| 00:03.58 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 00:08.40 | *** join/#asterisk grinder13 (~grinder@cpc2-sgyl2-0-0-cust1286.sgyl.cable.virginmedia.com) |
| 00:09.16 | grinder13 | hello! I would like to ask: is it possible to have encryption with IAX trunking? |
| 00:09.19 | ChannelZ | I don't think a sip reload invalidates all the peers, they stick around in the astdb |
| 00:11.39 | ChannelZ | or not. I just commented out one in 1.6.2.9 and reloaded and it doesn't show up in peers |
| 00:14.30 | *** part/#asterisk socomm (~socomm@99-121-209-85.lightspeed.irvnca.sbcglobal.net) |
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| 00:20.57 | russellb | grinder13: yes, as of asterisk 1.4, you can do that |
| 00:21.01 | russellb | you just set encryption=yes, heh |
| 00:26.00 | grinder13 | russelb, because I 've found an old thread in the mailing list which says that IAX trunk and encryption are mutually exclusive. also here: http://www.voip-info.org/wiki/view/IAX+encryption , I can see "trunk=no" |
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| 00:33.06 | p3nguin | I think sip was taking extra time to reload, since waiting a while made the entry finally go away, and I haven't been able to duplicate the behavior since. |
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| 00:50.46 | *** join/#asterisk rs100 (~redshift@c-68-80-54-179.hsd1.pa.comcast.net) |
| 00:51.26 | rs100 | Hi folks, I'm having an issue with a PRI - I can make inbound calls just fine; however, when I dial out I get a message saying "Everyone is busy/congested at this time (1:0/0/1)" |
| 00:54.16 | rs100 | I'd welcome any suggestions, really pulling my hair out on this one since inbound calls work just fine |
| 00:55.12 | WIMPy | Turn up verbose and debug and if that doesn't give you a hint, PB the output of a call attempt. |
| 00:57.34 | rustyclarkson | When a parked call times out, is it possible to change the values put into the Dial command? (identifier/timeout/etc) |
| 01:02.56 | rs100 | http://pastebin.com/d2XbJMkT |
| 01:03.00 | rs100 | from my above error |
| 01:04.18 | rs100 | with debug at 9 and verbose at 9 |
| 01:06.02 | WIMPy | Ok, not that informative. Try pri intensive debug. |
| 01:07.29 | rs100 | WIMPy: what's the command? pri set debug on is returning an error... |
| 01:07.40 | rs100 | ca-redlands-asterisk0*CLI> pri set debug on |
| 01:07.40 | rs100 | No such command 'pri set debug on' (type 'core show help pri set' for other possible commands) |
| 01:08.51 | rs100 | nevermind, found it |
| 01:09.02 | WIMPy | Err, 'pri intensive debug span 1' IIRC. Don't have dahdi enabled here. |
| 01:09.34 | rs100 | yeah, that's it |
| 01:09.41 | rs100 | it didn't seem to spam much when the call failed |
| 01:09.44 | rs100 | but the output is here |
| 01:09.54 | rs100 | http://pastebin.com/RaJ63Mvd |
| 01:10.58 | WIMPy | Indeed. No try. |
| 01:11.14 | WIMPy | Did you try to dial a group not defined in your dahdi config? |
| 01:11.49 | rs100 | No, one second and I will |
| 01:14.26 | rs100 | Same message if the group is one that is not defined |
| 01:14.36 | rs100 | http://pastebin.com/JBAzjbKJ |
| 01:14.41 | rs100 | that's my chan_dahdi for the pri |
| 01:17.07 | WIMPy | "dahdichan"? Wahts that? Shouldn't that be "Channel" or was it "channels"? |
| 01:18.01 | rs100 | its the keyword for using it in a separate dahdi.conf section |
| 01:18.05 | rs100 | I'll try switching it for kicks |
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| 01:20.08 | DiligaF | Hi, can someone help me with getting asterisk to relay dtmf tones? |
| 01:20.29 | p3nguin | What's the problem? |
| 01:22.00 | rs100 | http://pastebin.com/UdqMktgE |
| 01:22.08 | rs100 | New pastebin for the new chan_dahdi |
| 01:22.09 | rs100 | same issue |
| 01:22.21 | DiligaF | I have a Merlin Legend with a PRI attached to the asterisk system. for some reason I can call a conference bridge from a softphone and it works fine. But when I call from the Legend it does not. I am using dahdi |
| 01:23.19 | DiligaF | I have tried relaxdtmf=yes but that had no effect |
| 01:23.31 | p3nguin | What dtmfmode are you using? |
| 01:23.38 | rs100 | I'm getting a fast busy when it attempts to make the call, not sure if that's applicable |
| 01:24.35 | DiligaF | That would be a problem as I do not know what the Legend system uses. I dont know if it is in-band or out-of-band and I have tried both with no effect. |
| 01:24.46 | *** join/#asterisk alangarf (~alan@218.185.51.89) |
| 01:25.46 | DiligaF | rs100 are you using a Merlin Legend? |
| 01:26.16 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 01:27.38 | DiligaF | I also have a Merlin Legend tied to Switchvox and have no issues with dtmf |
| 01:28.12 | DiligaF | this one is getting to me and is a critical system that needs to work. |
| 01:30.09 | DiligaF | anyone?? |
| 15:09.12 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:09.12 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
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| 15:42.08 | *** join/#asterisk mweichert (~mweichert@216.16.254.34) |
| 15:42.28 | mweichert | hi, in my dialplan how to retrieve the DID dialed for a DAHDI channel? |
| 15:42.58 | tzafrir_laptop | duh, dude |
| 15:43.07 | tzafrir_laptop | the extension! |
| 15:43.43 | tzafrir_laptop | seriously, though: is it analog? If so: you don't really have a DID |
| 15:44.58 | wcselby | Naikrovek - you about? |
| 15:45.39 | wcselby | i'm looking at deploying a large number of polycom phones and I'm hoping to ease provisioning by setting a boot server value in DHCP, but I'm not sure which option I should use. Option 66 with the IP address of the FTP server should work by default, correct? |
| 15:45.51 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:47.13 | mweichert | tzafrir_laptop, when I use ${EXTEN}, I get "DAHDI/2-1" |
| 15:48.11 | tzafrir_laptop | mweichert, that's odd. This makes sense as the contents of ${CHANNEL} |
| 15:48.30 | *** join/#asterisk ddickenson (~chatzilla@67-198-0-5.static.grandenetworks.net) |
| 15:48.31 | [TK]D-Fender | wcselby: Yes |
| 15:48.37 | tzafrir_laptop | Can you pastebin your dialplan? |
| 15:48.48 | [TK]D-Fender | mweichert: No, you don't |
| 15:48.50 | wcselby | [TK]D-Fender - thanks :) |
| 15:49.05 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 15:50.25 | ddickenson | anyone who can help with asterisk realtime / mysql. I have followed setup info on voip-info.org and have everything as it is written but see nothing when issuing sip show peers at the cli and phones will not come up. I have done this before just in a test environment but has something changed on 1.6 that makes the setup different? |
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| 15:51.31 | mweichert | tzafrir_laptop, I can paste the relevant context. I'm using freepbx so the dialplan is quite large |
| 15:51.57 | [TK]D-Fender | mweichert: Show us the CALL. |
| 15:52.16 | [TK]D-Fender | ddickenson: sip show peers won't show you realtime peers |
| 15:52.20 | tzafrir_laptop | mweichert, what dahdi device is it? |
| 15:53.00 | ddickenson | I read if you did the realtime cache it would. |
| 15:53.16 | tzafrir_laptop | FreePBX has a convoluted way to generate a "DID" from a dahdi channel number, if you define it so. |
| 15:53.23 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 15:53.30 | Kobaz | show us the monies! |
| 15:53.43 | mweichert | http://pastie.org/1042641 |
| 15:54.14 | [TK]D-Fender | mweichert: Executing [5197725555@from-pstn:1] NoOp("DAHDI/1-1", "Catch-All DID Match - Found 5197725555 - You probably want a DID for <-- there you have it. |
| 15:54.15 | tzafrir_laptop | Executing [5197725555@from-pstn:1] |
| 15:54.21 | [TK]D-Fender | mweichert: 5197725555 |
| 15:54.43 | mweichert | yes... but in [pri-incoming] I have: Exten => s,n,SayDigits(${EXTEN}) |
| 15:55.06 | tzafrir_laptop | What is pri-incoming? How did you get there? |
| 15:55.08 | [TK]D-Fender | mweichert: the exten is obviously "s" because ${EXTEN} is where you ***ARE*** |
| 15:55.42 | [TK]D-Fender | [11:54]<mweichert>yes... but in [pri-incoming] I have: Exten => s,n,SayDigits(${EXTEN}) <-- this doesn't look back in time to see what it WAS |
| 15:57.38 | mweichert | [TK]D-Fender, ok - that makes sense. Hmm, I should maybe ask the freepbx guys if they set a channel variable for the incoming did |
| 15:57.58 | mweichert | [TK]D-Fender, is the dialplan, is there any way of listing a channel variables available? |
| 15:58.04 | mweichert | *in the dialplan... |
| 15:58.23 | [TK]D-Fender | mweichert: huh?! |
| 15:59.00 | mweichert | [TK]D-Fender, in the dialplan, in there any way that I can output to stdout or a logfile what channel variables are available? |
| 15:59.25 | [TK]D-Fender | mweichert: "core show channel [channel]" |
| 15:59.56 | wcselby | when I declare a global variable as YEAR = ${STRFTIME(${EPOCH},,%Y)} in the [globals] context, it's not evaluating the way I would it expect it to. |
| 16:00.08 | [TK]D-Fender | mweichert: First you are receiving a DID that you didn't set up in FreePBX. This is itself a problem. Go configure it and myabe you'll be able to do somethign with it later. |
| 16:00.33 | [TK]D-Fender | wcselby: Globals don't evalaute |
| 16:00.58 | Joe_CoT | so I'm not aware of having stun enabled for asterisk at all, but using the SVN version i keep getting "ast_sip_ouraddrfor: stun failed" messages. I have externip set, the ip is correct in sip settings. Any idea why it's trying to stun? |
| 16:01.17 | wcselby | It still gives me a timestamp, but it's not what I'm wanting |
| 16:02.26 | wcselby | it's giving me the default system format (i.e if I type date on the command line, or if I were to use ${STRFTIME(${EPOCH},,%c)) |
| 16:02.50 | wcselby | but okay |
| 16:02.51 | p3nguin | What do you want? |
| 16:03.06 | wcselby | I want just the year, hence the %Y |
| 16:03.20 | p3nguin | as in "2010"? |
| 16:03.24 | wcselby | yes |
| 16:03.27 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 16:03.42 | wcselby | i've got similar variables for month and day as well |
| 16:04.19 | wcselby | just wanted to save myself having to rewrite the entire strftime everytime |
| 16:04.41 | wcselby | hmmmm |
| 16:05.35 | p3nguin | Verbose(${STRFTIME(${EPOCH},,%Y)}) printed 2010 on my CLI. |
| 16:08.43 | *** join/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 16:09.47 | wcselby | p3nguin - it does on mine as well. but if you declare a global as YEAR = ${STRFTIME(${EPOCH},,%Y)}, then in your dialplan do Verbose(${YEAR}), it doesn't do that. It instead output the entire date string, i.e - Tue Jul 13 11:08:42 2010 |
| 16:10.11 | p3nguin | I'll test it that way. |
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| 16:11.56 | p3nguin | That's crazy. |
| 16:12.14 | wcselby | heh |
| 16:12.30 | wcselby | i would think I wouldn't get anything, since, as [TK]D-Fender said, globals don't evaluate that way |
| 16:12.41 | wcselby | well, he said they don't evaluate. |
| 16:12.45 | p3nguin | show globals says: YEAR=Tue Jul 13 11:11:44 2010 |
| 16:13.20 | [TK]D-Fender | I COULD BE MISTAKEN |
| 16:13.33 | wcselby | [TK]D-Fender - heh, never |
| 16:14.14 | p3nguin | It is certainly doing something, albeit not what seems logical. |
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| 16:24.31 | Joe_CoT | yeah, really don't understand. I don't have stun enabled anywhere, but I keep getting these "stun failed" messages. |
| 16:24.43 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 16:24.52 | [TK]D-Fender | isn't seeing backup |
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| 16:53.28 | *** join/#asterisk OppieT30 (~root@173-26-156-159.client.mchsi.com) |
| 16:54.32 | OppieT30 | Hello, I am running AsteriskNow and can't find where to change the default password for the freepbx user. |
| 16:56.43 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 16:59.26 | Joe_CoT | ok, so gave up on the svn version, back to 1.6.9. Here's the main issue i'm having: when I start asterisk, and I register with my provider, incoming calls are fine. after a little while, incoming calls aren't recognized as coming from that peer, and are sent to context default instead of the one i set. |
| 16:59.45 | Joe_CoT | any ideas on why that would happen? What asterisk is matching on? I have insecure set to invite,port |
| 17:03.32 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 17:05.51 | OppieT30 | Hmm. |
| 17:06.48 | [TK]D-Fender | OppieT30: #freepbx <- |
| 17:08.34 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
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| 17:23.54 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 17:23.56 | [sr] | howdy |
| 17:24.06 | [sr] | need a little help not direct asterisk related |
| 17:24.15 | OppieT30 | I can try. |
| 17:24.24 | [sr] | i have a port configured as extension on LCR, have a crossover ISDN cable |
| 17:24.45 | [sr] | but when i try to call that extension doesn't work, says unvaiable |
| 17:25.52 | OppieT30 | That is to far over my head. |
| 17:26.24 | OppieT30 | crossover ISDN. Never heard of that. Do you mean crossover ethernet cable? |
| 17:26.32 | [sr] | LCR show's the port as TE ptmp extension... should be NT ... |
| 17:26.41 | [sr] | OppieT30: no, != things |
| 17:29.15 | fauxalliance | OppieT30, Ethernet cross link cables don't work, because they use differen pairs than ISDN does. |
| 17:30.13 | Kobaz | [sr]: isdn is a layer 2 protocol... what you're referring to is a T1 crossover |
| 17:30.24 | [TK]D-Fender | IIRC you can't really use a crossover like that to place calls, only to test sync |
| 17:31.22 | [sr] | hum |
| 17:31.33 | [sr] | the schema will be a t1 crossover? |
| 17:31.58 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 17:32.17 | [sr] | right now my conventional phones are telling that this isdn extension is unavailable |
| 17:32.19 | fauxalliance | [sr], crosslink to isdn card in NT mode to telephone, then use a regular cable (ISDN or Ethernet) to your NT via the second port... the rest is all in the configuration of the software... |
| 17:32.55 | Kobaz | are you doing t1 or bri? |
| 17:33.06 | [sr] | wait fauxalliance |
| 17:33.28 | [sr] | Kobaz: the line cames from my siemens HI100 |
| 17:33.40 | Kobaz | and your plugging it into what |
| 17:33.56 | [sr] | my HFC-4S card, the ports are in NT mode now |
| 17:34.29 | Kobaz | it's BRI then |
| 17:34.35 | [sr] | yap |
| 17:35.08 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 17:35.28 | Kobaz | never did bri... I don't even know if the cabling is the same as t1 |
| 17:35.56 | Qwell | Kobaz: That is why it's called an ISDN cross-over cable. |
| 17:36.31 | [sr] | the cable schema i did was a ISDN crossover |
| 17:38.40 | Kobaz | Qwell: well since noone in this area really does bri... everyone calls it a t1 crossover |
| 17:38.47 | Kobaz | interesting |
| 17:39.09 | Qwell | call it whatever you like. you'll just be wrong. ;p |
| 17:39.48 | Kobaz | well, it's technically both |
| 17:39.53 | [sr] | hum |
| 17:39.56 | *** join/#asterisk decklar (~ross.inne@196.31.81.190) |
| 17:40.54 | [sr] | so T1 crossover != than ISDN crossover |
| 17:41.06 | Kobaz | it's the same thing |
| 17:42.28 | [sr] | wait wait |
| 17:42.31 | [sr] | i have one problem |
| 17:42.50 | [sr] | the isdn extension that cames from the siemens HICOM100 it's a one pair cable only |
| 17:42.58 | [sr] | so this crossover will do nothing to me.. |
| 17:43.12 | Qwell | one pair? that isn't ISDN. |
| 17:43.12 | [sr] | just confirmed that |
| 17:43.38 | Kobaz | one pair is generally analog or a propriatary digital extension |
| 17:43.58 | [sr] | it'll be the propriatary extension... like alcatel.. |
| 17:44.00 | [sr] | grrrrr |
| 17:44.07 | Kobaz | like siemens has it's own digital signalling |
| 17:44.12 | [sr] | so, i'll have no luck connection this to my HFC card, right? |
| 17:44.14 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:44.34 | Qwell | [sr]: Can you plug a telephone into an AC outlet? |
| 17:44.40 | Kobaz | hehe |
| 17:44.54 | Kobaz | Qwell: you *could* |
| 17:45.11 | telnettech | sr.... are you in US or canada? |
| 17:45.23 | [sr] | telnettech: Portugal |
| 17:46.04 | [sr] | Qwell: to damage the phone? :P |
| 17:46.25 | Talirk81 | lately we have been having alot of "carrier" issues with aretta, what sip providers do you guys use? |
| 17:46.36 | telnettech | sr......here is a link about ISDN cabling....look on page 5 for the pinout http://www.telos-systems.com/support/csb/TCSB_010103.pdf |
| 17:46.40 | Qwell | ~itsplist-us |
| 17:46.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 17:46.48 | Talirk81 | im starting to think aretta isnt a good as they used to be :( |
| 17:47.05 | *** join/#asterisk Banad (~banad@pool-74-101-134-245.nycmny.fios.verizon.net) |
| 17:47.13 | Talirk81 | well i know the offical list i mean personally what do people like |
| 17:47.18 | telnettech | sr......i think that is what you are looking for |
| 17:47.22 | Qwell | there is no "official" list. |
| 17:47.33 | Banad | Hi guys how can i create a cron job to reload sip all nights ? |
| 17:47.35 | Talirk81 | ok i call infobot the offical list :P |
| 17:47.36 | Qwell | The bot was rather clear on it.. |
| 17:47.49 | Talirk81 | but again i wasnt asking for the bots input |
| 17:47.56 | Talirk81 | but actual people impressions |
| 17:48.02 | [sr] | telnettech: nice doc, but doesn't have any one pair example |
| 17:48.04 | Qwell | you mean the actual people that update the bot? |
| 17:48.21 | Talirk81 | i meant as a commnity actvly in the room |
| 17:48.26 | Qwell | you mean the actual people that update the bot? |
| 17:48.42 | Talirk81 | the people who update the bot are not the only people in the room generally |
| 17:48.45 | Kobaz | Banad: man cron |
| 17:48.49 | Kobaz | man 5 crontab |
| 17:48.54 | telnettech | i never seen an ISDN cable with just 1 pair...whether BRI or PRI |
| 17:49.36 | Qwell | So you'd like the minority of the channel to give you providers simply because they aren't on the list? That seems silly. |
| 17:49.52 | [TK]D-Fender | Banad: asterisk -rx "sip reload" |
| 17:49.57 | Qwell | you might as well ask "Hi, what is the 8th best provider?" |
| 17:50.01 | Kobaz | hehe |
| 17:50.10 | [TK]D-Fender | Qwell: You don't even KNOW, do you? |
| 17:50.14 | [TK]D-Fender | :p |
| 17:50.17 | Kobaz | voicepulse is getting sucky these days |
| 17:50.25 | Kobaz | they now have a 10 dollar minimum monthly charge |
| 17:50.26 | Qwell | [TK]D-Fender: clearly not. or it would be on the list. |
| 17:51.02 | [sr] | telnettech: so my chances to make it work are zero... :( |
| 17:51.07 | Talirk81 | ops control a bot, so while their thoughts are valid , it doesnt mean only theirs count does it |
| 17:51.07 | Kobaz | they were nice to have as a backup voip provider, but now there's just too many other providers that are better and cheaper, and dont screw you with billing |
| 17:51.16 | Qwell | Talirk81: EVERYBODY controls the bot |
| 17:51.46 | Talirk81 | Qwell, you cant edit an entry made by someone else last time i checked unless you were an op |
| 17:51.51 | Qwell | Of course you can. |
| 17:52.34 | telnettech | sr..... not saying that, im just saying that I have never ran into a 1 pair ISDN cable....everything I have ever seen in my 10 years of telecom has been 2 pairs...the only difference is that T-1 is crossover and ISDN is straight thru |
| 17:52.40 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:52.57 | [TK]D-Fender | Talirk81: How was the Cretaceous anyway? ;) |
| 17:53.14 | Banad | <Kobaz> man 5 crontab any page to check it out ? |
| 17:53.21 | Banad | i am new on this field .... |
| 17:53.22 | Kobaz | Banad: what? |
| 17:53.42 | Qwell | Banad: Open a console. type this: man man |
| 17:53.44 | Banad | i need to create a cron job to reload sip every nihgt |
| 17:54.15 | Kobaz | telnettech: it depends on the devices in question whether it's straight through or crossover |
| 17:54.18 | Banad | No such command 'man' (type 'help man' for other possible commands) |
| 17:54.27 | Kobaz | Banad: not in your asterisk console |
| 17:54.28 | Banad | -sh: man: not found |
| 17:54.33 | fauxalliance | hrhrhr |
| 17:54.58 | Kobaz | Banad: you're not running trixbox or something like that, are you? |
| 17:55.01 | [TK]D-Fender | Banad: Not from * CIL <- |
| 17:55.03 | [TK]D-Fender | CLI |
| 17:55.17 | Banad | -sh: man: not found |
| 17:55.25 | Banad | no trixbox |
| 17:55.31 | [TK]D-Fender | bandFix your paths or install a sane OS |
| 17:55.37 | Kobaz | install a real linux distribution |
| 17:55.50 | Kobaz | find / -name man |
| 17:56.07 | [TK]D-Fender | Kobaz: Will fail if find isn't in his path either ;) |
| 17:56.13 | Kobaz | heh, yeah |
| 17:56.26 | Kobaz | Banad: does 'ls' even work? |
| 17:56.43 | Banad | of course |
| 17:56.48 | Kobaz | you never know |
| 17:57.01 | [sr] | telnettech: have to dig... but don't find nothing on google regarding to this |
| 17:57.12 | Kobaz | Banad: uname -a |
| 17:58.01 | Banad | Linux Fri Apr 23 03:16:12 EDT 2010 ppc unknown |
| 17:58.12 | hardwire | anybody use the Aastra 6757i CT? |
| 17:58.15 | Kobaz | ppc... fun |
| 17:58.23 | Kobaz | Banad: cat /etc/*release* |
| 17:58.29 | hardwire | can it really handle 9 calls at once? can I associate lines with wireless handsets? |
| 17:59.06 | Kobaz | Banad: it doesn't even have a kernel version... you're running some prebuilt asterisk distribution |
| 17:59.36 | Banad | not pretty sure about that |
| 17:59.49 | Kobaz | but we are |
| 18:00.08 | Kobaz | Banad: did your cat find anything? |
| 18:00.24 | [TK]D-Fender | Kobaz: Probably his mouse :) |
| 18:00.27 | Kobaz | heh |
| 18:01.08 | Banad | nothing yet |
| 18:01.16 | Kobaz | http://pics.nase-bohren.de/friends.jpg/1278540019 |
| 18:01.21 | evilbit | Apr 23rd, huh |
| 18:01.26 | Kobaz | Banad: cat /etc/*version* |
| 18:03.14 | telnettech | sorry wasnt able to help sr |
| 18:03.17 | Banad | cat: can't open '/etc/*version*': No such file or directory |
| 18:04.00 | fauxalliance | ughh, 'uname -a' |
| 18:04.09 | Kobaz | Banad: you do not have a normal linux system... we can't help you |
| 18:04.17 | Banad | :( |
| 18:04.27 | Kobaz | what did you install? |
| 18:04.28 | fauxalliance | Kobaz, define 'normal' GNU/Linux system? |
| 18:04.43 | Kobaz | fauxalliance: debian/redhat/ubuntu/etc you know.. the major players |
| 18:05.18 | fauxalliance | normal linux = major distribution... all clear |
| 18:06.04 | Kobaz | normal as in like... doesn't hide the kernel version... includes things like man... you know... the usual |
| 18:06.14 | fauxalliance | Banad, uname -a perhaps will enlighten us a little about your system... |
| 18:06.20 | Kobaz | fauxalliance: he did |
| 18:06.26 | Kobaz | Linux Fri Apr 23 03:16:12 EDT 2010 ppc unknown |
| 18:06.32 | fauxalliance | sounds like SCO System V |
| 18:06.32 | Kobaz | unknown kernel on a powerpc |
| 18:06.41 | Kobaz | fauxalliance: haha sco |
| 18:06.56 | fauxalliance | oh yea.. /bin/ls works... not sco. |
| 18:07.09 | Banad | i just need to run a cron job |
| 18:07.18 | Kobaz | Banad: do you even have an editor? |
| 18:07.23 | Kobaz | type nano |
| 18:07.24 | fauxalliance | read the man page for cron online? |
| 18:07.46 | fauxalliance | type whoami |
| 18:07.55 | Kobaz | yeah, you're probably not even root |
| 18:08.10 | Banad | i am root |
| 18:08.14 | Banad | i have nano too |
| 18:08.20 | fauxalliance | type 'bash' |
| 18:08.22 | Kobaz | nano /etc/crontab |
| 18:08.23 | Kobaz | have fun |
| 18:08.53 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 18:09.36 | Kobaz | Banad: it's going to be *very* painful for you to learn linux this way... you're probably missing more than half the tools that you would normally have |
| 18:09.39 | *** join/#asterisk pabelanger-lap (~pabelange@207.236.117.2) |
| 18:10.00 | fauxalliance | yep... GNU puts the 'utilities' in GNU/Linux |
| 18:10.51 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 18:12.31 | fauxalliance | Kobaz, probably more painful than playing 'twinkle twinkle little star' on vuvuzela... |
| 18:13.17 | grinder13 | hi! I 've used this guides for SIPS/SRTP: http://www.remiphilippe.fr/2010/05/30/sips-on-asterisk-sip-security-with-tls/ , http://www.remiphilippe.fr/2010/06/04/asterisk-srtp-installation-and-configuration/ my confs for my 2 servers: http://pastebin.com/8PNzQPyL , http://pastebin.com/P2Wvu0Nk and the log: http://pastebin.com/u19C4QJS call is not getting through from ServerA-->ServerB. any hints? |
| 18:15.40 | Banad | :) |
| 18:15.43 | Banad | thanks anyway |
| 18:25.57 | decklar | anyone know the DAHDI details for ISDN2a/ NT / B410p setup in South Africa? |
| 18:29.44 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
| 18:30.38 | sol | TLA FTW. |
| 18:32.37 | decklar | Anyone here setup an asteriskNOW box in SA recently? Using DAHDI/B410p over a NT2a? |
| 18:33.39 | p3nguin | San Antonio? |
| 18:33.44 | [TK]D-Fender | Picked by a man with one leg? |
| 18:33.52 | [TK]D-Fender | p3nguin: South America <- |
| 18:33.55 | p3nguin | oh |
| 18:34.24 | [TK]D-Fender | p3nguin: You fell for that? :0 |
| 18:34.44 | [TK]D-Fender | [14:25]<decklar>anyone know the DAHDI details for ISDN2a/ NT / B410p setup in South Africa? <- I'll give you a clue ;) |
| 18:35.00 | p3nguin | I didn't bother reading any previous messages. |
| 18:37.28 | *** join/#asterisk matahou_ (566afda1@gateway/web/freenode/ip.86.106.253.161) |
| 18:38.08 | *** join/#asterisk matagou_ (566afda1@gateway/web/freenode/ip.86.106.253.161) |
| 18:38.37 | sol | [Jul 13 10:33:27] NOTICE[2839] chan_sip.c: Registration from '"walters" <sip:walters@10.12.10.109>' failed for '10.12.10.110' - No matching peer found |
| 18:38.39 | sol | this is fun :) |
| 18:41.00 | raden | how secure is a SIP to SIP call ? |
| 18:41.20 | [TK]D-Fender | raden: Depends on what part you're securing |
| 18:41.22 | raden | both devices registered to same asterisk server |
| 18:41.30 | [TK]D-Fender | raden: And the devices. |
| 18:41.35 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 18:41.36 | dohd | and the networks? |
| 18:41.45 | [TK]D-Fender | yup |
| 18:41.48 | raden | aastra 9133i and pap2t |
| 18:41.56 | raden | over wan |
| 18:42.10 | dohd | wan is not secure in theory |
| 18:42.12 | [TK]D-Fender | raden: Actual conversation is pretty certainly wide open |
| 18:42.14 | dohd | in practice you might live |
| 18:42.24 | dohd | but if it's important if's something to keep in mind |
| 18:42.28 | dohd | it's not encrypted |
| 18:42.29 | raden | what would have to be done to secure the line ? |
| 18:42.38 | dohd | e.g. use vpn |
| 18:42.40 | [TK]D-Fender | raden: VPN |
| 18:42.48 | Banad | how to check the registration of the trunks ? |
| 18:42.48 | dohd | the regular 'data over untrusted media' stuff |
| 18:43.30 | dohd | or use iax if possible, that has encryption |
| 18:43.40 | [TK]D-Fender | Banad: sip show registry |
| 18:43.40 | raden | so AASTRA 9133I <-> DDWRT w/ open vpn <-> Asterisk server with open VPN SERVER ? |
| 18:43.41 | dohd | banad: sip show peers? |
| 18:43.47 | raden | something like that ? |
| 18:43.54 | dohd | raden: it depends |
| 18:44.12 | dohd | it starts with identifying the risks |
| 18:44.22 | dohd | and putting a value to those risks |
| 18:44.43 | Banad | <[TK]D-Fender> Banad: sip show registry --> I mean the reason why a trunk it doesnt connect |
| 18:44.48 | dohd | which parts of the infrastructure are vulnerable |
| 18:44.54 | dohd | what is your risk model |
| 18:45.06 | [TK]D-Fender | Banad: Who doesn't connect to who? |
| 18:45.20 | Banad | my trunk to the provider |
| 18:45.28 | Banad | it shows like auth sent |
| 18:45.35 | [TK]D-Fender | Banad: .....FFS who is calling who? |
| 18:45.37 | dohd | You will spend way more money on securing a secret multibilliondollarcompany meeting, than a call to your mother |
| 18:45.41 | Banad | i called and they told me everything is fine on their end |
| 18:45.48 | [TK]D-Fender | Banad: Look at the SIP DEBUG of your register attempts <- |
| 18:46.00 | [TK]D-Fender | Banad: PASTEBIN is your friend |
| 18:46.02 | [TK]D-Fender | ~pb |
| 18:46.03 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:46.32 | dohd | and who are your 'enemies'? Government? Competition? Law enforcement? |
| 18:46.35 | Banad | remember yesterday TK we talked about it |
| 18:47.07 | [TK]D-Fender | Banad: No. Pastebin it. |
| 18:47.10 | dohd | ok, <- back to work |
| 18:49.51 | decklar | who needs a challenge? :P |
| 18:50.48 | decklar | It's 8:49pm in South Africa and I need to get this working by tomorrow. I've posted all my conf's here: http://mybroadband.co.za/vb/showthread.php/249482-B410p-DAHDI-AsteriskNOW-1.7 |
| 18:52.59 | decklar | I'm struggling to understand where to put the relevent information in chan_dahdi etc - Except for the most important two functions dialling in or out everything is working :P |
| 18:53.12 | evilbit | lol |
| 18:53.36 | decklar | and i'm on my last heineken in this freezing cold office :P |
| 18:54.03 | Banad | Ok here it is TK --> http://pastebin.com/vwFVap47 |
| 18:54.21 | Banad | but right now its connected , it gets disconnected during the day.... |
| 18:54.32 | Banad | i have to reload sip to get connected again.... |
| 18:54.44 | *** join/#asterisk tc0nn (~tc0nn@preston.farecompare.com) |
| 18:56.14 | sol | I can't seem to get my phone to authenticate |
| 18:56.29 | sol | I even set *all* the names |
| 18:56.34 | sol | and the password field, to the same value |
| 18:56.50 | wcselby | p3nguin - not sure if you even care, but it seems the fix for the STRFTIME in the globals section was replacing the commas with pipes. |
| 18:57.09 | [TK]D-Fender | Banad: I do recall this now... not sure.. could be the remote side. |
| 18:57.13 | *** join/#asterisk eye-scuzzy (~light@sun28.ipfw.su) |
| 18:57.20 | tc0nn | sol - sip debug, assuming its sip... |
| 18:57.24 | [TK]D-Fender | Banad: Or random network conditions. |
| 18:57.35 | sol | yeah gonna try now |
| 18:58.07 | Banad | random network conditions ? |
| 18:58.25 | Banad | <[TK]D-Fender> Banad: I do recall this now... not sure.. could be the remote side. --> Called them and said everything is ok ... |
| 18:58.40 | tc0nn | sol - you using fromuser fromdomain or any of those? Keep the sip config simple to start |
| 18:58.46 | Banad | what do you mean random network conditions thats the only trunk that gets disconnected |
| 18:58.57 | Banad | during the day.... |
| 18:59.11 | sol | tc0nn; it's extremely simple, I just started, so it could be anything, could be a server misconfig |
| 19:00.57 | *** join/#asterisk jkroon (~jkroon@dsl-244-0-196.telkomadsl.co.za) |
| 19:02.08 | [TK]D-Fender | Banad: Check that your externip is still the same. Check your routing, firewalls, etc |
| 19:02.16 | telnettech | decklar.... your using Freepbx....there is a channel for it....they should be able to help you |
| 19:02.48 | sol | I'm using trixbox, and totally regretting it. Looks just like they wrapped freepbx in linux |
| 19:02.53 | sol | yeah, thanks. very helpful. |
| 19:02.55 | sol | hehe |
| 19:03.11 | decklar | Ok, thanks telnettech |
| 19:03.12 | Nugget | telnet is eeeeeeevil! |
| 19:03.16 | decklar | I'll ask there |
| 19:03.19 | rustyclarkson | When a parked call times out, is it possible to change the values put into the Dial command? (identifier/timeout/etc) |
| 19:03.34 | [TK]D-Fender | sol: "wrapped freepbx in linux" <- PARDON? |
| 19:03.49 | [TK]D-Fender | rusWhat dial command? |
| 19:04.03 | sol | [TK]D-Fender; looks like they installed linux, asterisk, freepbx and then ISO'd it |
| 19:04.17 | rustyclarkson | Executing [SIP/10023@park-dial:1] Dial("SIP/10018-100c7f00", "SIP/10023|30|") in new stack |
| 19:04.18 | [TK]D-Fender | sol: DUH |
| 19:04.36 | sol | [TK]D-Fender; I know that's what they would have to do. It's just that it seems that's ALL they did. |
| 19:04.38 | [TK]D-Fender | rustyclarkson: that is YOUR DIALPLAN being executed |
| 19:04.47 | [TK]D-Fender | rustyclarkson: Go fix your own extensions.conf |
| 19:04.49 | p3nguin | wcselby: Which branch/version are you using? I tested it on 1.4. |
| 19:04.50 | rustyclarkson | negatory, its default asterisk |
| 19:05.03 | [TK]D-Fender | rusyThere is no such thing as "default" |
| 19:05.08 | Naikrovek | can iax2 be encrypted within asterisk itself? create a secure trunk? |
| 19:05.10 | sol | I should have installed from scratch |
| 19:05.15 | [TK]D-Fender | rustyclarkson: * does what you tell it to. |
| 19:05.20 | [TK]D-Fender | Naikrovek: Yes |
| 19:05.21 | rustyclarkson | k, ill try to force it |
| 19:05.28 | Naikrovek | [TK]D-Fender: thought so, thank you |
| 19:05.55 | [TK]D-Fender | rustyclarkson: Nothing to "force". "Do or do not. There is no try" </yoday> |
| 19:06.46 | [TK]D-Fender | sol: For all your regret sure don't see you looking at your probelm. Which really doesn't have Trixbox/FreePBX, etc to blame for this |
| 19:07.30 | sol | [TK]D-Fender; Of course not. Just that it wouldn't have made a difference. |
| 19:07.48 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:07.48 | [TK]D-Fender | sol: Of course not because looking never helps to identify the source of a problem. |
| 19:08.10 | sol | [TK]D-Fender; I'm saying of course trixbox/FreePBX is not to blame. |
| 19:08.16 | Banad | <[TK]D-Fender> Banad: Check that your externip is still the same. Check your routing, firewalls, etc ------> Using a Dyndns same network setup as before when it was working fine |
| 19:08.18 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:08.31 | sol | maybe if you'd look at what I'm typing? |
| 19:08.31 | sol | :) |
| 19:08.44 | [TK]D-Fender | Banad: Past is unimportant. Prove everything NOW. |
| 19:08.59 | [TK]D-Fender | sol: I did. It was open ended. I took door #2 |
| 19:09.19 | [TK]D-Fender | sol: So feel free to do something about your issue |
| 19:09.29 | sol | I am? |
| 19:09.42 | sol | You're very presumptuous. |
| 19:10.05 | *** join/#asterisk valajbeg (~hamo@b202c73.pptp-gw51.cable-internet.GlobalNET.ba) |
| 19:10.35 | Naikrovek | sol: that will get you exactly nowhere |
| 19:10.48 | Naikrovek | i'm not saying that to gang up on you |
| 19:10.58 | sol | What won't? |
| 19:11.00 | [TK]D-Fender | sol: You're right. I presume you actually want to solve your issue as opposed to lamenting your installation method :) |
| 19:11.21 | sol | [TK]D-Fender; I am working on my issue while lamenting my installation method here :p |
| 19:11.28 | OppieT30 | I am using AsteriskNow. How do I change the default password for the freepbx user? |
| 19:11.35 | [TK]D-Fender | sol: Multi-tasking too... nice |
| 19:11.36 | Naikrovek | OppieT30: ask in #asterisknow |
| 19:11.41 | Banad | TK ..... is this ok ? --- > http://pastebin.com/YL4hn4xn |
| 19:11.46 | OppieT30 | Ok |
| 19:11.47 | [TK]D-Fender | OppieT30: Told you before ----> #freepbx |
| 19:11.51 | [TK]D-Fender | OppieT30: NOT SUPPORTED HERE |
| 19:11.52 | Naikrovek | or #freepbx |
| 19:12.58 | [TK]D-Fender | Banad: Looks fine from here... I say a WIN IP in your reg string. Just go verify that it is currently accurate. |
| 19:13.13 | [TK]D-Fender | Banad: and then check your firewalls / routing |
| 19:13.26 | Banad | what about this http://pastebin.com/8v3HkMd8 |
| 19:13.40 | Banad | <PROTECTED> |
| 19:13.57 | [TK]D-Fender | Banad: externhost=dynds <--- is this exactly what you put? |
| 19:14.00 | *** part/#asterisk OppieT30 (~root@173-26-156-159.client.mchsi.com) |
| 19:15.37 | Banad | no no there is my dynds name |
| 19:15.42 | Banad | whatever.ath.cx |
| 19:15.50 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 19:16.25 | [TK]D-Fender | Banad: then everything you've shown seems fine |
| 19:17.14 | Banad | <PROTECTED> |
| 19:18.23 | [TK]D-Fender | WAN* |
| 19:18.50 | [TK]D-Fender | Banad: Public. So your Dyndns seems to have resolved. Verify that it is still accurate. If it is then there are only firewalls left to check. |
| 19:19.55 | Banad | something is happening with that |
| 19:20.28 | [TK]D-Fender | Banad: meaning? |
| 19:21.18 | Banad | i just checked the log incoming outgoing and its empty |
| 19:21.21 | *** join/#asterisk bbkt-trix (~bbkt-trix@unaffiliated/bbkt-trix) |
| 19:21.27 | Banad | on the router... |
| 19:21.47 | dmz | hey i'm having problems doing blind transfers or any transfer to a wait queue; when i press # the system says "transfer" but no matter what # I hit it drops the caller into my voicemail; any hints? |
| 19:22.52 | *** part/#asterisk bbkt-trix (~bbkt-trix@unaffiliated/bbkt-trix) |
| 19:23.11 | p3nguin | My hint is to show us the dialplan and any other relevant configuration. |
| 19:23.16 | [TK]D-Fender | dmz: What model phone are you doing the transfer from? |
| 19:23.53 | dmz | i have a call that's going out to a cell phone (so I only have the # option) but i've also tested from my polycom; but from polycom I can do a real transfer so that works ok |
| 19:24.03 | *** join/#asterisk brezular (~brezular@adsl-dyn229.78-98-58.t-com.sk) |
| 19:24.21 | [TK]D-Fender | dmz: Could be the dtmf is flakey from your cell. Try calling to a landline instead |
| 19:24.33 | [TK]D-Fender | dmz: if that works better you'll have your culprit |
| 19:24.39 | dmz | tried that and tried different cell phones |
| 19:25.17 | dmz | any # I hit after I hit # (& it says "transfer" so it recognizes me hitting #) transfers to voicemail; and i thought it wanted a 2 digit entry so i must have misconfigured something somewhere |
| 19:26.59 | [TK]D-Fender | dmz: What did you set the transferconext to? |
| 19:27.41 | dmz | parkedcalls |
| 19:29.12 | p3nguin | Where does transfercontext go? I don't have it in sip.conf nor features.conf samples. |
| 19:30.10 | dmz | should be in features but it's not transfercontext it's "context" (unless there is one i'm missing" |
| 19:32.13 | p3nguin | That's not for transfers, though, it's only for parking. |
| 19:32.49 | dmz | well parking isn't working :) i can do *2 to do attended transfer |
| 19:33.11 | dmz | but i want to be able to park it so parking isn't working, sorry if i had wrong term |
| 19:33.33 | *** join/#asterisk mphill (~mphill@12.239.164.34) |
| 19:34.01 | p3nguin | Parking does use transfer, but transfers could work and parking still not work. |
| 19:34.10 | mphill | does asterisk 1.6 have timing issues inside virtualbox? |
| 19:34.34 | rustyclarkson | [TK]D-Fender: I've come up with a better question. When a parked call times out, Asterisk does a "   -- Added extension 'SIP/10023' priority 1 to park-dial" which will overwrite any of my more general "exten => _.,..." I create in park-dial. Therefore, is it possible to specify an extension of the form SIP/10023 in a context as "dialplan add extension SIP/10023,1,Playback(blue-eyed-polar-bear) into park-dial" does a " -- Added extension 'SIP |
| 19:35.38 | [TK]D-Fender | .... WTF |
| 19:35.47 | [TK]D-Fender | SIP/10023 is NOT an "extension". |
| 19:36.07 | [TK]D-Fender | rustyclarkson: What have you done to end up in this state |
| 19:36.30 | [TK]D-Fender | rustyclarkson: And you should not be messing with the parking context |
| 19:37.09 | dmz | well i have attended & unattended transfers working, i just had to fix the feature set buttons; now if I could figure out why parking isn't working ... |
| 19:38.33 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 19:43.19 | rustyclarkson | [TK]D-Fender: for parked calls there is a timeout attribute, parkingtime. (which i'm sure you know about) When a call times out, that is the command that asterisk does: "   -- Added extension 'SIP/10023' priority 1 to park-dial". Asterisk dynamically modifies the dialplan of that context and then ends up executing that line. I have no way of overwriting that line it adds dynamically as far as I can see. |
| 19:44.09 | [TK]D-Fender | rustyclarkson: pastebin youre features.conf, and the full CLI of a call attempt parking it. |
| 19:44.18 | *** join/#asterisk decklar (~ross.inne@196.31.81.190) |
| 19:44.19 | rustyclarkson | k |
| 19:46.50 | *** join/#asterisk Intel`` (~DND@ner-as26608.alshamil.net.ae) |
| 19:47.13 | Intel`` | guys need help i already configured my pri card but whenever i try to call externally i get: Channel 0/1, span 1 got hangup request, cause 63 |
| 19:49.24 | WIMPy | {0x3F, "Service or option not available, unspecified"}, |
| 19:51.15 | *** join/#asterisk AndyML (~alauppe@pool-173-49-137-72.phlapa.fios.verizon.net) |
| 19:52.57 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:53.05 | AndyML | check the ping-time limiters... |
| 19:53.09 | AndyML | wrong channels |
| 19:54.56 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.89.20) |
| 19:55.45 | AndyML | i'm sorry - what I wanted to consult with this channel about is a TDM800P w echo canceller. I have one implemented but all calls start with about 10 seconds of unmanagable echo that eventually gets cancelled out. |
| 19:55.57 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:56.29 | Intel`` | AndyML been there. i dont recommend analogs |
| 19:56.38 | Intel`` | which everyone agrees if im not mistaken |
| 19:56.43 | [TK]D-Fender | AndyML: disable echotraining |
| 19:56.52 | AndyML | an excerpt from my /etc/dahdi/system.conf follows - it is invoking the mg2 echo-canceller, which I was hoping someone could confirm is right |
| 19:56.55 | AndyML | fxsks=1 |
| 19:56.58 | AndyML | echocanceller=mg2,1 |
| 19:57.11 | [TK]D-Fender | AndyML: system.conf = irrelevent |
| 19:57.19 | [TK]D-Fender | AndyML: chan_dahdi is what counts here. |
| 19:57.20 | AndyML | thank [TK]D-Fender. noted. |
| 19:57.26 | [TK]D-Fender | AndyML: make sure ET is off |
| 19:57.46 | AndyML | [TK]D-Fender: chan_dahdi has "echocancel = yes |
| 19:57.49 | AndyML | " for every channel. |
| 19:57.52 | AndyML | i'll turn off training. |
| 19:58.37 | rustyclarkson | [TK]D-Fender: http://pastebin.com/dXXqWYRp Thanks |
| 19:59.13 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 19:59.20 | AndyML | interesting - it says that echotraining parameters do not apply to hardware echo cancellers. |
| 19:59.32 | AndyML | it = the chan_dahdi.conf sample. |
| 20:00.12 | [TK]D-Fender | hrm |
| 20:01.24 | AndyML | the reason i asked about mg2, is that 'dahdi show channel 1' shows echo cancellation set at 128 taps, but the hardware ec should be 1024. |
| 20:01.32 | AndyML | it made me think i wasn't using the hardware module. |
| 20:01.41 | [TK]D-Fender | rustyclarkson: I do not see anything looking vagule like what you pasted here earlier in there. |
| 20:02.00 | rustyclarkson | line 48 |
| 20:02.26 | rustyclarkson | which leads to line 51 |
| 20:02.28 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 20:02.40 | [TK]D-Fender | waitasec... |
| 20:02.44 | saxa | hi, anybody knows what this means ? http://pastebin.com/NBxVNu8u |
| 20:02.46 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:03.17 | [TK]D-Fender | <PROTECTED> |
| 20:03.23 | [TK]D-Fender | TRhis is some whacked shit |
| 20:03.26 | Intel`` | saxa the latency digium told me its normal |
| 20:03.37 | Intel`` | i dont know the last line |
| 20:05.19 | [TK]D-Fender | rustyclarkson: I'm wondering if this is an incomplete attended transfer <----------- |
| 20:05.27 | rustyclarkson | hmm |
| 20:05.30 | rustyclarkson | i can see that |
| 20:05.33 | [TK]D-Fender | rustyclarkson: Which is what it looks like |
| 20:05.42 | [TK]D-Fender | rustyclarkson: You have to complete the hand-off |
| 20:06.08 | rustyclarkson | k, ill go looking down that path |
| 20:06.13 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:06.13 | rustyclarkson | thanks for your help [TK]D-Fender |
| 20:06.36 | saxa | Intel``: I had some other kind of error also before I upgraded from dahdi 2.3.0 to 2.3.0.1 |
| 20:06.55 | saxa | let me find it, and this kind of error made freeze my machine |
| 20:07.53 | saxa | oh actualy its the one on the second line Intel`` |
| 20:08.36 | saxa | so, by the way the dahdi known as to stable to be used in production machines is which version ? |
| 20:10.28 | Intel`` | the latest one can be used in production |
| 20:10.41 | Intel`` | im using 2.3.0.1 in our office |
| 20:11.14 | Intel`` | i did not encounter your problem. are you using asterisknow or trixbox? |
| 20:13.47 | *** join/#asterisk Intel`` (~DND@ner-as26608.alshamil.net.ae) |
| 20:15.18 | *** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl) |
| 20:16.06 | saxa | Intel``: i'm using the asterisk compilled from source, 1.6.2.8 |
| 20:16.19 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:16.22 | [sr] | hi again :) |
| 20:18.27 | tc0nn | asterisk support was suggesting dahdi-linux-complete-2.3.0.1+2.3.0 a month ago.. assume its the same. |
| 20:22.48 | Intel`` | guys anyone explain what cause 63 means? |
| 20:23.28 | [TK]D-Fender | Intel``: You've been told under various nicks as to what it is, now ask your CARRIER |
| 20:23.49 | WIMPy | Intel``: Search for my previous line. |
| 20:24.13 | *** join/#asterisk wierdo (~jimmy@77.78.3.197) |
| 20:24.15 | tc0nn | Cause 63 means your line isn't configured right so the call handoff failed |
| 20:24.20 | tc0nn | in lamen's terms |
| 20:29.20 | Intel`` | so its the carrier's fault? |
| 20:30.06 | *** join/#asterisk gloin (me@unaffiliated/gloin) |
| 20:30.27 | tc0nn | not necessarily, just means your line and the carriers switch aren't matched. |
| 20:30.51 | gloin | hrm |
| 20:30.54 | *** part/#asterisk jkroon (~jkroon@dsl-244-0-196.telkomadsl.co.za) |
| 20:31.04 | gloin | what's the correct dtmf mode for recognizing keypresses from cellphones? |
| 20:32.35 | gloin | I've got a sip trunk outbound dialing, dtmf is recognized just fine when calling an extension on the same PBX as the SIP trunk, but when it calls a cellphone, you can press keys until the cows come home and * doesn't detect them |
| 20:33.26 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 20:34.08 | Naikrovek | gloin: RFC2833 works for me |
| 20:34.20 | Naikrovek | avoid in-band if possible |
| 20:34.28 | WIMPy | BTW: Who hat the highly confusing idea to dispaly cause codes in decimal? |
| 20:34.39 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 20:34.41 | gloin | hm |
| 20:34.51 | Intel`` | is there any span config of crs and hdb3? i know hdb3 but crs? |
| 20:35.20 | Intel`` | telco is no help. hmmm |
| 20:35.37 | Naikrovek | that's the telco initial response to everything |
| 20:35.50 | Naikrovek | "can i have phone service?" "who do you think we are, a phone company" |
| 20:35.52 | gloin | heh |
| 20:35.54 | Naikrovek | then you have to call back |
| 20:35.59 | gloin | and wait on hold |
| 20:36.21 | Intel`` | forever |
| 20:36.25 | gloin | their dime |
| 20:36.30 | p3nguin | your time |
| 20:36.36 | gloin | exactly |
| 20:36.43 | Naikrovek | my rhyme? |
| 20:36.45 | Naikrovek | no YOUR rhyme |
| 20:36.51 | gloin | heh |
| 20:36.54 | gloin | well |
| 20:37.08 | gloin | rfc2833 doesn't work on my verizon blackberry |
| 20:37.14 | gloin | signal doesn't get passed at all |
| 20:37.32 | gloin | bleepity |
| 20:37.34 | p3nguin | Verizon seems to have issues with DTMF and Asterisk. |
| 20:37.34 | gloin | bleep |
| 20:37.39 | [TK]D-Fender | checkout time, BBL |
| 20:37.50 | gloin | p3nguin: known workaround? |
| 20:38.00 | gloin | assume leaving verizon isn't an option |
| 20:38.14 | gloin | waves to the 2-inch-thick corporate Verizon bill for this month |
| 20:38.28 | p3nguin | I can call my cell voicemail, and only a small percent of the time can I retrieve my messages. |
| 20:40.20 | p3nguin | I just tried it again, and it accepted my password on the FIRST TRY! I must be living in a parallel universe. |
| 20:40.21 | gloin | well, that's odd |
| 20:40.32 | gloin | inband works, rfc2833 fails |
| 20:40.43 | Intel`` | btw aside from switchtype,signalling,ccs and hdb3, what do i need to know more from the telco? |
| 20:40.56 | gloin | will just go ahead and use inband despite Naikrovek's well-intentioned suggestion |
| 20:41.38 | p3nguin | Usually it ignores me completely, and suggests that maybe I need to enter in another mailbox number, ultimately apologizing for my having trouble and hanging up. |
| 20:42.08 | [sr] | hi WIMPy |
| 20:43.28 | p3nguin | Inband should usually be a last resort. If all other methods fail, you've earned it. |
| 20:44.06 | wierdo | gloin, mostly inband is setuped by providers who use G711, and RFC2833 for G729 for example |
| 20:45.05 | gloin | wierdo: I wonder if auto would work |
| 20:45.17 | p3nguin | Yeah, you cannot use inband if you don't use a 64 kbit codec. |
| 20:46.14 | gloin | ulaw or alaw |
| 20:46.17 | wierdo | gloin, don't know how asterisk setup on "auto" selects the dtmf method, maybe it will detect it correct |
| 20:46.19 | gloin | what's the objection to inband though? |
| 20:46.40 | p3nguin | It plays tones in the voice stream. |
| 20:46.43 | Naikrovek | the cell phone will send the DTMF to its provider out-of-band, and it will be reliably transmitted over the wire out of band until it gets to your provider which will then send it in band |
| 20:47.09 | gloin | hm |
| 20:47.10 | [sr] | WIMPy: do you know any adapter that could transform a siemens digital extension to an ISDN line? |
| 20:47.15 | gloin | why is this a bad thing? |
| 20:47.17 | Naikrovek | so if you run anything that isn't G711 or G722 or something crystal freakin' clear, the tones will get mangled and become unrecognizable |
| 20:47.30 | gloin | ah |
| 20:47.33 | Naikrovek | it's bad because it's failure prone |
| 20:47.45 | wierdo | ...like in G729 |
| 20:47.47 | Naikrovek | yes |
| 20:48.00 | Naikrovek | telephony compression is optimized for voice, and nothing else |
| 20:48.04 | WIMPy | [sr]: What way round? |
| 20:48.07 | Naikrovek | anything that isn't voice will become garble |
| 20:48.16 | gloin | rfc2833 was guaranteed fail |
| 20:48.21 | [sr] | WIMPy: PBX digital line => asterisk |
| 20:48.23 | gloin | as in it never worked |
| 20:48.32 | gloin | "auto" seems to work though |
| 20:48.35 | [sr] | WIMPy: BX digital line => asterisk (NT or TE mode) |
| 20:48.37 | Naikrovek | if you want a tone to make it across, don't compress the audio |
| 20:48.44 | WIMPy | [sr]: But if it's digital and not IP, it's most likely ISDN. |
| 20:48.54 | gloin | allow=ulaw |
| 20:48.55 | gloin | allow=alaw |
| 20:49.00 | gloin | that's all, folks |
| 20:49.04 | p3nguin | You shouldn't be allowing two codecs. |
| 20:49.08 | [sr] | WIMPy: it's a only one pair cable :( |
| 20:49.10 | p3nguin | Pick the one you want to use. |
| 20:49.18 | WIMPy | [sr]: Hipath? |
| 20:49.28 | p3nguin | disallowing all first. |
| 20:49.31 | [sr] | WIMPy: siemens HICOM 100 |
| 20:49.44 | WIMPy | [sr]: That's kust another physical interface, Up0. |
| 20:49.56 | [sr] | kust? |
| 20:50.07 | p3nguin | typos happem |
| 20:50.09 | WIMPy | [sr]: Don't know the small ones, but it's probably the same. |
| 20:50.15 | gloin | well, this seems to work now |
| 20:50.16 | WIMPy | just |
| 20:50.18 | gloin | heh |
| 20:50.25 | [sr] | p3nguin: :P |
| 20:50.44 | gloin | Asterisk => SIP/ShoreTel => PRI (yes, it's godawful ugly) |
| 20:50.49 | WIMPy | [sr]: You can get both standalone adapters as well as addon adapters fot te Up0 Phones to S0. |
| 20:51.32 | WIMPy | Still too warm to type :-( |
| 20:51.45 | [sr] | WIMPy: but this S0 won'«t be a only one pair cable right? |
| 20:52.11 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 20:52.12 | [sr] | WIMPy: i ask this 'cause i have a S0 extension with only one pair cable that i used to use to connect to the net using ISDN modems |
| 20:52.47 | WIMPy | S0 is four wires, Uk0/Up0 is two wires. |
| 20:53.18 | [sr] | hum...i'm remembering now..you're right sorry my error |
| 20:53.30 | WIMPy | I'm not sure if a "normal" NT whould work on a Siemens PBX. |
| 20:53.35 | *** join/#asterisk Godfather_ (~Godfather@35.Red-88-7-6.staticIP.rima-tde.net) |
| 20:53.52 | [sr] | in this case this has to be NT mode with S0, right? |
| 20:54.18 | WIMPy | The Siemens whould be in NT mode. |
| 20:54.37 | WIMPy | You can't use Up0 as uplink AFAIK. |
| 20:55.01 | [sr] | hum going to study that |
| 20:55.30 | [sr] | i cut'ed the wired in my S0 jack... when i changed company instalations... grrrrrrr |
| 20:55.40 | [sr] | thats gonna be my worst problem to find them |
| 20:57.37 | WIMPy | [sr]: If they supply phantom power, the pairs are easy to identify. And the two wires within a pair only needs to have the same polarity on all connected devices. |
| 20:58.41 | [sr] | WIMPy: hum cool, the cable that goes out the pbx is a 20 pair cable, 'n i have almost of them in use so i guess/hope it'll be easy |
| 20:59.49 | WIMPy | [sr]: Pray that whoever connected them got the colour coding right. |
| 20:59.55 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 20:59.57 | vader-- | hello |
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| 21:00.30 | [sr] | WIMPy: ops... have no clue.. but tomorow i'll know 'n i'll let you know :) |
| 21:00.41 | [sr] | thanks for the today lesson!! |
| 21:00.52 | vader-- | Alittle off topic but if i had get audio from a phone call/line to an audio mixing board and asterisk is my PBX what are some options i have? |
| 21:01.17 | vader-- | I have IP Phones that are available to me |
| 21:01.34 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 21:02.04 | vader-- | i need two phone lines, so i was thinking maybe two cisco ip phones and tap the headset line with a line tap from like radioshack? |
| 21:02.45 | nova911 | call recording using monitor not working for asterisk 1.6 well as working with asterisk 1.4 |
| 21:03.22 | kieppie | hi guys. I'm new to Asterisk, although I've been toying with VoIP in general for a bit now (mostly FS). I'm running/testing Askozia at the moment & must say, I really like it. |
| 21:03.23 | kieppie | I've set it up & been able to make outgoing calls OK, but having issues with incoming calls: get a asterisk console message of: "Using SIP RTP CoS mark 5" & "Using SIP VRTP CoS mark 6". |
| 21:03.23 | kieppie | any ideas, please? |
| 21:03.32 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:05.42 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 21:06.20 | [sr] | how much BW does a SIP call with video takes? |
| 21:06.27 | [sr] | 20, 30, 40kb/sec? |
| 21:08.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:10.26 | *** join/#asterisk astrutt (astrutt@pinky.ratman.org) |
| 21:12.00 | FILLVAIO3 | Guys, please help, > handle_request_invite: Call from '79157723456' to extension 'avcafe' rejected because extension not found. |
| 21:12.24 | KavanS | echo -e "SET VARIABLE DIALSTATUS ANSWER" <<< --- proper syntax for agi sh script to set a variable? |
| 21:12.52 | FILLVAIO3 | i have tryed extension named in extension.conf and sip.conf and no results :( |
| 21:16.26 | [TK]D-Fender | FILLVAIO3: You clearly don't have an exten to match that in the context that it is looking for it in. Go make one. |
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| 21:39.06 | [sr] | well |
| 21:39.09 | [sr] | im going to sleep |
| 21:39.14 | [sr] | you all be ok |
| 21:39.17 | [sr] | c ya |
| 21:39.23 | lost_soul | [sr]: g'ight |
| 21:39.26 | [sr] | merci |
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| 22:01.19 | Mish- | I have a SPA3000 hooked in to Asterisk, inbound calls to the PSTN number from outside work 50% of the time, other times I get an error in Asterisk, here is the error and the relevent config, can someone take a look and give me suggestions: http://pastebin.com/VTX3k28K |
| 22:09.29 | p3nguin | You aren't matching the right peer, plus the peer is misconfigured. |
| 22:09.46 | Mish- | p3nguin: I really appreciate this. |
| 22:10.01 | Mish- | So, suggestions or pointers on how to correct? |
| 22:11.20 | p3nguin | In your PSTN tab, which user name did you type in? |
| 22:11.59 | Mish- | "PSTN Line" tab username is: asterisk |
| 22:12.29 | Mish- | Inbound and outbound are working this second, but half the time I get the error I outlined for inbound calls. |
| 22:12.31 | p3nguin | Then change it to pstn-spa3k |
| 22:12.55 | p3nguin | Actually, hold on a second. |
| 22:13.57 | p3nguin | Who/what uses 033896060? |
| 22:14.46 | p3nguin | Is the IP address of the device 192.168.2.10? |
| 22:14.59 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:15.08 | p3nguin | Make sure the Port setting on the PSTN Line tab is 5061. |
| 22:15.40 | Mish- | It is. "(S0<:033896060>)" is the "Dial Plan 2:" I use to forward inbound PSTN calls to Asterisk. |
| 22:16.09 | p3nguin | So Line 1 uses 033896060? |
| 22:16.21 | p3nguin | as the user name |
| 22:16.35 | Mish- | Line 1 uses 500 as the username. |
| 22:16.57 | p3nguin | I'm asking who or what uses the user name of 033896060. |
| 22:17.09 | p3nguin | You have a peer entry for it, and your device was matching it. |
| 22:18.19 | p3nguin | I'm asking questions to help you, but you aren't giving me answers. |
| 22:18.28 | p3nguin | I'm losing interest quickly. |
| 22:20.53 | p3nguin | Your device needs only two peer entries: one for Line 1 on port 5060 and one for PSTN Line on port 5061. |
| 22:21.28 | p3nguin | Definitions of type=peer do not use the "username=" field, so stop using it. |
| 22:22.08 | Mish- | Right, makes sense, I'll go back and sort this out, thanks. |
| 22:23.07 | p3nguin | http://pastebin.com/YgDTb6mx |
| 22:23.58 | p3nguin | You could also set both to host=dynamic and make the device send registrations. |
| 22:25.16 | p3nguin | It would also be a really good idea to specify a context for the device entries. |
| 22:25.32 | p3nguin | Otherwise they will be using a general (probably default) context. |
| 22:32.22 | gloin | anyone here handy enough with bash to see what I'm doing here and suggest a better (more asterisky) way to achieve the same goal? |
| 22:32.35 | gloin | http://pastebin.ca/1900120 |
| 22:32.52 | KavanS | echo -e "SET VARIABLE DIALSTATUS ANSWER" <<< --- proper syntax for agi sh script to set a variable? |
| 22:33.07 | KavanS | getting this nasty broken pipe message :( |
| 22:33.30 | Mish- | p3nguin: Excellent advice and now works, with one small change which you may be able to explain, if I don't put "username=asterisk" in the "[pstk-spa3k]" I can't make outbound PSTN calls and get "Failed to authenticate" in the logs. |
| 22:34.52 | p3nguin | If you've changed the username in PSTN Line to pstk-spa3k, that takes care of the username. If type=peer, the username= field should be completely ignored. |
| 22:35.31 | p3nguin | Are you using an analog phone on Line 1, or are you using an IP phone connected via Asterisk? |
| 22:35.51 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 22:38.54 | *** join/#asterisk blaines (~blaines@95.sub-75-244-165.myvzw.com) |
| 22:40.21 | Mish- | p3nguin: Weird, not that I'm complaining it's all working now. |
| 22:40.41 | Mish- | I'm using soft phones for inbound and outbound calls, as well as an analog phone on extension 500. |
| 22:40.47 | Mish- | All working now. |
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| 22:43.25 | tehrabbitt-1 | hi guys, quick question... is there a list somewhere of all the different sounds I can use when making IVR menus? |
| 22:44.18 | p3nguin | ls -l /var/lib/asterisk/sounds |
| 22:44.21 | tehrabbitt-1 | i mean yes, I can "ls" the directory where they are stored, but there's over 8000 of them and there's no easy way of finding the folers |
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| 22:44.40 | p3nguin | There are no "folers" in there. |
| 22:44.44 | p3nguin | Not even directories. |
| 22:44.59 | tehrabbitt-1 | lol there are directories p3nguin .. lol |
| 22:45.06 | tehrabbitt-1 | silence is a directory |
| 22:45.16 | tehrabbitt-1 | there's a few others too |
| 22:45.27 | lost_soul | tehrabbitt-1: http://www.enicomms.com/cutglassivr/ |
| 22:45.31 | lost_soul | maybe try that |
| 22:45.35 | p3nguin | Oh, right. I forgot about those. |
| 22:46.28 | p3nguin | digits, silence, letters, et cetera |
| 22:46.33 | p3nguin | find /var/lib/asterisk/sounds -type d |
| 22:46.49 | lost_soul | ah, nvm... doesn't appear to be what it says |
| 22:47.08 | tehrabbitt-1 | ah |
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| 22:48.30 | p3nguin | I like to use ls -l /var/lib/asterisk/sounds/*keyterm*.wav to see possible choices of sounds containing a specific key term. |
| 22:50.06 | tehrabbitt-1 | is there a sound "if you know the extension" or something like it? |
| 22:51.19 | p3nguin | if-u-know-ext-dial |
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| 23:29.41 | p3nguin | WTF, dude, do you have to leave us on auto-join when your shit is broken? 229 people (in this channel alone) shouldn't have to put you on ignore. |
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