (i=gerben@unaffiliated/infernix) |
| 15:50.52 | *** join/#asterisk Slart (n=markus@212.85.89.50) |
| 15:51.07 | raden_work | [TK]D-Fender, when i talk to them they say i can have as many computers connected and do whatever i want with it |
| 15:51.15 | Slart | Hello everyone |
| 15:51.20 | infernix | i'm playing with chan_mobile but my home servers bluetooth dongle is out of reach for my mobile phone when i'm at my desk. |
| 15:51.25 | [TK]D-Fender | raden_work: and "whatever i want with the line" generally does not include "reselling" <------- |
| 15:51.40 | bartpbx | dahdi show channels gives me State "In Service" |
| 15:51.40 | [TK]D-Fender | raden_work: Please don't become a whiner over this. |
| 15:51.46 | infernix | i was thinking of installing asterisk and setting up an IAX trunk between my workstation and my server, and then doing chan_mobile on the asterisk setup on my workstation |
| 15:51.49 | [TK]D-Fender | raden_work: And come up with a VIABLE business plan. |
| 15:51.56 | tzafrir_laptop | coppice, hmm.... not here, and not at the name servers |
| 15:52.05 | *** join/#asterisk Gon (n=gon@141-15-20-190.adsl.terra.cl) |
| 15:52.10 | infernix | is this a smart thing to do or are there better ways? |
| 15:52.19 | coppice | from here the name doesn't resolve |
| 15:52.41 | mort_gib | bartpbx: That's what I found strange, straight off more than 20 minutes job though |
| 15:52.48 | tzafrir_laptop | the name servers are ns1.panelboxmanager.com and ns2.panelboxmanager.com |
| 15:52.49 | hrmphh | god damn level3 |
| 15:52.52 | hrmphh | finally fixed their shit |
| 15:52.55 | raden_work | [TK]D-Fender, trying to come up wtih a plan trying to figure out the playing field |
| 15:52.57 | hrmphh | had to reboot the FRoATM switch |
| 15:53.01 | hrmphh | impacting 68 customers |
| 15:53.13 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:54.12 | Slart | Can I use asterisk to control wireless phones (DECT)? or do you buy some kind of dect to ip switch and run it through that? |
| 15:55.02 | [TK]D-Fender | Slart: Latter |
| 15:55.17 | coppice | tzafrir_laptop: did you fix something? its working now, but it failed over the last few hours |
| 15:55.23 | Slart | [TK]D-Fender: ok, thanks |
| 15:55.36 | tzafrir_laptop | coppice, no, I didn't change anything |
| 16:00.55 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 16:04.05 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 16:04.53 | mort_gib | bartpbx: How long did you leave the lines connected for?? |
| 16:05.05 | bartpbx | some hours now |
| 16:05.33 | bartpbx | is there any way to see the raw isdn events from dahadi? |
| 16:05.40 | freh | Does anyone know how to use spandsp on asterisk 1.4.24 these days? |
| 16:05.47 | freh | I need to configure faxing |
| 16:06.03 | mort_gib | bartpbx: Don't know, are you SURE you are using the right channels ?? |
| 16:06.13 | freh | I can't seem to find the apps |
| 16:06.22 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 16:06.25 | _pepo_ | hi friends |
| 16:06.45 | jeff_phillips | Slart: yeah it would be nice if there was a DECT 6.0 base station that worked with SIP & let each handset be a seperate VoIP extension. |
| 16:06.49 | bartpbx | as i am dialing the group an see that he tries each channel i think so |
| 16:07.28 | _pepo_ | I am trying to connect two systems... but, when the first go to voicemail in the second, the second dont recognize the DTMF tones send by the first |
| 16:07.44 | freh | mort_gib: BTW, I configured asterisk 1.4.24 with mISDN and it seems to be working fine for now |
| 16:08.04 | mort_gib | bartpbx: In any case you should see incoming calls in the cli |
| 16:08.14 | mort_gib | freh: Yeah?? same card |
| 16:08.17 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 16:08.32 | freh | mort_gib: Digium B410P |
| 16:09.16 | _pepo_ | why my voicemail-asterisk dont recognize the DTMF tones? Do I need to configure something in sip.conf if the calls are comming from other system? |
| 16:09.54 | mort_gib | freh: Well, i went for 1.6 thinking I was better off with a Digium driver for a Digium card :-/ |
| 16:11.48 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 16:13.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:13.56 | mort_gib | bartpbx: What are you using ptp or PtMP |
| 16:14.06 | bartpbx | ptmp |
| 16:14.52 | mort_gib | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg220646.html |
| 16:15.33 | *** part/#asterisk bbryant (n=bbryant8@68.208.65.34) |
| 16:16.21 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 16:18.48 | Slart | jeff_phillips: indeed |
| 16:19.39 | *** join/#asterisk zaafouri (n=zaafouri@196.203.51.238) |
| 16:25.23 | freh | mort_gib: I was thinking the same, but it seems 1.6 is not good for production use yet |
| 16:27.10 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 16:31.07 | tzafrir_laptop | bartpbx, what do you mean by "raw ISDN events"? |
| 16:33.46 | bartpbx | rzafrir: the protocol messages |
| 16:33.59 | bartpbx | like pri debug |
| 16:34.07 | bartpbx | setup messages and so on |
| 16:35.12 | bartpbx | oh |
| 16:40.44 | rue_mohr | ~pb |
| 16:40.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:41.10 | rue_mohr | ^^ please fix the space immediatly after rafb |
| 16:42.16 | rue_mohr | http://paste.debian.net/31391/ << to confirm, the call will go to the first available line of the 3 |
| 16:42.48 | [TK]D-Fender | rafb sucks anyways |
| 16:43.46 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:43.52 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 16:44.07 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 16:44.17 | rue_mohr | well pastebin.com is always clogged |
| 16:45.00 | [TK]D-Fender | rue_mohr: Loads a hell of a lot faster for me than all theothers |
| 16:45.07 | rue_mohr | k |
| 16:45.18 | rue_mohr | will you please confirm that paste? |
| 16:45.40 | rue_mohr | or someone |
| 16:46.00 | [TK]D-Fender | rue_mohr: It'll do one after the other until one takes it |
| 16:46.44 | rue_mohr | k, then that dial will take the call to termination and the dials after wont be reached |
| 16:46.47 | rue_mohr | right? |
| 16:47.41 | [TK]D-Fender | rue_mohr: yes... |
| 16:47.56 | rue_mohr | ok, so, lets say their all busy, I should put a dotones(busy) and hangup after it, right? |
| 16:48.10 | [TK]D-Fender | rue_mohr: what do YOU want to do?this is basic Dial crap that has never changed.... and you've been using * how long |
| 16:48.23 | rue_mohr | cause otherwise it would repeat the extension programming |
| 16:48.37 | rue_mohr | no we havn't flipped the switch on this system yet |
| 16:48.42 | [TK]D-Fender | rue_mohr: Repeat? Pardon? |
| 16:48.54 | rue_mohr | I havn't been allowed, I had to come up with a speed dial systemt hat I just finished |
| 16:49.07 | rue_mohr | using *X |
| 16:49.15 | rue_mohr | where X is 0-9 |
| 16:49.33 | rue_mohr | I'm just checking things before we enable the system |
| 16:49.35 | harryv | how long for a oversized mail to asterisk-users to be approved? |
| 16:50.47 | rue_mohr | which? |
| 16:58.55 | rue_mohr | Congestion(), how convienient |
| 17:00.35 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-140-02.telkomadsl.co.za) |
| 17:01.54 | *** join/#asterisk cp5 (n=samy@72.37.252.206) |
| 17:02.51 | *** join/#asterisk mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 17:02.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:03.43 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:04.32 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 17:04.37 | carrar | w00t! |
| 17:06.55 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.rubicom.hu) |
| 17:08.32 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-552-1-76-145.w92-144.abo.wanadoo.fr) |
| 17:08.36 | *** join/#asterisk Sir-Gon (n=gon@141-15-20-190.adsl.terra.cl) |
| 17:09.04 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-552-1-76-145.w92-144.abo.wanadoo.fr) |
| 17:10.19 | *** join/#asterisk PDani (n=pekdanie@89.133.156.227) |
| 17:11.04 | infernix | is ztdummy really needed when using IAX |
| 17:11.04 | *** join/#asterisk docidu (n=eris@whthyt253-26.northwestel.net) |
| 17:11.41 | infernix | i mean the box has HPET but asterisk (1.4) still complains when using IAX that it needs zaptel timings |
| 17:13.06 | [TK]D-Fender | infernix: It is if you want IAX2 trunk mode |
| 17:13.16 | *** join/#asterisk bbryant (n=bbryant8@68.208.65.34) |
| 17:15.09 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 17:15.26 | j_kroon | infernix, i hear your argument and i fully agree. |
| 17:15.48 | j_kroon | however, just use dahdi_dummy with hpet as your backing timer for it and live with it. |
| 17:17.00 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 17:17.12 | watchy | is it hard to get oslec working with dahdi? |
| 17:17.17 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 17:18.00 | *** join/#asterisk Pegasus_RPG (n=chatzill@cpe-071-076-024-036.sc.res.rr.com) |
| 17:19.18 | j_kroon | watchy, no, just compile the modules, and load it, and set echocanceller=osclec,1-?? in /etc/dahdi/system.conf |
| 17:19.19 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:19.23 | Qwell | watchy: no, I think tzafrir_laptop made it pretty easy |
| 17:19.38 | watchy | ah ok |
| 17:20.22 | Pegasus_RPG | Is it possible to get a decent call on a 6kbps (yes 6000 baud) upstream speed satellite connection using Speex on a client on an asterisk system? (128kbps downstream isn't a problem) |
| 17:21.01 | watchy | man youd have bad latency on any sat connect |
| 17:21.03 | *** part/#asterisk HuntsMan (n=hunts@88-69-246-201.adsl.terra.cl) |
| 17:21.27 | Pegasus_RPG | Oh I do |
| 17:21.28 | Pegasus_RPG | but Skype is useable |
| 17:21.31 | watchy | wow |
| 17:21.32 | Pegasus_RPG | most of the time |
| 17:22.13 | Pegasus_RPG | I just want to be able to connect to the * system for all the benefits it provides (like general outcalling) |
| 17:23.15 | Pegasus_RPG | but so far, * isn't playing nice with Speex on Wengophone/QuteCom or X-Lite |
| 17:24.17 | Pegasus_RPG | (I mean I got it to work with X-Lite on a LAN, but over the satellite link, * doesn't see the X-Lite client) |
| 17:25.03 | Pegasus_RPG | it did just for a second or so, then the call dropped and I couldn't make any more calls |
| 17:27.03 | [TK]D-Fender | Pegasus_RPG: You can't even fit UDP overhead into 6000 baud let alone payload |
| 17:27.25 | Pegasus_RPG | so how is Skype doing it? Or was speedtest.net just wrong? |
| 17:27.38 | Pegasus_RPG | (THe connection is suposed to be 32kbps upstream) |
| 17:28.13 | [TK]D-Fender | Pegasus_RPG: How do you go from 600 baud to 32kbps all of a sudden? |
| 17:28.27 | [TK]D-Fender | Pegasus_RPG: Perhaps you'd like to start your description over in some consistent manner |
| 17:28.29 | Qwell | [TK]D-Fender: I suspect a random multiple of 8 was thrown in somewhere |
| 17:28.46 | [TK]D-Fender | 42 L- |
| 17:28.47 | harryv | the provider says he has 32k, in reality, not. |
| 17:29.40 | harryv | .. |
| 17:29.53 | Pegasus_RPG | OK, sorry. The provider says it's 128kbps down, 32k up. Speedtest.net shows 128k down, 6k up. |
| 17:30.16 | Pegasus_RPG | Skype works most of the time using it's SVOPP protocol |
| 17:30.43 | [TK]D-Fender | Pegasus_RPG: Nothing I know can survive that with overhead |
| 17:30.46 | Qwell | Pegasus_RPG: speedtest is clearly wrong |
| 17:31.05 | Pegasus_RPG | Qwell: that's encouraging :) |
| 17:31.53 | Pegasus_RPG | To further wrinkle things, the client is connecting via a VPN to the * server, then using the softphone over that |
| 17:32.17 | Qwell | umm...yeah |
| 17:33.02 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 17:33.11 | Pegasus_RPG | I imagine the VPN overhead is hurting quite a bit |
| 17:33.24 | *** join/#asterisk cesar_CR (n=cesar@201.201.176.2) |
| 17:33.35 | Pegasus_RPG | But when I set it up, I was under the impression that trying to expose SIP directly to the internet is a bad thing |
| 17:34.32 | [TK]D-Fender | Pegasus_RPG: ... |
| 17:34.34 | [TK]D-Fender | ~wglwat |
| 17:34.38 | jbot | i guess wglwat is well, good luck with all that |
| 17:34.44 | *** join/#asterisk lftsy (n=lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 17:35.55 | Pegasus_RPG | yeah, I'm starting to lose hope |
| 17:36.04 | Pegasus_RPG | thanks for your time and attention though. |
| 17:36.19 | Pegasus_RPG | mutters...stupid 3rd world countries |
| 17:37.11 | *** part/#asterisk Pegasus_RPG (n=chatzill@cpe-071-076-024-036.sc.res.rr.com) |
| 17:37.12 | tzafrir_laptop | watchy, managed to install oslec? |
| 17:37.32 | tzafrir_laptop | aparantly hope lost Pegasus_RPG |
| 17:37.45 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 17:41.43 | lftsy | Hello all, I have a small question not answered, is there someone how could help me please.. |
| 17:41.56 | lftsy | Do you think that is it easily possible to modify the C Dial function adding a parameter in order to change the number to be reach without modifying the Peer IP and Reg. Contact IP |
| 17:42.27 | lftsy | like a Dial(SIP/1003/1005,"??") that will use 1003 IP and Reg Contact rewriting only the number in the URI to reach without altering the IP |
| 17:43.01 | lftsy | I'll be very grateful if you has any clue how to do it.. |
| 17:43.54 | carrar | I can has the book |
| 17:43.58 | carrar | ~book |
| 17:43.59 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:44.58 | lftsy | So you think it's possible? |
| 17:45.29 | carrar | I can't understand your english |
| 17:45.56 | carrar | I mean what you are asking |
| 17:45.56 | *** join/#asterisk lftsy (n=lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 17:46.29 | *** part/#asterisk bartpbx (n=bartpbx@p5099e196.dip0.t-ipconnect.de) |
| 17:47.52 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 17:48.37 | *** join/#asterisk Melifaro (n=melifaro@melifaro.astel.ru) |
| 17:48.59 | Kobaz | i'm having problems with call parking: http://pastebin.com/m5ab9cb21 (i want a number that i can use the ami to do a redirect to, that will park a specific channel... so as a test i have 600X being parkng lots) |
| 17:51.48 | harryv | just had a technician by. he sad I had to set stroke/timing (rough translation) to 2048/incoming, and that atm mine was about 20hz below. tried skimming the docs. what would that be? |
| 17:52.17 | Qwell | harryv: context? |
| 17:53.01 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) |
| 17:53.09 | harryv | 2sec |
| 17:53.33 | zoid_99 | I need a Echo application that has a deleay time setting |
| 17:53.42 | zoid_99 | anybody know of one? |
| 17:53.47 | zoid_99 | like a buffered echo |
| 17:54.00 | Qwell | zoid_99: interesting... |
| 17:54.32 | harryv | Qwell: I have an ISDN-30 connection. It is dropping every 3-10 minutes (console output: http://sprunge.us/NFQN , PRI debug: http://sprunge.us/cgQI |
| 17:54.34 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 17:54.52 | harryv | the connection is fine, but he said I was out of sync with the central |
| 17:55.09 | harryv | ( configurations: http://sprunge.us/idCe |
| 17:55.14 | zoid_99 | Qwell: Yeah, sounds simple enough and I thought someone would have already built it :) |
| 17:55.19 | Qwell | zoid_99: ideally, one would modify the existing Echo application to add a delay param... I don't suspect it would be difficult |
| 17:56.03 | Qwell | zoid_99: I can't think of any way to do it without modifying existing stuff |
| 17:56.24 | zoid_99 | Qwell: I looked at echo and it simply reads/writes... |
| 17:56.58 | zoid_99 | Qwell: but it shouldn't be too difficult to add the param and the code |
| 17:58.26 | harryv | Qwell: I looked at the span timing param, doesn't seem to be that |
| 17:58.55 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 18:00.49 | [TK]D-Fender | Kobaz: Warning: Return Context Invalid, call will return to default|s <--------- |
| 18:01.01 | Kobaz | [TK]D-Fender: yes, i think i found a bug |
| 18:01.10 | *** join/#asterisk Sir-Gon (n=gon@141-15-20-190.adsl.terra.cl) |
| 18:01.32 | Kobaz | [TK]D-Fender: because the return context is correct, it's saying it's going to fall back to default, but it doesn't |
| 18:01.41 | Kobaz | [TK]D-Fender: so i think there's two bugs |
| 18:02.40 | [TK]D-Fender | Kobaz: I think you're not showing me everything so I naturally don't trust your setup is at all sane |
| 18:02.58 | Kobaz | [TK]D-Fender: that is everything |
| 18:03.02 | Kobaz | [TK]D-Fender: Return Context: (_cos_basic,6001,0) ID: 5506 |
| 18:03.08 | Kobaz | [TK]D-Fender: that's not the return context i gave it |
| 18:03.12 | [TK]D-Fender | Kobaz: I don't see your DIALPLAN |
| 18:03.21 | Kobaz | first of all, it's ignoring the return context i'm giving it |
| 18:03.48 | Kobaz | http://pastebin.com/m415d3780 |
| 18:03.51 | Kobaz | theres the big of dialplan |
| 18:03.52 | Kobaz | bit |
| 18:04.37 | Kobaz | ParkAndAnnounce(,10,,phonegroup_internal_override); |
| 18:04.38 | Kobaz | -- Return Context: (_cos_basic,6001,0) ID: 5506 |
| 18:04.48 | Kobaz | why is it ignoring the context i give it? |
| 18:05.16 | Kobaz | the last argument to ParkAndAnnounce is return_context |
| 18:07.58 | *** join/#asterisk Habile (n=chatzill@78.32.178.49) |
| 18:11.45 | [TK]D-Fender | Kobaz: core show application parkandannounce |
| 18:12.42 | Kobaz | ParkAndAnnounce(announce:template|timeout|dial|[return_context]): |
| 18:13.58 | *** join/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 18:16.11 | Ether_Man | If using Asterix as a server for just IAX and SIP softphones. Can the server be inside a NAT (ofc with ports forwarded to it). And will outside clients still be able to call the clients that are within that and vice versa? |
| 18:16.22 | Qwell | Ether_Man: What's Asterix? |
| 18:16.30 | Ether_Man | Asterisk* |
| 18:16.48 | Habile | Asterix was a ghoul I believe |
| 18:16.57 | [TK]D-Fender | Ether_Man: read up : |
| 18:16.59 | [TK]D-Fender | ~sipnat |
| 18:16.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:17.01 | [TK]D-Fender | ^^^^^^^^^ |
| 18:17.03 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 18:17.26 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 18:17.46 | [TK]D-Fender | Kobaz: The goto-style label to jump the call back into after timeout. Default <priority+1>. |
| 18:17.47 | iratik | I have a retarded question: And i've already googled "faul tolerance extensions.conf" .. I just need a call to go out on another trunk if the first trunk is full (outgoinglimit=5) ... hint..or point me to the right docs? |
| 18:17.49 | Ether_Man | [TK]D-Fender, that only explains about one or the other.. clients inside, OR outside. Not both :/ |
| 18:18.01 | zoid_99 | a gaul not a ghoul :) |
| 18:18.03 | [TK]D-Fender | Kobaz: this implies that its looking for a label, not a context. Looks like tis poorly worded |
| 18:18.22 | [TK]D-Fender | Ether_Man: Yes, its explains when BOTH are behind their own NATs |
| 18:18.39 | [TK]D-Fender | Ether_Man: And inside ones don't matter |
| 18:19.29 | Habile | ah - close - it was in my youth... |
| 18:20.32 | Kobaz | [TK]D-Fender: aah, okay |
| 18:20.36 | Kobaz | [TK]D-Fender: lemme try that |
| 18:21.17 | [TK]D-Fender | iratik: Just dial the 2nd. |
| 18:21.56 | *** part/#asterisk harryv (n=harry@67.207.147.205) |
| 18:21.58 | iratik | If the call succeeds on the first... won't the dialplan cause the call to be dialed again on the second? |
| 18:22.21 | [TK]D-Fender | iratik: No |
| 18:23.27 | iratik | So ... exten => _1NXXNXXXXXX,1,Dial(SIP/frthree/${EXTEN}); exten => _1NXXNXXXXXX,2,Dial(SIP/frfour/${EXTEN}); exten => _1NXXNXXXXXX,3,Hangup; ? |
| 18:24.17 | iratik | and that will work with [frthree]; outgoinglimit=5; so that on thet 6th call it will go out on four? .. testing |
| 18:24.20 | iratik | thanks btw |
| 18:25.38 | [TK]D-Fender | iratik: Yes |
| 18:27.36 | *** join/#asterisk AndyCrawford (n=andy@dynamic-65-161-142-80.tvscable.com) |
| 18:29.11 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 18:31.09 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:31.57 | *** join/#asterisk lidocaine (i=hose@larvae.bluemaggottowel.com) |
| 18:34.13 | *** join/#asterisk smooth_penguin (n=smoove_@59.95.2.144) |
| 18:34.38 | smooth_penguin | hey does asterisk depend on the soundcard of the machine its running on? |
| 18:34.46 | [TK]D-Fender | smooth_penguin: no |
| 18:34.53 | smooth_penguin | ok |
| 18:35.05 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 18:35.07 | smooth_penguin | so it can run just fine if there is no audio card? |
| 18:35.14 | smooth_penguin | on that machine? |
| 18:35.15 | [TK]D-Fender | smooth_penguin: Sure |
| 18:35.27 | smooth_penguin | ok thanks |
| 18:36.04 | lidocaine | so i'm running 1.4.24 copiled from source, and I've run across this bug http://bugs.digium.com/view.php?id=13222 which looks to be resolved in a revision way prior to the one 1.4.24 is based on. |
| 18:36.08 | lidocaine | should i reopen? |
| 18:36.14 | lidocaine | exact same behavior |
| 18:37.01 | lidocaine | or should i first checkout the latest 1.4.24 revision and attempt to reproduce with that before re-opening? |
| 18:37.08 | lidocaine | er 1.4 revision |
| 18:38.02 | Kobaz | [TK]D-Fender: setting return_context as a jump label doesn't work either |
| 18:38.22 | Kobaz | [TK]D-Fender: it's the exact same flow as the previous pastebin, and it tries to go to the wrong context |
| 18:39.48 | [TK]D-Fender | Kobaz: -- Return Context: (_cos_basic,6001,0) ID: 5506 <- clearly it wants to return to the same exten & context as the one calling it |
| 18:40.11 | [TK]D-Fender | Kobaz: and forget "context" as a parameter, it appears misworded. |
| 18:42.25 | Kobaz | k |
| 18:42.57 | Kobaz | i may have to monkey with the module to do what i want to do |
| 18:45.21 | [TK]D-Fender | Kobaz: Or just fix your dialplan |
| 18:46.44 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 18:47.30 | Kobaz | i don't think there is a way to do what i need to do, without either implementing my own parking, or modifying the module |
| 18:47.56 | *** join/#asterisk DelphiWorld (n=Miranda@41.221.19.173) |
| 18:48.03 | DelphiWorld | hello my friends |
| 18:48.17 | DelphiWorld | please any semple IVR application in asterisk ? |
| 18:49.36 | Kobaz | here's the scenerio: phone A is talking to phone B, phone A has a pc application that has a button to place the current call on park.... the park program does a redirect to a macro in the dialplan to place the call on park using parkandannounce.... since it is a redirect and not a transfer, parkandannounce now has no information about who parked the call... so the return exten is going to be phone B, ad the context that it came from... which is now what i wa |
| 18:50.12 | Kobaz | s/now what/not what/ |
| 18:50.22 | *** join/#asterisk Great_Anta_Baka (i=c4219f53@gateway/web/ajax/mibbit.com/x-4ccc904bdfb026a5) |
| 18:51.05 | Kobaz | i dont see any way to get the parked caller to timeout back to who put them on park, with this "out of band" method of parking |
| 18:51.05 | [TK]D-Fender | Kobaz: In your macro we can see where you want to send the call to, ans we can see where it si going. Where it goes, just do a friigen GOTO YOURSELF |
| 18:51.38 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:51.47 | Kobaz | arkAndAnnounce(,10,,phonegroup_internal_override); |
| 18:51.51 | Kobaz | that's what the macro does |
| 18:51.53 | [TK]D-Fender | DelphiWorld: what "application"? You do IVR's in your dialplan. |
| 18:52.33 | [TK]D-Fender | Kobaz: And it returns on the exten so make a priority label for it to return to and just GOTO |
| 18:52.39 | Kobaz | [TK]D-Fender: how do i control what exten ParkAndAnnounce will timeout back to? |
| 18:53.01 | [TK]D-Fender | Kobaz: it falls back to ITSELF apparently just just put the Goto in that exten |
| 18:53.07 | DelphiWorld | [TK]D-Fender: any semple IVR included ? |
| 18:53.34 | Kobaz | hmm |
| 18:53.58 | [TK]D-Fender | DelphiWorld: lookup "IVR tips" on the WIKi, and go read the book for dialplan basics |
| 18:54.00 | [TK]D-Fender | ~wikis |
| 18:54.01 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:54.02 | [TK]D-Fender | ~book |
| 18:54.03 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:55.51 | Kobaz | [TK]D-Fender: i don't see how that would work... the B caller can be from anywhere. ie: outside caller over a sip peer, they aren't going to have an entry in the context they came from |
| 18:56.03 | *** join/#asterisk Great_Anta_baka (n=tensai@dsl-245-171-245.telkomadsl.co.za) |
| 18:57.01 | Kobaz | ie: it will fall back to 18005558355@source_context |
| 18:57.07 | [TK]D-Fender | Kobaz: You know what... you need to seriously look at the CHANNEL that is originating this action and look at the variables * offers you so you can choose where to jump back to. |
| 18:57.24 | Kobaz | yeah |
| 18:57.27 | Kobaz | that's the problem |
| 18:57.46 | Kobaz | you can't chose where to jump back to, it will only jump back to source_exten@source_context, it seems |
| 18:57.49 | Kobaz | unless i missed something |
| 18:58.06 | [TK]D-Fender | Kobaz: DO YOUR OWN &^$#ING GOTO |
| 18:58.13 | [TK]D-Fender | Kobaz: what are you not getting? |
| 18:58.17 | Kobaz | heh |
| 18:58.33 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.24.187) |
| 18:58.37 | Kobaz | i can't preempt the implied goto that ParkAndAnnounce does by itself |
| 18:58.54 | [TK]D-Fender | Kobaz: No, but you can put YOURS where IT lands <- |
| 18:59.05 | [TK]D-Fender | Kobaz: GEEZ! |
| 18:59.07 | WHYS | Does Digium offer setup support. I have a 500+ phone installation to tackle, and while I can do a lot of the work I would like some help when I get stuck setting up a federation. Platinum support has 15 "incidents" per year, but I'm not sure what that covers. |
| 18:59.08 | Kobaz | yeah, but hmm |
| 18:59.19 | [TK]D-Fender | ~clubat Kobaz |
| 18:59.31 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.rubicom.hu) |
| 18:59.36 | Kobaz | okay here's the thing |
| 18:59.37 | [TK]D-Fender | WHYS: ... "Federation"? |
| 18:59.44 | WHYS | cuslter |
| 18:59.46 | Kobaz | i know i can do a goto from there |
| 18:59.55 | Kobaz | it's a matter of possible information loss |
| 19:00.04 | Kobaz | but i dunno, i'll have to think about this |
| 19:00.25 | *** join/#asterisk AndyML (n=alauppe@pool-96-245-116-32.phlapa.fios.verizon.net) |
| 19:00.28 | [TK]D-Fender | Kobaz: Don't go all hypothetical on us. When you have an actual problem, come and show us :) |
| 19:00.37 | Kobaz | i do have an actual problem |
| 19:00.48 | Kobaz | it's an actual problem with something that's not yet implemented |
| 19:00.51 | [TK]D-Fender | Kobaz: For that... #drphil |
| 19:00.58 | Kobaz | heh |
| 19:01.05 | AndyML | [TK]D-Fender: is there any reason to believe a dual-channel T1 interface can't be configured to use ni1 on one span, and dms100 on the other? |
| 19:01.08 | WHYS | .... ah, thats Cluster |
| 19:01.23 | Kobaz | andyml: no why would that be a problem |
| 19:01.25 | [TK]D-Fender | AndyML: none |
| 19:01.30 | AndyML | [TK]D-Fender: thanks |
| 19:02.11 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 19:06.17 | *** join/#asterisk path_ (n=path_@240-117-21-190.adsl.terra.cl) |
| 19:10.47 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 19:15.30 | watchy | man i just pluged in my dell blade |
| 19:15.34 | watchy | its so loud |
| 19:15.44 | watchy | i can hear it throughout our entire office |
| 19:16.37 | watchy | i bought it for my home but its way to loud |
| 19:16.53 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 19:25.05 | infernix | ok so I got chan_mobile working, but is it only usable for outgoing calls? |
| 19:25.16 | Qwell | infernix: nope |
| 19:25.24 | infernix | I'd like to route incoming calls into asterisk but nothing much is said about that in the docs |
| 19:25.33 | infernix | and if i call my phone, asterisk console doesn't show much either |
| 19:25.52 | infernix | funny enough it seems that asterisk answers on the first ring |
| 19:25.54 | lidocaine | why would you buy a rackmount server for your house? |
| 19:26.08 | infernix | but i've no clue where that call is being routed to, nor why it's answering |
| 19:26.22 | [TK]D-Fender | watchy: When I got my IBM x346 I told head-office that it was very quiet.... it their idea of "quiet" was a 747 on take-off at a distance of about 10' |
| 19:27.58 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:29.26 | *** join/#asterisk gsiener (n=gsiener@206.48.2.97) |
| 19:29.29 | AndyML | [TK]D-Fender: http://pastie.org/425703 - pri debug on span 2 (nortel side). Any ideas where the disconnect might be? |
| 19:29.30 | jeff_phillips | lidocaine: You don't have a server rack at your house??? |
| 19:29.59 | gsiener | Hi all. I'm looking for an ITSP that can provide incoming DIDs via IAX. Any recommendations? |
| 19:30.17 | AndyML | configs - http://pastie.org/425704 |
| 19:30.38 | [TK]D-Fender | AndyML: == Auto fallthrough, channel 'DAHDI/47-1' status is 'UNKNOWN' <-- sure... you ran out of dialplan |
| 19:31.17 | lidocaine | heh, no. i did back in the day, but then i got wise and just colocated everything but a small storage server and another dead quiet server in the basement. |
| 19:32.26 | infernix | Qwell: ah, my bad, extensions.conf issue |
| 19:32.36 | infernix | lots of sco_write() not ready, though :| |
| 19:34.59 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 19:36.15 | AndyML | [TK]D-Fender: it matched, failed, then ran out of dialplan... - here is the explaination of the dialplan-fail - Ext: 1 Cause: Incoming call barred (54), class = Service or Option not Available (3) ] |
| 19:36.23 | AndyML | what does that mean? |
| 19:36.40 | *** join/#asterisk Quintana (n=sylvain@aghnar.doowan.net) |
| 19:37.09 | [TK]D-Fender | AndyML: You had progress but didn't answer or do anything. You appear to have a DIALPLAN ERROR |
| 19:37.48 | AndyML | ugh, tnx |
| 19:39.50 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 19:39.58 | AndyML | [TK]D-Fender: would you say I need to Answer the call on span 1 before I dial span 2 with it? |
| 19:40.57 | [TK]D-Fender | AndyML: ..... I don't see you doing ANYTHING with it |
| 19:41.11 | [TK]D-Fender | AndyML: you NoOp and thats all. You don't effectively do anything |
| 19:41.22 | AndyML | exten => _XXXX,n,Dial(dahdi/g2/${EXTEN}) |
| 19:41.34 | AndyML | that is right after the NoOp() - maybe it didn't make it into the paste |
| 19:41.44 | [TK]D-Fender | AndyML: none of your dialplan did |
| 19:42.01 | [TK]D-Fender | AndyML: and that pattern is no good |
| 19:42.45 | [TK]D-Fender | AndyML: - Executing [16108882763@pri_descend:1] NoOp("DAHDI/47-1", "ld") in new stack <-- does this look like a 4 digit number to you? |
| 19:43.01 | AndyML | http://pastie.org/425713 |
| 19:43.38 | [TK]D-Fender | AndyLook @ 23... then look @ 24 |
| 19:43.46 | [TK]D-Fender | 22/23 rather |
| 19:43.53 | [TK]D-Fender | "." <--------- |
| 19:44.04 | AndyML | reload - i fixed that\ |
| 19:44.17 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 19:44.27 | [TK]D-Fender | AndyML: All I'm seeing are errors... show me something current and real |
| 19:44.46 | AndyML | no problem |
| 19:44.54 | [TK]D-Fender | AndyML: Don't tell me i can't trust what I do see, because i certainly don't trust what I can't |
| 19:45.57 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 19:46.21 | *** join/#asterisk PDani_ (n=pekdanie@89.133.156.227) |
| 19:46.56 | infernix | i give up on chan_mobile. android doesn't support rfcomm anyway >.< |
| 19:47.03 | AndyML | [TK]D-Fender: ok, this is an inbound call - applies to [pri_ascend] - http://pastie.org/425716 |
| 19:47.11 | *** join/#asterisk cesar_CR (n=cesar@201.201.176.2) |
| 19:49.14 | Qwell | infernix: if it doesn't support rfcomm, how did you get it connected? |
| 19:49.15 | [TK]D-Fender | AndyML: Ok, that is refused... setup your prilocaldialplan , etc to "unknown |
| 19:49.20 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 19:50.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:50.15 | infernix | Qwell: the one with the working connection (but broken sco audio) was my WM6.1 (htc kaiser) |
| 19:51.48 | Qwell | I'd be shocked if it didn't support HFP |
| 19:52.34 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:02.07 | Qwell | infernix: -- Bluetooth Device G1 initialised and ready. |
| 20:03.01 | infernix | Qwell: huh?? |
| 20:03.08 | infernix | Qwell: what version of bluez? |
| 20:03.33 | *** join/#asterisk Great_Anta_baka (n=tensai@dsl-245-171-245.telkomadsl.co.za) |
| 20:03.51 | Qwell | 3.36? |
| 20:04.53 | Qwell | -- Launching echo() on Mobile/G1-8e29 |
| 20:05.21 | infernix | !! |
| 20:05.43 | Qwell | "Dialing 800-466-4411" |
| 20:05.57 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:05.57 | Qwell | "Call in progress 00:10" |
| 20:06.17 | infernix | what kernel version? |
| 20:06.39 | Qwell | 2.6.26 |
| 20:06.46 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 20:08.18 | infernix | you're on debian lenny? |
| 20:08.46 | infernix | maybe, just maybe, its due to me using bnep0 as a tethering interface on my rooted G1 |
| 20:08.56 | infernix | im rather confused though |
| 20:09.08 | Qwell | quite likely :p |
| 20:09.18 | infernix | [Mar 24 19:12:26] DEBUG[12308] chan_mobile.c: connect() failed (112). |
| 20:09.42 | infernix | thats all i get when it attempts to set up rfcomm with my G1 |
| 20:10.31 | Qwell | untether it |
| 20:10.57 | Qwell | errno 12 is Address already in use |
| 20:10.59 | Qwell | 112* |
| 20:12.20 | infernix | well i think i'll have to switch back to stock kernel |
| 20:12.29 | infernix | i compiled bnep in |
| 20:14.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:16.52 | infernix | Qwell: well no thats not it, i had that disabled already |
| 20:17.03 | infernix | Qwell: are you paired with another bt headset on your G1? |
| 20:17.13 | infernix | im paired with my carkit, although its not connected |
| 20:17.52 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 20:20.52 | Qwell | infernix: I cleared all pairings first |
| 20:23.24 | infernix | Qwell: you got a popup on your G1 to enter pin? |
| 20:23.35 | Qwell | yes |
| 20:25.04 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 20:25.55 | infernix | oh man |
| 20:25.57 | infernix | sucks |
| 20:26.01 | infernix | typo in HW addr |
| 20:26.14 | Qwell | >_> |
| 20:27.14 | infernix | <PROTECTED> |
| 20:27.40 | infernix | enters PIN |
| 20:28.22 | infernix | well, that fails with errno 111 |
| 20:28.33 | Qwell | 115 = Network is down, 111 = Connection refused |
| 20:28.35 | infernix | Qwell: where did you look up those errnos? |
| 20:28.42 | Qwell | http://www.toshima.ne.jp/~maoyam/show_errno_message/cygwin-gcc-errno.txt |
| 20:28.53 | infernix | ah, cool |
| 20:28.56 | Qwell | 111 sounds like a bad pin |
| 20:29.14 | Qwell | When I entered the pin on the phone, I got a popup on my desktop and entered the same thing there |
| 20:29.54 | infernix | fires up bluetooth-applet |
| 20:32.31 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 20:33.08 | ZX81 | hi all, same question as last week - what circumstances could cause an IAX2 peer to show as OK at one end and unknown at the other end? |
| 20:33.27 | ZX81 | even though it shows IP etc |
| 20:34.14 | infernix | got two pairing attempts, now stuck with 115 |
| 20:34.17 | infernix | reboots the lot |
| 20:37.36 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-32aef24bcf429637) |
| 20:42.32 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:43.57 | infernix | argh |
| 20:48.50 | jblack | ZX81: turn on qualify on both sides |
| 20:51.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:54.30 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:55.35 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 20:55.45 | _BBV_ | 1 |
| 20:55.47 | *** part/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 20:57.40 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:00.41 | *** join/#asterisk bartpbx (n=bartpbx@62.216.165.71) |
| 21:01.06 | bartpbx | hello, in my spare time I'm working on implementign t.38 in our network |
| 21:01.27 | infernix | Qwell: you're using default bluetooth config files (hcid.conf etc)? |
| 21:01.34 | bartpbx | and i have some questinon on the t.38 impl in 1.4 |
| 21:02.04 | bartpbx | is it possible to have the rtp stream flow through the asterisk? |
| 21:02.57 | Qwell | infernix: looks like it |
| 21:03.07 | bartpbx | currently any t.38 ends in an end to end reinvite. But our ss7 gatway is only availible throug from our asterisk server |
| 21:03.11 | goodjoke | I rebuilt my system a couple days ago... since then, all calls to certain phones get dropped as soon as they are answered. These phones can make outgoing calls, but all incoming calls (even extension to extension) calls get dropped when answered. I have replaced one of the extensions with a new phone on the same network drop and things are fine. |
| 21:03.13 | bartpbx | not from the client ip |
| 21:03.15 | mnicholson | infernix, what version of bluez are you using? |
| 21:03.45 | goodjoke | here is a log of the error when i try a call that fails.. http://mibbit.com/pb/MWPmSt |
| 21:04.40 | infernix | mnicholson: 3.36, in debian sid |
| 21:05.23 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:05.26 | infernix | already tried switchign hcid.conf to auto and setting predefined PIN |
| 21:05.47 | infernix | but whether i enter correct or incorrect pin, keep getting errno 111 |
| 21:07.43 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 21:08.20 | infernix | yay paired |
| 21:08.28 | infernix | what a nightmare |
| 21:08.41 | infernix | seems my bluetooth setup on my workstation is borked |
| 21:09.26 | infernix | works!!! |
| 21:09.32 | infernix | Qwell, thanks :D |
| 21:09.40 | ZX81 | jblack: qualify is on on both sides otherwise it would be unmonitored |
| 21:10.08 | mnicholson | infernix, yeah, pairing is tricky, im thinking about adding pairing support to chan_mobile so the user does not have to worry about it |
| 21:10.35 | Qwell | mnicholson: O.o |
| 21:12.33 | ThoMe | hello |
| 21:12.35 | ThoMe | backup*CLI> meetme list 400 |
| 21:12.35 | ThoMe | User #: 01 01786323765 01786323765 Channel: SIP/2244892e2-086c7e30 (unmonitored) 00:03:03 |
| 21:12.40 | ThoMe | why i have two numbers? |
| 21:12.46 | ThoMe | "01786323765 01786323765" ? |
| 21:12.54 | ThoMe | where i can find a field names? |
| 21:13.06 | ThoMe | 01 (userid) 01786323765 (callerid) .. ? |
| 21:13.31 | Qwell | ThoMe: callerid number and name |
| 21:13.47 | ThoMe | Qwell: ah ok. and the first always callerid? |
| 21:13.58 | Qwell | both are callerid |
| 21:14.10 | ThoMe | Qwell: i use the manager-console. i can't not in comma seperated? |
| 21:15.16 | ThoMe | or xml? |
| 21:18.26 | *** join/#asterisk Great_Anta_baka (n=tensai@196-209-178-64-wrbs-esr-2.dynamic.isadsl.co.za) |
| 21:20.57 | goodjoke | any suggestions for my call disconnect problem? http://mibbit.com/pb/MWPmSt |
| 21:21.41 | bartpbx | goodjoke: what type of devices are you unsing |
| 21:21.46 | *** join/#asterisk Great_Anta_baka (n=tensai@196.33.159.83) |
| 21:21.53 | goodjoke | polycom and linksys |
| 21:22.04 | goodjoke | the phones in question are polycom 601s |
| 21:22.22 | bartpbx | and on both you get this error in codec string? |
| 21:23.03 | goodjoke | my phone is a linksys...that always works |
| 21:23.22 | goodjoke | but if i call from any phone (polycom or my linksys) i get that codec error |
| 21:23.37 | goodjoke | and the call gets disocnnected as soon as the person picks up |
| 21:23.55 | goodjoke | i've replaced the poly 601 with a polycom speaker phone and it works fine |
| 21:24.29 | goodjoke | so that would seem to me that there is something with the phone...but i did firmware upgrades and still have the issues |
| 21:24.35 | mnicholson | hmm, what does asterisk not like about the codec string? |
| 21:25.08 | mnicholson | goodjoke, what version of asterisk are you using? |
| 21:25.09 | goodjoke | i can call other polycom phones that work fine and i am pretty sure that I do not get that codec error |
| 21:28.35 | goodjoke | mnicholson: not gonna lie.. it is trixbox 2.6.2.2.. but i am in the cli |
| 21:28.55 | goodjoke | so... basically means that I am not sure what build of asterisk it is based on |
| 21:28.59 | mnicholson | goodjoke, hmm |
| 21:32.22 | *** join/#asterisk IOU (n=ryan@121.73.80.73) |
| 21:33.05 | *** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com) |
| 21:34.07 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:35.07 | *** join/#asterisk PDani (n=pekdanie@catv-89-133-156-227.catv.broadband.hu) |
| 21:35.39 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 21:36.03 | mnicholson | goodjoke, "Error in codec string 'eo 0 sip 34 99'" asterisk is not expecting the eo there |
| 21:36.57 | goodjoke | mnicholson: OK... i have no idea what that means |
| 21:38.01 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 21:38.43 | [TK]D-Fender | It means "WTF, hasn't this guy figured out he should have enabled full sip debug by now?" |
| 21:39.08 | goodjoke | apprently that is what it means.. :) |
| 21:40.08 | mnicholson | goodjoke, find out why the polycom has eo in the media string, which probably looks like this "m=audio 12345 RTP/AVP eo 0 sip 34 99". That will fix your issue. |
| 21:40.27 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 21:44.46 | mnicholson | goodjoke, use sip debug to capture that packet and pastebin it |
| 21:45.21 | goodjoke | sip set debug, command not found |
| 21:45.40 | mnicholson | just do 'sip debug' |
| 21:46.18 | goodjoke | same |
| 21:46.20 | *** join/#asterisk Great_Anta_baka (n=tensai@dsl-245-151-145.telkomadsl.co.za) |
| 21:46.35 | mnicholson | hmm, 'sip<tab>' |
| 21:46.38 | mnicholson | what does that give you |
| 21:47.07 | mnicholson | you may have to just use wireshark/tcpdump to do it |
| 21:47.23 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 21:47.34 | mnicholson | 'sip ?' should show debug in the list that pops up |
| 21:48.07 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:48.11 | goodjoke | 1 sec |
| 21:48.16 | goodjoke | got it, getting log |
| 21:48.19 | *** join/#asterisk shindig_ (n=matt@138.172.188.72.cfl.res.rr.com) |
| 21:50.01 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 21:50.50 | goodjoke | says log it too large for pastebin..just trying to find what you need |
| 21:50.53 | goodjoke | SIP Response message for INCOMING dialog BYE arrived |
| 21:51.16 | mnicholson | look for the OK response to the INVITE message |
| 21:51.32 | brunner | Is there any technical reason why using a media gateway to convert PRI's to SIP would be better than just buying TDM PCI cards? |
| 21:51.38 | mnicholson | and you may want to strip out any public IP addresses and hostnames before you post it |
| 21:52.25 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 21:52.33 | goodjoke | http://mibbit.com/pb/AlkfoF |
| 21:52.41 | goodjoke | should be all internal ips |
| 21:53.18 | *** join/#asterisk CapriCoRN^80 (i=administ@209.8.41.157) |
| 21:54.13 | mnicholson | goodjoke, that's the wrong OK message |
| 21:54.59 | mnicholson | it should have CSeq: XXXXX INVITE or something in it |
| 21:57.30 | *** join/#asterisk WindBack (i=jorge@201-212-51-44.cab.prima.net.ar) |
| 21:57.39 | goodjoke | http://mibbit.com/pb/4rzSm3 |
| 21:59.47 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 22:01.27 | mnicholson | goodjoke, that was not a successful call either |
| 22:05.12 | *** join/#asterisk xirdal (n=xirdal@203-mo7-3.acn.waw.pl) |
| 22:05.30 | ecret | when asterisk receives a sip call and uses exten => _X.,4,Dial(SIP/othersip), does asterisk still act as a go between or are packets sent from the original caller to the destination sip? |
| 22:06.27 | *** part/#asterisk xirdal (n=xirdal@203-mo7-3.acn.waw.pl) |
| 22:07.09 | WindBack | I have an strange problem. I can found yet the solution. I have a tdm800p with 2 Quad FX0 modules. I have four analog lines to connect my Asterisk PBX to the PSTN. This PBX is giving service for three different group of extension. One group of extensions are using the first analog line to in/out the PSTN. The socond group of extensions are using the second analog line to in/out the PSTN and finally the third group of extensions are usin |
| 22:07.09 | WindBack | g the THIRD AND FOURTH analog lines to in/out the PSTN. The last group of extensions use the fourth and third lines doing a pooling. The problem is in this lines: Very few times this lines keep unhunguped at the same time. At the first time I tougth it was a problem in the card, but I changed the ports and the problem contine. Another thing to say is: In this group of extensions there are a fax machine. Any Ideas?? |
| 22:07.14 | *** join/#asterisk CapriCoRN^80 (i=administ@207.176.6.154) |
| 22:08.10 | [TK]D-Fender | ecret: depends |
| 22:12.14 | ecret | [TK]D-Fender: my setup is pretty basic. http://pastebin.com/d678982c9 Is there a sip.conf property of some sort that sets this? |
| 22:12.45 | ecret | i read through sip.conf and could not discern one |
| 22:12.53 | jsmith | ecret: Asterisk still acts as a go-between (back-to-back user agent, in technical terms). If you have re-invites enabled (canreintvite=yes in sip.conf), then the media can go directly between the endpoints, but the signaling will still go through Asterisk |
| 22:13.06 | [TK]D-Fender | ecret: "canreinvite" |
| 22:13.42 | [TK]D-Fender | ecret: depends if your endpoints CAn or not... NAT interferes with this, etc |
| 22:14.16 | ecret | jsmith, [TK]D-Fender: thanks! |
| 22:18.55 | *** join/#asterisk PDani_ (n=pekdanie@catv-89-133-156-227.catv.broadband.hu) |
| 22:21.09 | WindBack | please can anybody helpme? |
| 22:26.02 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-133-195.dsl.stlsmo.sbcglobal.net) |
| 22:26.26 | *** part/#asterisk CapriCoRN^80 (i=administ@207.176.6.154) |
| 22:26.28 | *** join/#asterisk CapriCoRN^80 (i=administ@207.176.6.154) |
| 22:26.29 | *** join/#asterisk Great_Anta_baka (n=tensai@196.33.159.83) |
| 22:26.42 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:27.30 | LemensTS | hey all. i need to create a call queue via a php webpage. Im not sure the best way to do this, by creating an queues_additional.conf and updating it and including it in queues.conf........? |
| 22:29.36 | LemensTS | need to do the same thing with sip users |
| 22:29.36 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 22:31.05 | [TK]D-Fender | LemensTS: "Yes" |
| 22:32.13 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 22:33.14 | mmlj4 | what does this webpage need to do? |
| 22:34.04 | LemensTS | have to create a new queue from the webpage. than have to create a sip user and assign a queue to them. |
| 22:34.30 | mmlj4 | doable |
| 22:34.47 | LemensTS | well yea freepbx/tribox/a@h all do it :D |
| 22:34.57 | mmlj4 | I generate my sip users, extensions and polycom files via perl, so... |
| 22:35.22 | LemensTS | u just do it in a seperate file? |
| 22:35.32 | mmlj4 | yes, which I include |
| 22:35.37 | LemensTS | then send asterisk reload sip in ami? |
| 22:35.43 | [TK]D-Fender | who cares if its an included file or one big one? |
| 22:35.49 | mmlj4 | I don't do it realtime, so no |
| 22:35.50 | [TK]D-Fender | No functional difference |
| 22:36.24 | mmlj4 | [TK]D-Fender: because I wouldn't want to generate all of sip.conf, just the entries |
| 22:36.38 | [TK]D-Fender | the entries = sip.conf |
| 22:37.03 | mmlj4 | plus other stuff... fine, that's simple... extensions.conf is another matter |
| 22:37.05 | LemensTS | TK: id rather not have the webpage modifying the config file that has all the sip settings for the server |
| 22:37.56 | mmlj4 | my scripts are cli, not web, but there's no real difference |
| 22:38.18 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 22:38.26 | denon | ~book |
| 22:38.27 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:38.38 | f0ner00t | Hello. I'm trying to install asterkisk. But My module zaptel is failing. |
| 22:38.44 | LemensTS | @denon, does it show in the book how to do it? |
| 22:39.00 | mmlj4 | I need to locate my other book |
| 22:39.00 | denon | no, I was getting the url for someone else :) |
| 22:39.00 | [TK]D-Fender | LemensTS: "it"? |
| 22:39.23 | mmlj4 | LemensTS: with this, you're on your own |
| 22:39.29 | f0ner00t | Any idea why Zaptel module is failing? |
| 22:39.55 | [TK]D-Fender | f0ner00t: You've shown us NOTHING. Do you think we're psychic? |
| 22:40.04 | mmlj4 | f0ner00t: not unless you give us info |
| 22:40.42 | LemensTS | mmlj4: yea just wanted to get some pointers before i started this and found out there was a better way i should do it. |
| 22:41.17 | brunner | Is there any reason I wouldn't be able to use more than one Sangoma A108 card in a single machine? |
| 22:42.00 | denon | well, it is sangoma .. |
| 22:42.03 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 22:42.03 | mmlj4 | LemensTS: i think you have everything you need, asterisk-wise... you're left with a PHP problem |
| 22:42.25 | brunner | denon: what does that mean? |
| 22:42.28 | mmlj4 | sangoma-- |
| 22:42.58 | brunner | why shouldn't I buy sangoma? |
| 22:43.16 | f0ner00t | Hold on I |
| 22:43.22 | f0ner00t | I'm going to reinstall the package. |
| 22:43.33 | denon | brunner: you can buy whatever you'd like, just remember who has more experience with asterisk |
| 22:44.00 | mmlj4 | also, sangoma requires other drivers besides zapte;/dahdi |
| 22:44.06 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:44.11 | brunner | denon: okay, well is it difficult to use several digium cards in the same box? is there any reason why I couldn't use four or more? |
| 22:44.16 | [TK]D-Fender | Hrm... |
| 22:44.26 | [TK]D-Fender | LOL |
| 22:44.38 | [TK]D-Fender | brunner: Digium's site tells you its not recommended to use more than *2* |
| 22:45.15 | brunner | well that's not any more useful than the sangoma card, then |
| 22:45.23 | brunner | media gateways are so expensive =/ |
| 22:45.24 | f0ner00t | Setting up zaptel (1:1.4.11~dfsg-3) ... |
| 22:45.25 | f0ner00t | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
| 22:45.29 | f0ner00t | Thats what I'm getting. |
| 22:45.37 | f0ner00t | Sorry I am not calling Cleo. |
| 22:46.45 | f0ner00t | :) |
| 22:47.19 | [TK]D-Fender | brunner: 2 x A108d is 16 ports... how many do you need? |
| 22:47.47 | brunner | [TK]D-Fender: 16 will do it, but it's not clear that it's a good idea to use two in the same box. I couldn't find anything on their site about it. |
| 22:48.33 | mmlj4 | f0ner00t: which card are you trying to use? |
| 22:49.19 | f0ner00t | mmlj4. I'm not actually using a card. I just wanna do straight sip. Can't I use asterisk without an card? |
| 22:49.26 | mmlj4 | sure |
| 22:49.30 | f0ner00t | Do I need zaptel if i'm not using fxs0 |
| 22:49.39 | WindBack | I have an strange problem. I can found yet the solution. I have a tdm800p with 2 Quad FX0 modules. I have four analog lines to connect my Asterisk PBX to the PSTN. This PBX is giving service for three different group of extension. One group of extensions are using the first analog line to in/out the PSTN. The socond group of extensions are using the second analog line to in/out the PSTN and finally the third group of extensions are usin |
| 22:49.40 | WindBack | g the THIRD AND FOURTH analog lines to in/out the PSTN. The last group of extensions use the fourth and third lines doing a pooling. The problem is in this lines: Very few times this lines keep unhunguped at the same time. At the first time I tougth it was a problem in the card, but I changed the ports and the problem contine. Another thing to say is: In this group of extensions there are a fax machine. Any Ideas?? |
| 22:49.41 | f0ner00t | fxs / fx0 / t1 / pri. |
| 22:49.43 | mmlj4 | are you doing any conferencing? |
| 22:50.12 | f0ner00t | mmlj4 I was thinking about just using it as a test. I'm in telecomm and I work with voip / t1 switches all day long. |
| 22:50.32 | mmlj4 | then don't even load zaptel.conf |
| 22:50.38 | f0ner00t | Cool cool. |
| 22:50.55 | f0ner00t | What is the best way to configure the asterisk? |
| 22:51.06 | mmlj4 | there is no best way |
| 22:51.08 | [TK]D-Fender | f0ner00t: vi |
| 22:51.16 | mmlj4 | [TK]D-Fender++ |
| 22:51.18 | f0ner00t | ahh VI the conf files huh. |
| 22:51.59 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 22:53.28 | f0ner00t | There isn't any gui interface LOL. |
| 22:53.35 | mmlj4 | they exist |
| 22:53.43 | [TK]D-Fender | And they all own your ass |
| 22:53.49 | mmlj4 | that they do |
| 22:54.04 | mmlj4 | ask LemensTS, he's writing one now |
| 22:54.17 | Habile | yeah but saved me having to actually do any thinking ;) |
| 22:54.23 | f0ner00t | LOL |
| 22:54.31 | f0ner00t | I guess I should go through the config than. |
| 22:54.35 | mmlj4 | thinking is hard! ask barbie |
| 22:54.38 | f0ner00t | and edit some conf files. |
| 22:54.52 | f0ner00t | I can use IPKALL and gator right? |
| 22:55.25 | Habile | IPKALL works fine |
| 22:56.31 | f0ner00t | Thank you |
| 22:56.48 | mmlj4 | why? |
| 22:57.40 | Habile | because asterisk is the #1 software PBX - that's why |
| 22:57.46 | Habile | isn't it? |
| 22:58.05 | mmlj4 | ipkall? |
| 22:58.07 | mmlj4 | gator? |
| 22:58.19 | Habile | oh ... freebie inbounds that's all I use IPKALL for |
| 22:59.25 | Habile | is that a bit lame? |
| 23:01.48 | f0ner00t | What if I don't have a dialing plan cuz there will be no outbound. |
| 23:02.24 | [TK]D-Fender | f0ner00t: If you have no dialplan you have no INBOUND either |
| 23:02.53 | carrar | dialplans are over rated! |
| 23:03.11 | [TK]D-Fender | shoots Roger Ebert |
| 23:03.15 | [TK]D-Fender | carrar: How about now? |
| 23:03.48 | carrar | hop on the Harley and just visit the person instead of calling them! :) |
| 23:04.09 | [TK]D-Fender | ok, Off to play with sharp & pointy things :) |
| 23:04.10 | [TK]D-Fender | BBL |
| 23:04.39 | *** join/#asterisk docidu (n=eris@whthyt253-26.northwestel.net) |
| 23:04.41 | carrar | mowing the lawn? |
| 23:04.46 | f0ner00t | [TK]D-Fender: That acutally makes sense. |
| 23:04.53 | f0ner00t | I've works on Mitel200sx before. |
| 23:04.53 | f0ner00t | LOL |
| 23:05.39 | prakriti | we have two seperate working asterisk systems. Each has a different number and its own set of extensions and voicemail boxes. |
| 23:05.54 | prakriti | Is there any way to have one install basically act like two? |
| 23:06.05 | mmlj4 | prakriti: do what? |
| 23:06.31 | mmlj4 | oh, hrm, yes, it's possible |
| 23:06.31 | prakriti | I want to move both numbers to the same box, but still have all the extensions etc seperated. |
| 23:06.44 | mmlj4 | use different contexts |
| 23:06.59 | mmlj4 | unless that's deprecared now |
| 23:07.53 | prakriti | we have asterisk/static/config/index.html runnng |
| 23:08.05 | prakriti | and we would even want to seperate those |
| 23:08.53 | *** join/#asterisk matt_keys (n=matt_key@h88.17.40.69.dynamic.ip.windstream.net) |
| 23:08.59 | mmlj4 | i have no idea what that is.... what are you running? |
| 23:10.00 | prakriti | oh, i thought that was a standard web interface |
| 23:10.11 | prakriti | some digium|asterisk web config interface |
| 23:10.30 | matt_keys | I'm having problems getting a Grandstream GSW4104 to register. I enabled debug logging and it's showing SIP message 403 |
| 23:11.02 | matt_keys | GXW4104* |
| 23:11.07 | f0ner00t | So zaptel is needed for sip conferencing? |
| 23:13.00 | keith4_ | ztdummy, at least |
| 23:13.03 | matt_keys | before the 403 message, it says: Packet Dropped During Privision: INVITE sip:150@192.168.1.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK50c2b6b1;rport |
| 23:14.33 | matt_keys | If I pastebin the log can someone help me out? I've been at this all day and I don't seem to be having any luck |
| 23:17.58 | f0ner00t | So if the module crashed on Zaptel should I try installing it manually? |
| 23:18.21 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:21.26 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 23:25.10 | mmlj4 | f0ner00t: do you have USB enabled on that box? |
| 23:25.12 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 23:25.41 | seb- | is sip.conf where you first set up asterisk to accept a call for testing? |
| 23:25.58 | mmlj4 | seb-: calls are accepted in extensions.conf |
| 23:26.06 | mmlj4 | phones are set up in sip.conf |
| 23:27.11 | seb- | tessier_: mmlj4: thanks |
| 23:27.37 | seb- | mmlj4: i mean thanks to you |
| 23:28.24 | seb- | mmlj4: i have a cheapo grandstream POTS->VOIP converter at home...it has a device id? |
| 23:28.28 | f0ner00t | mmlj4. Of course USB is enabled. |
| 23:29.17 | mmlj4 | f0ner00t: zaptel's ztdummy relies on USB for timing, usually |
| 23:29.22 | mmlj4 | why I asked... |
| 23:29.36 | mmlj4 | ok, you're loading it wrong, or it's compiled incorreclty |
| 23:29.51 | f0ner00t | mmlj4 I'm using the debian apt-get install zaptel. |
| 23:30.53 | mmlj4 | etch? lenny? |
| 23:31.05 | mmlj4 | but again, you might be loading it wrong |
| 23:31.29 | mmlj4 | I can tell you that I tried lenny on my last project, and asterisk segfaulted every time I tried to start it |
| 23:31.33 | mmlj4 | YMMV |
| 23:31.46 | f0ner00t | Lenny. |
| 23:32.16 | f0ner00t | Asterisk is running. I need to get Zaptel to run. |
| 23:39.40 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:40.37 | seb- | is the "register => ..." line in sip.conf only for OUTGOING or for INCOMING as well? |
| 23:42.31 | f0ner00t | You do not appear to have the sources for the 2.6.26-1-686 kernel installed. |
| 23:42.39 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 23:42.45 | f0ner00t | What am I missing? |
| 23:42.50 | mmlj4 | you're compiling now? |
| 23:43.21 | f0ner00t | I wanted to try to compile Zaptel since my apt-get install Zaptel is getting module not found. |
| 23:43.54 | mmlj4 | you don't have the kernel sources installed |
| 23:43.56 | mmlj4 | wait one |
| 23:44.43 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 23:45.10 | f0ner00t | What do you mean wait one? |
| 23:45.29 | mmlj4 | one moment |
| 23:45.45 | mmlj4 | apt-get install kernel-devel kernel-headers |
| 23:45.59 | f0ner00t | Thank you for getting those packages for me :) |
| 23:46.20 | mmlj4 | apt-cache search kernel | grep kernel |
| 23:46.29 | mmlj4 | do an apt-gt update frist |
| 23:46.30 | mmlj4 | first |
| 23:46.38 | f0ner00t | This should be the point that you say asterisk is too advanced for ya |
| 23:46.39 | f0ner00t | LOL |
| 23:46.48 | f0ner00t | Yea I wil do a update first. |
| 23:47.28 | mmlj4 | well, if you're compiling zaptel, you need to compile asterisk too |
| 23:47.34 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 23:47.42 | f0ner00t | Yea. That makes sense. |
| 23:47.55 | mmlj4 | apt-get remove all the zaptel and asterisk stuff before compiling |
| 23:47.56 | f0ner00t | I think asterisk compiled fine. |
| 23:48.06 | VaGoNeTaS | guys, i got a little problem here... |
| 23:48.07 | f0ner00t | Ahh I'll just try the apt-get way. |
| 23:48.10 | f0ner00t | Its much easier. |
| 23:48.23 | VaGoNeTaS | im sure that u can help me |
| 23:49.29 | mmlj4 | untar asterisk.whatever.tar.gz and zaptel, then stick those sources in /usr/include/asterisk and /user/include/zaptel (those exact paths, ignore the version numbers) |
| 23:49.35 | NovceGuru | fuck |
| 23:49.37 | f0ner00t | Kernel-devel does not exsist. |
| 23:49.55 | mmlj4 | compile zaptel first, then asterisk... do a make menuselect to verify zaptel's included |
| 23:49.59 | mmlj4 | lower-case |
| 23:50.09 | f0ner00t | I did lower care. |
| 23:50.12 | f0ner00t | case. |
| 23:50.15 | mmlj4 | then I dunno |
| 23:50.24 | f0ner00t | apt-get install kernel-devel |
| 23:50.24 | f0ner00t | Reading package lists... Done |
| 23:50.24 | f0ner00t | Building dependency tree |
| 23:50.24 | f0ner00t | Reading state information... Done |
| 23:50.24 | f0ner00t | E: Couldn't find package kernel-devel |
| 23:50.27 | VaGoNeTaS | ok this is the situation |
| 23:50.49 | mmlj4 | what does /etc/apt/sources.list say? |
| 23:50.58 | VaGoNeTaS | i've just installed Asterisk 1.4.24 with Dahdi linux 2.1.0.4 + Dahdi tools 2.1.0.2 |
| 23:51.03 | VaGoNeTaS | everything was perfect |
| 23:51.19 | VaGoNeTaS | till i've realized that i've forgot to install the libpri ... |
| 23:51.29 | f0ner00t | deb http://ftp.us.debian.org/debian/ lenny main |
| 23:51.29 | f0ner00t | deb-src http://ftp.us.debian.org/debian/ lenny main |
| 23:51.29 | f0ner00t | deb http://security.debian.org/ lenny/updates main |
| 23:51.29 | f0ner00t | deb-src http://security.debian.org/ lenny/updates main |
| 23:51.29 | f0ner00t | deb http://volatile.debian.org/debian-volatile lenny/volatile main |
| 23:51.32 | f0ner00t | deb-src http://volatile.debian.org/debian-volatile lenny/volatile main |
| 23:51.37 | VaGoNeTaS | crap, man im writting here... |
| 23:51.38 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 23:51.48 | f0ner00t | Sorry VaGoNeTaS. |
| 23:51.52 | VaGoNeTaS | ty |
| 23:51.57 | mmlj4 | VaGoNeTaS: so go back and install that, then dahdi, then asterisk, again |
| 23:52.04 | VaGoNeTaS | yep dude |
| 23:52.05 | VaGoNeTaS | i did |
| 23:52.06 | mmlj4 | do make clean whererever appropriate |
| 23:52.14 | VaGoNeTaS | but... the "dahdi" command inside the Asterisk console is gone |
| 23:52.24 | *** part/#asterisk CapriCoRN^80 (i=administ@207.176.6.154) |
| 23:52.31 | VaGoNeTaS | u know? |
| 23:52.35 | mmlj4 | volatile? testing? bleeding? |
| 23:52.38 | VaGoNeTaS | so i can see "dahdi show status" |
| 23:52.50 | VaGoNeTaS | but "dahdi command is gone" |
| 23:53.00 | mmlj4 | VaGoNeTaS: make menuselect when compiling asterisk, if it'll let you |
| 23:53.06 | mmlj4 | not sure if that applies to * 1.6 |
| 23:53.07 | VaGoNeTaS | did it |
| 23:53.13 | VaGoNeTaS | and this is 1.4 |
| 23:53.15 | VaGoNeTaS | not 1.6 |
| 23:53.20 | mmlj4 | core show channeltypes |
| 23:53.35 | f0ner00t | mmlj4 any idea? |
| 23:53.36 | mmlj4 | ah. |
| 23:54.02 | mmlj4 | f0ner00t: unless you're running testing or unstable, no |
| 23:54.16 | f0ner00t | Nope stable. |
| 23:54.21 | mmlj4 | no idea |
| 23:54.27 | f0ner00t | Wouldn't run unstable till i'm a pro. |
| 23:54.32 | VaGoNeTaS | i know, but if the "dahdi" command is gone inside the Asterisk console means that something is working wrong |
| 23:54.32 | mmlj4 | except my only experience with lenny was segfaults |
| 23:54.57 | mmlj4 | VaGoNeTaS: I don't think asterisk got recompiled with dahdi support |
| 23:55.18 | VaGoNeTaS | the command was included before recompiling with libpri |
| 23:55.54 | VaGoNeTaS | i admin several asterisk machines |
| 23:55.55 | VaGoNeTaS | look |
| 23:56.24 | VaGoNeTaS | asterisk-hp1*CLI> dahdi show status |
| 23:56.24 | VaGoNeTaS | Description Alarms IRQ bpviol CRC4 |
| 23:56.24 | VaGoNeTaS | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 23:56.24 | VaGoNeTaS | T4XXP (PCI) Card 0 Span 2 OK 0 0 0 |
| 23:56.24 | VaGoNeTaS | T4XXP (PCI) Card 0 Span 3 OK 0 0 0 |
| 23:56.31 | VaGoNeTaS | see? |
| 23:57.09 | f0ner00t | I wish I knew why that don't work. |
| 23:57.21 | f0ner00t | It doesn't look like the kernel exsists. |
| 23:57.21 | VaGoNeTaS | me too |
| 23:57.23 | VaGoNeTaS | :s |
| 23:57.53 | VaGoNeTaS | the thing is that it was working before |
| 23:57.58 | VaGoNeTaS | 4 hours ago |
| 23:58.05 | VaGoNeTaS | before recompiling all this stuff with libpri |