| 00:01.01 | Jeekay | I have a dialplan to dial inoc-dba for me.. it is correctly sending the call to SIP/111@inoc-dba, but then says:chan_sip.c:1973 create_addr: No such host: inoc-dba] |
| 00:01.10 | [hC] | any of you guys had to debug network jitter on RTP on a lan? |
| 00:01.13 | [hC] | using ethereal, even? |
| 00:01.18 | Jeekay | but I have [inoc-dba] in my sip.conf, and 'sip show peers' shows the inoc host |
| 00:01.27 | [hC] | im seeing some interesting results but not sure what to do with them |
| 00:03.11 | justinu|laptop | ethereal jitter measurement freaks out if there's rfc2833 in the rtp stream |
| 00:03.20 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 00:09.30 | De_Mon | if dial fails with busy or other error, does asterisk go to the next priority or hang up? |
| 00:09.49 | De_Mon | the j option appears to make it jump to +101, so.. whats the default behavior? |
| 00:10.16 | benjk | busy is not an error |
| 00:10.42 | Jeekay | Anyone here on inoc? I would like to test my setup |
| 00:11.36 | benjk | you can use the g flag to let Dial() continue, then check DIALSTATUS when it returns |
| 00:11.50 | benjk | no need for jumping |
| 00:13.17 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:14.40 | aptura | ,NoOp,$(CALLERIDNAME) |
| 00:15.02 | aptura | that is under the wait(1) application yet it causes my incomming did call to fail. |
| 00:15.27 | aptura | err wait,1 application. |
| 00:16.05 | aptura | What would cause the incomming call to fail in a case like this. I comment it out renumber the priorities and the call goes though. |
| 00:17.55 | benjk | you will find that you need to provide detail if you want answers |
| 00:18.26 | De_Mon | so if Dial reutrns busy the channel is just hugup unless an option is specified. sounds good |
| 00:18.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:19.12 | *** join/#asterisk Nitrus^ (n=Nitrus_@72-34-76-86.skyriver.net) |
| 00:19.36 | benjk | depends, if you want to dial another extension in case it was busy, or go to voicemail etc tec, then obviously its no got |
| 00:19.41 | benjk | no good |
| 00:19.48 | benjk | also Asterisk is a liar |
| 00:20.01 | benjk | its not always busy when it claims its busy |
| 00:20.11 | hmmhesays | unless you have another priority in your dp after dial |
| 00:20.35 | benjk | in fact without looking at the detail, the console is almost never telling the truth |
| 00:20.53 | De_Mon | eh? |
| 00:21.15 | Nitrus^ | im having a problem implementing calling rollover. i want to have incoming calls to a certain extension then attempt to call the cellphone, and if there is no answer go to voicemail. i have s,1,Dial(extension) then s,2,Dial(cellphone), then s,3,Voicemail... but if the person's cellphone is off it will go straight to their voicemail on their cell phone i want to avoid connecting a call to the phone if it picks up without ringing first. is t |
| 00:21.24 | benjk | everything is always "nobody is available to take your call" |
| 00:21.42 | hmmhesays | Nitrus^: not much of a way around that |
| 00:21.47 | benjk | no matter what the real reason is |
| 00:21.49 | Nitrus^ | benjk: i got rid of my channel bridging problem |
| 00:22.05 | benjk | cool |
| 00:22.16 | Nitrus^ | got rid of vonage and FXO |
| 00:22.20 | benjk | heh |
| 00:22.29 | benjk | analog sucks |
| 00:22.37 | Nitrus^ | agreed |
| 00:23.18 | Nitrus^ | hmmhesays: is there a way to determine how long a dial command took to execute a call? |
| 00:23.24 | Nitrus^ | to get a connection that is |
| 00:23.25 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 00:25.11 | hmmhesays | not as far as I know |
| 00:25.59 | Jeekay | How do you get asterisk to re-read voicemail.conf? |
| 00:26.04 | Nitrus^ | reload |
| 00:26.18 | aptura | benjk heh yea thats the downside of vm..vm chacing. |
| 00:26.28 | Nitrus^ | or reload app_voicemail.so |
| 00:26.32 | Nitrus^ | if you want to just do that |
| 00:26.48 | Nitrus^ | alternatively hit reload <tab> <tab> for the list of autocompletes |
| 00:28.07 | Jeekay | I have a voicemail defined, [default] 100=>100,me,a@b.c |
| 00:28.11 | Jeekay | but I can't log in with 100/100 |
| 00:28.23 | Jeekay | the numbers coming up on the console are correct (password '100' for user '100') |
| 00:28.28 | Jeekay | any idea why ? |
| 00:28.31 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:29.02 | Nitrus^ | should work |
| 00:29.14 | Nitrus^ | ls /var/spool/asterisk/voicemail/default |
| 00:29.17 | Nitrus^ | make sure that box is there |
| 00:29.46 | Nitrus^ | i have spaces before and after the => i dont think that matters though ,i believe the lexer strips white space |
| 00:29.51 | Jeekay | it is |
| 00:29.56 | Jeekay | though there isnt a directory in there called 100 |
| 00:29.59 | Jeekay | there is one for 1234 |
| 00:30.05 | Jeekay | does it autocreate or do I need to do something manually? |
| 00:30.18 | Nitrus^ | it gets created when a voicemail gets left for that user |
| 00:30.26 | Jeekay | hm |
| 00:30.33 | Jeekay | could it be that i cant log in until a voicemail has been left? |
| 00:30.35 | Nitrus^ | or if you change options |
| 00:30.41 | Nitrus^ | but being that you cant even login you wont be able to do that |
| 00:30.45 | Jeekay | hm |
| 00:30.56 | Jeekay | my entry looks identical to the (working) 1234 |
| 00:31.05 | Jeekay | do I need any particular magic in sip.conf maybe? |
| 00:31.14 | Nitrus^ | do asterisk -vvvvr |
| 00:31.29 | Nitrus^ | then when you're in the CLI do reload app_voicemail.so |
| 00:31.42 | Nitrus^ | are you running * as root? |
| 00:31.44 | Jeekay | yeah |
| 00:31.49 | Jeekay | it just says 'parsing conf: found' |
| 00:31.57 | Jeekay | except with more words.. nothin to indicate a failure anywhere |
| 00:32.13 | Nitrus^ | whats your VoicemailMain params? |
| 00:32.19 | Jeekay | hmm.. my line for this extension in sip.conf says 'mailbox=100' |
| 00:32.25 | Jeekay | the other one says 'mailbox=1234@default' |
| 00:33.03 | Jeekay | didnt fix it |
| 00:33.15 | Nitrus^ | i believe without the @context default is implied |
| 00:33.24 | Jeekay | how bizarre. |
| 00:33.32 | Nitrus^ | but try that |
| 00:33.37 | Jeekay | just did |
| 00:33.48 | Nitrus^ | does it work? |
| 00:33.50 | Jeekay | nope |
| 00:34.03 | Nitrus^ | what does your VoicemailMain() command look like? |
| 00:34.16 | Nitrus^ | exten => x,1,VoicemailMain(); ? |
| 00:34.16 | Jeekay | exten => #65,1,VoicemailMain |
| 00:34.25 | Nitrus^ | try passing it @default |
| 00:34.31 | Nitrus^ | you said 1234 works though right/ |
| 00:34.35 | Jeekay | yeah |
| 00:34.41 | Jeekay | that may have been before some hack'n'slash though |
| 00:34.48 | Nitrus^ | strange... |
| 00:35.01 | Jeekay | hm |
| 00:35.12 | Jeekay | my voicemail line for this exten is x,2,Voicemail(x) |
| 00:35.16 | Jeekay | the old one was Voicemail(ux) |
| 00:35.30 | Jeekay | guess thats only for callers |
| 00:35.46 | Nitrus^ | ux sends them to your unavailable message |
| 00:35.49 | Jeekay | ah |
| 00:35.50 | Nitrus^ | bx sends the to your busy |
| 00:35.54 | luke-jr_ | ... |
| 00:36.01 | luke-jr_ | was Asterisk 1.4 really released 6 days ago? |
| 00:36.13 | Jeekay | this is my vm.conf line and console output: |
| 00:36.21 | Jeekay | 8192 => 8192,test,test@localhost |
| 00:36.25 | Jeekay | -- Incorrect password '8912' for user '8912' (context = default) |
| 00:36.46 | Nugget | I prefer Asterisk 1.π |
| 00:36.47 | Jeekay | oh |
| 00:36.51 | Jeekay | dear god i'm dumb |
| 00:37.20 | tmccrary | Are there any desktop phones that have bluetooth built in for headsets? |
| 00:37.24 | tmccrary | Sip based |
| 00:37.28 | Nitrus^ | does it work with the correct pass? |
| 00:37.29 | Nitrus^ | lol |
| 00:37.31 | Nitrus^ | and login |
| 00:37.36 | Jeekay | quite.. |
| 00:37.37 | Jeekay | sorry :) |
| 00:37.41 | Nitrus^ | no worries |
| 00:37.54 | Nitrus^ | that's why we have people to proof read |
| 00:38.46 | Jeekay | ah well. time for bed i think :) |
| 00:38.52 | Jeekay | dont suppose you're on inoc for a quick test? |
| 00:39.37 | Nitrus^ | neg |
| 00:39.45 | Jeekay | it was worth a try :) |
| 00:39.52 | Nitrus^ | yeah im at work |
| 00:39.55 | Nitrus^ | screwing around |
| 00:42.13 | *** join/#asterisk [shodan] (n=shodan@ip114.99-113-216.pppoe4.joliette.intermonde.net) |
| 00:45.07 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 00:59.54 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:05.00 | [hC] | son of a... my ip601 is claiming invalid app error after a reboot. |
| 01:07.43 | *** join/#asterisk Jason559-55 (n=Jason@adsl-75-10-110-241.dsl.frs2ca.sbcglobal.net) |
| 01:08.14 | Jason559-55 | Hi is anyone in here? |
| 01:08.33 | C6Vette | absolutly NOT... Not a person.... No one... :) |
| 01:08.48 | Jason559-55 | sorry, I'm using Xchat and it isn't showing me a list of people in the room. ;) |
| 01:09.04 | Jason559-55 | Oh, there it is, 279. ;) sorry |
| 01:09.58 | Jason559-55 | I'm pretty new to Asterisk, I was wondering if someone could recommend an FX0 (hope that's the correct term for a way to connect an Asterisk box to an analog line) unit... |
| 01:10.13 | Jason559-55 | I've been told to try a Sipura 3000 |
| 01:10.36 | De_Mon | Jason559-55 are you looking for a phone? |
| 01:10.45 | De_Mon | isn't sipura 3000 a phone? |
| 01:10.53 | intralanman | De_Mon: it's an ATA |
| 01:11.03 | Jason559-55 | no, I've found an IP phone. But I need a way to connect my asterisk box to my SBC phone line |
| 01:11.03 | intralanman | one fxo and one fxs |
| 01:11.11 | C6Vette | http://www.digium.com/en/products/hardware/tdm400p.php |
| 01:11.22 | intralanman | a 3000 could do it |
| 01:11.30 | C6Vette | yes a 300 would work |
| 01:11.41 | De_Mon | Thats, interesting. |
| 01:11.48 | intralanman | i'd go with pci for the asterisk box personally |
| 01:11.51 | C6Vette | and its cheaper. But its a separate unit. Which has its advantages/disadvantages |
| 01:11.53 | De_Mon | why would someone use an ATA over a tdm? |
| 01:11.57 | Jason559-55 | looked at a 400p, seemed pricy for what I needed, I only have one phone line |
| 01:12.10 | *** join/#asterisk ltd (n=z@202-161-18-11.dyn.iinet.net.au) |
| 01:12.15 | intralanman | you could use a (shhhhhhhhh) x100p |
| 01:12.23 | De_Mon | I use a x100p myse.. |
| 01:12.29 | intralanman | i do too |
| 01:12.38 | Jason559-55 | I'm unsure what a x100p is, who makes it? |
| 01:12.43 | intralanman | i haven't had any problems whatsoever with mine |
| 01:12.47 | C6Vette | I left mine in the computer i donated to good will |
| 01:12.51 | Jason559-55 | Also, an easy setup would be a plus. ;) |
| 01:12.53 | intralanman | oh goodness |
| 01:12.54 | intralanman | lol |
| 01:12.59 | De_Mon | Jason559-55 it's a generic card you can get for $8 which may, or may not work ;) |
| 01:13.02 | intralanman | it's pretty easy to set up |
| 01:13.14 | intralanman | that's very true |
| 01:13.17 | De_Mon | assuming the card is useable hehe |
| 01:13.26 | C6Vette | just search for x100p on ebay |
| 01:13.28 | De_Mon | so far I haven't any any problems using x100p's |
| 01:13.55 | intralanman | digitnetworks.com or ebay |
| 01:14.05 | intralanman | ebay should really be first i guess |
| 01:14.14 | intralanman | i dunno |
| 01:14.28 | intralanman | it's like searching for dinosaur fossils |
| 01:14.30 | intralanman | lol |
| 01:14.32 | Jason559-55 | Thanks for the referral. I know this isn't connected to Asterisk@home, but I assume it would be compatible with that as well? |
| 01:14.54 | intralanman | should be |
| 01:15.06 | intralanman | as long as they didn't screw asterisk up that bad when they made it |
| 01:15.28 | intralanman | s/it/@home/ |
| 01:15.38 | Jason559-55 | got it |
| 01:15.56 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 01:16.12 | Jason559-55 | Can anyone recommend a seller in California who sells these? (x100p), eager to get started... |
| 01:16.24 | C6Vette | I setup an @home server just to see how long it would take for someone to hackit... 4 hours and the pc was taken over. :) |
| 01:16.32 | Nugget | heh |
| 01:16.44 | intralanman | it took that long?!?!? where do you live? |
| 01:16.45 | intralanman | lol |
| 01:17.12 | Jason559-55 | The seller I found is in VA |
| 01:18.57 | C6Vette | I see some on on ebay selling from cal. but they want $43 shipped. |
| 01:19.12 | intralanman | it could be worse |
| 01:19.26 | intralanman | not much, but it could be |
| 01:19.31 | C6Vette | Guess it depends on how bad you want it fast |
| 01:19.34 | Jason559-55 | :) thanks |
| 01:19.51 | Jason559-55 | I guess I can use the extra time to get used to the system |
| 01:20.09 | Jason559-55 | I still can't seem to get outoing SIP working with voipstunt... |
| 01:20.14 | intralanman | yeah, look into enum, that'll take a day or two |
| 01:20.16 | intralanman | lol |
| 01:20.33 | C6Vette | nat problem |
| 01:20.35 | intralanman | speaking of ENUM..... does "anyone" know much about it? |
| 01:21.10 | Jason559-55 | not familar with enum |
| 01:21.28 | Jason559-55 | NAT, is that basically problems with the router? |
| 01:23.13 | C6Vette | http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 01:23.49 | Jason559-55 | thank you, I will do some reading and hopefully find a solution. |
| 01:25.28 | *** join/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 01:29.03 | *** join/#asterisk pbx1 (n=pbx1@124.106.141.64) |
| 01:29.04 | ANTILOCAS | hello |
| 01:29.14 | ANTILOCAS | anybody here uses Grandstream HT 496? |
| 01:29.29 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:32.22 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:33.08 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 01:35.37 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 01:41.02 | ANTILOCAS | using my Grandstream HT496 ATA i can hear the other person but the other person cant hear me. Any advice? |
| 01:44.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:49.18 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 01:52.53 | *** join/#asterisk tlow (n=tlow@voip.terrorist.net) |
| 01:53.22 | [TK]D-Fender | profounded : PM |
| 01:53.59 | tlow | one of my asterisks stopped logging in sexy colors, i think hes sad. |
| 01:56.16 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 02:04.23 | *** part/#asterisk tlow (n=tlow@voip.terrorist.net) |
| 02:04.46 | *** join/#asterisk misc-- (n=blah@203.215.120.228) |
| 02:05.34 | misc-- | hi I was wondering if anyone has used smsq over pstn in australia? |
| 02:06.17 | _DAW | anyone here using the aastra 480i ct? |
| 02:06.47 | *** part/#asterisk gJon (n=jellis@ip72-204-240-138.ph.ph.cox.net) |
| 02:11.45 | Druken | nope, i use the 9133.... |
| 02:12.15 | Druken | _DAW: wuts the problem ? |
| 02:12.17 | [TK]D-Fender | Druken : How do yuo find it? I'm thinking I should learn more about Aastra... |
| 02:12.49 | Druken | [TK]D-Fender: remember the old meridian business sets? |
| 02:12.55 | tmccrary | with a telco, if I have analog lines, can the telco set it up so one number rings all the lines? |
| 02:13.09 | tmccrary | like 555-5555 will use any available line |
| 02:13.17 | Druken | tmccrary: yes, it's called a hunt group |
| 02:13.19 | Druken | or rollover |
| 02:13.41 | _DAW | Druken: I get a goofy double ringing if I dont use ,r in * when dialing out. |
| 02:13.51 | tmccrary | Would I need centrex for that or can that be done with just POTS lines? |
| 02:14.02 | Druken | ahh, yeah... welcome to voip :) |
| 02:14.15 | Druken | tmccrary: pots... |
| 02:14.38 | _DAW | I havent run into that issue before. I mostly use polycom. |
| 02:14.48 | tmccrary | Also I mean dialing inward, not out (I know asterisk can do hunt groups) |
| 02:14.56 | tmccrary | like when a third part calls you |
| 02:16.02 | Druken | _DAW: well, it's basically the PHONE gives you a ring, but for some reason, your also getting the PRI's ring... |
| 02:16.42 | Druken | tmccrary: uhmm.. well, you said telco... and i would assume your telco wouldn't have access to your box... |
| 02:17.12 | tmccrary | I mean, can I have one phone number ring all of my analog lines. |
| 02:17.28 | Druken | yes, it's called a huntgroup or rollover... |
| 02:17.32 | tmccrary | ok, thanks |
| 02:17.47 | _DAW | tmccrary: just ask your provider, they will know what you are asking for. |
| 02:18.00 | tmccrary | right |
| 02:18.09 | tmccrary | I see you have never dealt with SBC/ATT |
| 02:18.30 | ManxPower | _DAW, you have a SIP issue, not a ring issuel |
| 02:18.48 | tmccrary | ;) |
| 02:19.21 | ManxPower | AT&T has been good to us. |
| 02:19.47 | ManxPower | _DAW, the problem with ,r is that it overrides what you SHOULD be hearing and plays a ringing sound. |
| 02:19.53 | tmccrary | Wow, you are lucky |
| 02:20.05 | _DAW | ManxPower: is there a usual reason for this problem? |
| 02:20.18 | ManxPower | tmccrary, they were the first of our carriers to have the lines back up after Katrina |
| 02:20.32 | ManxPower | _DAW, It's been discussed on the mailing lists. |
| 02:20.54 | _DAW | Ill research it further.. thanks |
| 02:20.59 | tmccrary | They tend to tell me "the line is fine" and then it still doesn't work. But on the second trouble ticket, man they fix it that time. |
| 02:21.00 | ManxPower | _DAW, What interface to PSTN? |
| 02:21.12 | ManxPower | tmccrary, THAT is ALL carriers |
| 02:21.14 | _DAW | PRI |
| 02:21.28 | ManxPower | _DAW, sounds like you need priindication=outofband or something like that. |
| 02:22.02 | *** join/#asterisk nitrus^ (n=nitrus@ip70-187-148-2.oc.oc.cox.net) |
| 02:22.13 | nitrus^ | anyone here us any of the asterisk web guis? |
| 02:22.57 | tmccrary | Does anyone know of any desktop SIP phones that have bluetooth, so you can use a BT headset with it? |
| 02:23.30 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 02:25.10 | nitrus^ | is freepbx pretty good? |
| 02:26.30 | misc-- | does anyone know what this means, it appears in my logs: pbx_spool.c: Delaying retry since we're currently running 'X?X??' |
| 02:29.27 | ManxPower | misc--, I suspect any notice 'X?X??' is in indication of a system crash, since that is garbage |
| 02:29.30 | tmccrary | Does anyone here use Polycom 501 phones? Are the displays backlit? |
| 02:30.56 | stubert | tmccrary: good question, I have 20 of them, and have never looked at them in the dark |
| 02:31.13 | misc-- | ManxPower: no crash, I'm getting it repeatedly |
| 02:31.25 | *** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid) |
| 02:31.27 | CpuID2 | sup ppls |
| 02:31.50 | [TK]D-Fender | tmccrary : No, only the IP 650 has a backlight |
| 02:32.33 | tmccrary | Can anyone recommend any backlit phones other than that one? |
| 02:32.42 | tmccrary | btw thx d-fender :) |
| 02:32.59 | wunderkin | cool, finally a poly with a backlight, cool |
| 02:33.06 | _DAW | does anyone here think the IP 650 is worth $100 more than the 601? |
| 02:33.11 | [TK]D-Fender | tmccrary : The only other model woth thinking about is the Linksys SPA-942. |
| 02:33.20 | [TK]D-Fender | _DAW : No. |
| 02:33.34 | _DAW | :) |
| 02:33.41 | [TK]D-Fender | _DAW : not until they show the profitability of the USB port on it as well. |
| 02:33.57 | tmccrary | It looks like a cisco... but is it really a sipura in cisco-like linksys clothing? |
| 02:34.00 | CpuID2 | hmm, anyone here know the best option in terms of CDR with asterisk queues? |
| 02:34.00 | litage | is it possible to ring multiple sip friends simultaneously if a call comes in on an FXO? |
| 02:34.10 | CpuID2 | standard cdr doesnt store the answering party |
| 02:34.11 | [TK]D-Fender | tmccrary : Seriously though if you're in North America, screw everything but Polycom. |
| 02:34.24 | tmccrary | Why is that? |
| 02:34.34 | litage | [TK]D-Fender: why specifically if you're in north america? |
| 02:34.41 | CpuID2 | oh hang on my bad |
| 02:34.43 | ANTILOCAS | using my Grandstream HT496 ATA i can hear the other person but the other person cant hear me. Any advice? |
| 02:34.44 | CpuID2 | it does :) |
| 02:35.09 | tmccrary | I recently purchased a Polycom 301, it's an extremely nice phone for the money (like 90 bucks). So I am fairly fond of polycom |
| 02:35.27 | stubert | litage: yes SIP/100&SIP/102&SIP/104 |
| 02:35.28 | _DAW | and the 301 is pretty crappy in comparison to the 501 and up |
| 02:35.33 | tmccrary | But I wonder why they would be the best option in north america |
| 02:36.18 | tmccrary | I've gotten a couple of Linksys IPW-5000 or whatever, those are *interesting*... :) |
| 02:36.31 | litage | stubert: and if one of those phones picks up the call, the others will stop ringing? |
| 02:36.39 | tmccrary | yeah |
| 02:36.39 | stubert | litage: yes |
| 02:36.46 | litage | stubert: awesome, thanks |
| 02:37.24 | tmccrary | I've been thinking about writing some changes so that accounts can have "aliases" so you don't need to do that if one person uses multiple phones. |
| 02:37.56 | tmccrary | I've always found doing things the way described above as kind of cludgy for that purpose |
| 02:38.05 | stubert | I do it as a global in the dialplan my self |
| 02:38.40 | stubert | Pretty much an alias... <g> |
| 02:38.43 | tmccrary | yeah, but if you're using an alternate backend, things like that are much trickier |
| 02:38.55 | stubert | True... |
| 02:39.35 | tmccrary | and then I can add that to my other patch i wrote for asterisk, notching my personal total to TWO ;) |
| 02:39.50 | stubert | heh... |
| 02:40.18 | tmccrary | well, assuming this one would get accepted. The other was a bug fix, this is a little more political I think |
| 02:40.33 | [TK]D-Fender | litage : Its just that outside of N/A Polycom pricing CAN be a lot more than other phones that might be considered over it |
| 02:41.11 | [TK]D-Fender | _DAW : Yeah the 301 isn't as nice for screen ,etc but it does the job well and cheap. |
| 02:41.24 | [TK]D-Fender | _DAW only reall thing you lose is speakerphone. |
| 02:41.35 | _DAW | 430 is not too much more |
| 02:41.35 | tmccrary | yeah, I've tried SNOM, Grandstream and at the price, the 301 seems like 10x the phone |
| 02:42.00 | [TK]D-Fender | _DAW : I only suggest the IP430 for those doing PoE deployments, otherwise the 501 is worth a few extra $ |
| 02:42.04 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 02:42.06 | aptura | TK what would cause a incomming DID to die on the cli when the application $(CALLERIDNAME) is on the second priority after answer? |
| 02:42.22 | tmccrary | pb |
| 02:42.31 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 02:42.32 | [TK]D-Fender | aptura : Pastebin the CLI output of that call and the associated dialplan |
| 02:42.33 | aptura | Perhaps my DID provider does not support names? |
| 02:42.41 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-138-7-77.dsl.irvnca.pacbell.net) |
| 02:42.42 | _DAW | oops |
| 02:42.57 | aptura | Well, there is only two lines from the DID when the issues happens. |
| 02:43.11 | aptura | But I can do this. |
| 02:43.17 | [TK]D-Fender | IP 430 is the perfect "new company" general purpose PoE phone. |
| 02:43.27 | [TK]D-Fender | ~pb |
| 02:43.29 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 02:43.37 | tmccrary | yeah, I'm about to order some :) |
| 02:44.24 | [TK]D-Fender | tmccrary : Keep in mind my viewpoint as to its target audience. The IP 501 *IS* nicer. |
| 02:44.40 | aptura | also, is MusicWaitOnHold After Dial application or After the Background(Pls-wait-connect-call) sound file? |
| 02:46.54 | *** join/#asterisk shodan (n=shodan@ip213.99-113-216.pppoe4.joliette.intermonde.net) |
| 02:47.18 | tmccrary | D-Fender, do you know if the PoE adapter for the 501 is the same as the one for the 500? |
| 02:47.34 | tengulre | http://www.megacomm.com.cn/download/asteriskmanager.rar version: 1.2.5 |
| 02:50.03 | [TK]D-Fender | tmccrary : Should be. |
| 02:50.20 | [TK]D-Fender | tmccrary : Most on-line descriptions should confirm it |
| 02:51.53 | tmccrary | thx |
| 02:52.30 | aptura | TK my music on hold did work but can it in this case work by putting the caller on hold while the extension is ringing? |
| 02:52.58 | tengulre | what 's the DND of Zap mean? |
| 02:53.37 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 02:54.02 | [TK]D-Fender | aptura : You mean you want callers to hear MoH instead of ringing when they call you? |
| 02:55.34 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:57.52 | aptura | yes |
| 02:59.37 | aptura | and? |
| 03:04.40 | denon | anyone from AU awake? |
| 03:04.58 | aptura | denon australia if thats what your refering. |
| 03:05.03 | denon | yeah |
| 03:05.10 | denon | you are? |
| 03:05.19 | aptura | I just saw the televised broadcast of Steve Irwins memorial service from there. |
| 03:05.30 | denon | eh .. |
| 03:05.35 | denon | not quite what I was wondering :) |
| 03:05.43 | denon | I'm saying I wonder if anyone is in the channel here .. from AU |
| 03:07.21 | aptura | :) |
| 03:07.59 | aptura | Found a interesting site of royalty free music that can be used as alternatives to those supplied in the add-ons |
| 03:08.48 | Nugget | I just used Soundtrack to make some vanilla background music. |
| 03:10.00 | tmccrary | vanilla ice background music I hope |
| 03:10.02 | aptura | sounddogs.com has some nice MOH music |
| 03:10.11 | Nugget | police on the scene, you know what I mean! |
| 03:12.53 | tmccrary | yes |
| 03:12.59 | tmccrary | hip hop |
| 03:13.09 | tmccrary | hip hop anonymous |
| 03:16.44 | ANTILOCAS | i wanna call someone in USA so he can tell me if he can hear me |
| 03:16.48 | ANTILOCAS | any volunteer? |
| 03:19.02 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 03:21.11 | CpuID2 | meh just call some 1800 number and ask the operator :) |
| 03:22.21 | aptura | I think there is a free service that will do that. somone can also create a dial plan that can include incomming and a echo statement for you. |
| 03:25.12 | hmmhesays | time for beer |
| 03:26.03 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:33.34 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 03:42.27 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:46.40 | *** join/#asterisk NetNut404_remote (n=NetNut40@166.195.31.95) |
| 03:52.03 | *** join/#asterisk bmg505 (n=leon@c1-21-4.rndf.isadsl.co.za) |
| 03:58.34 | *** join/#asterisk P4C0 (n=paco@201.224.103.235) |
| 03:59.34 | *** part/#asterisk NetNut404_remote (n=NetNut40@166.195.31.95) |
| 03:59.36 | P4C0 | hello, doesn anyone knows what's a good timeout before hanging up? the equivalent to 3 or 4 rings maybe? |
| 04:00.02 | CunningPike | P4C0: For what purpose? |
| 04:00.57 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 04:02.05 | P4C0 | CunningPike for some strange reason if I call someone and don't answer in a while, the call fails... it seems that I wait longer than my sip provider, so when I picked up my voip provider doesn't send me the ACK or notification that the call was picked up... it's strange cause I'm using ethereal an sip message from voip provider just never cames |
| 04:04.30 | P4C0 | so I was thinking that maybe if I limit the time it will work better... I'm using 30 for my voip provider outgoing calls and 100 for calls to extensions |
| 04:09.14 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 04:14.49 | *** join/#asterisk juice (n=juice@mo-76-0-45-81.dhcp.embarqhsd.net) |
| 04:26.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.15 | aptura | CunningPike got my extention to work..kinda. :) |
| 04:29.29 | CunningPike | aptura: Oh ya? |
| 04:30.00 | aptura | The musiconhold works...but the Dial does not dial the phone untill the time I set for music on hold expires. |
| 04:30.31 | aptura | want the Dial application to dial the phone while the caller is listening to the music. |
| 04:31.21 | benjk | Dial(...,,m) |
| 04:31.46 | aptura | so put the m on the end of the dial statment? |
| 04:31.55 | aptura | interesting |
| 04:31.58 | benjk | m instead of r, m for music |
| 04:32.00 | aptura | okay will try |
| 04:32.15 | benjk | show application Dial on the CLI will tell you all the flags |
| 04:32.24 | benjk | *CLI> show application Dial |
| 04:33.17 | aptura | okay never knew that existed thanks :) |
| 04:33.50 | aptura | I know cli fairly well thanks |
| 04:33.53 | aptura | :) |
| 04:36.40 | nitrus^ | anyone have a working script to get all my conf files into mysql? |
| 04:38.15 | aptura | benjk, thanks for the heads up. |
| 04:38.18 | Lyfe | hahaha.. anyone here ever heard a commercial for Pasquinelli Homes? (guess this might only apply to the midwest US.) They use the same sound file called fpm-calm-river.mp3 found in the default asterisk hold music for background music in the commercial :) |
| 04:38.25 | aptura | worked well |
| 04:38.49 | aptura | ohh really |
| 04:38.54 | aptura | but who created the file? |
| 04:39.03 | benjk | which file? |
| 04:39.19 | aptura | Lyfe just said it |
| 04:39.33 | Lyfe | no idea, far as i knew the hold music in asterisk was free to use for anything. |
| 04:40.00 | benjk | it is royalty free music from Freeplay Music |
| 04:40.10 | benjk | royalty free means you pay only once |
| 04:40.20 | Lyfe | interesting. |
| 04:40.49 | benjk | type "royalty free music" into Google |
| 04:41.09 | benjk | you'll see quite a number of companies offering royalty free music |
| 04:41.35 | aptura | Now I need to do radio station streaming media in place of music on hold for something different. |
| 04:41.36 | aptura | :) |
| 04:41.45 | benjk | that's illegal |
| 04:41.58 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 04:42.06 | benjk | you need a broadcasting license |
| 04:42.13 | benjk | fairly expensive |
| 04:42.19 | De_Mon | for music on hold? |
| 04:42.32 | benjk | music on hold is broadcasting |
| 04:42.39 | wunderkin | Lyfe, i've heard someone use one of the * moh files on a commercial before too.. |
| 04:42.46 | benjk | if you use a radio stream, it is re-broadcasting |
| 04:42.49 | aptura | streaming media music on hold |
| 04:42.56 | De_Mon | heh. I know quite a few people that have a radio hooked into their home phone system for hold music |
| 04:43.04 | [TK]D-Fender | quite true. Interesting how many companies rebroadcast local radio stations illegally for MoH |
| 04:43.11 | benjk | that doesn't mean it is legal though |
| 04:43.19 | benjk | they will need a broadcasting license |
| 04:43.23 | Lyfe | wunderkin: i'm just amazed. mostly that i knew where i'd heard it before. |
| 04:43.30 | benjk | if they get caught and sued it can be extremely costly |
| 04:43.35 | benjk | like ruining you totally |
| 04:43.36 | [TK]D-Fender | benjk : True, but sub 1% have one |
| 04:43.36 | aptura | I have seen a few pbxs with cheap radios hooked to them. |
| 04:43.38 | De_Mon | actually, I think my dentist does that too, I wonder if they have a broadcasting license... |
| 04:43.49 | aptura | nortel meridians with cheap radios. |
| 04:43.55 | *** join/#asterisk droops (n=root@adsl-147-224-61.jan.bellsouth.net) |
| 04:43.58 | benjk | the dentist will have paid for the license |
| 04:44.05 | benjk | like your local supermarket |
| 04:44.33 | benjk | the radios can be used if you have purchased the license |
| 04:45.09 | nitrus^ | god damn freepbx |
| 04:45.11 | benjk | its from several thousand dollars per year to millions of dollars per year depending on the size of your audience |
| 04:45.14 | nitrus^ | that shit hosed my extensions.conf |
| 04:45.43 | benjk | you can check with a music royalty collection society |
| 04:45.51 | benjk | ASCAP is one of them |
| 04:45.56 | benjk | in the US |
| 04:46.57 | benjk | in some countries you also are obliged to fill out so called cue sheets and submit them |
| 04:47.11 | benjk | basically those are reports what tracks you were playing when and for how long |
| 04:47.55 | De_Mon | 99.1 all day =p |
| 04:48.03 | Lyfe | backups are your friend. :) |
| 04:49.07 | aptura | benjk, talking about paying for a licence if you sell a pbx to a dentist office? |
| 04:49.36 | benjk | De_Mon, if you happen to be in a place where the rule is to fill out the cue sheets and submit them, then you will have to do that, it won't be sufficient to tell them which radio station you are rebroadcasting, though luck |
| 04:50.24 | droops | i asked at our local am station |
| 04:50.32 | [TK]D-Fender | nitrus^ : .... DUH. thats what it does. |
| 04:50.41 | droops | he pays about 300$ a month for rights to broadcast the music |
| 04:50.49 | benjk | if you want to sell a PBX to a business, you can buy a CD (or download) from a royalty free music publisher and just charge the cost for that to your customer |
| 04:51.09 | benjk | they can then use that music as they please for music on hold and never have to pay again |
| 04:51.20 | Lyfe | yeap. |
| 04:51.25 | Lyfe | i've seen those. |
| 04:51.32 | benjk | I use music from Shockwave Sound |
| 04:51.41 | benjk | a CD is typically 100 USD |
| 04:51.52 | benjk | you can get it as download too |
| 04:52.07 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 04:52.15 | benjk | I then charge the money back to my customers and pass the license document to them |
| 04:52.33 | benjk | www.shockwave-sound.com |
| 04:52.44 | benjk | they have a pretty decent portfolio |
| 04:53.09 | De_Mon | I need to come up with a way to transfer a call from my music on hold, to the other systems music on hold without it being so obvious. |
| 04:53.36 | De_Mon | like a please stay on the line msg and then drop them in new music... |
| 04:54.14 | tmccrary | You mean like fading in and out? |
| 04:54.22 | benjk | its a lot cheaper to pay once 100 USD (per customer) than paying 1000+ USD per year (per customer) |
| 04:54.37 | *** join/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 04:54.59 | De_Mon | benjk why not include royalty free music with the pbx? |
| 04:55.26 | De_Mon | they want better music I bet ;P |
| 04:55.28 | w32 | trixbox is fucking useless |
| 04:55.32 | benjk | the 100 USD *is* for the royalty free music |
| 04:55.56 | De_Mon | tmccrary that might work |
| 04:55.59 | benjk | if you use it for a business, you have to make your customers buy their own license (one time license) |
| 04:56.26 | benjk | you cant just buy a license for yourself and then redistribute it commercially |
| 04:56.32 | De_Mon | what do you call music that is just free |
| 04:56.38 | *** join/#asterisk eiko (n=tv@ip68-228-32-72.tc.ph.cox.net) |
| 04:56.42 | De_Mon | public domain music |
| 04:56.50 | benjk | royalty free music publishers typically have two licensing options: |
| 04:57.18 | benjk | 1) pay once and distribute with one single project, non-profit |
| 04:57.35 | benjk | in which case the number of users doesn't matter |
| 04:58.00 | De_Mon | sounds like you could buy the music for the asterisk project and give it to yer customers that way |
| 04:58.02 | benjk | 2) pay on a per user basis with any number of projects, profit or non-profit |
| 04:58.18 | benjk | De_Mon, not if you charge for your work |
| 04:58.28 | benjk | and not if they use it for business |
| 04:58.35 | De_Mon | hmmm |
| 04:58.42 | benjk | using Asterisk as a business PBX is for-profit use |
| 04:58.55 | Lyfe | so, is the first way the reason that asterisk comes with 3 songs? |
| 04:59.22 | wunderkin | 1) install asterisk, 2) steal underpants, 3) ?, 4) profit |
| 04:59.23 | benjk | Digium probably purchased the three tracks on the non-profit scheme |
| 04:59.46 | benjk | which means they can redistribute them as part of one single project (Asterisk), non-profit |
| 04:59.54 | Lyfe | wunderkin: been watching southpark, huh? :) |
| 04:59.59 | De_Mon | I need to dig up the Terms of Use on the music im using |
| 05:00.00 | wunderkin | heh :D |
| 05:00.09 | benjk | at the same time, they will pay for each commercial distribution of Asterisk |
| 05:00.15 | wunderkin | bring a towel! |
| 05:00.25 | benjk | but the fees are very reasonable with royalty free music |
| 05:00.34 | Lyfe | but, how do they know if you use asterisk in a business environment? |
| 05:00.45 | benjk | you can buy a single track for about 25-30 USD |
| 05:00.53 | benjk | and then just charge that back to your customer |
| 05:01.04 | benjk | you get a license document from the music publisher |
| 05:01.04 | aptura | how many times has a customer brought up the question "well if your a asterisk consultant and its free why dont we download and install it ?" anyone heard that? |
| 05:01.11 | benjk | you pass the license document to your customer |
| 05:01.29 | Lyfe | i mean digium. they build asterisk, but there's no questions or whatnot when you install asterisk, specifically when a business does it in-house. |
| 05:01.51 | De_Mon | RIAA randomly raids business searching for unlicensed music |
| 05:02.02 | benjk | yes they do |
| 05:02.07 | De_Mon | you tell a buddy, and he rats you out |
| 05:02.09 | aptura | really |
| 05:02.11 | benjk | that's why you need to pass the license document on |
| 05:02.20 | drray | I don't see how the RIAA has legal authority to do that |
| 05:02.25 | Lyfe | does it simply not matter, since digium's not making anything out of it, or does the busines need to go get their own hold music? |
| 05:02.40 | benjk | the nice thing about royalty free music is that the RIAA has no ownership in it |
| 05:03.16 | benjk | royalty free music is music that is recorded by independent studios, independent musicians, using independent or dead composer's compositions |
| 05:03.34 | benjk | that's why a lot of the stuff is classical music |
| 05:03.53 | benjk | written before the collection society system was established in the 1920s |
| 05:03.59 | Lyfe | mhmm. |
| 05:04.15 | benjk | if you are best friends with say Bob Dylan |
| 05:04.17 | De_Mon | you shouldn't need a license to play music written by someone 200 years ago |
| 05:04.26 | benjk | and Bob Dylan writes a song specifically for you |
| 05:04.48 | benjk | and you play it yourself with your own guitar in your own recording studio |
| 05:04.52 | benjk | and you put it on your PBX |
| 05:05.02 | benjk | you will have to buy a broadcasting license from ASCAP |
| 05:05.06 | benjk | no way around that |
| 05:05.33 | Lyfe | ahh, the things i never expect to learn in #asterisk. :P |
| 05:05.38 | benjk | if Bob Dylan writes himself a song for his own PBX |
| 05:05.50 | benjk | he will need to purchase a broadcasting license from ASCAP |
| 05:06.02 | Lyfe | sucks to be bob dylan. |
| 05:06.07 | benjk | heh |
| 05:06.23 | benjk | but then again, he can afford the license |
| 05:06.33 | De_Mon | whats a license cost? |
| 05:06.35 | *** part/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 05:06.36 | [TK]D-Fender | benjk : Do I need a license if I sing in an IVR recording? This is getting very messy you know.... |
| 05:07.07 | De_Mon | [TK]D-Fender if you wrote and sing the song yourself, yes. |
| 05:07.07 | benjk | if you sing a song written by a composer who is a member of a collection society like ASCAP, then yes |
| 05:07.33 | benjk | if you write the song yourself, and you are not a member of a collection society, then no |
| 05:07.36 | [TK]D-Fender | De_Mon : From what benjk just alluded to, no. |
| 05:07.57 | [TK]D-Fender | benjk : AH, that could apply. |
| 05:08.04 | benjk | if you sing a song by Franz Schubert who died long before collection societies were founded, then no |
| 05:08.40 | benjk | if you sing a song that's in the public domain, like "composer unknown" then no |
| 05:08.52 | [TK]D-Fender | benjk : Similar topic comes up in the transctiption by "interpretation" of songs back to sheet music. while not always accurate there mere attempting of which seems to be actionable. |
| 05:09.20 | benjk | indeed |
| 05:09.31 | benjk | the music rights holders are vultures |
| 05:09.40 | benjk | they try to get they hands on everything |
| 05:09.45 | [TK]D-Fender | Welcome to "Stupid World", population : US |
| 05:10.23 | benjk | In the US, some classical works which did not fall under the collection society system have been reassigned to the collection societies |
| 05:10.40 | benjk | talk about theft |
| 05:10.47 | benjk | theft from the public domain |
| 05:11.43 | Qwell | <benjk> if you write the song yourself, and you are not a member of a collection society, then no |
| 05:12.04 | benjk | yes |
| 05:12.05 | Qwell | benjk: Why did you qualify the "and are not a member"? Would you get sued if you weren't? heh |
| 05:12.37 | [TK]D-Fender | Qwell : I think the group maintains the rights in perpetuity... |
| 05:12.45 | benjk | many composers choose to become members of collection societies (ie ASCAP) because it means they get a cut of the revenues |
| 05:13.07 | Qwell | benjk: Yes, but would that mean that the composer would get fined/sued/whatever if he sung HIS OWN song? |
| 05:13.13 | benjk | if they are members, then all the music they write in their lifetime will automatically fall under the collection society scheme |
| 05:13.20 | Qwell | You qualified it, making it sound as though that were the case |
| 05:13.30 | benjk | if he sings in his bathroom, then not |
| 05:13.39 | Qwell | but we're talking IVRs |
| 05:13.43 | Qwell | or MoH, or whatever |
| 05:13.47 | benjk | but if it is public performance, then he is liable to pay the royalties to the collection society |
| 05:13.55 | Qwell | even for his own composition? |
| 05:14.02 | benjk | yes |
| 05:14.10 | Qwell | That's...pathetic? |
| 05:14.13 | benjk | MOH is public performance |
| 05:14.34 | benjk | that;s how the system is set up |
| 05:14.45 | [TK]D-Fender | ok, well this has been enlightening. But bed beckons.... later all |
| 05:15.00 | wasim | yawn ... morning |
| 05:16.18 | benjk | hi wasim |
| 05:16.49 | wasim | ugh .. too much work piled up, i should go back to bed :) |
| 05:16.56 | benjk | heh |
| 05:17.28 | benjk | everybody is getting screwed |
| 05:17.38 | Qwell | except the execs |
| 05:19.10 | denon | the execs take a risk .. |
| 05:19.11 | benjk | all the more reason to buy royalty free music from royalty-free music publishers |
| 05:19.20 | Strom_C | NEVAR |
| 05:19.21 | denon | when they put $20M of cash behind New Kids on the Block .. and the flop .. |
| 05:19.30 | denon | that's $20M they've gotta make up elsewhere |
| 05:19.42 | denon | the/they |
| 05:19.49 | benjk | from composers who are long dead, or from composers who chose not to become members of collection societies |
| 05:22.12 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:26.58 | *** join/#asterisk ast_freak (n=ast_frea@68-112-143-168.dhcp.stcd.mn.charter.com) |
| 05:29.20 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 05:40.25 | Bobcat991966 | hello all, anybody got time to help me with a dialplan issue. I have created a voice recognition script and it works fine as long as I do a Dial SIP/XXX but if I try to use a goto command to the ext-local context it craps out. Here is my pastebin http://pastebin.ca/177111 |
| 05:41.05 | Bobcat991966 | Sorry I meant to say gosub no goto |
| 05:42.06 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 05:42.46 | aptura | thats great |
| 05:43.12 | aptura | Bobcat991966 I am looking into popping up a Netsend message on a windows machine when a cid comes in on asterisk. |
| 05:43.24 | aptura | trying to figure out this perl script |
| 05:43.48 | Bobcat991966 | Good luck, I dont understand perl |
| 05:44.09 | Bobcat991966 | Im having a heard enought time just doing scripting in Asterisk config files |
| 05:44.51 | Bobcat991966 | But I do have a bunch of perl books if I ever get the time to read them :) |
| 05:45.34 | aptura | yea |
| 05:46.03 | Bobcat991966 | is file around tonight |
| 05:46.11 | Qwell | Bobcat991966: he went to bed |
| 05:46.46 | Bobcat991966 | ahhh, owell maybe if I did the same thing I could figure out what was wrong with my sript after a good nights sleep |
| 05:46.49 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 05:46.56 | Bobcat991966 | thanks Qwell |
| 05:47.33 | Bobcat991966 | Im sure it something really simple but I cannot seem to figure it out |
| 06:05.48 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 06:07.30 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:11.24 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:12.53 | tessier | Anyone know of any resources for tuning the QoS in a Linux router to prioritize voip calls? |
| 06:13.03 | tessier | I used to use wondershaper for this but it seems to not do a thing on my latest installation of it |
| 06:13.08 | tessier | Haven't used it for a couple years. |
| 06:14.55 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 06:27.35 | stephane | jour |
| 06:31.44 | aao_pwner | i have a noobish question.. |
| 06:31.50 | aao_pwner | with Asterisk, do i need Zaptel too? |
| 06:31.55 | tuck3r | depends |
| 06:31.59 | aao_pwner | on what? |
| 06:32.21 | tuck3r | do you want to use meetme or IAX trunking and of cource you need it for zaptel hardware |
| 06:32.22 | wunderkin | the phase of the moon |
| 06:32.34 | wunderkin | and that |
| 06:32.39 | aao_pwner | umm, i dont have hardware for it.. |
| 06:32.44 | aao_pwner | all i have is a server and internet |
| 06:32.47 | aao_pwner | and i wanna host a PBX |
| 06:32.48 | aao_pwner | lol |
| 06:32.58 | tuck3r | ok, so the real question is do you want to use meetme confs? |
| 06:33.07 | aao_pwner | uhh.... do i NEED to |
| 06:33.15 | tuck3r | do you want to? |
| 06:33.19 | aao_pwner | i dont even know what thoes are dude |
| 06:33.20 | aao_pwner | lol |
| 06:33.34 | tuck3r | conferences |
| 06:33.39 | aao_pwner | yes. |
| 06:33.46 | aao_pwner | do i need any special hardware for that? |
| 06:33.51 | tuck3r | no |
| 06:33.55 | aao_pwner | ok.. |
| 06:33.55 | tuck3r | you need zaptel |
| 06:34.16 | aao_pwner | ok |
| 06:35.46 | aao_pwner | well |
| 06:35.52 | aao_pwner | im having problems compiling zaptel erm ... |
| 06:35.55 | aao_pwner | ya.. with libnewt |
| 06:36.15 | tuck3r | linux > freebsd |
| 06:36.24 | aao_pwner | you better duck. |
| 06:36.29 | tuck3r | hah |
| 06:36.30 | aao_pwner | it depends what you want |
| 06:36.31 | aao_pwner | lol |
| 06:36.42 | aao_pwner | distro wars are pointless so im not gonna start |
| 06:36.42 | aao_pwner | lol |
| 06:36.53 | tuck3r | i have never had to compile libnewt |
| 06:37.01 | aao_pwner | each distro (Except ubuntu) has its advantages |
| 06:37.03 | wasim | you need it for zttool |
| 06:37.10 | tuck3r | but the ebuild seems to always want to compile it |
| 06:37.13 | aao_pwner | wtf is zttool |
| 06:37.13 | aao_pwner | lol |
| 06:37.20 | tuck3r | wasim: ah, now i get it |
| 06:37.27 | tuck3r | aao_pwner: it's for zaptel cards |
| 06:37.31 | aao_pwner | http://pastebin.ca/177147 |
| 06:37.39 | aao_pwner | so.. what do i need to do to get this to work lol |
| 06:38.10 | tuck3r | bitch at the zaptel maintainer for fbsd? |
| 06:38.27 | tuck3r | or libnewt for that matter |
| 06:38.33 | aao_pwner | i just cvsuped the ports tree lol |
| 06:38.55 | tuck3r | i might find that funny if i used bsd... |
| 06:39.51 | aao_pwner | i sent it out to a mailing list |
| 06:39.51 | Lyfe | did you install newt?.. it is in the ports tree, for that matter. |
| 06:39.55 | aao_pwner | well.. |
| 06:39.57 | aao_pwner | when i try and do that lol |
| 06:40.14 | aao_pwner | pastebin'n it |
| 06:40.22 | Lyfe | oh yeah, you were here the other day having problems trying to even install newt, weren't you? |
| 06:40.23 | aao_pwner | http://pastebin.ca/177148 |
| 06:40.37 | aao_pwner | Lyfe, ya i was... |
| 06:41.16 | Lyfe | *cough*. cd /usr/ports/devel/newt && make install |
| 06:41.25 | aao_pwner | Lyfe, i tried that |
| 06:41.40 | Lyfe | and you get the same output as in this? |
| 06:41.50 | aao_pwner | not the exact same, im pastebining it now |
| 06:41.56 | aao_pwner | http://pastebin.ca/177149 |
| 06:41.56 | Lyfe | ok. |
| 06:43.16 | Lyfe | hmm.. try reinstalling slang? (devel/slang) |
| 06:43.59 | aao_pwner | trying now |
| 06:44.08 | aao_pwner | i THINK i already have it |
| 06:44.09 | aao_pwner | but hold on |
| 06:44.25 | Lyfe | that's why i asked about 'reinstalling' it. |
| 06:44.35 | aao_pwner | well i probably tried that lol |
| 06:44.39 | aao_pwner | but hold on |
| 06:44.45 | Lyfe | it shouldn't have tried compiling neewt without it, but anyway. |
| 06:44.56 | aao_pwner | it says it found it lol |
| 06:45.03 | aao_pwner | in the zaptel install |
| 06:45.22 | aao_pwner | ===> libslang-1.4.9 is already installed |
| 06:45.33 | tuck3r | "hmm.. try reinstalling slang" |
| 06:45.44 | tuck3r | key word part is "re" |
| 06:45.47 | aao_pwner | i am right now lol |
| 06:45.55 | aao_pwner | i did make clean && make distclean && make install clean |
| 06:46.06 | aao_pwner | well, thats what im doing right now |
| 06:46.10 | Lyfe | how about a "make deinstall" :) |
| 06:46.16 | aao_pwner | i did that too |
| 06:46.21 | Lyfe | ok, just checking. |
| 06:46.23 | aao_pwner | forgot to say that |
| 06:46.23 | aao_pwner | lol |
| 06:46.27 | EyeCue | portupgrade -rRf <portname |
| 06:46.39 | aao_pwner | EyeCue i just did a cvsup |
| 06:46.40 | aao_pwner | on all the ports |
| 06:46.43 | Lyfe | one of these days i will familiarize myself with portupgrade :) |
| 06:46.50 | aao_pwner | and i just cvsup'd the whole box |
| 06:46.59 | aao_pwner | and build the world and just updated the whole box |
| 06:47.08 | aao_pwner | ===> Registering installation for libslang-1.4.9 |
| 06:47.08 | aao_pwner | ===> Cleaning for libslang-1.4.9 |
| 06:47.08 | aao_pwner | [root@aaopwner /usr/ports/devel/libslang]# |
| 06:47.40 | Lyfe | ok, time for newt again. |
| 06:47.45 | aao_pwner | sameerror |
| 06:47.48 | aao_pwner | same error* |
| 06:48.35 | Lyfe | what's sad is that's a test case. :P |
| 06:49.00 | aao_pwner | ? |
| 06:49.02 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:49.26 | Lyfe | that compile error is over a test case for newt. |
| 06:49.41 | aao_pwner | which means what? |
| 06:49.41 | aao_pwner | lol |
| 06:49.42 | aao_pwner | i dunno |
| 06:50.02 | aao_pwner | http://www.mail-archive.com/asterisk-bsd@lists.digium.com/msg01270.html |
| 06:50.03 | aao_pwner | ahh |
| 06:50.29 | Lyfe | interesting. |
| 06:50.43 | Lyfe | ncurses conflicts with newt? very interesting. |
| 06:51.06 | aao_pwner | pretty not cool |
| 06:51.19 | aao_pwner | OH |
| 06:51.20 | aao_pwner | and now it works |
| 06:51.21 | aao_pwner | >:| |
| 06:51.25 | aao_pwner | WHAt a gaylorn |
| 06:51.25 | tuck3r | how can you not have ncurses tho |
| 06:51.27 | aao_pwner | D |
| 06:51.42 | Lyfe | i don't have ncurses on many systems. |
| 06:51.49 | aao_pwner | well... i use it for tmsnc |
| 06:52.52 | tuck3r | i can't live without make menuconfig :) |
| 06:53.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:54.06 | aao_pwner | err |
| 06:54.09 | aao_pwner | another noob question |
| 06:54.10 | aao_pwner | [root@aaopwner /usr/ports/misc/zaptel]# /usr/local/etc/rc.d/zaptel start |
| 06:54.10 | aao_pwner | zaptelkldload: can't load /usr/local/lib/zaptel/zaptel.ko: File exists |
| 06:54.11 | aao_pwner | lol |
| 06:54.22 | EyeCue | kldstat |
| 06:54.23 | aao_pwner | ok |
| 06:54.25 | EyeCue | it'll be loaded |
| 06:54.39 | aao_pwner | oh.. |
| 06:54.39 | aao_pwner | ok |
| 06:54.45 | aao_pwner | interesting |
| 06:54.58 | EyeCue | it does work on freebsd :) |
| 06:55.14 | aao_pwner | FreeBSD is pretty fancy actually |
| 06:55.16 | aao_pwner | i like it a lot |
| 06:55.20 | EyeCue | *nods* |
| 06:55.22 | aao_pwner | started on it like 3 years ago |
| 06:55.29 | aao_pwner | when i was 11 |
| 06:55.31 | tuck3r | bsd is fancy? |
| 06:55.46 | aao_pwner | truck3r yes it is. |
| 06:55.49 | EyeCue | fancy != gimmick |
| 06:55.52 | aao_pwner | tuck3r |
| 06:55.53 | aao_pwner | lol |
| 06:55.53 | EyeCue | fancy = refined :) |
| 06:56.01 | tuck3r | vista is fancy |
| 06:56.02 | aao_pwner | plus.. |
| 06:56.06 | aao_pwner | www.trustedbsd.org |
| 06:56.09 | aao_pwner | makes it pretty hot |
| 06:56.41 | tuck3r | i just love the new logo |
| 06:57.03 | tuck3r | makes me want to use it, but last time I tried the installer crashed on me :( |
| 06:57.15 | EyeCue | awesome, the nsa sponsors trustedbsd |
| 06:57.29 | tuck3r | and SELinux |
| 06:57.31 | tuck3r | Go NSA! |
| 06:57.41 | aao_pwner | yep |
| 06:57.41 | aao_pwner | lol |
| 06:58.14 | tuck3r | CIA fails |
| 06:58.21 | aao_pwner | what they sponsor ubuntu? |
| 06:58.26 | aao_pwner | LoL |
| 06:58.28 | tuck3r | they wont |
| 06:58.35 | aao_pwner | good |
| 06:58.36 | aao_pwner | ubuntu sucks |
| 06:58.43 | tuck3r | back that one up |
| 06:58.45 | aao_pwner | people are like its good for starting on... |
| 06:58.48 | aao_pwner | and im like.. |
| 06:58.51 | aao_pwner | i stareted on freebsd |
| 06:58.52 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 06:59.00 | aao_pwner | you dont learn anything about a *nix like envirnment with a GUI |
| 06:59.01 | aao_pwner | that's why |
| 06:59.02 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 06:59.09 | tuck3r | tell that to me |
| 06:59.14 | aao_pwner | i just did |
| 06:59.14 | tuck3r | I started on fedora |
| 06:59.25 | aao_pwner | i started on FreeBSD and FC4 |
| 06:59.26 | aao_pwner | actually |
| 06:59.26 | aao_pwner | lol |
| 06:59.26 | tuck3r | and now I know plenty of nix |
| 06:59.38 | Lyfe | 'cept most people don't *want* to learn anythinga bout a unix environment. which is what a mac is for, i'd say :0 |
| 06:59.41 | Lyfe | :) |
| 06:59.44 | CpuID2 | hehe ya :) |
| 06:59.48 | aao_pwner | lol Lyfe |
| 06:59.50 | CpuID2 | most ppl just wanna use it and dont wanna learn bout it |
| 07:00.01 | aao_pwner | Sad, Sad thing. |
| 07:00.09 | aao_pwner | i wanan buy a MAc book |
| 07:00.13 | aao_pwner | I dont see any point though |
| 07:00.18 | aao_pwner | for looking-cool and being unix based |
| 07:00.22 | aao_pwner | but you can do that with any noteook |
| 07:00.28 | aao_pwner | notebook*& |
| 07:00.28 | tuck3r | OSX is fbsd based |
| 07:00.37 | tuck3r | OSX kicks ass |
| 07:00.37 | Lyfe | nice hardware, great OS. and it. just. works. |
| 07:00.38 | aao_pwner | i know |
| 07:00.50 | aao_pwner | can you partiton windows on it? |
| 07:00.51 | aao_pwner | i mean... |
| 07:00.54 | aao_pwner | ya.. can you? |
| 07:00.56 | tuck3r | yes |
| 07:00.58 | Lyfe | yes, you can. |
| 07:01.00 | aao_pwner | is it hard? |
| 07:01.04 | tuck3r | no |
| 07:01.04 | aao_pwner | oh wait.. |
| 07:01.06 | Lyfe | i don't recommend actually wanting to use windows on it, but anyway. |
| 07:01.07 | aao_pwner | grub will do all |
| 07:01.11 | tuck3r | but why would you |
| 07:01.15 | aao_pwner | ya... |
| 07:01.17 | aao_pwner | its a Mac.. |
| 07:01.23 | aao_pwner | you minus well go buy a regular ass laptop |
| 07:01.24 | tuck3r | and no grub isn't used |
| 07:01.25 | aao_pwner | for half as much |
| 07:01.30 | aao_pwner | and 200% more functionallity |
| 07:01.34 | aao_pwner | but minus the good looks |
| 07:01.34 | tuck3r | aao_pwner: please cut the bullshit |
| 07:01.40 | aao_pwner | lol |
| 07:01.48 | Lyfe | actually, given the hardware, apple's not really out of the typical price range. |
| 07:01.52 | tuck3r | Macs are not overpriced anymore |
| 07:02.52 | Lyfe | my biggest beef with apple is that once they discontinue a model, it's practically imposible to get that model (at a hopefully lower cost) |
| 07:03.37 | tuck3r | mine is I want leopard now! |
| 07:04.00 | Lyfe | i've seen a pre-release. you don't want it yet, you want it when it's as stable as 10.4 |
| 07:04.15 | tuck3r | you mean 10.5.5? lol |
| 07:05.29 | Lyfe | only things i want from it right away are Time Machine and the new text-to-speech stuff. (mostly so i can get a good recording for asterisk IVR's without having to get someone else to do it) :) |
| 07:06.18 | tuck3r | has anybody tried that new voice recognition from digium? |
| 07:06.46 | Lyfe | didn't know they had it, even. |
| 07:07.10 | tuck3r | http://www.digium.com/en/products/software/lumenvox.php |
| 07:07.21 | tuck3r | they need to put a test number up |
| 07:07.37 | *** join/#asterisk deeLer (n=deeLer@41.14-78-194.adsl-static.isp.belgacom.be) |
| 07:07.53 | Lyfe | interesting. |
| 07:07.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:08.08 | Lyfe | welp, i'm not exactly someone in the market for that, so .. :P |
| 07:08.28 | tuck3r | i'm not but you never know |
| 07:09.07 | Lyfe | $600 for the full version doesn't strike me as a terrible price. |
| 07:09.18 | tuck3r | The LumenVox Speech Engine is purchased by channel (eg. per port). When ordering, please be sure to update the quantity to match the number of active channels needed for your machine. |
| 07:09.52 | Lyfe | oh. |
| 07:09.59 | *** join/#asterisk yuski0206 (n=yuski020@211.148.221.46) |
| 07:10.00 | Lyfe | well, nevermind, that can become very price. |
| 07:10.02 | Lyfe | pricey, even |
| 07:10.13 | tuck3r | Yes, i just saw that |
| 07:10.50 | *** join/#asterisk Slowking_Man (n=chris@wikipedia/Slowking-Man) |
| 07:11.16 | yuski0206 | hi |
| 07:12.38 | Slowking_Man | Hello. |
| 07:12.54 | Slowking_Man | Sorry, just passing by; I was finding out why someone else couldn't join. |
| 07:13.06 | tuck3r | you have to be authed with freenode |
| 07:13.20 | Slowking_Man | Yes, I'm helping him. |
| 07:13.38 | tuck3r | he needs to be authed with freenode |
| 07:13.54 | Slowking_Man | I'm aware of that. |
| 07:14.07 | tuck3r | ok... |
| 07:14.13 | yuski0206 | talk about me? |
| 07:14.19 | tuck3r | no |
| 07:14.29 | yuski0206 | sorry |
| 07:18.26 | tuck3r | i'm off, nighty night |
| 07:18.45 | Lyfe | g'night. |
| 07:19.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:20.08 | *** part/#asterisk Slowking_Man (n=chris@wikipedia/Slowking-Man) |
| 07:20.44 | puzzled | morning |
| 07:21.20 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:24.25 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 07:31.22 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 07:31.28 | zouzou | test |
| 07:45.49 | *** join/#asterisk DanTMG (n=danielga@203-214-135-194.perm.iinet.net.au) |
| 07:53.14 | *** join/#asterisk shtoom (n=godson@202.149.32.50) |
| 07:54.51 | hank | moin |
| 07:57.57 | hank | im trying to install asterisk to a debian unstable using the asterisk-bristuff package. |
| 07:58.37 | hank | shouldnt there be a kernel module (*.ko) in there? |
| 08:02.28 | *** join/#asterisk ltd (n=z@202-161-18-11.dyn.iinet.net.au) |
| 08:08.23 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 08:09.59 | deeLer | hank : no, same with zaptel ... you need to compile them yoursel |
| 08:09.59 | deeLer | f |
| 08:10.08 | deeLer | use module-assist to help you |
| 08:10.33 | hank | deeLer: huh? ok... you know the package name? |
| 08:10.54 | deeLer | try apt-cache search module-assist |
| 08:10.55 | hank | i cant find anything apart from asterisk-bristuff which looks like bristuff |
| 08:11.01 | hank | i know module-assistant ;) |
| 08:11.14 | hank | i mean the source to the modules. |
| 08:11.16 | deeLer | you'l need to download the bristuff source code from the web |
| 08:11.34 | hank | deeLer: and that will work with module-assistant? nice... |
| 08:11.41 | deeLer | i hope so |
| 08:11.47 | hank | deeLer: i assume you mean from junghanns.net? |
| 08:11.54 | deeLer | zaptel did work that way |
| 08:12.12 | deeLer | i'm not familiar with the bristuiff package , sorry |
| 08:12.28 | justdave | there's a couple packages mentioned in the asterisk-bristuff package description |
| 08:12.53 | hank | hmm yes zaptel would work because there is a zaptel-source package... |
| 08:12.59 | deeLer | indeed |
| 08:13.18 | justdave | ok, one. chan-capi for CAPI-compatible ISDN cards |
| 08:13.47 | hank | i have a billion usb isdn adapter which would work with bristuff or misdn afaik |
| 08:14.01 | justdave | the "supported by Linux" addendum on the end of one of them implies if you have the drivers for the device... |
| 08:15.00 | justdave | there's an asterisk-chan-misdn package |
| 08:15.19 | hank | yes i know but misdn sucks atm and i was recommended to use bristuff some days ago here. |
| 08:20.53 | hank | 'sigh' |
| 08:22.10 | hank | are there any isdn cards known to work out of the box with debian, a standard debian kernel and the debian asterisk package? |
| 08:23.06 | deeLer | hank : fritz card isdn (avn) |
| 08:23.39 | deeLer | someone willing to assist me with MP3 playback on asterisk ... i hear no sound at all , tried different MP3 bitrates |
| 08:24.25 | hank | 'sigh' i dont like this company... but ill go try. thx |
| 08:26.10 | *** join/#asterisk Jason559-55 (n=Jason@adsl-75-10-122-136.dsl.frs2ca.sbcglobal.net) |
| 08:27.01 | Jason559-55 | I was wondering if anyone can help me diagnose the reason I'm unable to test my asterisk system by trying to call from one extension to another via soft phones and it "times out"... |
| 08:27.37 | deeLer | <PROTECTED> |
| 08:27.37 | deeLer | Sep 20 12:21:57 NOTICE[5814]: app_mp3.c:111 timed_read: Poll timed out/errored out with 0 |
| 08:28.00 | deeLer | mpg123 plays this thing without trouble tru the pc speakers |
| 08:28.26 | deeLer | i'd like mp3 music on hold :) !! |
| 08:31.28 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 08:31.46 | Jason559-55 | nobody? |
| 08:34.40 | *** join/#asterisk laichzeit (n=immortal@dsl-145-164-35.telkomadsl.co.za) |
| 08:35.53 | laichzeit | Hi, can anyone refer me to documentation or something explaining how I can adjust the volume of calls going over PSTN (SIP phone has no handset-in volume adjustment)? |
| 08:36.08 | laichzeit | I already have TXgain to 3.0 |
| 08:46.18 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:48.10 | *** join/#asterisk linagee (n=na@unaffiliated/linagee) |
| 08:49.01 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 08:49.10 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 08:51.01 | linagee | NICE. lol |
| 08:51.12 | linagee | who says trixbox is bad. lol |
| 08:51.18 | linagee | i just installed it in 9 minutes! :-D |
| 08:51.31 | X-Rob_ | Uh |
| 08:51.32 | Jason559-55 | I'm still trying to figure out trixbox |
| 08:51.33 | X-Rob_ | I say trixbox is bad |
| 08:51.38 | X-Rob_ | and I wrote freepbx |
| 08:51.40 | X-Rob_ | 8) |
| 08:51.46 | linagee | X-Rob_: 9 minutes. :-D isn't that awesome? lol |
| 08:51.47 | X-Rob_ | Well trixbox 1.2 is awful, anyway |
| 08:51.58 | linagee | X-Rob_: vmware on a super fast server, using the .ISO to boot from. :-D |
| 08:52.08 | Jason559-55 | I would use asterisk, but I need a helpful installer, anyone know of one? |
| 08:52.11 | X-Rob_ | linagee, well, you still have to recompile asterisk and zaptel |
| 08:52.16 | linagee | X-Rob_: not at all |
| 08:52.20 | X-Rob_ | No, you do, really. |
| 08:52.24 | X-Rob_ | it's broken in trixbox 1.2 |
| 08:52.34 | linagee | X-Rob_: trixbox 1.2 is fscked... :( |
| 08:52.43 | X-Rob_ | Are you a bot? |
| 08:52.44 | linagee | X-Rob_: that's why i just did a reinstall. i just made an instance of 1.1 |
| 08:52.52 | linagee | X-Rob_: no |
| 08:53.17 | X-Rob_ | Disagreeing with me in one sentance and then totally changing your tune and agreeing with me in the next is somewhat disconcerting. |
| 08:53.20 | linagee | X-Rob_: you said asterisk and zaptel. my SIP phone can't even log in for some odd reason.. |
| 08:53.35 | X-Rob_ | linagee, yes. because asterisk and zaptel is broken in trixbox 1.2 |
| 08:53.40 | linagee | lol. |
| 08:53.49 | linagee | X-Rob_: who did that? heh |
| 08:54.01 | X-Rob_ | agillis, the guy who put trixbox together. |
| 08:54.20 | linagee | X-Rob_: so the best thing to do is just use trixbox 1.1 and update to the latest |
| 08:54.26 | X-Rob_ | Yeah. |
| 08:54.32 | X-Rob_ | Don't forget to upgrade your freepbx |
| 08:54.43 | linagee | yep. 2.1.2? |
| 08:54.45 | *** part/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 08:55.03 | X-Rob_ | yeah |
| 08:55.17 | linagee | you use freepbx? :-O |
| 08:55.19 | sxpert-work | morning |
| 08:55.21 | X-Rob_ | I wrote it |
| 08:55.21 | X-Rob_ | 8) |
| 08:55.25 | linagee | X-Rob_: cool |
| 08:55.28 | X-Rob_ | well |
| 08:55.32 | linagee | helped? |
| 08:55.37 | X-Rob_ | I wrote the later bits of it anyway. I'm the lead developer. |
| 08:56.18 | linagee | i guess you can still say trixbox in under an hour. if you count updating stuff, setting up extensions, educating the user. :) |
| 08:56.43 | *** join/#asterisk Cyt (n=danielcy@athedsl-121035.otenet.gr) |
| 08:57.15 | linagee | X-Rob_: i think most people in here hate freepbx. they are masochists or something |
| 08:57.30 | X-Rob_ | the ones that hate it are the ones that don't understand asterisk |
| 08:57.46 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 08:57.51 | X-Rob_ | just because freepbx has a convoluted (but well documented) dialplan, they dont' understand it. And people fear what they don't understand. |
| 08:58.00 | linagee | i think freepbx + knowing how to manually edit extensions.conf and such is the most powerful way to use asterisk. |
| 09:00.47 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 09:02.41 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 09:04.52 | laichzeit | freepbx just needs some technical documentation |
| 09:05.24 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 09:09.04 | *** join/#asterisk bmg505 (n=leon@c1-19-10.rndf.isadsl.co.za) |
| 09:09.50 | *** join/#asterisk S^P (n=ss@fiber-gw3-stc-39.brain.net.pk) |
| 09:13.38 | deeLer | i've installed trixbox in 30 mins or so .... once i got it up and running, i'd wanted to do the same with a install from scratch on debian .... you learn more with a vi and all the config files, but it's harder, no doubt.... depends on what you want |
| 09:17.06 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:17.45 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 09:21.04 | *** join/#asterisk SToRM1k (n=anunah@80.92.19.27) |
| 09:23.54 | SToRM1k | hi, tell me, please, what status query format for manager API in asterisk v.1.2.8?? |
| 09:25.52 | RoyK | ~docs |
| 09:25.54 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:25.54 | RoyK | ~book |
| 09:25.57 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:25.57 | RoyK | ~rtfm |
| 09:26.00 | jbot | rtfm is, like, Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 09:26.53 | *** join/#asterisk gardo (n=gardo@124.104.34.199) |
| 09:26.59 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:27.35 | SToRM1k | thx |
| 09:29.12 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 09:29.23 | zouzou | hi |
| 09:29.28 | zouzou | i need some help |
| 09:29.39 | zouzou | ? |
| 09:30.59 | deeLer | just ask |
| 09:31.30 | linagee | what does 488 on the phone mean? |
| 09:32.09 | deeLer | where is that '488' ? |
| 09:32.15 | linagee | on a grandstream's screen |
| 09:32.25 | zouzou | i m not able to connect asterisk to a postgres database |
| 09:32.32 | linagee | :( :( |
| 09:32.36 | zouzou | i got the message |
| 09:32.39 | linagee | 488 = bad codec. damn |
| 09:32.52 | zouzou | could not connect to the database server conection refused |
| 09:32.56 | linagee | i removed PCM from my codec pick list. the 488 went away when i added it back. :( |
| 09:33.14 | deeLer | zouzou, check the acl's on your postgres |
| 09:33.16 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 09:33.17 | linagee | stoopid reliable PCM. what with it's bloat and all. :( |
| 09:33.28 | zouzou | is the server accepting tcp/ip connection on port 5432? |
| 09:34.05 | linagee | i know that voicepulse is supposed to support at least iLBC, why isn't that working? :( |
| 09:34.20 | zouzou | i can connect to the postgre through the basho by typing psql command |
| 09:34.23 | SToRM1k | what is "ActionID: blah"? |
| 09:35.20 | linagee | does IAX2 only support PCM? :( |
| 09:36.36 | sxpert-work | linagee: IAX2 supports anything |
| 09:36.47 | sxpert-work | it's a container protocol |
| 09:37.00 | linagee | sxpert-work: then why isn't this working. :( |
| 09:37.08 | linagee | sxpert-work: voicepulse supports iLBC, right? |
| 09:37.09 | sxpert-work | what do I know ? |
| 09:37.45 | linagee | sxpert-work: it says so on their page: http://connect.voicepulse.com/Specifications.aspx |
| 09:38.19 | linagee | sxpert-work: so why do i get a 488 on the phone when i try iLBC? :( |
| 09:40.06 | fourcheeze | linagee: does the user have it enabled? |
| 09:40.10 | linagee | damnit. :( |
| 09:40.17 | linagee | fourcheeze: user have it enabled? |
| 09:40.33 | linagee | in voicepulse's interface? i don't think there |
| 09:40.36 | linagee | 's an option |
| 09:40.52 | linagee | or in the extension button on freepbx? |
| 09:41.29 | *** join/#asterisk arcy (n=arcanum@ppp4-6.adsl.forthnet.gr) |
| 09:42.08 | fourcheeze | linagee: I've got no idea about your setup, but you need to check which codecs are available to the client in iax.conf |
| 09:42.17 | linagee | sec |
| 09:42.58 | linagee | there's three. ulaw, alaw, and gsm. wow that's silly. hah |
| 09:43.33 | E-bola | Anybody have an example dialplan where u set the callerid/number before u dial out? |
| 09:43.58 | linagee | interesting... there's a voicepulse section, it should pick that up... |
| 09:44.39 | linagee | fourcheeze: it still hates me. :( |
| 09:44.50 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 09:45.09 | fourcheeze | hmm |
| 09:45.18 | fourcheeze | so phone connects to * and * connects to voicepulse? |
| 09:45.22 | linagee | fourcheeze: it's in there though |
| 09:45.26 | linagee | fourcheeze: yep |
| 09:45.33 | fourcheeze | which is saying 488? |
| 09:45.36 | linagee | fourcheeze: i added it in the general section also |
| 09:45.39 | sxpert-work | what is 488 ? |
| 09:45.40 | linagee | fourcheeze: the phone says 488 |
| 09:45.43 | linagee | not sure |
| 09:45.45 | linagee | an error message? |
| 09:45.57 | linagee | it goes away when i tell the phone it can use ulaw/alaw |
| 09:45.58 | fourcheeze | check the console when you make a call |
| 09:46.16 | fourcheeze | sxpert-work: 488 is codec not available |
| 09:46.24 | sxpert-work | ah... |
| 09:46.34 | linagee | sip debug? |
| 09:46.38 | linagee | yikes. lotsajunk |
| 09:46.53 | fourcheeze | shouldn't need to have debug on |
| 09:47.15 | fourcheeze | are you in console mode? asterisk -rvvvvvvvvvvvvvvvvvvvc ? |
| 09:47.37 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:47.46 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:47.52 | backblue | hi* |
| 09:47.57 | stoffell | anyone experience with VoIP assessment tools? more specifically, NetIQ's products.. ? |
| 09:48.12 | linagee | that's weird... |
| 09:48.16 | backblue | anyone know which it's the name of the siemens rdis phones? |
| 09:48.21 | linagee | fourcheeze: voicepulse is telling me this: "Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x511 (g723|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x0 (nothing)" |
| 09:48.24 | backblue | ups, not rdis |
| 09:48.34 | linagee | so i have to tell it peer somewhere. |
| 09:48.37 | linagee | instead of us.... (?) |
| 09:49.00 | linagee | 488 means "not acceptable here" |
| 09:49.01 | linagee | "SIP/2.0 488 Not acceptable here" |
| 09:49.08 | SToRM1k | Why Asterisk doesnt reply to my Status Query in Manager? |
| 09:50.00 | linagee | er, wait, my iax.conf says "type=peer" |
| 09:51.13 | sxpert-work | shouldn't that say "friend" or something ? |
| 09:51.41 | linagee | "The SIP gateway generates this response if media negotiation fails. " |
| 09:51.54 | linagee | i guess media negotiation failed... weird. |
| 09:52.23 | hank | i am using debian unstable asterisk and asterisk-chan-capi. i followed the readme coming the asterisk-chan-capi and now asterisk wont start anymore giving this message: [chan_capi.so]Sep 20 11:51:50 WARNING[12147]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_pickup_call |
| 09:52.32 | linagee | wtf???? per google. "Voicepulse _has_ ilbc turned on, but it will only work if I disallow GSM. |
| 09:52.32 | linagee | " |
| 09:52.36 | linagee | that's just silly! lol |
| 09:53.17 | *** join/#asterisk kaimis (n=pwn@80.233.233.240) |
| 09:53.35 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 09:54.29 | sxpert-work | linagee: voicepulse's setup is borken |
| 09:54.55 | sxpert-work | linagee: only allow ilbc for that particular peer :D |
| 09:55.16 | linagee | nope. i took out gsm. still doesn't work. :-/ |
| 09:55.28 | backblue | hank: missing any library? ldd /usr/lib/asterisk/modules/chan_capi.so |
| 09:55.28 | sxpert-work | hmmm... |
| 09:55.31 | backblue | and check it. |
| 09:55.35 | linagee | sxpert-work: yeah, i guess i could try ilbc only |
| 09:55.40 | fourcheeze | linagee: first off try to make it work within * |
| 09:55.49 | linagee | sxpert-work: i just want something other than pcm. |
| 09:55.52 | linagee | anything! :( |
| 09:55.55 | fourcheeze | try listening to music on hold or recording a message with ilbc |
| 09:56.00 | sxpert-work | linagee: try using the * console to call out |
| 09:56.02 | fourcheeze | check that it's actually using it |
| 09:56.04 | linagee | pcm sucks. it eats like 22 kilobytes/sec or something |
| 09:56.14 | fourcheeze | more like 80 |
| 09:56.17 | linagee | sxpert-work: how do i do that? no sound card |
| 09:56.22 | linagee | fourcheeze: 80 kilobytes? |
| 09:56.27 | sxpert-work | linagee: you only have sip stuff ? |
| 09:56.28 | linagee | are you sure not 80 kiloBITS? |
| 09:56.29 | sxpert-work | ouch |
| 09:56.31 | fourcheeze | ahh yeah |
| 09:56.35 | fourcheeze | kbits/sec |
| 09:56.41 | linagee | :) |
| 09:56.49 | backblue | not 80, it's 64. |
| 09:57.00 | linagee | sxpert-work: connection to voicepulse is iax2. phone is sip. (grandstream POS) |
| 09:57.23 | sxpert-work | hmmm... |
| 09:57.30 | linagee | backblue: i think the doc says something like, "stated 64, could jump to 80 just to screw with you" |
| 09:57.38 | linagee | (on voip-info) |
| 09:57.39 | backblue | hum ok |
| 09:57.44 | backblue | whatever |
| 09:57.51 | linagee | in more technical terms of course. :) |
| 09:58.10 | sxpert-work | linagee: in fact, it's 64kbit/s for the voice data itself, and 80 when you count rtp overhead |
| 09:58.17 | linagee | aha. there you go |
| 09:58.27 | linagee | sxpert-work: that sucks. that's a LOT of overhead. lol |
| 09:58.31 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 09:58.34 | *** join/#asterisk hotroot (n=michael@pD9E95FB3.dip.t-dialin.net) |
| 09:58.35 | linagee | 16 kilobits? hah |
| 09:58.35 | backblue | well overhead you will have in all of them. |
| 09:58.36 | zouzou | what s the best free GUI to manage Asterisk????? |
| 09:58.39 | sxpert-work | linagee: that's why there's IAX trunks :D |
| 09:58.53 | backblue | zouzou: CLI. |
| 09:59.02 | linagee | you could fit VOICE with some codecs in 16 kilobits. lol |
| 09:59.08 | zouzou | ;-) |
| 09:59.08 | sxpert-work | linagee: in which multiple channels are sent in the same rtp packet |
| 09:59.14 | backblue | sxpert-work: they suck |
| 09:59.17 | backblue | i hate g729 |
| 09:59.25 | sxpert-work | backblue: not talking about g729 |
| 09:59.29 | fourcheeze | backblue: why do you hate it? |
| 09:59.39 | sxpert-work | because it's non free is a good start |
| 09:59.43 | backblue | because, it makes me loose quality. |
| 09:59.50 | hank | backblue: hm linux-gate.so.1 => (0xffffe000) is the only filename i couldnt find on that system. |
| 09:59.51 | sxpert-work | that too |
| 09:59.56 | fourcheeze | anyone using g726-32 ? |
| 10:00.00 | backblue | hank: so, you have a problem. |
| 10:00.04 | sxpert-work | my voip provider is ALAW only... |
| 10:00.14 | backblue | and in a provider, you cannot loose quality. |
| 10:00.25 | sxpert-work | which is not a problem, as I get 13M/s down, 1M/s up ADSL2+ |
| 10:00.29 | backblue | sxpert-work: that it's bad, it should suport all |
| 10:00.39 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 10:00.46 | backblue | and give the correct suport to the client, and explain, why use which. |
| 10:01.02 | hank | backblue: uhm afaict not really: http://lists.us.dell.com/pipermail/linux-poweredge/2004-October/016899.html |
| 10:01.08 | kaimis | hello, I want to know about asterisk gui configuration tools, any suggestions? |
| 10:01.21 | hank | backblue: it says this library is virtual and provided by the 2.6 kernel. one cannot see it. |
| 10:01.44 | sxpert-work | backblue: the voip provider is the same company as the DSL provider which will soon be installing Gbit ethernet over fiber to everyone :D |
| 10:01.46 | backblue | so, search where that symbol should be. |
| 10:01.52 | backblue | have you done, depmod -ea |
| 10:01.53 | backblue | ? |
| 10:02.02 | linagee | wtf?? :( |
| 10:02.09 | backblue | sxpert-work: ehehe, lucky guy. |
| 10:02.23 | linagee | does this phone support anything but ulaw/alaw? does it lie about it's options on the pulldown? |
| 10:03.04 | backblue | linagee: why dont you contact the phone manufacture, and ask him? |
| 10:03.08 | sxpert-work | backblue: heh |
| 10:03.27 | hank | backblue: how do i search for a symbol? i have done it now but asterisk still doesnt start. (in case thats what the command was supposed to do) |
| 10:03.49 | sxpert-work | backblue: they'll be investing 1B eur in the next 5 years for that (the stock market didn't like it, but they don't give a shit, the owner owns 66% of the company) |
| 10:04.11 | backblue | sxpert-work: what country are you? |
| 10:05.48 | sxpert-work | .fr |
| 10:06.06 | sxpert-work | provider is http://adsl.free.fr (site in french, obviously) |
| 10:06.09 | backblue | hank: i think ast_pickup_call, should be on the asterisk librarys, or this is capi modules related |
| 10:06.28 | backblue | sxpert-work: well, i'm from portugal, and we have not that luck. |
| 10:07.04 | sxpert-work | backblue: time to start a similarly modeled provider :D |
| 10:07.33 | backblue | sxpert-work: we have adsl2+ here, to 24Mb. |
| 10:07.36 | sxpert-work | backblue: instill some competition in the market there... |
| 10:09.46 | linagee | can't the phone talk ulaw to my asterisk box and the asterisk box convert it to a better protocol? |
| 10:11.51 | fourcheeze | sure |
| 10:12.10 | fourcheeze | backblue: what upload speeds do you have? |
| 10:12.24 | backblue | fourcheeze: where? |
| 10:12.53 | backblue | hank: why are you trying to use capi? |
| 10:13.11 | fourcheeze | backblue: upload speeds on your adsl2 |
| 10:13.33 | fourcheeze | here we can get 8Mb downloads but only about 800kbit upload |
| 10:13.44 | fourcheeze | and that's if you're very lucky |
| 10:13.47 | backblue | fourcheeze: hooo, if it is 24Mb, it has 4Mb upload i think. |
| 10:13.54 | hank | backblue: no special reason... what would you recommend? |
| 10:14.06 | backblue | hank: what card do you have? |
| 10:14.48 | hank | backblue: i have the choice: a billion usb adapter using a hfcsusb chipset or a fritz A1. |
| 10:15.09 | backblue | hank: what do you need? BRI? PRI? |
| 10:15.11 | hank | backblue: atm im trying with the fritz because i was told it'd work with debian unstable and debian asterisk out of the box. |
| 10:16.28 | hank | backblue: uhm... i know what those acronyms mean but i had no idea i could choose. eventually i want to setup a system to handle some of our business' calls. but atm i just want to get some experience with asterisk. |
| 10:16.48 | *** join/#asterisk Cyt (n=danielcy@athedsl-121035.otenet.gr) |
| 10:16.57 | backblue | hank: go with hfc cards, and use misdn |
| 10:17.21 | hank | yeeehaaa great lets go in a circle and go back where i started... |
| 10:17.53 | hank | backblue: let me just plugin the other computer again. i already have a kernel patched with misdn... |
| 10:17.56 | backblue | well, if you want to use that, go heade, but i cant help you more than i did, since i have never used suck a system. |
| 10:18.05 | nitrus^ | what text-to-speech engines work with asterisk? |
| 10:18.12 | backblue | kernel patched? misdn it's modules. |
| 10:18.23 | backblue | you can use it, in every kernel. |
| 10:18.39 | hank | backblue: its kernel modules. so one can build them in-tree |
| 10:19.18 | hank | backblue: so you cant help me with misdn? |
| 10:19.23 | backblue | misdn i can |
| 10:19.24 | stoffell | backblue: as you told me a few days ago.. i also used misdn to test.. and i'm gonna get rid of my bristuff as soon as possible :p |
| 10:19.27 | backblue | i use it every day. |
| 10:19.41 | hank | ah cool :) its pretty fscking bad documented imho... |
| 10:19.46 | backblue | stoffell: well done! ;) |
| 10:20.09 | stoffell | backblue: still in testing, but after i get the iaxmodem+hylafax config, i'll do it in production also :) |
| 10:20.13 | backblue | hank: misdn? yes, it is. |
| 10:20.45 | backblue | i have a big provider solution i have made with iaxmodem + hylafax + asterisk + misdn |
| 10:20.52 | backblue | it works like a charm. |
| 10:21.05 | hank | nice :) |
| 10:21.33 | hank | ok i have the modules loadable. do i need anything else for misdn to work? |
| 10:21.48 | stoffell | backblue: nice, i'm using iaxmodem+hylafax in small environment, works nice.. |
| 10:21.59 | stoffell | backblue: will it work even "better" with misdn instead of bristuff? |
| 10:21.59 | *** join/#asterisk s0lid (n=jlq@61.28.130.145) |
| 10:22.00 | hank | ok, asterisk-chan-misdn of course ;) |
| 10:22.38 | backblue | hank: you say it lacks documentation, but you have not read, the one it is online, it says everything you need, it only dont speaks with the last new options. |
| 10:23.02 | backblue | you have /etc/init.d/misdn-init? if yes, do /etc/init.d/misdn-init scan |
| 10:23.11 | backblue | and see if it detects your card. |
| 10:23.43 | *** join/#asterisk sakimustafa (n=sakimust@202.133.14.226) |
| 10:24.02 | backblue | stoffell: i have made some developing in that solution, because we have a billing system on the top of it, that it's the same billing system for voice & faxing. |
| 10:24.24 | backblue | it has some improvements to work our way. |
| 10:24.42 | backblue | but in a small enviorment, it's easy to install, and easy to use. |
| 10:25.00 | backblue | and you dont need all my developing to do it. |
| 10:25.31 | stoffell | nice, the only thing i'm not (yet) sure of, if it will work as good, or even better, then bristuff? |
| 10:25.34 | nitrus^ | can loquendo be used with asterisk as a TTS enginer? |
| 10:25.48 | backblue | hank: you have installed misdn from apt? maybe stoffell can help you in that part, since he has done it, i will take my breakfast now. |
| 10:25.59 | hank | backblue: i have read it but not taken it seriously because it talks about 2.6.0 |
| 10:26.08 | stoffell | i have read also that you can use misdn parameters in the dialplan to do no echo-cancellation, needed for fax, so looks like it's gonna be okay |
| 10:26.29 | backblue | stoffell: it will work, just like the same, faxing it's not a problem with bristuff. |
| 10:26.43 | backblue | the problem with bristuff, it's his bad implementation of isdn protocol. |
| 10:26.58 | backblue | hank: and you are not using 2.6? |
| 10:27.14 | stoffell | backblue: ah, and could that "bad implementation" be the reason of "some" problems here and there? (like in seldom situations no good faxes, etc..) |
| 10:27.33 | backblue | stoffell: yes, you need, i can help you that too. !e:vt0 i think. |
| 10:27.40 | backblue | but i have to check |
| 10:28.50 | backblue | stoffell: i have not gone so far with bristuff, bristuff take me so much troubles in the begining, i started looking a solution, to that, and starting speak with beronet, and the main developer of misdn, and see how much better it is. |
| 10:29.11 | backblue | i use misdn for every all our implementations here |
| 10:29.29 | backblue | even E1 cards, i use them with misdn, no zaptel, only for ztdummy :) |
| 10:29.37 | hank | backblue: i am but do you know how old 2.6.0 is? |
| 10:29.49 | stoffell | ah, even for E1's you use mIsdn? lol, didn't know that was possible |
| 10:29.59 | backblue | hank: they speak about 2.6.0, as a release 2.6, not 2.6.0 :P |
| 10:30.21 | backblue | stoffell: yes they have a quite good E1 cards. |
| 10:31.05 | stoffell | backblue: i only used the "older" 1port E1's of digium (without echo can) untill yet, so not so much experience in that |
| 10:31.19 | backblue | the misdn driver it's groing up, i have see, and helped to correct many crucial bugs, but they get corrected, and very good corrected, so its ok. |
| 10:31.22 | stoffell | (also purchased them at beronet, though) |
| 10:31.25 | x86 | http://pastebin.ca/177306 |
| 10:31.36 | stoffell | indeed |
| 10:31.52 | backblue | stoffell: E1 with echo cancel onboard, are bad for faxing. |
| 10:32.02 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 10:32.07 | backblue | i dont use E1 cards from digium, i use the ones from beronet. |
| 10:32.12 | stoffell | backblue: hm, oh yes, that's true... |
| 10:32.36 | backblue | that's why, beronet cards dont have echo cancel onboard :P |
| 10:32.47 | stoffell | loool x86 |
| 10:32.55 | x86 | stoffell: haha |
| 10:32.57 | stoffell | tnx for sharing that :p |
| 10:33.01 | x86 | np |
| 10:33.25 | x86 | thought some people might find humor in that :P |
| 10:33.33 | hank | hmpf where do i get misdn-init for debian? |
| 10:33.38 | x86 | i was laughing so hard i could hardly breathe ;) |
| 10:33.43 | stoffell | x86: sure do ;) |
| 10:34.00 | backblue | hank: http://www.beronet.com/downloads/install-asterisk.tar.gz |
| 10:34.04 | backblue | use this script on debian |
| 10:34.22 | stoffell | backblue: misdn is even possible on astlinux, wich is also nice.. |
| 10:34.26 | backblue | and you will get EVERYTHING installed, including asterisk and all the famaly |
| 10:34.42 | backblue | if you only want misdn stuff, use this one -> http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz |
| 10:35.08 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 10:35.16 | backblue | stoffell: i dont care and i dont even know astlinux |
| 10:35.18 | backblue | :) |
| 10:35.33 | backblue | if it is good, it as to work in everywhere. |
| 10:35.59 | backblue | btw, dont use hfc cards with SiS chipsets. you will get troubles. |
| 10:36.11 | stoffell | backblue: hm, where's a good place to buy hfc based cards? |
| 10:36.19 | backblue | stoffell: beronet. |
| 10:36.27 | stoffell | backblue: ok, thanks.. |
| 10:36.40 | backblue | or, you can buy us! :) |
| 10:37.10 | backblue | and use the suport :P |
| 10:37.11 | stoffell | where are you located anyway? |
| 10:37.15 | backblue | stoffell: portugal. |
| 10:37.27 | stoffell | backblue: ah, i'm in belgium |
| 10:38.10 | stoffell | backblue: so you use the hfc singple s0 card of beronet? (38 euro's) |
| 10:39.01 | *** join/#asterisk darkpadden (n=wes@adsl-066-156-076-204.sip.asm.bellsouth.net) |
| 10:40.40 | hank | 'grml.org' |
| 10:41.08 | hank | aha now it compiles... |
| 10:42.30 | RoyK | http://karlsbakk.net/fun/detention.gif |
| 10:44.23 | hank | lol |
| 10:45.06 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 10:46.38 | hank | 'grml.org' why does this fscking script want to write to /usr/include/asterisk??? |
| 10:47.50 | s0lid | what do you mean by mkintf: Channel 16 is reserved for D-channel? |
| 10:48.06 | s0lid | my carrier said that my d-channel should be 17 so i set the dchannel to 17 |
| 10:48.06 | RoyK | s0lid: not reserved. usually used for the dchan |
| 10:48.20 | RoyK | that is quite unusual |
| 10:48.25 | RoyK | 16 is normal |
| 10:48.35 | s0lid | RoyK: i got this error from asterisk console |
| 10:48.42 | s0lid | RoyK: i ran it asterisk -cvg |
| 10:48.45 | RoyK | try to use 16, then |
| 10:48.47 | RoyK | ~pb |
| 10:48.52 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 10:49.01 | s0lid | RoyK: the problem is my carrier set the dchannel to 17 |
| 10:49.06 | s0lid | RoyK: i'll try thaty wit |
| 10:49.07 | hank | ok too much now... fsck those debian asterisk packages... |
| 10:49.31 | RoyK | hank: apt-get remove --purge asterisk |
| 10:49.35 | RoyK | and install from sources |
| 10:49.43 | hank | ack thats what im gonna do |
| 10:49.50 | hank | it was worth a try |
| 10:50.32 | s0lid | RoyK: this what i got from dchannel 16 |
| 10:50.33 | s0lid | mkintf: Signalling requested on channel 16 is PRI Signalling but line is in Unknown signalling 896 signalling |
| 10:52.13 | s0lid | RoyK: ok i forgot to configure the channel in zapata.conf it's now working but i got this error |
| 10:52.14 | s0lid | <PROTECTED> |
| 10:52.33 | *** join/#asterisk Cyt (n=danielcy@athedsl-17987.otenet.gr) |
| 11:00.53 | RoyK | s0lid: no idea |
| 11:00.56 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 11:02.08 | backblue | anyone here, knows the protocolo from the siemens gigaset phones? |
| 11:10.43 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:15.18 | s0lid | RoyK: ok thanks bro |
| 11:16.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:17.50 | Dr-Linux | anybdoy ever use Gammu SMS program? |
| 11:26.28 | *** join/#asterisk _Problem_ (n=lokesh@estrela.nortenet.pt) |
| 11:31.23 | *** join/#asterisk jrprado (i=jrprado@201.47.13.55.adsl.gvt.net.br) |
| 11:31.34 | *** join/#asterisk bmg505 (n=leon@c1-218-7.rndf.isadsl.co.za) |
| 11:34.16 | hank | ok that is done... |
| 11:34.32 | hank | now i probably need documentation for /etc/asterisk/misdn.conf, right |
| 11:34.41 | hank | s/$/?/ |
| 11:34.49 | hank | s/$/\?/ |
| 11:41.58 | *** join/#asterisk jraprado (i=jrprado@201.47.13.55.adsl.gvt.net.br) |
| 11:44.17 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 11:55.35 | *** part/#asterisk jraprado (i=jrprado@201.47.13.55.adsl.gvt.net.br) |
| 11:57.40 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 12:00.04 | *** join/#asterisk InforSOLutions (i=HydraIRC@201.47.13.55.adsl.gvt.net.br) |
| 12:00.29 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 12:00.35 | hank | hmm asterisk couldnt load chan_oss.so because there is no oss.conf. where do i get one? |
| 12:01.08 | backblue | hank: dont load that. |
| 12:01.18 | backblue | forget oss you probably dont need it |
| 12:01.34 | hank | backblue: how do i tell asterisk not to load it? |
| 12:01.46 | backblue | edit modules.conf |
| 12:01.51 | backblue | and read the file |
| 12:02.51 | hank | 'g' ok where do i get a modules.conf? ;) |
| 12:03.20 | DarKnesS_WolF | hank: did u make examples? |
| 12:03.36 | hank | ah duh... yep just saw that myself. sorry |
| 12:04.10 | zouzou | if i have to reinstall asterisk |
| 12:04.27 | zouzou | can i just copy back the old configuration files? |
| 12:04.41 | backblue | zouzou: same release? |
| 12:04.45 | zouzou | yes |
| 12:04.50 | backblue | so, why not? |
| 12:05.05 | zouzou | i need to install the zaptel module |
| 12:05.18 | backblue | so do it |
| 12:05.20 | zouzou | i didn t install at the first time |
| 12:05.47 | zouzou | i reda that i have to reinstall asterisk if i want to install zaptel |
| 12:06.19 | zouzou | ? |
| 12:06.23 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 12:07.19 | hank | do i need chan_phone? |
| 12:07.45 | hank | hm i think i dont |
| 12:08.08 | s0lid | how come i can't change my d-channel to 17? |
| 12:08.19 | s0lid | when i do this i get this error -channel 16 is not in HDLC/FCS mode. See /etc/zaptel.conf |
| 12:08.39 | s0lid | my provider set their dchannel to 17 that's why i need to set it to 17 |
| 12:08.44 | s0lid | is there a way to tweak this? |
| 12:08.54 | drray | it can be whatever you set it to |
| 12:09.24 | hwt | hi. |
| 12:10.11 | hwt | i have several interfaces that there is traffic on. but when an invite comes in on one of the other interfaces, it always shows the primary IP address. |
| 12:10.16 | hwt | anyone know how to fix this? |
| 12:10.19 | hwt | SIP only. |
| 12:12.20 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 12:12.46 | *** join/#asterisk gardo (n=gardo@124.104.34.199) |
| 12:13.15 | hwt | maybe setting defaultip will help |
| 12:13.36 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:14.10 | hank | chan_misdn.c:4074 load_module: Unable to initialize mISDN |
| 12:14.11 | hank | any ideas? 'sigh' |
| 12:14.44 | hank | that comes directly before the error and after asterisk loading chan_misdn: mISDN_close: fid(9) isize(131072) inbuf(0xa76ac008) irp(0xa76ac008) iend(0xa76ac008) |
| 12:20.14 | hwt | hm, no, that's not right. |
| 12:21.03 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 12:22.28 | mattfletcher | hello all, i am trying to set up a tdm400p with two fxs and two fxo ports. at the moment i only care about getting one pstn line connected up. i am having real problems getting it to go though, the lights refuse to come on, even though i believe i have installed everything correctly |
| 12:23.41 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:23.55 | mattfletcher | I followed the instructions at http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu |
| 12:23.57 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 12:26.15 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:27.27 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 12:27.31 | mut | i billed like 200 customers for all their tollfree calls |
| 12:28.16 | mattfletcher | I saw the instruction to add the line KERNEL="zap0-9*", NAME="zap/%n" to a udev configuration file. Is this line correct, or do the numbers/star need expanding to something? |
| 12:29.22 | Druken | mut: whoops... |
| 12:29.44 | mut | yea vrry woops |
| 12:30.29 | Druken | i'm sure ya got a few calls... WTF am i paying for TOLLFREE calls for? cancle my fucken service! |
| 12:31.18 | mut | yea |
| 12:31.28 | mut | no one threatened to cancel tho |
| 12:31.44 | Druken | :) well, that's good |
| 12:31.46 | mut | phone service for $8/month and free intralata calling ya can't complain too much |
| 12:32.15 | Druken | how in hell can you afford to do it for $8 a month? |
| 12:33.22 | Druken | unless the $8 a month is simply for account maint. |
| 12:40.58 | nortex | Has anyone had problems with making Zaptel SVN r 1470? I get errors in strcompat.c and want to see if is just me. |
| 12:48.08 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 12:49.43 | *** join/#asterisk wirehead_ca (n=ads@mail.tethco.com) |
| 12:49.52 | wirehead_ca | hello all |
| 12:50.03 | wirehead_ca | i'm knee deep in a IAX2 'oddity' |
| 12:50.18 | wirehead_ca | anyone care to throw their thoughts in? |
| 12:51.38 | wirehead_ca | i'm able to register with my IAX2 provider upon bootup no problem. But if the 60 second timeout is reached and the asterisk machine gets unregistered, only a reboot will re-register with my provider. I have a fixed ip, and the proper port forwarding setup. My provider is running 1.2.10 and i'm running 1.2.11 (trixBox 1.2) |
| 12:51.58 | wirehead_ca | restarting asterisk with the /etc/init.d won't do it |
| 12:52.14 | wirehead_ca | i can watch the attempts on the ppp0 interface of my linksys with tcpdump |
| 12:52.19 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 12:52.24 | wirehead_ca | my provider says its my end |
| 12:52.36 | wirehead_ca | i'm stumped |
| 12:52.53 | wirehead_ca | i've been on this two weeks now beating it, trying various trixboxes ,and freePBX installs/upgrades |
| 12:52.57 | wirehead_ca | is anyone here? |
| 12:53.13 | wirehead_ca | hello? |
| 12:53.19 | tzanger | wirehead_ca: you need packet dumps |
| 12:53.27 | tzanger | showing you trying to register and not getting a response |
| 12:53.39 | tzanger | and also showing that the (re) register packet that is failing is identical to the register packet that works upon bootup |
| 12:53.56 | wirehead_ca | understood |
| 12:53.58 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 12:54.03 | tzanger | then give that information to your provider and saying "can you tell me why #2 doesn't work where #1 does, when it looks identical?" |
| 12:54.15 | wirehead_ca | my providers only response to is offer us a SIP account instead |
| 12:54.18 | wirehead_ca | heh |
| 12:54.32 | wirehead_ca | how much worse is SIP for a 5 line trunk vs. IAX? |
| 12:54.47 | wirehead_ca | i'm not entirely confident in their ability to support IAX2 |
| 12:55.00 | tzanger | wirehead_ca: who's that? |
| 12:55.04 | wirehead_ca | is what i'm experiencing common? rare? (WITH iax) |
| 12:55.09 | wirehead_ca | they are... unlimitel.ca |
| 12:55.15 | wirehead_ca | they have example IAX configs etc |
| 12:55.27 | tzanger | unlimitel is a top-notch company, I have several DIDs with them and terminate on-net to them daily |
| 12:55.30 | wirehead_ca | the guys in the freepbx forums said their examples are out of date etc, using non-existing parameters |
| 12:55.31 | tzanger | I wonder what's going on there |
| 12:55.32 | Druken | lots of people here use them... |
| 12:55.33 | Druken | hehe |
| 12:55.39 | tzanger | but yes definitely provide packet dumps and see |
| 12:55.39 | wirehead_ca | tzanger |
| 12:55.49 | wirehead_ca | tzanger you using IAX with them or SIP? |
| 12:55.56 | tzanger | perhaps ask them what version of asterisk they are using and if they're using anything custom in their chan_iax2 |
| 12:55.59 | tzanger | wirehead_ca: both |
| 12:56.09 | wirehead_ca | so the freePBX guys blew smoke up my A$$ about them, they seemed okay to me hence my choosing them |
| 12:56.41 | wirehead_ca | the service has been great, but the re-registration is an issue, i can't have a phone system rebooted every time our internet hickups ;) |
| 12:56.41 | tzanger | heh |
| 12:56.48 | tzanger | wirehead_ca: totally agreed |
| 12:56.56 | tzanger | but get the packet dumps to them and see if they can tell what's going on |
| 12:57.00 | tzanger | *especiallY* if they're identical |
| 12:57.17 | wirehead_ca | i can't even get them on the phone |
| 12:57.23 | wirehead_ca | i've send them tcpdump logs |
| 12:57.29 | wirehead_ca | i've sent them my iax2 debug logs |
| 12:57.34 | wirehead_ca | showing before and after |
| 12:57.35 | wirehead_ca | etc |
| 12:57.50 | wirehead_ca | what kind of support are you able to get from there? is there an inside guy you deal with? |
| 12:58.29 | *** join/#asterisk acehunky (n=acehunky@59.184.10.178) |
| 12:58.38 | tzanger | wirehead_ca: I just email stephan monette and response is generally VERY fast |
| 12:58.42 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:58.57 | wirehead_ca | do you have his addy? |
| 12:59.00 | wirehead_ca | address... |
| 12:59.05 | wirehead_ca | i only have their sales email |
| 12:59.25 | tzanger | techsupport@unlimitel.ca |
| 12:59.46 | wirehead_ca | this is good, someone who's used this guys |
| 12:59.49 | acehunky | anyone out here who has worked on dundi ? |
| 12:59.55 | wirehead_ca | how long have you had accounts with them tzanger? |
| 13:00.03 | *** join/#asterisk dashu (n=root@p549C6A95.dip.t-dialin.net) |
| 13:00.16 | wirehead_ca | and what are you using with them? TrixBox? What version of asterisk? etc. |
| 13:00.18 | tzanger | over a year |
| 13:00.27 | mattfletcher | I'm having problems getting my TDM400P to work. Could someone take a look at these lspci and dmesg outputs and see if there is a solution: http://pastebin.ca/177414 |
| 13:00.27 | tzanger | asterisk svn trunk |
| 13:00.32 | tzanger | and before that, asterisk cvs head |
| 13:00.49 | dashu | anyone knows why my asterisk isnt hanging up automatically ? :o |
| 13:00.52 | wirehead_ca | so pretty bleeding edge stuff then. On top of debian? RedHat? |
| 13:01.25 | tzanger | mattfletcher: [42949387.520000] Freshmaker failed register test |
| 13:01.37 | tzanger | either the card is dead or you have some very fucked up PCI problems (unlikely) |
| 13:01.43 | dashu | - CAPI Hangingup |
| 13:01.43 | dashu | <PROTECTED> |
| 13:01.43 | dashu | <PROTECTED> |
| 13:01.43 | dashu | <PROTECTED> |
| 13:01.43 | tzanger | contact digium tech support and be prepared to RMA |
| 13:01.50 | tzanger | wirehead_ca: slackware |
| 13:01.55 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 13:02.03 | *** join/#asterisk Nitesh (n=Nitesh@65.48.63.178) |
| 13:02.31 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:02.35 | wirehead_ca | Tzanger, would SIP be just as good as IAX2? |
| 13:02.40 | wirehead_ca | to use with unlimitel? |
| 13:02.46 | mattfletcher | thanks tzanger |
| 13:02.54 | wirehead_ca | i've read about security issues with the IAX2 stack |
| 13:02.57 | tzanger | if you're using trunking you'll use more bandwidth with SIP, but if you're not trunking it's more or lesss the same |
| 13:03.04 | wirehead_ca | it seems better at bandwidth use and NAT friendly.... |
| 13:03.06 | tzanger | wirehead_ca: rumours and speculation abound |
| 13:03.13 | wirehead_ca | mmm |
| 13:03.19 | wirehead_ca | at most we'd use five channels |
| 13:03.19 | *** join/#asterisk remiss (i=bofh@225.84-48-68.nextgentel.com) |
| 13:03.23 | wirehead_ca | we're a small company/office |
| 13:03.54 | wirehead_ca | how much more b/w with SIP? Say 20% more? |
| 13:04.12 | s0lid | can i use b8zs on E1 in framing? |
| 13:04.34 | s0lid | !ping |
| 13:05.04 | *** join/#asterisk sakimustafa (n=sakimust@202.133.14.226) |
| 13:05.40 | tzafrir | try adding the boot option pci=noacpi |
| 13:05.43 | kaldemar | s0lid: b8zs is for T1 and hdb3 for E1, thinks me. |
| 13:06.17 | tzafrir | I believed it has solved a problem with a TDM400P (rev E?) on one of our systems |
| 13:06.21 | s0lid | kaldemar: yeah that's what i thought too but my provider set their E1 to b8zs |
| 13:06.31 | s0lid | they're using lucent for their switches |
| 13:06.32 | creativx | wirehead_ca: theres a page on voip info about bw usage |
| 13:06.43 | wirehead_ca | thx creativx |
| 13:07.02 | wirehead_ca | is SIP more "reliable" then IAX? |
| 13:07.07 | tzanger | wirehead_ca: no |
| 13:07.13 | wirehead_ca | are they on par? |
| 13:07.24 | tzafrir | wirehead_ca, SIP can use TCP, which is more reliable ;-) |
| 13:07.28 | wirehead_ca | hahaha |
| 13:07.32 | wirehead_ca | smart a$$ |
| 13:07.39 | tzafrir | but then again, asterisk does not support it |
| 13:07.47 | wirehead_ca | reliable is nice but getting packets in rapid sequence is better in voip ;) |
| 13:08.09 | wirehead_ca | asterisk does not support SIP? |
| 13:08.13 | s0lid | tzafrir: yeah that's what i thought too :) thanks for the reply! |
| 13:08.19 | wirehead_ca | wha? :) |
| 13:09.29 | coppice | officially it seems * does not support SIP, since some recent SIP documents appear to a SIP implementation must support TCP. Of course, that currently means most things do not support SIP. :-) |
| 13:09.48 | wirehead_ca | several smartasses in here ;) |
| 13:10.02 | mut | thats why they're here |
| 13:10.04 | wirehead_ca | haha ;) |
| 13:10.08 | Druken | welcome to #asterisk..... |
| 13:10.09 | mut | don't want dumbasses workin on it do ya |
| 13:10.20 | wirehead_ca | well SIP uses TCP for the control end right? and UDP for the data/audio streams non? |
| 13:10.38 | tzafrir | SIP *is* the control protocol. |
| 13:10.46 | tzafrir | can be TCP or UDP. |
| 13:10.52 | coppice | not in most current implementations |
| 13:10.57 | tzafrir | The data streams are RTP |
| 13:11.04 | wirehead_ca | RTP is UDP is it not? |
| 13:11.12 | coppice | yep |
| 13:11.27 | tzafrir | coppice, now look what you did to oej with those talks |
| 13:11.27 | coppice | RTP over TCP would be fun :-) |
| 13:11.32 | Druken | rtp == real time packets, best to not care if they get there or not... :) |
| 13:11.33 | wirehead_ca | no kidding |
| 13:11.45 | wirehead_ca | which means UDP :) |
| 13:12.19 | wirehead_ca | tzafrir |
| 13:12.32 | wirehead_ca | what are you using as a firewall between your * boxen and say unlimitel? |
| 13:12.52 | wirehead_ca | as its one other possible issue, mind you my tcpdupms show mine is working fine.... |
| 13:13.12 | Druken | if the first gets threw, why wouldn't the rest? |
| 13:13.25 | wirehead_ca | drunken - exactly! |
| 13:13.31 | wirehead_ca | i haven't a damn clue |
| 13:13.37 | wirehead_ca | only rebooting the box will re-register |
| 13:13.54 | wirehead_ca | right now its spitting out POKE, PING, LAGRQ constantly |
| 13:13.58 | wirehead_ca | Iax2 |
| 13:14.09 | Druken | that's normal... |
| 13:14.11 | wirehead_ca | i can see them flying out of the WAN side |
| 13:14.23 | wirehead_ca | but my box lost its registration when i dropped/brought up the wan |
| 13:14.29 | wirehead_ca | only rebooting will re-register |
| 13:14.30 | *** join/#asterisk acehunky (n=enterux@59.184.10.178) |
| 13:14.34 | wirehead_ca | with the remote host |
| 13:14.47 | wirehead_ca | if use the init.d script and stop/start that won't do it |
| 13:14.55 | wirehead_ca | if i issue the 'iax2 reload' command, that won't do it. |
| 13:14.59 | wirehead_ca | i can see the traffic stream out |
| 13:15.23 | wirehead_ca | is there a behavioural difference in asterisk, first starting/setting up an IAX2 trunk vs. re-registering? |
| 13:15.28 | wirehead_ca | thats whta i need to know first |
| 13:15.40 | wirehead_ca | if there is, then i'm leaning towards my provider or trixbox |
| 13:15.47 | wirehead_ca | if not, then perhaps a firewalling issue at my end |
| 13:16.05 | wirehead_ca | i have no other theory at this point, and i'm burnt out after beating on this for over a week |
| 13:16.57 | Druken | you say it onl craps out after a timeout no ? |
| 13:17.07 | wirehead_ca | yes |
| 13:17.22 | wirehead_ca | it goes to hell once that 60 seconds passes and its not getting a response from the remote IAX host |
| 13:17.35 | Druken | why not take the easy way out and highten the timeout? |
| 13:17.37 | wirehead_ca | then i can watch running "iax2 show registry" and seee it retry, timeout etc |
| 13:17.56 | wirehead_ca | druken thats a double edged sword |
| 13:18.01 | wirehead_ca | what if our wan drops for 5 min |
| 13:18.05 | wirehead_ca | or next time 30 min |
| 13:18.14 | wirehead_ca | i can't have staff reboot the phone system every time that happens |
| 13:18.25 | Druken | why would your lan be dropping all the time? |
| 13:18.27 | wirehead_ca | ideally when the wan comes up, after 60 seconds we should be re-registered |
| 13:18.35 | *** join/#asterisk neoalex (n=neoalex@user-0cdfjlp.cable.mindspring.com) |
| 13:18.39 | wirehead_ca | it doesn't |
| 13:18.41 | wirehead_ca | but |
| 13:18.46 | wirehead_ca | this is a corporate phone system |
| 13:18.51 | wirehead_ca | and its a Fixed IP DSL line |
| 13:19.00 | wirehead_ca | i'd rather have the thing work as its suppposed to |
| 13:19.21 | Druken | aight, but i have ip dedicated dsl at my office, and it NEVER drops... |
| 13:19.37 | wirehead_ca | well nothing lasts forever ;) |
| 13:20.06 | wirehead_ca | i've got linux servers here with uptimes in the year(s), i'd like to see the voip linux box work the same |
| 13:20.06 | Druken | well, the connection time is going on a good 3-4 months at least... |
| 13:20.15 | Druken | i never have to reboot the router.... hehe |
| 13:20.27 | wirehead_ca | besides if we start to farm out multip providers rebooting the voip box kills them all |
| 13:21.17 | wirehead_ca | who here is awake? how can give me a show of hands they use IAX2 with a VOIP provider and it re-registers all good and nice. Please tell me your setup (distro, version, or package like Trixbox etc) |
| 13:21.17 | dashu | is there a german asterisk channel ? :o |
| 13:22.08 | Druken | can't say i use register with iax... hehe cause i have the static's, my providers just dump the calls to me |
| 13:22.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:22.29 | wirehead_ca | so they have a config at their end to do so? |
| 13:22.32 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:22.39 | wirehead_ca | they contact your box to register you? |
| 13:22.51 | wirehead_ca | is there a term for that |
| 13:22.51 | Druken | yeah, instead of dynamic in the block, they have my ip |
| 13:23.05 | wirehead_ca | perhaps i can ask my provider to just do that, but i need to know how to ask for that ;) |
| 13:23.36 | wirehead_ca | i'd rather stick with IAX over SIP for trunking, it seems much better suited and efficent for it.... |
| 13:24.28 | dashu | is anyone of u using asterisk with isdn and mmh :o can coach me a bit ? :p |
| 13:26.01 | dashu | T_T im just trying a really simple thing atm but it just doesnt work im calling my phone with my mobile and i just want that the asterisk thing hangs up after 1 ring when he sees that it is my number |
| 13:26.06 | acehunky | anyone using Dundo |
| 13:26.08 | acehunky | Dundi |
| 13:26.22 | dashu | to hang up* |
| 13:26.29 | Druken | i dundo :) |
| 13:26.49 | acehunky | Druken: aaah you worked with dundi ? |
| 13:27.22 | Druken | no, i was going for the whole i dunno thing... but obviously it didn't work :) |
| 13:27.52 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 13:28.36 | blaylock | is it possible to do BLF over multiple locations? |
| 13:28.57 | Druken | blf ? |
| 13:29.06 | blaylock | like from one asterisk box to another where the phone systems are linked together |
| 13:29.10 | blaylock | Busy Lamp Field |
| 13:29.36 | blaylock | shows whether someone is on the phone or not through via hints |
| 13:29.41 | Druken | ewh... i highly doubt it... |
| 13:29.57 | blaylock | yeah thats what i thought |
| 13:30.08 | blaylock | i was hoping for some kind of add on feature or something |
| 13:30.14 | stoffell | dashu: what is mmh? |
| 13:30.20 | blaylock | which would communicate that information |
| 13:30.21 | neoalex | how do I connect to a sip provider vi stun in asterisk? |
| 13:30.37 | dashu | mmmh is just mmmh word of thinking :p |
| 13:30.48 | Druken | stun is for nat... to help you find your own ip |
| 13:31.14 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:31.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:31.16 | dashu | like well or etto in japanese :p |
| 13:31.18 | stoffell | dashu: ok:) i think you need the ex-girlfriend "protection" do to what you want (look for ex-girlfriend on voip-info.org) |
| 13:31.22 | stoffell | lol |
| 13:31.29 | dashu | XD lol |
| 13:31.36 | neoalex | i know my ip... and I am behind nat |
| 13:31.46 | dashu | k ill look for it :p |
| 13:32.57 | *** part/#asterisk hotroot (n=michael@pD9E95FB3.dip.t-dialin.net) |
| 13:35.46 | acehunky | Anyone who has successfully setup Dundi over here ? |
| 13:35.51 | acehunky | can help me |
| 13:39.13 | Druken | pretty sad... i had to put the furnace on this morning.... |
| 13:39.27 | Druken | hello autumn |
| 13:39.42 | blaylock | does anyone here manage multi location asterisk implementations? |
| 13:40.03 | blaylock | for a single company that is |
| 13:40.04 | *** join/#asterisk esculapio_ (n=The@200.88.44.66) |
| 13:40.08 | acehunky | blaylock: thats exactly what Dundi for .. I have been trying to hunt for people who have done thsi over here :( |
| 13:40.25 | acehunky | I am having quite some problem with that .. |
| 13:40.35 | acehunky | blaylock: how many locations your company has btw ? |
| 13:40.41 | blaylock | Dundi is used to get BLF information accross multiple locations? |
| 13:41.00 | blaylock | well this is research for a customer prospect |
| 13:41.27 | blaylock | we're trying to figure out how to make the phone systems work as if they were one big one |
| 13:41.32 | acehunky | Busy Lamp Field .. have no idea what that is .. |
| 13:41.32 | blaylock | over multiple locations |
| 13:41.36 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 13:41.40 | blaylock | using IAX2 |
| 13:41.56 | acehunky | the idea with asterisk is not to have one big one ...instead break it apart into multiple ones |
| 13:42.02 | blaylock | BLF notifies buddy list members when someone is on the phone |
| 13:42.06 | acehunky | talking 2 each other over Dundi |
| 13:42.12 | blaylock | using hint syntax in the dial plan |
| 13:42.51 | *** join/#asterisk pigpen2 (n=mark@daikoku.ccnbi.net) |
| 13:43.20 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 13:43.24 | angryuser | [TK]D-Fender: hi;) |
| 13:44.07 | acehunky | using IAX2 to interconnect too many asterisk systems aint a good idea .. |
| 13:45.31 | Nitesh | To all, |
| 13:45.31 | Nitesh | Thanks for helping to solve ma dtmf problem... I had to put dtmfmode=auto and type=peer. And I also found out that remote GW/Carrier also matters... |
| 13:45.31 | Nitesh | Thanks... |
| 13:45.38 | blaylock | acehunky, we have about 15 IAX2 connections here |
| 13:45.45 | acehunky | 15 :O |
| 13:45.48 | acehunky | One way ? |
| 13:46.09 | blaylock | no, a bunch of different customers talking back to our PBX |
| 13:46.18 | acehunky | well thats fine .. |
| 13:46.33 | blaylock | oh i see what you were saying |
| 13:46.41 | blaylock | dont have a chain of IAX connections |
| 13:46.44 | acehunky | i m talking of case where you want to make all locations feel to be in a single PBX system |
| 13:46.49 | blaylock | yeah that would be ick |
| 13:46.55 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:47.01 | acehunky | yeah .. he he he .. |
| 13:47.25 | blaylock | well thats how businesses want their phone system to operate in the end though |
| 13:47.56 | blaylock | be able dial single extensions and get someone accross the country |
| 13:48.03 | blaylock | instead of dialing some special number |
| 13:48.13 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:48.40 | jeremy_g | blaylock:that can be done |
| 13:49.02 | jeremy_g | blaylock:u shud read on location servers |
| 13:49.04 | acehunky | thats exactly what DUNDi is .. and i am not able to make two systems pick up extensions from each other :( |
| 13:49.24 | acehunky | jeremy_g: how to do that using Dundi ? got any idea ? |
| 13:49.46 | jeremy_g | acehunky:right now too busy with a customer's cdr problems.. |
| 13:50.27 | acehunky | jeremy_g: :( okkk |
| 13:54.01 | acehunky | Dundi dundi dundi ...man this stuff is makin me go crazy .. i seriously doubt it it has ever worked for any one ! |
| 13:54.15 | creativx | Dundee, Crocodile. |
| 13:54.31 | acehunky | yeah .. crocodile .. |
| 13:54.57 | blaylock | Dundi is confusing |
| 13:55.03 | blaylock | the documentation is vauge |
| 13:55.13 | [TK]D-Fender | angryuser: Hi |
| 13:55.20 | creativx | is it documented through source code? |
| 13:55.31 | acehunky | well the mailing list also seems to be dead |
| 13:55.38 | acehunky | no one is ever there on #dundi |
| 13:55.55 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 13:56.32 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 13:56.32 | acehunky | creativx: have you ever encountered to make a simple two extension system work on dundi ? |
| 13:56.41 | [TK]D-Fender | DUNDI - The best idea nobody gives a shit about |
| 13:56.46 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 13:56.47 | blaylock | can anyone explain what Dundi is for? |
| 13:57.02 | creativx | acehunky: afraid not. havent even touched dundi |
| 13:57.04 | Makenshi | dundi is like a peer-to-peer dial plan |
| 13:57.10 | Makenshi | personally i prefer e164 |
| 13:57.15 | acehunky | DUNDi™ is a peer-to-peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as the remarkably simple and concise ENUM standard), DUNDi is fully-distributed with no centralized authority whatsoever. |
| 13:57.16 | Makenshi | but it is a great concept |
| 13:57.35 | acehunky | thats what dundi.com says .. this site is always down or pointed to thevoice.digium.com |
| 13:57.43 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 13:57.47 | Seyr | I setup DUNDi once and got it working with multiple extensions on each box calling eachother |
| 13:57.54 | Seyr | just dont have any client that needs/wants it |
| 13:58.02 | blaylock | ok but what does it do? |
| 13:58.09 | blaylock | publish numbers? |
| 13:58.17 | acehunky | well i got a client here who needs it between their three offices |
| 13:58.17 | charles___ | it's going to be necessary as soon as people start interconnecting their asterisks |
| 13:58.30 | blaylock | instead of having static extensions listed in the dial plan it becomes dynamic? |
| 13:58.36 | kaldemar | it publishes local numbers to other defined dundi peers |
| 13:58.36 | acehunky | their employees keep on roaming .. |
| 13:59.00 | Seyr | a device can register on any of them and still be found |
| 13:59.13 | acehunky | thats right Seyr ! |
| 13:59.33 | creativx | you have user1 with extension 501 on server1, then user2 on server2 dials 501 and calls user1 because dundi knows the route? |
| 13:59.33 | Seyr | it works as advertised, i just never see anyone using it |
| 13:59.54 | nortex | acehunky, One of the guys at txaug.net posted a whitepaper on Dundi in a cluster, it may not be what you are trying to do, but he gave good examples. |
| 14:00.30 | kaldemar | creativx: yes, server1 tells server2 that it has 501 on it. |
| 14:00.33 | Seyr | i believe i I got mine working from just the info on vopi-info |
| 14:00.42 | acehunky | nortex: i actually checked out his presentation and did my config exactly as that of his .. but dundiLookup returns empty string :( |
| 14:00.42 | Seyr | links on there you can follow off to examples |
| 14:00.49 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:01.11 | Seyr | voip-info that is |
| 14:01.43 | Seyr | anyone here ever used the "Cyberdata PoE VoIP Speaker" for paging? |
| 14:02.13 | Seyr | or any other PoE device for paging? |
| 14:02.22 | creativx | kaldemar: then i've understood it right |
| 14:02.22 | creativx | hehe |
| 14:02.44 | acehunky | here is my problem : http://pastebin.ca/171439 |
| 14:04.22 | blaylock | interesting |
| 14:04.45 | blaylock | i wonder if Dundi will also publish BLF information then |
| 14:06.16 | acehunky | the thing that worrys me is .. dundi.com is down most of the time .. very little documentation ... and a dead mailing list and irc channel .. |
| 14:06.44 | nortex | Has anyone run app_followme on 1.2? |
| 14:07.00 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 14:07.08 | [TK]D-Fender | Seyr: Well my Polycom phones are PoE and I have paging enabled on them, sop technically YES |
| 14:07.39 | FreezeS | hello people |
| 14:07.43 | nortex | acehunky, I think the offical website is dundi.info |
| 14:07.58 | FreezeS | I've got a lot of desynchronisations on my digium PRA card |
| 14:07.59 | FreezeS | Sep 20 17:08:22 NOTICE[1581]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 14:08.20 | FreezeS | it's setup to use the timing of the carrier, not mine |
| 14:08.31 | [TK]D-Fender | FreezeS: Pastebin "cat /proc/interrupts" |
| 14:08.33 | [TK]D-Fender | ~pb |
| 14:08.40 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:08.50 | [TK]D-Fender | FreezeS: And also your zaptel.conf |
| 14:08.57 | FreezeS | span=1,0,0,ccs,hdb3,crc4 |
| 14:09.15 | [TK]D-Fender | FreezeS: : that line does not use carrier timing |
| 14:09.34 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 14:09.49 | [TK]D-Fender | FreezeS: :span=1,1,0,ccs,hdb3,crc4 <--- this does |
| 14:09.49 | Seyr | [TK]D-Fender: I should have specified "poe speakers" :-) but then again, maybe your Polycom has a speakerphone.. so you could have answered yes then as well :-) |
| 14:09.49 | FreezeS | oh, damn |
| 14:09.50 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 14:09.54 | sakimustafa | any tutorial to install asterisk on FC 2 |
| 14:09.54 | FreezeS | let's see |
| 14:10.00 | [TK]D-Fender | Seyr: Thats exactly what it does. It answers incoming calls period. |
| 14:10.11 | nortex | acehunky, I know the feeling though, I'm trying to set it up between several internal server to reduce some work i my dialplan, but I get the same lack of results. If I get it working I'll post it somewhere, probably voip-info. |
| 14:10.34 | acehunky | nortex: that would be great .. |
| 14:10.45 | FreezeS | and, I see that the card uses the same interrupt as the soundcard: 18: 11570854 IO-APIC-level SiS 7012, wcte11xp |
| 14:10.49 | acehunky | i am reading through the presentation @ txaug.net .. |
| 14:10.49 | FreezeS | how can I change that ? |
| 14:10.58 | [TK]D-Fender | Seyr: That device you're looking at is just a 1-way SIP phone. nothing more |
| 14:11.00 | acehunky | but they seem to talk of something other then dundi |
| 14:11.16 | [TK]D-Fender | FreezeS: In your BIOS. You need to get it on its own IRQ |
| 14:11.27 | acehunky | nortex: http://asterisk.pbx.in/ you can post it here too :) |
| 14:11.43 | Seyr | i'm looking for just PoE speakers that auto-answer.. One of the latest upgrades to * or FreePBX borked my paging, so we are looking at a speaker based paging system |
| 14:12.35 | Seyr | [TK]D-Fender: but have you or anyone you know used the ones from CyberData? or any others that are known to be decent? |
| 14:12.36 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 14:13.21 | FreezeS | thanx [TK]D-Fender |
| 14:13.34 | [TK]D-Fender | Seyr: No I've heard nothing on them specifically. People have hot-wired other paging systems onto BT-100 series phones before. The best way IMO is the gat an analog pagin unit and run it on an ATA. |
| 14:14.22 | Seyr | that company actually has some of those as well. just plug the speaker into the device and it makes it SIP.. so its just an ATA that auto-answers |
| 14:14.42 | Seyr | but i kinda like the idea of a PoE speaker. i just hope it works :-) |
| 14:14.59 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 14:16.10 | [TK]D-Fender | Seyr: Less flexible and less powerful for sure. hope it does the job. |
| 14:17.52 | zouzou | where can i found the nissinf meetme apllication |
| 14:17.52 | zouzou | ? |
| 14:18.21 | zouzou | where can i found the missing meetme application? |
| 14:18.37 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-134-155-184.dsl.irvnca.pacbell.net) |
| 14:18.45 | nettie | Hi guys, I created a $STAFF variable in my dialplan. STAFF=SIP/poly1&SIP/poly2&SIP/poly3&SIP/poly4&SIP/utstar01 I would like to achieve the same result using an extracontext jump, somethign liek goto context,201,202,203,204,205,206 because every extensions in the local context is bound to a macro which checks various things, from call forwarding to voicemail etc.. anyone can give me some hints to do it please? thanx in advance. |
| 14:20.41 | Seyr | [TK]D-Fender: Less points of failure and the only cabling is additional CAT5, is what I am looking at. Buying an analog paging system and ATAs, versus buying a couple of PoE VOiP speakers. |
| 14:20.48 | [TK]D-Fender | nettie: Look at it this way : Imagine the caller being sent off liek that. Whose voicemail should they hear? * can't do 10 things at a time and decide which branch to follow. |
| 14:21.27 | Seyr | VoIP even |
| 14:21.47 | [TK]D-Fender | Seyr: Could be you are restricted in that sense. I'd say buy one and try it. If you don't like it that buy a bunch of low-end Polycom phones and mount them as combo Paging / Emergency phones :) |
| 14:22.38 | Seyr | [TK]D-Fender: Good point there. Instead of having speakers in the ceiling in the halls, just put a few phones on small tables or whatnot and use them |
| 14:22.42 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:23.00 | Seyr | looks like you put em out there for clients, so looks good as well |
| 14:23.16 | nettie | [TK]D-Fender: that's right but, my problem is mostly related to call forwarding |
| 14:23.36 | nettie | [TK]D-Fender if I configure call forwarding on a single extension |
| 14:23.47 | blitzrage | I imagine a bunch of people here have 7960's -- I'm trying to make the first 3 lines register to one proxy, and the last 3 lines register to another. In SIPDefault do I just do proxy1_address: my_proxy1, proxy4_address: my_proxy2 ? (with the appropriate addition lines needed for the other 4 lines) |
| 14:24.07 | [TK]D-Fender | blitzrage: ! ! ! |
| 14:24.19 | blitzrage | [TK]D-Fender: ! ! ! |
| 14:26.44 | nettie | [TK]D-Fender and the variable points to the the sip devices |
| 14:26.52 | nettie | directly |
| 14:27.01 | Dr-Linux | anybody every use opensource SMS gateway? like Gammu etc? |
| 14:27.12 | Dr-Linux | s/every/ever |
| 14:27.24 | nettie | instead of the extensions it will bypass the call forwarding check |
| 14:27.43 | [TK]D-Fender | nettie: You can try using a local channel, but I shudder at the thought of how it'll react to multiple channels preparing to answer.... |
| 14:28.09 | [TK]D-Fender | nettie: calllist=Local/100@context&Local/101@context&Local/102@context |
| 14:28.57 | Remenic | I have two hardware SIP phones (yuxin and grandstream) connected to my asterisk box. When I call either one, and never pick up, the phone that is calling never stops calling, even though the other phone stopped ringing... |
| 14:29.05 | Remenic | anyone know what could be causing that? |
| 14:29.11 | mut | wow |
| 14:29.16 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:29.16 | mut | my cdr screwed up |
| 14:29.28 | mut | it recorded a call was like 20 hours long |
| 14:29.42 | mut | 73000 some odd seconds |
| 14:30.39 | jeremy_g | mut:wow |
| 14:32.59 | mut | just short of 1 million cdr records |
| 14:33.00 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:33.03 | mut | and thats the first that messed up |
| 14:33.27 | mut | nto too bad i spose |
| 14:35.08 | [TK]D-Fender | mut: Who do you work for again? |
| 14:35.18 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:36.06 | mut | someone o_O |
| 14:36.15 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 14:40.47 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 14:46.26 | *** part/#asterisk stephane_ (n=stephane@nostromo.cabale.net) |
| 14:46.26 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 14:46.33 | *** join/#asterisk stephane_ (n=stephane@nostromo.cabale.net) |
| 14:46.54 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 14:48.33 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.116) |
| 14:49.05 | *** join/#asterisk shtoom (n=godson@59.184.10.178) |
| 14:51.30 | tamp4x | is it possible to run asterisk on 2 bind ports? |
| 14:52.01 | [TK]D-Fender | tamp4x: Typically * binds to all ports |
| 14:52.08 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 14:52.18 | Juggie | you mean interfaces d-fender |
| 14:52.21 | Juggie | he means ports. |
| 14:52.27 | [TK]D-Fender | Err... oops :) |
| 14:52.33 | [TK]D-Fender | askjhslakjdhflkasdf |
| 14:52.42 | Juggie | i assume he wants to listen for sip on two dif ports or something |
| 14:53.56 | *** join/#asterisk nerdz (n=carpe@200.140.28.80) |
| 14:53.59 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:54.16 | tamp4x | yes juggie |
| 14:54.35 | Juggie | i dont know if thats possible w/ chan sip, but you can surely do it with iptables |
| 14:54.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:54.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:55.26 | tamp4x | with iptables? forwarding another port to 5060? |
| 14:55.31 | Juggie | correct |
| 14:56.34 | tamp4x | but how will it know to forward back to that clinet |
| 14:56.39 | tamp4x | via that port |
| 14:56.48 | tamp4x | client |
| 14:58.40 | Juggie | hmmmmm |
| 14:58.42 | Juggie | i guess it woudnt. |
| 14:59.15 | Juggie | why do you need a second port? |
| 14:59.37 | tamp4x | certain cable companies are rate limiting 5060 |
| 14:59.45 | Juggie | hah |
| 14:59.48 | Juggie | so? |
| 15:00.11 | Juggie | 5060 is the control channel |
| 15:00.20 | tamp4x | ao we are losing customers |
| 15:00.22 | Juggie | it should contain very little traffic that would require rate limiting |
| 15:00.40 | tamp4x | yes, though they can screw with it so it is difficult to hold registration |
| 15:00.55 | Makenshi | i really, really hate isps that do that |
| 15:00.57 | Juggie | drop your registration timeout to be something low. |
| 15:01.09 | Makenshi | if the mailman started ripping bits out of my mail i'd be livid |
| 15:02.56 | Juggie | tamp4x, the best thing i could recomend would be to do a second compile of * with special options to give it a new /etc/ dir |
| 15:02.57 | *** join/#asterisk Tr4d3r (n=Tr4der@200.71.51.147) |
| 15:03.03 | Tr4d3r | Hello there ;) |
| 15:03.05 | Juggie | say, /etc/asterisk-grrisp |
| 15:03.31 | tamp4x | yea i was going to put asstricks on another machine |
| 15:03.41 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:03.45 | Juggie | run it very minimal with chan_sip and chan_iax so you could take sip calls on the port you want and trunk them over iax to another asterisk process |
| 15:03.49 | Juggie | yes either or would work |
| 15:03.58 | Juggie | another machine would be easier, but either will work |
| 15:04.05 | Tr4d3r | hey.. i need to configure a server with 200 analog extensions ... i don't know what to do because the TDM2400P only aceppt 24 for each card ... :S |
| 15:04.20 | Juggie | Tr4d3r, you need whats called a channel bank. |
| 15:04.32 | Tr4d3r | Ho ok... |
| 15:04.41 | Juggie | basically, you would put say 8 t1 ports in your server |
| 15:04.49 | Tr4d3r | some manufacturer? |
| 15:04.52 | Juggie | and then connect them all to a channel banks(s) |
| 15:05.18 | Juggie | http://www.voip-info.org/tiki-index.php?page=Asterisk+Channel+Bank |
| 15:05.29 | Tr4d3r | ok ... i understand ... the asterisk to the channel bank with t1 connection ... and the channel bank have the analog ports ... correct? |
| 15:05.30 | L|NUX | hello every one |
| 15:05.32 | L|NUX | can some one tell me what switchtype used by korean isdn 30 ? |
| 15:05.33 | Tr4d3r | Thanks |
| 15:05.37 | Juggie | Tr4d3r, correct. |
| 15:05.42 | Tr4d3r | THX! |
| 15:05.56 | L|NUX | Juggie : can you help me brother |
| 15:06.08 | Juggie | dont ask for help, just ask your question |
| 15:06.45 | Juggie | Tr4d3r, make sure you buy the right interface too.... eg FXO vs FXS |
| 15:06.47 | L|NUX | tell me what switchtype used by korean isdn 30 ? |
| 15:06.53 | L|NUX | becuase i have configured sangoma card with PRI in it |
| 15:07.05 | L|NUX | but when i do call on DID nothing comes on asterisk |
| 15:07.11 | Juggie | L|NUX, i have no idea, i'm in canada. |
| 15:07.25 | L|NUX | Oh |
| 15:07.39 | Juggie | make sure you have the right pri type |
| 15:07.41 | *** join/#asterisk [jwb] (n=me@schizophrenia.paravolve.net) |
| 15:07.43 | Juggie | eg T1/E1/J1 |
| 15:07.53 | L|NUX | E1 |
| 15:08.00 | L|NUX | it E1 |
| 15:08.18 | Juggie | no alarms? |
| 15:08.34 | L|NUX | when i do pri show span 1 |
| 15:08.43 | L|NUX | on asterisk cli it will show me output |
| 15:08.58 | Juggie | whats the status: |
| 15:08.59 | Juggie | line. |
| 15:09.06 | L|NUX | wait |
| 15:09.18 | L|NUX | Status: Provisioned, Up, Active |
| 15:09.34 | Juggie | ok, well looks like your hard ware level link is up. |
| 15:09.39 | L|NUX | but Provider told me it Korean ISDN not EuroISDN |
| 15:09.42 | Juggie | is your d channel up? |
| 15:09.44 | nortex | acehunky, You still around? |
| 15:10.28 | L|NUX | Channel 16: D-channel (Default) (Slaves: 16) |
| 15:10.34 | L|NUX | when i type ztcfg -vvvv |
| 15:11.18 | [TK]D-Fender | Tr4d3r: Do not waste time with Channel banks and T1 cards. That adds a lot of unneecessary load onto * and increases your risk of failure. Get multiple 24-port SIP gateways. |
| 15:11.26 | Juggie | do you see any messages like, -- B-channel 0/1 restarted on span 1 on your conosle? |
| 15:11.49 | [TK]D-Fender | Tr4d3r: Channel banks is a pretty bad idea for that scaling |
| 15:11.51 | L|NUX | wait |
| 15:12.07 | Juggie | [TK]D-Fender, i very much agree that he should go ip, but, he asked for analog information. |
| 15:12.19 | file | you can do voice over IP? |
| 15:12.20 | L|NUX | nope |
| 15:12.21 | file | wow |
| 15:12.43 | [TK]D-Fender | Juggie: Yes he wants the EXTENSIONS analog, but that has nothing to do with getting the multiplexing source into * |
| 15:13.32 | Juggie | i guess he could use a large ATA as well, touche. |
| 15:13.51 | [TK]D-Fender | Tr4d3r: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-28397405952.htm |
| 15:14.00 | [TK]D-Fender | Tr4d3r: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-29013176064.htm |
| 15:14.13 | [TK]D-Fender | Tr4d3r: I'd suggest the latter, but try and get a better price on it. |
| 15:17.16 | Juggie | [TK]D-Fender, do you know of any hardware devices that will take t1 and give me sip/iax |
| 15:17.52 | [TK]D-Fender | Juggie: http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-40409868032.htm |
| 15:18.53 | Juggie | hmmm |
| 15:19.08 | Juggie | kind of expensive when you compare the cost of a t1card+system to the hardware gateway |
| 15:19.46 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:20.30 | jeremy_g | is there no cdy_mysql.conf in asterisk-1.2.10 |
| 15:20.36 | jeremy_g | -10 |
| 15:20.43 | Juggie | there is |
| 15:20.47 | Juggie | if you install asterisk-addons |
| 15:20.53 | Juggie | and do make samples from within that source |
| 15:21.01 | jeremy_g | thanks |
| 15:21.40 | Tr4d3r | [TK]D-Fender: Thanks ... i'm reading ;) |
| 15:22.55 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 15:24.18 | *** part/#asterisk acehunky (n=enterux@59.184.10.178) |
| 15:24.31 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:24.46 | supjigatr | Any idea why a mp3 for MOH would sound slurred and slow? |
| 15:24.58 | Juggie | is it your own mp3? |
| 15:25.09 | [TK]D-Fender | Juggie: Yes it is pricy but us scales to certain solutions. Picture a true SIP proxy & billing environment with * where the central system may have redundant boxes to your to the termination gateways to cirumvent failures..... |
| 15:25.13 | supjigatr | One a studio sent me for MOH. |
| 15:25.20 | supjigatr | Sounds fine in winamp |
| 15:25.26 | [TK]D-Fender | Juggie: For a simple PBX though it be a total waste |
| 15:25.38 | tzafrir | supjigatr, do you use native moh or mpg123? |
| 15:25.39 | Juggie | [TK]D-Fender, indeed... |
| 15:25.45 | supjigatr | mpg123 is running. |
| 15:25.46 | Juggie | supjigatr, is it VBR? |
| 15:25.51 | [TK]D-Fender | supjigatr: Make sure its not a VBR recording and has no ID3 tag |
| 15:25.57 | supjigatr | How do I tell? |
| 15:26.04 | Juggie | the bitrate in winamp will change |
| 15:26.10 | supjigatr | Is there a lame command to convert it? |
| 15:26.11 | Juggie | while playing |
| 15:26.14 | [TK]D-Fender | supjigatr: Play it in winamp and look at the bitrate |
| 15:26.16 | Juggie | probally but i dont know it off hand |
| 15:26.25 | Juggie | you can also do file info in winamp |
| 15:26.32 | Juggie | and it will say VBR or CBR |
| 15:26.39 | tzafrir | or use file(1) in linux |
| 15:27.25 | tzafrir | For starters, asterisk uses 8kHz mono |
| 15:27.30 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 15:28.59 | tzafrir | when you use mpg123, you waste cpu cycles on decoding and downsampling the mp3 file in real-time |
| 15:29.08 | supjigatr | How do I convert? |
| 15:29.11 | tzafrir | decode and downsample it to wav / slinear |
| 15:29.19 | supjigatr | Oh and no ID3 and it is a CBR file. |
| 15:29.55 | *** join/#asterisk ktf2 (n=ktf@83.71.163.137) |
| 15:30.08 | tzafrir | mpg123 for the basic conversion and sox for the downsampling |
| 15:30.37 | *** join/#asterisk afrosheen (n=afroshee@txprotoa2.august.net) |
| 15:30.40 | supjigatr | Is there a simple command to convert? And do I just put that file in the MOH directory? |
| 15:30.47 | Juggie | Tr4d3r, http://www.icstel.com/topgate.html |
| 15:30.50 | *** join/#asterisk saftsack (n=saftsack@p54A7D255.dip.t-dialin.net) |
| 15:31.03 | Juggie | 48port gateway |
| 15:31.07 | afrosheen | yesterday asterisk went down on it's own...how can I find in the logs where it died and what killed it? |
| 15:31.19 | ktf2 | hi all, I need some help getting 2 asterisk servers to talk to each other. Bascially I want to do, (normal phone)->Sip Provider->Asterisk1->Asterisk2->sip phone |
| 15:31.59 | ktf2 | I have the Normal phone->Sip Provider->Asterisk1 working and I have Asterisk2->Sip Phone working |
| 15:32.05 | afrosheen | ktf2, you'll probably just set up an iax trunk between the two and use a shared dialplan |
| 15:33.02 | supjigatr | K doing a mpg123 to wav yeilds a native format but it still sounds bad. |
| 15:33.03 | ktf2 | afrosheen, I can't get an IAX trunk up and going... (note asterisk1 is in my DMZ and asterisk2 is behind my firewall, all ports open at the moment) |
| 15:33.12 | [TK]D-Fender | Juggie: Never heard of them, no pricing listed and the can't spell! |
| 15:33.26 | afrosheen | iax has no problems traversing firewalls usually |
| 15:33.58 | [TK]D-Fender | Juggie: No listing for CWCID, Conferencing, etc..... it scares me... |
| 15:34.38 | Juggie | i just found it, i didnt say it was good :) |
| 15:35.30 | ktf2 | afrosheen, I think I must be going wrong in my extensions then... when I ring the sip provider, he passes the call onto asterisk1, without an extension? So in my extensions.conf, how to I route that call to a particular exten on asterisk2? |
| 15:35.55 | ktf2 | afrosheen, basically, I can asterisk1 to act like a Sip Proxy, and route all calls to asterisk2 |
| 15:36.13 | afrosheen | ktf2, sounds like you're using another box just to avoid NAT, am I right? |
| 15:36.23 | *** join/#asterisk mick_linux (n=chatzill@adsl-067-035-111-237.sip.bct.bellsouth.net) |
| 15:37.38 | mick_linux | anyone know how i can encrypt VoIP calls? |
| 15:38.01 | ktf2 | afrosheen, no basically I don;t want to open my firewall straight into the inside of my network, I want the call to go into the DMZ and then from the DMZ into the internal network (2 layers of firewall) |
| 15:38.21 | smackus | is it possible to add an external telephone number to the queues with Add Queue Member from the CLI>? I have tried a few things like add queue member Zap/8285200 and Zap/g1/8285200 and Zap/8285200@context, etc.. nothing seems to be working. is this even possible? |
| 15:38.30 | afrosheen | ktf2, we solved that by dual-homing our server and leaving one port on the outside of any firewall |
| 15:40.10 | ktf2 | afrosheen Hmm don't think we will be allowed do that :( |
| 15:40.25 | afrosheen | that's unfortunate, we did it mainly to avoid dual nat |
| 15:41.06 | *** join/#asterisk acsmedic (n=acsmedic@12.165.173.5) |
| 15:43.30 | [TK]D-Fender | ktf2 : So you want a border SIP server to receive all calls and forward them on to the inside? |
| 15:45.03 | *** join/#asterisk bernardovieira (n=bernardo@vorlon.intellivox.com.br) |
| 15:45.04 | *** join/#asterisk rosivelt (n=rosivelt@201008239182.user.veloxzone.com.br) |
| 15:45.11 | smackus | I also did AddQueueMember(helpdesk) in the extensions.conf and then dialed that extension from my cell phone. it added the device, but it added it as Zap/48. Is there a way to make it so that it will add a cell phone to the queue using AddQueueMember? |
| 15:45.19 | ktf2 | Fender, Yep! |
| 15:45.56 | Qwell | smackus: AddQueueMember(helpdesk, Zap/1/5551234) |
| 15:45.59 | Qwell | or whatever |
| 15:46.07 | smackus | 1 instead of g1? |
| 15:46.18 | Qwell | hence the "or whatever" |
| 15:46.40 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 15:47.00 | smackus | does it work the same way from the CLI? Add Queue Member Zap/g1/5551234 to helpdeks? because I tried it that way and it did not work. |
| 15:47.23 | smackus | the reason i would like to do it with add queue member is because I am using a php page to run the login |
| 15:47.25 | Qwell | smackus: it does |
| 15:47.50 | smackus | ok... let me try again. |
| 15:47.51 | smackus | thanks |
| 15:48.59 | jeremy_g | i did this svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-addons and got addons that didnt contain cdr_mysql |
| 15:49.03 | jeremy_g | ??why |
| 15:49.33 | Qwell | jeremy_g: because you didn't checkout asterisk-addons, you checked out asterisk |
| 15:50.02 | *** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 15:50.57 | jeremy_g | Qwell:i got other addons like cdr_mysql, read again |
| 15:51.06 | jeremy_g | s/cdr_mysql/cdr_pgsql |
| 15:51.12 | Qwell | jeremy_g: because you didn't checkout asterisk-addons, you checked out asterisk |
| 15:52.20 | rollergrrl | Is 1.4 supposed to make it easier to compile apps? Because it looks confusing as hell to me. |
| 15:53.08 | jeremy_g | Qwell::P |
| 15:53.19 | jeremy_g | Qwell:are you retarded or just acting like one , i clearly see asterisk-addons downloaded into my dir |
| 15:53.24 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:53.31 | file | jeremy_g: you checked out the 1.2 asterisk branch |
| 15:53.38 | jeremy_g | yup |
| 15:53.40 | Qwell | No, you see a directory NAMED asterisk-addons |
| 15:53.42 | file | and told SVN to make a directory named asterisk-addons and put it in it |
| 15:53.47 | Qwell | That doesn't mean it is |
| 15:54.01 | jeremy_g | i accept my mistake |
| 15:54.04 | jeremy_g | thanks file and Qwell |
| 15:54.20 | Qwell | svn co http://svn.digium.com/svn/asterisk/trunk WMD |
| 15:54.31 | Qwell | Doesn't mean I have bombs and such on my filesystem :p |
| 15:54.47 | Druken | anyone got a working app_dbodbc.so module with 1.2.12 ? |
| 15:54.54 | Qwell | Druken: just use func_odbc :p |
| 15:55.11 | Druken | it'll let me do odbc calls in the dialplan ? |
| 15:55.15 | Qwell | mmhmm |
| 15:55.21 | Qwell | as function calls |
| 15:55.45 | Druken | i used to use odbcget..... |
| 15:55.52 | *** join/#asterisk rokerr (n=rowan@38.99.162.187) |
| 15:56.00 | Qwell | func_odbc is way cooler than any of the other odbc things out there |
| 15:57.01 | Druken | ok, but does it have something like the odbcget ? |
| 15:57.09 | Qwell | What is odbcget? |
| 15:57.23 | file | nothing will drive them away! |
| 15:57.57 | Druken | well, i used it in the dialplan, to grab variables out of the odbc database... for call setup etc |
| 15:58.25 | Qwell | Druken: yes, it's just function calls |
| 15:58.28 | Qwell | very simple, very robust |
| 15:58.41 | file | very functional |
| 15:58.47 | acsmedic | Hello - has anyone know how to resolve dtmf issue with polycom phones calling into voicemail? thanks in advance |
| 15:58.52 | Dr-Linux | brb -- dinner time |
| 15:59.13 | [TK]D-Fender | acsmedic: Maybe you could actually describe the issue and show us your peer config.... |
| 15:59.20 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 15:59.37 | jayk- | does asterisk come with a fastbusy macro? how do i play the fastbusy tone? |
| 16:00.39 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 16:00.57 | Tr4d3r | <[TK]D-Fender: i will need 10 channel banks and 3 quad t1 cards?? there is not other choice? |
| 16:01.26 | [TK]D-Fender | Tr4d3r: Didn't you just follow those links to the SIP gateways I gave you? |
| 16:01.30 | mattfletcher | can anyone tell me a good place to buy an ata with 2 fxo ports (no fxs needed) to use to bring two pstn lines into an asterisk server |
| 16:01.35 | Tr4d3r | Yes i do |
| 16:01.41 | Tr4d3r | 24 fxs |
| 16:01.48 | Tr4d3r | or i dont understand |
| 16:01.50 | Tr4d3r | lol |
| 16:02.00 | [TK]D-Fender | Tr4d3r: You don't need ANY T1 cars or channel banks with those. so just 10 of those gateways |
| 16:02.16 | Tr4d3r | Haaa Ok... they connect directly to the lan? |
| 16:04.02 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 16:06.03 | [TK]D-Fender | Tr4d3r: Yes. So All I would suggest is running it all on Gigabit |
| 16:06.33 | Qwell | 240 fxs? ugh... |
| 16:07.02 | mattfletcher | No-one? All I need is a cheap 2 port PSTN to Ethernet ATA - which I believe is an FXO unit. I'm UK-based so a UK seller would be best |
| 16:07.06 | *** join/#asterisk pbx1 (n=pbx1@124.106.141.64) |
| 16:07.34 | [TK]D-Fender | mattfletcher: 2 x Linksys SPA-3102 |
| 16:07.35 | Qwell | mattfletcher: a couple of the spa3xxx's |
| 16:08.32 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F325.versanet.de) |
| 16:08.32 | mattfletcher | That's what I thought. So there are no units with more than one FXO port? Also I don't need FXS if that is possible |
| 16:08.53 | tzafrir | mattfletcher, not of the cheapo ATAs |
| 16:09.06 | pbx1 | Hey guys. I'm having issues with dropped calls. I am not behind Doing NAT. What reasons would their be for dropped calls? |
| 16:09.28 | [TK]D-Fender | mattfletcher: Every other SIP gateway for FXO is MUCH more expensive |
| 16:09.52 | mattfletcher | how much more? can you point me towards a few? |
| 16:10.12 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:10.14 | *** part/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:10.16 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:11.27 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F325.versanet.de) |
| 16:11.53 | [TK]D-Fender | mattfletcher: Here's a US link, but as you can see... more than double the cost : http://www.voipsupply.com/index.php?cPath=94_286_120 |
| 16:13.12 | pbx1 | the dropped calls are happening mostly from answered queue calls. They seem to drop at random times, but not all calls drop at the same time. any thoughts? |
| 16:15.10 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 16:15.20 | mattfletcher | Thanks [TK]D-Fender, I assume you mean the SPA-400 of that list |
| 16:15.24 | KranZ | would checking out trunk be the way to test 1.4? |
| 16:15.30 | FlatFoot | [TK]D-Fender: the SPA-3102 is it a good device ? |
| 16:15.54 | mick_linux | btw: is the iaxy supposed to get really hot? |
| 16:16.42 | mick_linux | i'm having some issues w/ getting no dial tone (randomly), then i only get a busy signal (whenever i pickup the phone), and now i get a sound that can only be described as excrusiating |
| 16:16.54 | mick_linux | i'm wondering if it is the heat |
| 16:19.18 | mick_linux | now i'm back to no dialtone |
| 16:19.51 | mick_linux | now it works |
| 16:20.39 | FlatFoot | mick_linux: is the offending item produced by B.Gates Enterprises ? |
| 16:21.10 | mick_linux | Digium |
| 16:21.39 | mick_linux | and it usually kicks ass |
| 16:21.51 | mick_linux | my guess is that an admin was playing w/ the pbx |
| 16:21.53 | FlatFoot | can't be made by B.Gates then |
| 16:22.06 | mick_linux | Mark Spencer ;) |
| 16:22.17 | mick_linux | who btw is a really cool guy |
| 16:22.21 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:22.23 | FlatFoot | someone sensible then |
| 16:22.44 | mick_linux | ya, but i doubt that it is the iaxy that is the problem |
| 16:23.08 | mick_linux | the hardware, application, and company are great |
| 16:23.30 | mick_linux | the admins who sometimes futz w/ the pbx.... well - no comment ;) |
| 16:23.38 | mick_linux | but that is on my side ;) |
| 16:24.14 | FlatFoot | sounds like some of our blokes , it's the "what happens if i pull this cable" work ethic |
| 16:24.36 | brodiem | can anyone offer some insight on troubleshooting dropped calls on a PRI interface? The PRI debug logs show what appears to be a normal disconnect, and if I look in my * logs for HDLC messages, there are generally a couple of "HDLC Bad FCS (8)" messages every day, but the timestamps don't correspond even close to when the dropped call occurs |
| 16:24.57 | mick_linux | FlatFoot: LOL |
| 16:25.03 | mick_linux | sooo very true |
| 16:25.38 | *** join/#asterisk saftsack (n=saftsack@p54A7FEDB.dip.t-dialin.net) |
| 16:26.38 | [TK]D-Fender | FlatFoot: Yeah its a great little box. |
| 16:27.02 | FlatFoot | good looking for a solution for Emergency Number failover |
| 16:27.04 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 16:27.17 | [TK]D-Fender | mattfletcher: Actually the SPA-400 looks like the best deal of the bunch, but I can't corroborate its effectiveness with * |
| 16:27.37 | brodiem | anyone? |
| 16:27.41 | FlatFoot | ah well night all time for tea |
| 16:28.41 | [TK]D-Fender | brodiem: Pastebin "cat /proc/interrupts" |
| 16:28.54 | [TK]D-Fender | brodiem: And your zaptel.conf |
| 16:30.11 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 16:30.27 | brodiem | [TK]D-Fender, thanks: http://pastebin.ca/177681 |
| 16:30.44 | mick_linux | i can't seem to find any info on encrypting VOIP calls |
| 16:31.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:31.20 | brodiem | [TK]D-Fender, span 2 (25-47) is the PRI |
| 16:31.41 | afrosheen | mick_linux, we saw problems like this with dying fxo modules on our tdm400 |
| 16:31.41 | brodiem | i guess that was obvious though, lol |
| 16:32.04 | mick_linux | afrosheen: sure you mean me? |
| 16:32.07 | afrosheen | mick_linux, about encryption, generally you shove the calls through a hardware vpn |
| 16:32.09 | [TK]D-Fender | brodiem: you have 2 cards AND a USB all sharing 1 IRQ! |
| 16:32.19 | afrosheen | mick_linux, yeah you were talking about getting weird noises and rapid busies |
| 16:32.20 | [TK]D-Fender | brodiem: Massive and obvious problem. |
| 16:32.34 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 16:32.37 | mick_linux | afrosheen: ooh, ok :) |
| 16:32.48 | afrosheen | mick_linux, here's some coffee buddy |
| 16:32.55 | mick_linux | heh |
| 16:33.44 | anonymouz666 | If I have 2 ATAs and a telephone with display, using SIP when I place call to other ATA, the ATA will pass the callerID to the display? |
| 16:33.49 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:34.33 | brodiem | [TK]D-Fender, thanks, I wonder if there is a kernel option I can set at compile time to change the irq mapping? |
| 16:34.42 | brodiem | or should i just try disabling usb to at least get the usb off of there |
| 16:34.56 | [TK]D-Fender | anonymouz666: If * has CID to pass on and the ATA supports it. |
| 16:35.12 | [TK]D-Fender | brodiem: This is something you should be doing in your BIOS. |
| 16:35.32 | brodiem | [TK]D-Fender, i'm not sure if it matters, but previously we were using a channelized T1 (just plain fxs) on span1 and never had an issue, it was only after switching to using the PRI |
| 16:35.35 | jeremy_g | what is asterisk call management cdr for? |
| 16:36.14 | jeremy_g | and how do i change cdr mode e.g. i have configured the cdr_mysql.conf and all the db,table...,installed modules, now how do i tell asterisk to please log to my mysql db instead of cdr file |
| 16:36.20 | afrosheen | ~cdr |
| 16:36.22 | jbot | hmm... cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 16:36.29 | brodiem | [TK]D-Fender, also, i'm using intel e1000 which I've heard of problems with, but doesn't look to be any IRQ misses |
| 16:36.36 | jeremy_g | i know that. |
| 16:36.38 | jeremy_g | :D |
| 16:36.46 | brodiem | unless the problems with e1000 extend beyond that |
| 16:36.48 | afrosheen | well you asked what it was for :_ |
| 16:36.54 | [TK]D-Fender | brodiem: You're running out of wrong things to do on that server you know..... |
| 16:37.00 | jeremy_g | how to change cdr mode |
| 16:37.03 | *** join/#asterisk saftsack (n=saftsack@p54A7FEDB.dip.t-dialin.net) |
| 16:37.05 | afrosheen | mode? |
| 16:37.05 | [TK]D-Fender | brodiem: E1000 is a no-no |
| 16:37.26 | jeremy_g | cdr mode { mysql,pgsql,odbc...} current its plain >cdr status |
| 16:37.34 | jeremy_g | currently |
| 16:37.53 | brodiem | [TK]D-Fender I knew about e1000, but the guys at dell told me they don't use them anymore, but of course it came with e1000. It's been very stable though, and only experiencing dropped calls a couple months ago after moving to a PRI |
| 16:37.58 | *** join/#asterisk saftsack (n=saftsack@p54A7FEDB.dip.t-dialin.net) |
| 16:38.49 | [TK]D-Fender | brodiem: I was in the same boat when my company when live with *. Constant bouncing, fax problems, echo, the works. |
| 16:39.20 | brodiem | [TK]D-Fender fax and echo are problems for me too =/, could both of those be contributed by e1000? |
| 16:39.48 | brodiem | the echo really only came about w/ the PRI too |
| 16:39.58 | brodiem | always far end echo |
| 16:40.55 | afrosheen | jeremy_g, I'm digging around for info... |
| 16:41.17 | afrosheen | jeremy_g, you installed the asterisk-addons correct? |
| 16:42.03 | afrosheen | jeremy_g, http://www.voip-info.org/wiki-Asterisk+cdr+mysql has instructions |
| 16:42.28 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:42.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:42.43 | jeremy_g | afrosheen:no, just had to load the cdr module from asterisk :) (its happening to me third time that i am forgetting to load the relevant module) awww |
| 16:42.54 | afrosheen | yeah that's listed in the wiki ;) |
| 16:43.01 | jeremy_g | hehe |
| 16:43.39 | *** join/#asterisk phacka (n=maros@chello082119111107.chello.sk) |
| 16:43.45 | phacka | quit |
| 16:43.45 | afrosheen | has anyone had problems with 1.2.1.1 and chan_zap acting up? |
| 16:45.49 | *** join/#asterisk phacka_ (n=phacka@chello082119111107.chello.sk) |
| 16:46.29 | phacka_ | hello everybody |
| 16:46.53 | mercestes | Ohaiu Gozaimasu. |
| 16:46.56 | phacka_ | i am trying to set E1 PRI numbering type on outgoing calls |
| 16:47.28 | *** join/#asterisk souphead (n=soup@dns2.cascom.ca) |
| 16:47.45 | phacka_ | but it seems asterisk sends national format regardless how i set internationalprefix/national prefix |
| 16:48.19 | phacka_ | it is Digium TE410P (2nd Gen) |
| 16:48.23 | mercestes | phacka_: From memory, did you set the switchtype in zapata.conf for your e1?? |
| 16:48.37 | phacka_ | yes, to euroisdn |
| 16:48.43 | phacka_ | that's what we have here |
| 16:49.02 | mercestes | Did you do a full reboot after changing it? |
| 16:49.26 | phacka_ | mercestes: i tried setting 00 and + for international prefix |
| 16:49.44 | phacka_ | mercetes: yes, multiple restarts |
| 16:49.50 | mercestes | I haven't had to do much E1 stuff so I'm trying to cover the obvious..;) |
| 16:50.35 | phacka_ | problem is that no matter how i set up asterisk, pbx on the other side sees national format only |
| 16:51.03 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.6) |
| 16:51.50 | JT | phacka_: did you set both zapata.conf and zaptel.conf to euroisdn? |
| 16:52.02 | phacka_ | JT:yes |
| 16:52.16 | JT | ran ztcfg, etc? |
| 16:52.45 | phacka_ | JT: of course, defaultzone=de |
| 16:52.55 | JT | hmm |
| 16:52.59 | JT | i have no idea then |
| 16:53.14 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 16:53.14 | phacka_ | JT: do you have it working ? |
| 16:53.18 | JT | no |
| 16:54.51 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 16:54.52 | mercestes | phacka_: http://copilotconsulting.com/mail-archives/asterisk.2005/msg00614.html |
| 16:54.55 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 16:54.59 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 16:55.25 | mercestes | What version of asterisk, phacka_? |
| 16:56.45 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 16:57.13 | phacka_ | mercestes: 1.2.7.1 |
| 16:57.40 | phacka_ | mercestes: and i need it for outgoing CLID |
| 16:58.04 | hwt | has anyone experienced the same problems as in this case? http://bugs.digium.com/view.php?id=5747&nbn=4 |
| 16:58.10 | hwt | is there a patch available? |
| 16:58.18 | hwt | (asterisk peering with nortel mcs5200) |
| 16:58.35 | mercestes | YOu can set the outbound CID before you send it, can't you? |
| 16:58.54 | mercestes | set(CallerID(Number) = 00$CallerID(number) |
| 17:00.50 | phacka_ | mercestes: it works |
| 17:01.19 | phacka_ | mercestes: but pbx still sees number format as national |
| 17:02.22 | phacka_ | mercestes: so if you set 0036234984 and try to call abroad, national prefix will be added |
| 17:04.05 | phacka_ | i guess i'try to upgrade asterisk and libpri and we'll see |
| 17:07.03 | mercestes | Try 1.2.9 |
| 17:07.19 | mercestes | That's "stable" for me....1.2.10 I rolled back from and I Haven't tried 1.2.11. |
| 17:07.24 | mercestes | You could give it a go if you wish tho... |
| 17:08.07 | bkw__ | 1.2.9 has a security bug |
| 17:08.13 | bkw__ | not recommended to run that one |
| 17:08.14 | mercestes | Oh? |
| 17:08.20 | mercestes | Give 1.2.11 a shot then. |
| 17:08.20 | bkw__ | yes |
| 17:08.34 | phacka_ | i'll go with 1.2.12.1 |
| 17:08.35 | bkw__ | wait its not a bug... its a "feature" |
| 17:08.38 | Qwell | or you could run 1.2.12.1... |
| 17:08.46 | bkw__ | yo Qwell |
| 17:08.52 | Qwell | bkw__: You rang? |
| 17:10.00 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:10.05 | flujan | hi guys... :) |
| 17:11.05 | flujan | I need help with asterisk. :D I need to execute a different dialplan based in the EXTEN lenght |
| 17:11.14 | flujan | is it possible? to have a IF in the dial plan? |
| 17:12.08 | mercestes | GotoIf |
| 17:14.47 | mercestes | YOu could also do different commands using different types of pattern matching. |
| 17:14.49 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 17:14.53 | mercestes | _xxx,1,Do(this) |
| 17:15.01 | mercestes | _xxxx,1,Do(this) etc. |
| 17:18.10 | *** join/#asterisk redox- (n=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com) |
| 17:20.36 | *** join/#asterisk astar` (n=astar@ANantes-154-1-91-86.w86-210.abo.wanadoo.fr) |
| 17:22.32 | flujan | hi merceste, but to evaluate the exten variable size... ? is it possible? something like _.X(GotoIf ${EXTEN}.length =3, Dial(SIP/${EXTEN) else, Dial(Zap/g1/{EXTEN}) |
| 17:23.01 | [TK]D-Fender | GotoIf($[${LEN(${MYVAR})}<5]?short:long) |
| 17:23.09 | [TK]D-Fender | flujan: See above |
| 17:23.28 | flujan | :) HI [TK]D-Fender . :) |
| 17:23.55 | flujan | and the short and long statements... How can I substitute it? |
| 17:24.31 | [TK]D-Fender | flujan: Substitute with what? Just jump to another priority and do whatever the hell you want |
| 17:24.50 | [TK]D-Fender | GotoIf($[${LEN(${MYVAR})}<5]?10:20) |
| 17:25.02 | flujan | [TK]D-Fender, so the short and long are the priority number... :) |
| 17:25.08 | flujan | thanks i will try it |
| 17:25.23 | mercestes | doesn't even have to be a number..could use short and long |
| 17:25.25 | [TK]D-Fender | flujan: Or labels. You need to go read THE BOOK. |
| 17:25.31 | [TK]D-Fender | ~book |
| 17:25.32 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:27.39 | *** join/#asterisk jmoore (n=jmoore@65.172.31.40) |
| 17:27.46 | flujan | ok [TK]D-Fender thanks for the help |
| 17:29.24 | wunderkin | who is the mix networks guy? is that blizrage? |
| 17:29.27 | wunderkin | er |
| 17:31.29 | wunderkin | anyone else with a voip provider? what about their brothers? :) |
| 17:32.42 | *** join/#asterisk Nitesh (n=Nitesh@65.48.63.178) |
| 17:32.59 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 17:33.30 | jmoore | I've been using asterisk for a few weeks now in a lab enviroment. I'm not considering replacing an existing avaya PBX with asterisk. Have any of you move from a Com PBX? If so what feature do you not have an asterisk that you wish were available? |
| 17:35.13 | jayk- | does asterisk come with a fastbusy sound byte? how do i play the fastbusy tone if a line is busy? |
| 17:35.30 | stoffell | jayk-: you could use the "playtones" function.. |
| 17:35.42 | stoffell | jmoore: uhm, it's free? extra features != extra cost |
| 17:35.57 | *** join/#asterisk Das_Tech (n=DasTech@d47-69-194-47.col.wideopenwest.com) |
| 17:36.16 | jayk- | k |
| 17:36.31 | Das_Tech | anyone here good with polycom config files I need to know where to set the dns |
| 17:37.25 | *** join/#asterisk plumbus (n=plumbus@unaffiliated/plumbus) |
| 17:40.43 | *** join/#asterisk Ebola (n=Ebola@user-54463be2.lns2-c12.dsl.pol.co.uk) |
| 17:41.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:48.10 | *** join/#asterisk rosivelt (n=rosivelt@201008239182.user.veloxzone.com.br) |
| 17:49.16 | rosivelt | hi all |
| 17:49.19 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 17:49.30 | bhrobinson | has anyone used the free 729 codec? |
| 17:50.41 | rosivelt | anyone help me with x100p and hangup? |
| 17:53.09 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 17:53.25 | plumbus | rosivelt, www.asteriskguru.org |
| 17:54.16 | rosivelt | i already make there |
| 17:54.30 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 17:54.46 | rosivelt | none of choice there resolve my problem |
| 17:54.51 | THX2000 | Anyone know why touchtones wouldn't be recognized on incoming calls when using the ulaw codec, but work fine when using gsm? |
| 17:55.34 | [TK]D-Fender | Das_Tech: You don't set that in config files. Thats bootrom territory. you either set it in DHCP or in manual settings on the phone itself before booting the application. |
| 17:56.31 | [TK]D-Fender | THX2000: Maybe you were using inband with ulaw and the quality was shaky and with GSM you're forced to use an out-of-band format whose transmission is guaranteed |
| 17:56.40 | *** join/#asterisk sakimustafa (n=sakimust@202.133.14.226) |
| 17:57.27 | THX2000 | is there a way i can force an out-of-band w/ ulaw? |
| 17:57.39 | THX2000 | relaxdtmf=no maybe? |
| 17:58.48 | rosivelt | i have trunk of 2 asterisks then when one of parts disconnect the other dont detect and the line stay busy |
| 17:58.49 | Qwell | THX2000: dtmfmode=rfc2833 |
| 17:59.05 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:59.08 | THX2000 | i set that for my trunk in sip.conf and no dice |
| 17:59.17 | Qwell | THX2000: set it on both sides |
| 18:00.24 | THX2000 | Maybe im misunderstanding the solution, but if this is for an incoming call how would i do it for both sides? |
| 18:00.50 | *** join/#asterisk jhildebrand (n=hildjj@corp-fw-main.jabber.com) |
| 18:01.57 | jhildebrand | i understand that the gtalk stuff isn't released yet, but i'm getting a core dump when trying to use it. perhaps it's a configuration issue? |
| 18:02.05 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:02.15 | TripleFFFF | how you actualy specifiy a trunk verison you want ?when you Checkout |
| 18:02.26 | jhildebrand | i just grabbed head. |
| 18:02.37 | jhildebrand | yesterday-ish. |
| 18:02.38 | TripleFFFF | i mean lets say i want revision X from trunk |
| 18:02.41 | TripleFFFF | how can i do that |
| 18:03.28 | jhildebrand | TripleFFFF: sorry, thought you were asking about my issue. have you already checked out, or are you checking out from scratch? |
| 18:04.16 | jhildebrand | if it's new, just "svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk" for example. |
| 18:04.28 | TripleFFFF | hehee |
| 18:04.29 | TripleFFFF | thanks |
| 18:04.35 | jhildebrand | you can go to the http url in a browser to see what your options are. |
| 18:06.05 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:08.33 | pjz | anyone mess with PoE with polycoms? |
| 18:08.56 | [TK]D-Fender | pjz : Yes. |
| 18:09.01 | pjz | I've got a netgear switch (FSM7352PS) that's supposed to do PoE, and a polycom 501 which is supposed to take PoE |
| 18:09.09 | rosivelt | how can i configure one PABX with asterisk? (hangup) |
| 18:09.12 | pjz | ...but I plug it in and get nada |
| 18:09.49 | [TK]D-Fender | pjz: You need their special cable to do PoE on it. it has an in-line circuit for PoE negociation. |
| 18:10.04 | pjz | [TK]D-Fender: PoE *negotiation* ? |
| 18:10.12 | [TK]D-Fender | pjz: Did yours come with a funny Cat5 wire with a jack for a power brick in it? |
| 18:10.20 | pjz | [TK]D-Fender: yeah |
| 18:10.38 | *** join/#asterisk ardor (n=vircuser@las-static-66.18.135.148.mpowercom.net) |
| 18:10.46 | [TK]D-Fender | pjz: Correct, 802.3af PoE has to be negociated between the switch & device. You don't just throw power down the line you know.... you could fry stuff.... |
| 18:11.07 | pjz | [TK]D-Fender: so you're saying I need a different special cable? |
| 18:11.07 | [TK]D-Fender | pjz: Then you don't have the PoE cable, and no you can't make it. |
| 18:11.14 | [TK]D-Fender | pjz: Exactly |
| 18:11.34 | pjz | [TK]D-Fender: hrm. presumably there's someplace that sells them standalone? |
| 18:12.54 | [TK]D-Fender | pjz : Plenty : http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-38040166656.htm |
| 18:14.18 | pjz | [TK]D-Fender: ah! thanks |
| 18:16.57 | pjz | [TK]D-Fender: same as http://www.voipsupply.com/product_info.php?products_id=699 ? |
| 18:17.00 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:17.13 | ardor | I'm wondering if its possable confence IAX channels and still have some level of control during the confernce. Some leve of Control, I would like to beable to sell that User x hit keypad 5 and beable to play back a Sound File |
| 18:19.05 | ardor | I looked at the meetme background AGI, and it looks like that would work if I was using zap channels instead of IAX. |
| 18:19.25 | CunningPike | Does anyone know if the GOTO_ON_BLINDXFER variable works in 1.2, or is it only in Trunk? |
| 18:19.31 | *** join/#asterisk jeffgus (n=jeffgus@38.119.60.2) |
| 18:19.50 | ardor | and I watched the call from the manager and I dont see when the user presses keys. |
| 18:21.21 | [TK]D-Fender | pjz : Yup |
| 18:22.23 | pjz | [TK]D-Fender: Thanks |
| 18:23.51 | mercestes | ~The Book |
| 18:23.54 | mercestes | ~the book |
| 18:24.01 | [TK]D-Fender | ~book |
| 18:24.03 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:24.03 | mercestes | Damnit, J-bot.... |
| 18:24.04 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 18:24.08 | mercestes | Yay, thanks, D-Fender. |
| 18:24.15 | [TK]D-Fender | mercestes: ywc |
| 18:26.13 | rosivelt | anyone expert in hangup? please!! |
| 18:26.45 | mercestes | I hang up on lots of people, rosivelt, what's the issue? |
| 18:26.50 | intralanman | sure, i hand up on people all the time |
| 18:26.51 | intralanman | lol |
| 18:26.58 | intralanman | dammit, i'm too slow |
| 18:27.43 | rosivelt | <PROTECTED> |
| 18:27.56 | mercestes | ... |
| 18:27.59 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 18:28.14 | mercestes | How long does the line stay hung? |
| 18:28.42 | rosivelt | sometimes is short sometimes forever |
| 18:28.57 | mercestes | Could try ending with a "Hangup()" after the dial/voicemail, etc. |
| 18:29.03 | mercestes | SHouldn't call the next line until "Dial" is done. |
| 18:29.13 | stoffell | rosivelt: using bristuff? |
| 18:29.20 | mercestes | Wait, I don't think * keeps going in exten's when you hang up on Dial. |
| 18:30.03 | rosivelt | wait |
| 18:30.41 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:30.46 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:30.52 | intralanman | kinda depends which side hangs up |
| 18:33.26 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-cbfae1ae448b3e76) |
| 18:36.12 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 18:38.58 | rosivelt | stoffell:what's bristuff? |
| 18:39.41 | rosivelt | i use x100p |
| 18:40.05 | anonymouz666 | X100P suckz ballz |
| 18:40.13 | *** join/#asterisk gJon (n=jellis@ip72-204-240-138.ph.ph.cox.net) |
| 18:41.27 | rosivelt | i have in each asterisk one x100p connected with one pabx traditional |
| 18:42.26 | stoffell | rosivelt: u don't know it so you don't use it ;) it's okay |
| 18:43.21 | rosivelt | ok |
| 18:45.54 | *** part/#asterisk rokerr (n=rowan@38.99.162.187) |
| 18:46.43 | aptura | who is rob lilo |
| 18:46.54 | Qwell | aptura: founder of pdpc/freenode |
| 18:47.15 | aptura | I see is he moving on to something new? |
| 18:47.21 | Juggie | no, he died |
| 18:47.26 | aptura | ohh sorry |
| 18:47.33 | aptura | natrual or accident? |
| 18:47.44 | Juggie | accident |
| 18:47.47 | aptura | mmm |
| 18:47.51 | *** part/#asterisk mercestes (n=merceste@216.54.143.2) |
| 18:48.18 | rosivelt | mercestes: to use hangup after dial doesn't resolv |
| 18:48.34 | Juggie | aptura, http://en.wikipedia.org/wiki/Rob_Levin |
| 18:48.36 | aptura | Just like when I watched the tribute to Steve Irwin last night it bothered me alot. I am more of a outdoor person who loves adventure then here at the terminal :) |
| 18:49.19 | *** join/#asterisk jeffgus (n=jeffgus@38.119.60.2) |
| 18:49.45 | KranZ | aptura: did you cry? |
| 18:49.52 | aptura | no |
| 18:50.05 | afrosheen | just a little |
| 18:50.07 | KranZ | don't lie |
| 18:50.15 | aptura | but just wattery eyed |
| 18:50.18 | afrosheen | wow, look at the tears on this 'ere crock |
| 18:50.27 | afrosheen | 'e's a big one too |
| 18:50.28 | aptura | He made me laugh so many times |
| 18:50.28 | aptura | :) |
| 18:50.55 | KranZ | i watched it too, he was one of the few good men |
| 18:50.58 | aptura | I drive a land cruiser just like he does :) Its new for me cannot wait to take it out back |
| 18:51.29 | KranZ | ~trunk |
| 18:51.31 | jbot | [trunk] my trunk my trunk; my lovely asterisk trunk (check it out) |
| 18:51.39 | aptura | He came to oregon I think to find rattle snakes then to florida at pensacola to show off his crocks to the Army rangers. |
| 18:51.44 | hmmhesays | wow |
| 18:51.51 | [TK]D-Fender | KranZ: EW! |
| 18:52.01 | KranZ | heh |
| 18:52.11 | rosivelt | jbot: do you know deal with hangup? |
| 18:52.12 | jbot | yes, I know deal with hangup. |
| 18:52.16 | *** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 18:52.25 | [TK]D-Fender | KranZ: That slutty bitch does nothing but annoy me now.... |
| 18:52.56 | aptura | I just wonder how the fate of the australia zoo will do without him he was the universe there. |
| 18:52.57 | KranZ | fergie? |
| 18:53.04 | [TK]D-Fender | KranZ: Correct |
| 18:53.13 | justdave | is it possible to set up MeetMe options for a conference room on a per-room basis rather than having to specify them on the dial plan for the room you dial into? |
| 18:53.14 | KranZ | yeah, she's so hot, but lost appeal recently |
| 18:53.15 | Qwell | aptura: life goes on |
| 18:53.20 | justdave | (such as from meetme.conf) |
| 18:53.27 | aptura | true |
| 18:53.29 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 18:53.36 | rosivelt | <PROTECTED> |
| 18:54.08 | KranZ | [TK]D-Fender: would checking out trunk be close to trying 1.4? |
| 18:54.14 | Qwell | KranZ: very |
| 18:54.32 | [TK]D-Fender | KranZ: Wouldn't know... I only use FTP releases. |
| 18:54.50 | justdave | we have lots of conference rooms, and a single extension number you dial to get any of them, then you dial your conference room number. If I set up another extension with different options on it, then people could bypass options by using the other extension |
| 18:55.08 | rosivelt | jbot: when one ramal disconnect the other doesn't detect |
| 18:55.17 | [TK]D-Fender | justdave: Then don't let meetme as the room, YOU do it in your dialplan instead |
| 18:55.18 | Cubber | I have a question about an error I am getting when I try to access the online module repository in freepbx, no one on the #freepbx channel has been able to help me, and web searches have not yieled any results. |
| 18:55.22 | [TK]D-Fender | ask* |
| 18:55.34 | Cubber | The error is a bit big so I wanted to make sure it is ok to post here before I did |
| 18:55.40 | aptura | lilo did the right thing to ride a bike as I do this alot but have come in close contact with cars before. |
| 18:56.03 | [TK]D-Fender | Cubber: Don't expect much help here.... go ask on a FreePBX forum somewhere |
| 18:56.10 | *** part/#asterisk Das_Tech (n=DasTech@d47-69-194-47.col.wideopenwest.com) |
| 18:56.12 | Cubber | ok thanks |
| 18:56.20 | justdave | [TK]D-Fender: yeah, so that means setting up about 100 extensions for conference rooms. bah :) |
| 18:56.38 | *** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 18:56.39 | KranZ | justdave: dont want dynamic meetme? |
| 18:57.09 | justdave | KranZ: hmm? |
| 18:57.18 | KranZ | nvm |
| 19:01.38 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:02.37 | justdave | hmm, ok. so I can set up a dialplan for the current extension so that instead of sending to MeetMe it prompts for a room number then transfers to the direct-dial extension for that room |
| 19:03.01 | justdave | then I can have a prefix-match extension that uses that part after the prefix as the room number, with the general settings |
| 19:03.11 | justdave | and override specific extensions for the ones we want special settings on |
| 19:03.29 | justdave | that could work |
| 19:04.57 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 19:06.05 | [TK]D-Fender | justdave: wow... and you figured it out ALL by yourself! |
| 19:06.25 | *** join/#asterisk hfern (n=hfern@adsl-71-159-128-187.dsl.rcsntx.sbcglobal.net) |
| 19:06.31 | justdave | heh |
| 19:07.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:07.53 | hfern | hello, I'm using * under Gentoo Linux and the latest available version is 1.0.11. I'm interested in upgrading to 1.2.11, but it's still masked. Does anyone have experience with newer versions under Gentoo? |
| 19:09.17 | [TK]D-Fender | hfern: Screw Ports & packages and jsut compile like everyone else |
| 19:09.24 | *** join/#asterisk kamileon (n=kamileon@c-71-207-204-15.hsd1.al.comcast.net) |
| 19:09.44 | intralanman | hfern: i use gentoo myself |
| 19:09.53 | intralanman | and [TK]D-Fender has it right |
| 19:09.56 | *** join/#asterisk pagec (n=pagec@64-252-108-252.adsl.snet.net) |
| 19:09.57 | intralanman | screw portage |
| 19:10.01 | *** join/#asterisk clive- (n=pirch@dsl-145-28-211.telkomadsl.co.za) |
| 19:10.11 | intralanman | just get the source and compile it yourself |
| 19:10.13 | pagec | intralanman: portage is great, what is the problem? |
| 19:10.21 | intralanman | it is great |
| 19:10.23 | intralanman | for some things |
| 19:10.35 | [TK]D-Fender | Just like binary packages on most distros.... |
| 19:10.36 | pagec | intralanman: o, ok, i thought you ment in general lol |
| 19:10.50 | intralanman | nah, in general it's awesome |
| 19:11.01 | intralanman | that's my main reason for picking gentoo |
| 19:11.05 | Qwell | except when the maintainers of baselayout screw up |
| 19:11.13 | clive- | Hi all, I have a question, , .compiling zaptel gives me an error that kernel ources are not found, but "yum install kernel-source" does not seem to work....any suggestions ? |
| 19:11.14 | intralanman | heheh |
| 19:11.22 | Qwell | That's happened to me TWICE in like 3 months :p |
| 19:11.34 | [TK]D-Fender | clive-: Make sure you've got the headers as well |
| 19:11.36 | hfern | So 1.2.11 is working fine under Gentoo? |
| 19:11.37 | Juggie | clive-, uname -a, paste it here. |
| 19:11.41 | Qwell | They move a script from one place to another, and forget to update the net.lo config, heh |
| 19:11.49 | clive- | headers....:?? |
| 19:11.55 | Juggie | clive-, uname -a, paste it here. |
| 19:11.56 | pagec | intralanman: cannot be the same install setup (ok like 3 lines difference) for ppc64 and x86 and ppc64/32ul |
| 19:12.07 | *** join/#asterisk plumbus (n=plumbus@unaffiliated/plumbus) |
| 19:12.07 | clive- | Juggie: Linux localhost.localdomain 2.6.9-42.ELsmp #1 SMP Sat Aug 12 09:39:11 CDT 2006 i686 i686 i386 GNU/Linux |
| 19:12.22 | Juggie | clive-, yum install kernel-smp-source |
| 19:12.31 | Juggie | er, kernel-smp-devel |
| 19:12.33 | Juggie | i mean |
| 19:12.43 | Juggie | well it may differ on your distro, but centos is devel |
| 19:13.02 | Juggie | which, you appear to be running anyways :) or RHEL4 |
| 19:13.11 | clive- | Juggue I have done that...very strange |
| 19:13.28 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 19:13.45 | pagec | anyone here using Polycom 501s and have a problem where only the person using the phone get horrible sound? and this problem only happens sometimes? the caller on the other end hears them fine, they just cannot hear the caller well |
| 19:13.56 | Juggie | clive-, centos or rhel4? |
| 19:13.59 | clive- | juggie I have the kernel-smp-devel installed |
| 19:14.03 | clive- | its Centos 4.4 |
| 19:14.36 | Juggie | clive-, please pastebin your output from zaptel make into pastebin.a |
| 19:14.39 | Juggie | *pastebin.ca |
| 19:14.47 | *** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:15.29 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 19:16.06 | *** join/#asterisk SZ^ (n=eparker@cerberus1.smartechcorp.net) |
| 19:16.37 | SZ^ | Anyone available for what I hope is a simple question? |
| 19:16.46 | Juggie | dont ask to ask |
| 19:17.00 | SZ^ | I wasn't asking to ask, I was seeing if anyone was actually awake here |
| 19:17.04 | clive- | Juggie http://pastebin.ca/177837 |
| 19:17.24 | Juggie | sometimes, so shoot |
| 19:17.26 | SZ^ | Thoght it was a lil rude to just jump in and start right up aking questions, sorry :) |
| 19:17.47 | Juggie | clive-, rpm -qa|grep kernel |
| 19:17.51 | Juggie | pastebin what you see. |
| 19:18.04 | SZ^ | I'm using AddQueueMember to add members to a queue. Is there a way to auto log them out if they do not answer? |
| 19:18.26 | Qwell | SZ^: No, but you can use autopause |
| 19:18.43 | Qwell | which means they won't get any calls |
| 19:19.11 | clive- | Juggie: http://pastebin.ca/177842 |
| 19:19.27 | SZ^ | Qwell: Interesting... So if they are the only one in the queue, and autopause is setup, will it then see noone in queue and send the user to voicemail if configured that way? |
| 19:19.37 | Qwell | SZ^: pretty much |
| 19:19.46 | [TK]D-Fender | pagec: The only negative feedback I've gotten on Polycom audio quality is on IP 430's with their stock firmware. What are you running on your IP 501? |
| 19:20.00 | SZ^ | Qwell: Thanks for the info, might be back if I can't figure it out :) |
| 19:21.45 | Juggie | clive-, your kernel-smp-devel version does not match your installed kernel version |
| 19:22.49 | clive- | Juggie , thanks...hmm, wondering how to get the same version |
| 19:24.43 | *** join/#asterisk Kongsted (n=Kongsted@0x55530b2e.adsl.cybercity.dk) |
| 19:24.47 | hmmhesays | ugh, women drive me insane |
| 19:24.54 | Kongsted | hey |
| 19:24.56 | intralanman | clive-: recompile your kernel using the version you have of the source |
| 19:25.18 | Juggie | clive-, you should just run a yum update and upgrade everything to the latest version |
| 19:25.32 | intralanman | yeah, or that |
| 19:25.32 | Kongsted | someone in here have a big know-how around queue's??? :-) |
| 19:25.49 | [TK]D-Fender | hmmhesays: No, just into lumber :) |
| 19:26.00 | hmmhesays | heh |
| 19:26.08 | clive- | Juggie...sounds like a god idea, recompilingthe kernel myself is a bit daunting |
| 19:26.18 | Juggie | indeed. |
| 19:26.19 | hmmhesays | Kongsted: what do you want to do? |
| 19:26.33 | clive- | Juggie, thanks |
| 19:26.34 | hmmhesays | eh, kernel compiles aren't tough |
| 19:26.40 | Kongsted | I want a priority queue... |
| 19:26.49 | intralanman | nah, i try to do one a day if i can ;) |
| 19:26.59 | clive- | hmmmhesays lol |
| 19:27.15 | hmmhesays | what are we laughing at? |
| 19:27.40 | Kongsted | The phone have to start ringing at #1, and a few sec. later at #2, and again a few sec. later #3 and so on.... |
| 19:27.56 | Juggie | clive-, yum install kernel-smp-devel-2.6.9-42.EL.x86_64 |
| 19:28.13 | hmmhesays | i don't know if you can do that with app queue |
| 19:28.16 | Juggie | er, remove _64 since your not running 64bit |
| 19:28.29 | Kongsted | app queue? |
| 19:29.10 | Juggie | clive-, yum install kernel-devel-2.6.9-42.0.2.EL |
| 19:29.10 | Jeekay | if I set 'callerid' in my sip phone definition, do I need to also do the SetCIDName() thing in my dialout dialplan? |
| 19:29.11 | Juggie | should do it |
| 19:29.23 | hmmhesays | you shouldn't |
| 19:29.41 | hmmhesays | oh man tasha yar just died |
| 19:29.44 | Juggie | Jeekay, no. |
| 19:30.43 | Jeekay | Next question :) if I have both a register => in my sip.conf with a password and an authname, and I have my sip definition of [voipprovider], what is the effect of putting 'userame' and 'secret' in that [voipprovider] sip section? |
| 19:30.50 | pagec | [TK]D-Fender: bootrom 3.2.1 app 1.6.7 |
| 19:31.08 | [TK]D-Fender | pagec: is it on all calls? |
| 19:31.09 | hmmhesays | Jeekay: same as with a sip phone |
| 19:31.21 | ManxPower | Jeekay, register is used for registration, the [voiporovider] is used for incoming or outgoing calls |
| 19:31.21 | Jeekay | Which is? :) |
| 19:31.25 | pagec | [TK]D-Fender: no, it is intermittant, mostly not, but when it is, it is on the whole call |
| 19:31.26 | intralanman | Jeekay: they might make you auth on invites too |
| 19:31.29 | clive- | Juggie I just did a yum update....have to wait for 47megs to trickle down to me here in south africa |
| 19:31.31 | Jeekay | but you dont send auth details on every invite do you? |
| 19:31.40 | clive- | lol |
| 19:31.41 | Juggie | clive-, ok good luck. |
| 19:31.41 | intralanman | with some providers you do |
| 19:31.45 | Jeekay | more to the point, if they did require that would I expect to see an error different to '404' if I try to invite ? |
| 19:31.47 | clive- | Thanks for your help |
| 19:31.50 | [TK]D-Fender | pagec: Is this with cals involving the PSTN only? What about polycom-polycom calls? |
| 19:31.51 | Juggie | np |
| 19:31.51 | ManxPower | Jeekay, yes. ALL a registration does is tell the remote server what your ip address is. |
| 19:31.58 | Jeekay | oh cool |
| 19:32.03 | ManxPower | NORMALLY the auth info would be the same for the two. |
| 19:32.18 | Juggie | if your ip is static and your sip provider supports it you can just define your ip w/ them |
| 19:32.21 | Juggie | and not register |
| 19:32.37 | Juggie | so registration is optional. |
| 19:32.46 | pagec | [TK]D-Fender: they don't do poly to poly so I couldn't tell you that. they have a completely digital connection and this happens to calls that terminate to cell phones too (so there is no analog at all) |
| 19:33.05 | Jeekay | so if i define username and secret in the [voipp] section, those auth details are sent with every invite? |
| 19:33.10 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 19:34.01 | [TK]D-Fender | pagec: How does it get to the cell phone? That mind you sounds like a large number of points of failure.... next time a call comes in that sounds bad, put that call on hold, start another to another Polycom. If its good, then go back and listen. If thats still bad, transfer the call to another Polycom. |
| 19:34.53 | pagec | [TK]D-Fender: it also happens iwth land lines, though i didn't try using another line to call out and see what happens |
| 19:35.45 | *** join/#asterisk ssseven (n=telefoni@ACD05B56.ipt.aol.com) |
| 19:35.54 | ssseven | Hi, can I use asterisk as an E1-ISDN-Switch, which transparently switches B and D channels across multiple E1-lines? |
| 19:35.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:36.10 | [TK]D-Fender | pagec: Well I just gave you a specific scenario as a guide to trouble shooting any audio problem to help ID the actual source. I've never heard of anything intermittant about SIP phone audio quality that wasn't network congestion related. That of course isn't supposed to be the phones problem anyways |
| 19:36.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:37.58 | Jeekay | looking at the packet capture of y invites, there is no authentication being sent in them |
| 19:38.23 | intralanman | Jeekay: then don't put the auth details in the peer |
| 19:38.33 | ssseven | My question is too trivial? |
| 19:39.17 | Jeekay | then how do I auth with each invite? |
| 19:39.17 | stoffell | ssseven: you want take incoming ISDN and get them out on PRI ? |
| 19:39.19 | pagec | [TK]D-Fender: ty, i will try that. it is virtually impossible that it is network congestion (unless the switch is on the fritz or somethign) |
| 19:39.52 | ManxPower | pagec, how about duplex issues |
| 19:39.52 | intralanman | Jeekay: they challenge you, if they deem it appropriate |
| 19:39.57 | Jeekay | ahh |
| 19:40.05 | Jeekay | that's the bit i had misunderstood :) thanks |
| 19:40.08 | pagec | ManxPower: what do you mean? |
| 19:40.45 | pagec | ManxPower: Asante PoE switch running 100mbs full duplex |
| 19:41.10 | ManxPower | pagec, switch thinks it's full duples, device things is is 1/2 duplex |
| 19:41.13 | ssseven | I want E1_Out -> ASTERISK_E1_IN and then ASTERISK_E1_OUT -> PUBLIC_SWITCH in a way that all progress/proceeding/ISDN-messages are directed from PUBLIC_SWITCH to E1_OUT as if E1_Out is directly connected with PUBLIC_SWITCH. |
| 19:41.27 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 19:41.43 | pagec | i have nearly an identical configuration with another asterisk pbx, but using 601 phones (that server does have a digium card instead of using ztdummy, but i have uhci usb so i assume it is fine) |
| 19:42.11 | ssseven | and vice-versa. |
| 19:42.35 | pagec | ManxPower: interesting thought, how do i see what the phone thinks the duplex is? |
| 19:42.45 | ManxPower | pagec, no idea |
| 19:43.02 | ManxPower | pagec, using polycom? |
| 19:43.11 | pagec | ManxPower: polycom 501s |
| 19:43.25 | ManxPower | pagec, turn OFF CDP in the boot config |
| 19:43.26 | afrosheen | my favorite |
| 19:43.37 | Qwell | polycoms use CDP? |
| 19:43.52 | ManxPower | Qwell, by default |
| 19:43.55 | Qwell | weird |
| 19:44.23 | pagec | ManxPower: is CDP is the .cfg files anywhere or do I actually need use the phones buttons to do it? |
| 19:44.43 | ManxPower | pagec, phone buttons |
| 19:45.13 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 19:45.13 | stoffell | phone buttons? can't you do that with provisioning? |
| 19:45.18 | pagec | ManxPower: what does CDP stand for? |
| 19:45.23 | pagec | stoffell: i was hoping :( |
| 19:45.25 | Qwell | pagec: Cisco Discovery Protocol |
| 19:45.36 | pagec | Qwell: oh, i c |
| 19:46.46 | pagec | yeah grep cdp * on the phones config files shows only log strings, ah well, guess i'll have to do it manually |
| 19:46.55 | Qwell | pagec: grep -i? |
| 19:47.13 | ManxPower | pagec, CDP happens even before the duplex is set |
| 19:47.50 | pagec | ManxPower: ah, the boot logs do show that CDP is diabled for every boot |
| 19:48.10 | ManxPower | pagec, check to be sure. |
| 19:48.30 | ManxPower | losing audio frequently is a classis issue of a CDP discovery happening during a call |
| 19:48.56 | *** join/#asterisk sting3r_ (n=sting3r@63.99.54.130) |
| 19:49.14 | sting3r_ | can someone please help me with cepstral integration |
| 19:49.17 | pagec | ManxPower: for the whole call or just for a small part of it? |
| 19:49.37 | ManxPower | SMALL PART |
| 19:50.17 | pagec | ManxPower: yeah this is a problem that lasts from the beginning to the end of the call in question, which makes me think that transient issues (like network quality) aren't at fault |
| 19:51.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:51.47 | pagec | ManxPower: something that happens intermittantly on a per call basis :( |
| 19:56.04 | ssseven | Nobody knows whether asterisk can switch b and d channels transparently over several E1? For example: One channel of one E1 is switched to another channel of another E1 in a way that all b and d channel messages remain untouched and are just switched over like with a Layer-2-switch? |
| 19:58.03 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 20:02.04 | clive- | ssseven there is a product called voxstream (I think) which can do that |
| 20:02.43 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 20:03.58 | ssseven | Thank you for this hint. I hoped asterisk can do it natively as it should not be too complicated as it is stupid data-copying with very minor modifications. |
| 20:04.44 | clive- | ssseven maybe asterisk can do that, I am not sure |
| 20:05.39 | Corydon-w | It might be able to do that, but only with NFAS |
| 20:06.09 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 20:06.16 | c4t3l | has anyone effectivley used ENUM for VOIP calling? |
| 20:06.33 | ssseven | What a pity! |
| 20:06.48 | ssseven | It would have saved me many many $$$ |
| 20:06.56 | Corydon-w | ssseven: You can't use NFAS? |
| 20:07.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:07.30 | Corydon-w | with a primary/backup D channels? |
| 20:07.41 | intralanman | c4t3l: yes |
| 20:08.42 | Corydon-w | ssseven: that's just how it works, really. A PRI circuit cannot switch to channels not under its control |
| 20:08.44 | ssseven | I have to ask whether I can define one D-chan for many E1. |
| 20:08.57 | Corydon-w | ssseven: that's a question for your telco provider |
| 20:09.14 | *** join/#asterisk PeteAtTCC (i=pete@dslB-3.tcc.on.ca) |
| 20:09.18 | Corydon-w | ssseven: Asterisk can do that, yes, by using NFAS |
| 20:09.36 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:10.01 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 20:10.10 | ssseven | Thank you! Have you any hint how to do it in the dialplan (nono, I don't need a step-by-step guide), just an asterisk command name to enable nfas in asterisk? |
| 20:10.28 | PeteAtTCC | Hi guys, thanks in advance for any answers I may receive. I am looking at using Asterisk as a dedicated Voicemail solution - nothing else. There will be no SIP phones on the inside, no extensions (other than mailboxes). Is there some decent documentation for this? |
| 20:10.30 | Corydon-w | ssseven: it's an option in zapata.conf |
| 20:10.53 | ssseven | Thank you, I will dive into then docs. |
| 20:11.30 | PeteAtTCC | It will be connected to a softswitch using a SIP line, which will host mailboxes for several people. I would like to set up the mailboxes to use the same number as in the incoming calling number, so people do not need to enter a mailbox number. |
| 20:12.40 | PeteAtTCC | I have been looking for a few days for documentation on using asterisk as a dedicated voicemail solution, but most docs seem geared at using internal extensions for checking VM. |
| 20:13.01 | PeteAtTCC | I hate to ask for help if there is documentation, so I am really just looking for pointers as to where to start looking effectively, thanks again. |
| 20:13.01 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-974d725ff587f514) |
| 20:13.02 | hwt | has anyone experienced the same problems as in this case? http://bugs.digium.com/view.php?id=5747&nbn=4 |
| 20:13.05 | hwt | (asterisk peering with nortel mcs5200) |
| 20:13.12 | hwt | is there a patch available? |
| 20:13.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-30.red.bezeqint.net) |
| 20:13.25 | hwt | or is it hard to make a patch with some basic c skills? |
| 20:13.58 | Corydon-w | PeteAtTCC: please read the variables documentation |
| 20:14.24 | PeteAtTCC | Corydon-w Will do, thx! |
| 20:15.33 | bkruse | hwt: it depends on the extent of the problem your making the patch for. sometimes it is an easy fix |
| 20:18.36 | hwt | bkruse: i basically just want it to send a 200 OK instead of a 401. |
| 20:19.22 | *** join/#asterisk lorinc (n=ang@caracas-2543.adsl.interware.hu) |
| 20:21.27 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 20:28.21 | hwt | i have another problem too. |
| 20:28.26 | hwt | I have a setup with two SER proxies, that works as registrars and SIP proxies. The ATA-boxes reinvites an Asterisk server to reach the PSTN. |
| 20:28.36 | hwt | The Asterisk server uses two different gateways to reach the PSTN (through SIP), and the different gateways are selected through dialplan matching in Asterisk. |
| 20:28.47 | hwt | The different gateways are reached through different logical network interfaces, that carries the traffic on the Internet, a localnet and a VPN. |
| 20:28.52 | hwt | (all egress traffic) |
| 20:29.04 | hwt | The problem occurs when an invite is sent through all the gateways but the "primary" (the Internet one, with the network default route). In the SDP message, it |
| 20:29.07 | hwt | will always tell the gateway on the other end the IP address of the first network interface. |
| 20:29.10 | hwt | How can I fix this behaviour? |
| 20:29.21 | hwt | SIP only setup. |
| 20:29.28 | mercestes | Hey, the Polycom Training has this nifty view of hte config files where it reads them as the XML elements they are....what do I open the Polycom config files with to view them in that manner? |
| 20:31.01 | [TK]D-Fender | mercestes: You need whatever software they are using. |
| 20:31.47 | mercestes | [TK]D-Fender: BRILLIANT! Thanks! I'll DL it now. |
| 20:32.01 | [TK]D-Fender | </sarcasm> |
| 20:32.03 | [TK]D-Fender | :D |
| 20:32.29 | mercestes | *nodnod* hehehe..:) |
| 20:32.58 | *** part/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 20:33.07 | [TK]D-Fender | mercestes: If you're on windows go to www.fileforum.com and look for an XML browser |
| 20:33.12 | [TK]D-Fender | (editor) |
| 20:33.17 | hwt | how is this solved in least cost call routing scenaros? it ought to be possible? |
| 20:35.18 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 20:36.45 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 20:38.58 | [TK]D-Fender | ok, heading home. BBIAB |
| 20:43.05 | bkruse | hwt: i see what you mean |
| 20:44.50 | hwt | bkruse: do you know how to fix it? |
| 20:44.59 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:45.00 | hwt | i can't find any relevant options in sip.conf |
| 20:49.06 | fall0ut | hrm |
| 20:49.16 | PakiPenguin | hi there |
| 20:49.18 | fall0ut | anybody do any billing/info working with AMA (BAF) records? |
| 20:49.24 | PakiPenguin | can someone please take a look at this http://pastebin.com/790834 |
| 20:49.26 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 20:49.57 | PakiPenguin | the call never goes to the middle of the queue ( channel 3 , 4 ,5 ) always uses channel 1 , otherwise it drops the call to the congestion line , if channel 1 is busy |
| 20:56.51 | *** join/#asterisk xomplex (i=jumbo@HSE-Montreal-ppp338551.sympatico.ca) |
| 20:57.10 | xomplex | can I ask from some help? |
| 20:57.13 | xomplex | *for |
| 20:57.26 | CunningPike | ~suggestions |
| 20:57.27 | jbot | well, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ... |
| 20:57.46 | *** join/#asterisk SplasPood (n=jwb@208.122.10.30) |
| 20:58.26 | xomplex | does anyone know how to configure trixbox for vbuzzer.com ? |
| 20:58.45 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:59.42 | xomplex | it would be a huge boost for guys from Toronto to be able to use it - free calls to toronto |
| 21:00.59 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 21:02.43 | xomplex | someone tried it with no luck > http://www.vbuzzer.com/forum/viewtopic.php?t=315&start=15 |
| 21:03.34 | mercestes | xomplex: #freepbx |
| 21:03.38 | mercestes | ~freepbx |
| 21:03.40 | jbot | i guess freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 21:03.41 | afrosheen | yeah oops |
| 21:03.58 | xomplex | ok tks |
| 21:04.01 | mercestes | np |
| 21:04.03 | *** join/#asterisk Samoied (n=Samoied@201.21.232.88) |
| 21:04.15 | xomplex | lol, microsoft BOB |
| 21:05.09 | mercestes | :) |
| 21:06.21 | xomplex | ok but gui or no gui, i think there are problems with asterisk and vbuzzer |
| 21:06.36 | xomplex | i shouldn't use the word trixbox *yuck* but i'm honest |
| 21:07.20 | *** join/#asterisk crochat (i=crochat@84-74-146-28.dclient.hispeed.ch) |
| 21:08.05 | *** join/#asterisk azzie (n=az@azzie.net) |
| 21:08.23 | mercestes | It's still trixbox..not asterisk. |
| 21:08.33 | *** join/#asterisk coolhp (n=Toshiba@modemcable239.6-80-70.mc.videotron.ca) |
| 21:08.38 | mercestes | it's asteriskesque.... |
| 21:08.40 | mercestes | granted. |
| 21:08.50 | coolhp | Good afternoon all. |
| 21:08.56 | mercestes | sup, coolhp? |
| 21:09.15 | coolhp | I am having a very strange issue with 2 boxes trying to talk using IAX2 trunking. |
| 21:09.26 | coolhp | I get... how should I put it... One Way Chopping ? |
| 21:09.36 | coolhp | Basically, Audio from A to B is perfect. |
| 21:09.52 | coolhp | Audio from B to A is chopping... I hear a fraction of audio every second. |
| 21:10.10 | coolhp | (Running SVN Trunk) |
| 21:10.26 | mercestes | Given the nature of IAX2 I would eliminate the IAX2 channel between the two boxes and focus on isolating one box or the other. |
| 21:10.35 | mercestes | Have you tried terminating a call from each box outside of the IAX2 trunk? |
| 21:10.43 | coolhp | Yup |
| 21:10.51 | mercestes | perfect I'm assuming? |
| 21:10.56 | coolhp | Of course :-) |
| 21:10.58 | coolhp | hehehehehe |
| 21:11.02 | mercestes | had to ask. |
| 21:11.06 | coolhp | That's why this is driving me insane ;-) |
| 21:11.30 | mercestes | yea, I'm there now..;) What codecs are you using on IAX and are your peers SIP, ZAP, or something else? |
| 21:11.46 | coolhp | ulaw |
| 21:12.00 | coolhp | On all legs. |
| 21:12.03 | mercestes | To SIP ulaw peers?? |
| 21:12.16 | coolhp | Yeah |
| 21:12.28 | coolhp | Basically, the Setup goes : |
| 21:12.35 | mercestes | ... and you can call these same end point peers (both ways) from within the boxes with no problems?? |
| 21:12.55 | coolhp | SIP User(A) -> Asterisk Server ---- IAX ---> Asterisk Server -> PRI |
| 21:13.07 | coolhp | You are correct. |
| 21:13.22 | coolhp | The SIP leg of the call works great. |
| 21:13.27 | mercestes | I *hate* to point at IAX...can you setup a sipconnect between the two boxes and try it via SIP?? |
| 21:13.29 | coolhp | The TDM leg is fine. |
| 21:13.52 | coolhp | I could :-)... I was hoping I wouldnt need to because of nat traversal :-) |
| 21:14.09 | mercestes | What is IAX running across? T1's? |
| 21:14.59 | coolhp | Dedicate Internet E1 -> Internet -> Dedicate 100Mbit Fast Ethernet. |
| 21:15.09 | coolhp | The Internet leg is actually ok. |
| 21:15.19 | coolhp | The same E1 is being used to carry SIP calls to Vonage. |
| 21:15.23 | coolhp | And they all work like a charm. |
| 21:15.24 | mercestes | IAX2 just runs over 4569 tho.... |
| 21:15.33 | mercestes | yea, but the path from * box1 to * box2 is different. |
| 21:16.02 | mercestes | if you can get into the routers, check for errors on the routers...and try to run ping tests between teh two boxes to make sure some 3rd party backbone carryier isn't screwing you. |
| 21:16.05 | coolhp | Yeah... thought about that too... I ran some bandwidth stress testing. |
| 21:16.15 | coolhp | Checked. |
| 21:16.21 | mercestes | do an mtr between the two boxes. |
| 21:16.30 | mercestes | box 1 mtr box 2 and box2 mtr box1 |
| 21:16.38 | coolhp | Checking |
| 21:16.48 | mercestes | make sure they're simliar routes as well. |
| 21:19.06 | coolhp | Routes are similar. |
| 21:19.12 | coolhp | Although not identical |
| 21:19.17 | coolhp | But quite close. |
| 21:19.29 | mercestes | clean pings and low jitter on both routes? |
| 21:19.40 | coolhp | 40ms RTD. |
| 21:19.45 | coolhp | max 10ms jitter |
| 21:20.26 | mercestes | *thinks* |
| 21:20.42 | coolhp | (I think I just hit the winner in the tough cookie department :-) ) |
| 21:21.03 | *** part/#asterisk azzie (n=az@azzie.net) |
| 21:21.26 | mercestes | Yea....it would be utterly rediculous for me to suggest that IAX2 has some port/nat issue unless Mr. Spencer lied to us. Because everything should be running across 4569 |
| 21:21.57 | coolhp | And it is.... I actually see 2 way traffic. |
| 21:22.00 | coolhp | Oh... |
| 21:22.13 | coolhp | Jitterbuffer=on/off makes no difference. |
| 21:22.24 | coolhp | Trunk=on/off makes no difference either. |
| 21:22.32 | mercestes | what type of phones is it?? |
| 21:22.39 | *** join/#asterisk Jason559-55 (n=Jason@adsl-75-10-104-33.dsl.frs2ca.sbcglobal.net) |
| 21:22.58 | file | outgoing it might not be on port 4569, if your NAT is assigning it some other outbound port that it expects packets back on |
| 21:23.02 | coolhp | Tried X-Lite / Cisco 7940 / Idefisk (so see if it might have been the SIP side) |
| 21:23.29 | mercestes | file: true, it would nat to a seperate port, but it should be fairly standard Inet traffic then, not this sip port debauchery that goes on. |
| 21:23.39 | coolhp | Agreed file : However, that would affect the call itself... not just choppy audio. |
| 21:23.49 | *** join/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 21:23.52 | Jason559-55 | I was wondering if anyone can shed some light on a problem I'm having... I've setup and seem to have asterisk working.. but, when I attempt to conntect via a softphone (x-lite) it says "registering" for awhile and returns error 408, time out... Any suggestions on how I could resolve this? |
| 21:24.22 | coolhp | Jason : What does your asterisk console say ? |
| 21:24.53 | file | choppy audio could mean a few things, depends on the circumstances |
| 21:25.10 | Jason559-55 | something about 8776 terminates |
| 21:25.23 | Jason559-55 | opps, terminated not terminates |
| 21:25.53 | coolhp | Jason : That usually means its an authentication problem of some sort... verify your sip.conf file. |
| 21:26.19 | coolhp | I'm going to try something : |
| 21:26.27 | coolhp | Going to try and register a phone with * Box 2 |
| 21:26.33 | coolhp | And call the phone on * Box 1 |
| 21:26.38 | coolhp | Sip -> IAX -> Sip |
| 21:27.22 | mercestes | alright |
| 21:28.13 | Jason559-55 | I just tried connecting again and receive on on screen message on the asterisk server. those messages could have been because the server is "open" via DMZ ? |
| 21:28.57 | Jason559-55 | My sip.conf is pretty basic, can I post it here or should I do that somewhere else? |
| 21:29.57 | mercestes | pastebin.ca is suggested/preferred. |
| 21:29.58 | *** join/#asterisk linagee (n=na@unaffiliated/linagee) |
| 21:30.58 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 21:31.08 | Jason559-55 | Ok, here it is: http://pastebin.ca/177973 |
| 21:32.37 | *** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) |
| 21:32.41 | diablopico | hello |
| 21:33.10 | *** join/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 21:33.45 | Jason559-55 | mercestes, did you see that pasted file? |
| 21:33.51 | diablopico | is anyone familiar with a call working perfectly till about 5 or 6 minutes are past , then getting really bad jitter ? |
| 21:35.14 | X-Rob_ | IT's the damn kangaroos. |
| 21:35.16 | X-Rob_ | in asterisk. |
| 21:35.27 | X-Rob_ | they get bored after a while, then start hopping around, causing jitter. |
| 21:35.54 | diablopico | is there any way to tame the kangaroos ? |
| 21:36.19 | diablopico | i just need to keep them from causing jitter |
| 21:36.21 | [TK]D-Fender | Jason559-55 : Please read the channel topic |
| 21:37.45 | diablopico | really ,, can i get a straight answer ? |
| 21:38.26 | hmmhesays | not much you can do about jitter on the public internet |
| 21:38.47 | hmmhesays | traceroute and find out if you have a bad router somewhere |
| 21:38.52 | Telamon | diablopico: Does it happen with local calls (sip <-> sip on the LAN of your Asterisk box)? |
| 21:38.54 | coolhp | Ok... same problem when using SIP -> IAX -> SIP but this time its intermittent. |
| 21:39.24 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 21:39.36 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 21:41.32 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
| 21:42.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:42.40 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:45.09 | *** part/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 21:45.15 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 21:45.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:45.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:46.55 | hmmhesays | i don't want, anybody else, when I think about you I touch myself |
| 21:47.11 | file | hmmhesays: good to know |
| 21:51.44 | CunningPike | I KNEW IT! hmmhesays is really Britney Spears |
| 21:52.40 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 21:53.41 | Strom_C | that was the Divinyls, silly |
| 21:54.31 | hmmhesays | lol |
| 21:55.30 | [TK]D-Fender | A song to which one only finds fat chicks dancing to *shudder* |
| 22:00.45 | bhrobinson | question.... I have incoming sip lines that are using my ZAP card. What is the exten-> info I need it to put in to use the next avail zap channel? |
| 22:00.52 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:01.21 | [TK]D-Fender | bhrobinson : assign your zap channels into a group and use the group to pick a line |
| 22:01.26 | ManxPower | bhrobinson, in /etc/asterisk/zapata.conf you put channels into groups, then you Dial(Zap/g1/whatever) |
| 22:01.26 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 22:01.45 | ManxPower | \/me looks at [TK]D-Fender |
| 22:01.53 | [TK]D-Fender | :D |
| 22:01.56 | bhrobinson | what file do I place the groups |
| 22:02.15 | bhrobinson | thanks |
| 22:02.16 | [TK]D-Fender | bhrobinson : ManxPower's response answered that :) |
| 22:02.21 | bhrobinson | saw |
| 22:02.50 | bhrobinson | will that be the group= line? |
| 22:03.40 | bhrobinson | mine says group=1, so is that g1? |
| 22:03.45 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 22:03.48 | diclophis | howdy all |
| 22:04.19 | diclophis | so whats the latest and greatest in the community? |
| 22:04.32 | nortex | g1 or G1 depending on the order (asending or descending) you want to pick up. |
| 22:05.35 | diclophis | what is the order thing...? |
| 22:05.50 | diclophis | what are the factors that decide what order you might want to use? |
| 22:05.54 | ManxPower | remember the group= line MUST be above the channel= lines you want it to apply to. |
| 22:06.17 | diclophis | since it takes the first free line in the group... why does it matter if it starts at the top of the list or bottom? |
| 22:07.22 | nortex | diclophis, Not sure of a good answer for that, just that it is an option. |
| 22:07.44 | diclophis | hmm |
| 22:07.57 | diclophis | myself.. i mix it up in the dialplan |
| 22:08.30 | diclophis | anyone attend von in boston? |
| 22:09.02 | bhrobinson | awesome guys!!! |
| 22:09.04 | nortex | I do it on a server that handles my PRI, the out going calls I set to G1 because I'm paranoid of the telco and my asterisk box picking up the same channel at the same time :) |
| 22:09.06 | bhrobinson | that worked perfect |
| 22:10.06 | nortex | diclophis, Didn't make VON, but there were some cool press releases and articles. |
| 22:10.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:11.28 | diclophis | yea the digium appliance looks sweet |
| 22:11.38 | diclophis | hey.. so whats with caller id.. and cellphones |
| 22:12.25 | diclophis | for some reason the caller id is not showing up as the real cell phone number the call came from |
| 22:15.23 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:16.23 | russellb | Asterisk SVN-branch-1.4-r43376M built by russell @ badunkadunk on a i686 running Linux on 2006-09-20 22:15:08 UTC |
| 22:17.48 | file | russellb: I love the name of that box |
| 22:17.53 | russellb | file: me too :D |
| 22:18.12 | russellb | i like the version tag |
| 22:18.50 | file | :D |
| 22:18.54 | C6Vette | WAIT... 1.4????? WHERE WHERE WHERE????? |
| 22:19.02 | russellb | :-p |
| 22:19.07 | file | NOWHERE! |
| 22:19.10 | russellb | nowhere yet ... beta soon ... |
| 22:19.19 | file | you saw nothing |
| 22:19.35 | russellb | and by soon, I mean within the next hour :) |
| 22:19.38 | C6Vette | awwww.....!!!!!!?????? |
| 22:20.22 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:21.57 | *** join/#asterisk SplasPood (n=jwb@208.122.10.36) |
| 22:22.11 | *** join/#asterisk asternick (i=asdas@222.126.38.77) |
| 22:23.22 | asternick | guys please help on this thing i have a server behind nat and got 2 client one which is in remote and the other is inside the NAT |
| 22:23.36 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:24.04 | asternick | when ever i call the client inside the NAT they can't hear abything |
| 22:24.08 | asternick | when ever i call the client inside the NAT they can't hear anything |
| 22:24.42 | asternick | is there anyone around? |
| 22:24.42 | Druken | that would be a problem with your nat/firewall |
| 22:24.55 | C6Vette | yup that will happen. Dont use nat. |
| 22:24.58 | asternick | what should i do |
| 22:25.09 | Druken | not put asterisk behind a nat.... |
| 22:25.22 | C6Vette | sip doesnt play well with nat. |
| 22:25.28 | [TK]D-Fender | Druken : How very useful.... |
| 22:25.39 | aptura | asternick check port forwarding and sip.conf in general. |
| 22:25.44 | Druken | [TK]D-Fender: why thank you :) |
| 22:25.56 | [TK]D-Fender | </sarcasm> |
| 22:25.57 | asternick | what are the ports used by asterisk |
| 22:26.03 | [TK]D-Fender | almost forgot the closing tag! |
| 22:26.37 | Druken | :) |
| 22:27.19 | asternick | aptura can i ask what shall be the ports that i should forwards |
| 22:27.32 | Druken | hey [TK]D-Fender: what was it i was told to use for odbc earlier? |
| 22:27.36 | Druken | i'm having a brain fart |
| 22:28.54 | [TK]D-Fender | Druken : Sorry, taht was my work IRC client no logs here |
| 22:29.17 | Druken | ahh, damn.... |
| 22:29.26 | [TK]D-Fender | asternick : 5060-5070, 10000-20000 all UDP. Pleas pastebin the [general} section of your sip.conf |
| 22:30.01 | asternick | oh thank yopu for this thing |
| 22:30.07 | asternick | just gonna archive this thing |
| 22:30.17 | [TK]D-Fender | BRB |
| 22:30.36 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:30.50 | TripleFFFF | can one http://www.virbiage.com/cubix.php#techspecs download and test that softphone ? when its |
| 22:31.03 | TripleFFFF | running its like driving my mouse nuts |
| 22:31.11 | TripleFFFF | just wnna make sure its only me or eveyone |
| 22:32.06 | [TK]D-Fender | ~pb |
| 22:32.07 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 22:32.20 | russellb | lol! |
| 22:32.33 | [TK]D-Fender | TripleFFFF : Firefly (under all its name) has been one buggy POS since it came out. |
| 22:32.46 | TripleFFFF | yeah i see |
| 22:32.54 | TripleFFFF | its a RBPOS |
| 22:33.03 | TripleFFFF | im looking for a core to make our softphone on |
| 22:33.24 | TripleFFFF | and cant find 1 that works.. onyl one i remmebr oging right.. was OLD xlite.. back in the days where htey had coders.. |
| 22:33.32 | TripleFFFF | since eyebeam they just shit |
| 22:36.26 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 22:38.06 | Druken | hmm... |
| 22:38.15 | Druken | func_odbc... looks interesting.... |
| 22:38.56 | TripleFFFF | ./me nick not hamered ;) |
| 22:41.13 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
| 22:42.14 | souphead | hi |
| 22:43.05 | Druken | anyone have experince with func_odbc ? |
| 22:43.48 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:44.08 | souphead | my pci te110p card has stopped displaying it's LEDs on the back of the computer after I shutdown and restarted my asterisk box. Now when I do a ztcfg it comes up with a span not available error... |
| 22:44.14 | Druken | wiki doesn't have dick for information on func |
| 22:44.22 | Druken | func_odc |
| 22:44.57 | Corydon-w | func_odbc |
| 22:45.27 | russellb | Corydon-w: ;) |
| 22:45.41 | Corydon-w | russellb: a little bit ;-) |
| 22:46.11 | Druken | Corydon-w: feel like sharing the .conf file? i don't have one... |
| 22:46.23 | Corydon-w | There's a sample one in configs/ |
| 22:47.17 | Druken | uhmm... let me guess, it's only available in trunk ? |
| 22:47.48 | Corydon-w | There's a backport for 1.2 at http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 22:48.10 | [TK]D-Fender | souphead : modprobe zaptel and your card |
| 22:48.17 | Corydon-w | Oddly, the backport has more functionality than trunk |
| 22:51.18 | asternick | hey another question have anyone of uses MVP130 as their FXO and FXS module? |
| 22:53.08 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:53.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:55.01 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
| 22:55.18 | AsteriskMonkey | anyone here played with set paging ? |
| 22:56.35 | [TK]D-Fender | AsteriskMonkey : As in? |
| 22:57.13 | AsteriskMonkey | as in i want to page mutiple sets like intercom |
| 22:57.18 | AsteriskMonkey | :) Aastra 480is' |
| 22:57.37 | [TK]D-Fender | AsteriskMonkey : Well I've done single-phone pages. multi-phone can be done easily enough |
| 22:57.41 | AsteriskMonkey | SetVar(_ALERT_INFO=info=alert-autoanswer) |
| 22:57.44 | AsteriskMonkey | yes i know |
| 22:57.48 | AsteriskMonkey | ? |
| 22:57.48 | souphead | i did a modprobe wct1xxp and it comes up with the error on span 1, no such device or address (6) |
| 22:57.54 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 22:58.54 | [TK]D-Fender | AsteriskMonkey : Thats the means of bringing in the phone. There is an * app taht is supposed to drag a whole bunch in that I haven't tested yet. |
| 22:58.59 | AsteriskMonkey | [TK]D-Fender: you have example of multiphone? i have the wierdest bug happening.. when i try and do multi ones like dial(sip/123,12&sip/124,12) I get a prompt from asterisk to record my name really whacky |
| 23:02.40 | [TK]D-Fender | AsteriskMonkey that format is bad... |
| 23:04.42 | *** join/#asterisk daysmen3 (n=primus@host86-141-241-150.range86-141.btcentralplus.com) |
| 23:05.38 | Druken | dial(sip/123&sip/124,12) is the correct format |
| 23:06.32 | AsteriskMonkey | oh ? |
| 23:06.49 | AsteriskMonkey | ok thanks |
| 23:06.51 | AsteriskMonkey | :D |
| 23:08.35 | Druken | Corydon-w: thanks for the information.... i'm just debating if i want to make those changes or not... :) |
| 23:08.53 | Druken | i'm anything but a programmer, and HATE changing code..... |
| 23:12.44 | *** join/#asterisk re-pete (n=repete@24.96.201.72) |
| 23:12.48 | *** part/#asterisk gJon (n=jellis@ip72-204-240-138.ph.ph.cox.net) |
| 23:13.15 | *** join/#asterisk Rhiliam (n=c37d6f4@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 23:14.12 | Rhiliam | Hello, I am trying to use Monitor to record all incoming calls on an incoming context. I am having an issue, whereby it only seems to record sporadically. I have checked all the file permissions etc, and everything looks to be fine. I have posted the snippet as follows http://pastebin.ca/178071 |
| 23:16.17 | [TK]D-Fender | Rhiliam : what is ${CALLRECORD7500} ? |
| 23:16.34 | Rhiliam | a variable I have set at the top of extentions.conf |
| 23:16.43 | [TK]D-Fender | show it. |
| 23:17.02 | Rhiliam | CALLRECORD7500=/data/asterisk/recordings/9054347500/${TIMESTAMP} |
| 23:17.18 | [TK]D-Fender | Rhiliam : pastebin the whole thing |
| 23:17.22 | Rhiliam | k |
| 23:18.07 | Rhiliam | http://pastebin.ca/178074 |
| 23:18.19 | Rhiliam | I didn't pastebin the who file, as it has passwords etc.. |
| 23:18.25 | Rhiliam | whole file |
| 23:19.16 | [TK]D-Fender | Rhiliam : that is a CONSTANT, not a variable. |
| 23:19.34 | [TK]D-Fender | Rhiliam : Can you show the CLI output of a call against that. |
| 23:19.56 | Rhiliam | What command would I use to see it |
| 23:20.20 | [TK]D-Fender | please just pastebin the output of a call to that context |
| 23:20.51 | Rhiliam | I know what you are asking, I am not sure how to get the information. I am new to asterisk, so not sure how to get CLI output. |
| 23:21.20 | [TK]D-Fender | Do you not know how to access the * CLI? |
| 23:21.39 | Rhiliam | yes, asterisk -r |
| 23:21.45 | Rhiliam | hang on |
| 23:21.53 | [TK]D-Fender | Rhiliam : Thats what I want. * CLI output of some calls |
| 23:22.08 | [TK]D-Fender | Rhiliam : I want to see what Monitor is being called with |
| 23:22.48 | Rhiliam | k, place an incoming call, but nothing showed up. How do I increase the verbosity of the CLI |
| 23:22.59 | [TK]D-Fender | "set verbose 10" |
| 23:24.07 | Rhiliam | http://pastebin.ca/178083 |
| 23:25.49 | [TK]D-Fender | hmmm |
| 23:26.23 | Rhiliam | actually, I just made another call, and it seemed to overwrite the first one... |
| 23:27.13 | Rhiliam | Looks, like to TIMESTAMP is not changing during a new call... |
| 23:27.59 | Rhiliam | hence your comment about it being a constant and not a variable. I will change it now |
| 23:28.32 | *** join/#asterisk sb_mx (n=sb_mx@200.67.149.101) |
| 23:28.40 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 23:30.10 | Rhiliam | Problem fixed - thanks. |
| 23:32.19 | Rhiliam | Onto the second problem :) - I seem to have an unusual amount of static on the analog phones which are running through a Linksys PAP2T. The calls are crystal clear when plugged into the wall jack. Is there any settings within asterisk that deal with static? |
| 23:32.23 | *** join/#asterisk Heim|away (n=Heimidal@phpbb/styles/heimidal) |
| 23:33.57 | *** join/#asterisk apardo (n=apardo@87.217.146.172) |
| 23:34.34 | linagee | how do i define who the operator is with freepbx/asterisk? |
| 23:34.44 | Qwell | linagee: see topic |
| 23:35.26 | linagee | :( |
| 23:35.28 | Nugget | There's no such thing as "the operator" in asterisk. Qwell is giving you good advice. |
| 23:35.42 | [TK]D-Fender | Rhiliam : If you call from 1 port on the PAP2 to the other you hear this static? |
| 23:41.56 | hmmhesays | fan fscking tastic, i finally got asterisk running on this mipsel box |
| 23:42.35 | [TK]D-Fender | abso-fsking-lutely |
| 23:43.01 | diclophis | mipsel? |
| 23:44.09 | re-pete | ! |
| 23:44.25 | hmmhesays | diclophis: yeah |
| 23:44.39 | hmmhesays | whoa, when did asterisk get a config script? |
| 23:45.21 | hmmhesays | LOL |
| 23:45.54 | file | Edouard Montpetit! (no I have no clue why I just said that) |
| 23:46.35 | Strom_C | [TK]D-Fender: I decided to listen to someone else's music just now, so I turned off my turntable (which was playing "would I lie to you?" by Eurythmics) and turned on a webcast, which after about thirty seconds, played "would I lie to you?" by Eurythmics :) |
| 23:46.49 | file | Strom_C: creepy |
| 23:46.57 | Strom_C | uber. |
| 23:46.58 | JunK-Y | file: because u totally insane? :) |
| 23:47.11 | file | JunK-Y: that's true |
| 23:47.33 | hmmhesays | i'm listening to some guns n roses |
| 23:47.59 | [TK]D-Fender | file : Lotta good restaurants on that street :) |
| 23:48.31 | [TK]D-Fender | Strom_C : And now another tune from "Broken Record" ! |
| 23:48.42 | Strom_C | heh |
| 23:48.51 | Strom_C | is it "Would I Lie to You?" |
| 23:49.17 | file | careful, he might lie to you |
| 23:50.40 | Strom_C | now would he say something that isn't true? |
| 23:50.42 | Ryushin | So is vinyl better than CD's? |
| 23:51.00 | [TK]D-Fender | Ryushin : Better for WHAT? |
| 23:51.03 | *** join/#asterisk dasenjo (n=dasenjo@c4178-123.impsat.com.co) |
| 23:51.05 | Strom_C | Ryushin: not necessarily...I was just playing it because I was studying where best to mix the 12" single into something else |
| 23:51.22 | Ryushin | better quality of sound. |
| 23:51.56 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 23:52.06 | Ryushin | From everything that I've read, it's only better for the first 4 playings, after that, the needle has already damaged it. Unless you're using a turntable that uses a laser to read the groves. |
| 23:52.07 | [TK]D-Fender | Ryushin : Depends on a certain point of view. CD's don't degrade like most other formats. However the dynamic range of a digital recording pales to an ordinary cassette |
| 23:52.12 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 23:52.29 | Strom_C | certainly not when I buy the records for fifty cents each at Amoeba :) |
| 23:52.51 | [TK]D-Fender | Ryushin : Eddie Van Halen passed a studio tape off to a reviewer and is sounded 10x better than the final release |
| 23:53.04 | linagee | can someone help me with an asterisk dialplan line? |
| 23:53.07 | linagee | exten => o,1,GotoIf($["foo${FROM_DID}" = "foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1) |
| 23:53.10 | Strom_C | [TK]D-Fender: what year though? |
| 23:53.20 | linagee | would this dial "1"? |
| 23:53.26 | hmmhesays | bah why can't I erase in my vi |
| 23:53.27 | joat | anyone know of any stores in the DC area that sells voip equipment? |
| 23:53.38 | linagee | joat: mail order. ;) |
| 23:53.49 | joat | other than mail order :) |
| 23:54.04 | [TK]D-Fender | linagee : Your DID is wrong. |
| 23:54.28 | Ryushin | Well, was it true studio tape. 8mm? That would probably be superior. But I can't say with the new equipment being produced. I don't know enough if that is the case anymore. If you take the digital recording out to enough resolution, it will have to have better quality than analog tape. |
| 23:54.28 | linagee | [TK]D-Fender: the context of this is, user presses '0' for operator (i think o = operator) |
| 23:55.08 | [TK]D-Fender | Ryushin : No I mean a NORMAL tape! |
| 23:55.22 | Ryushin | Cassette tape? |
| 23:55.51 | [TK]D-Fender | linagee : it either jumps to "s" in [from-pstn] or to the value of ${FROM_DID} if it is not blank |
| 23:55.59 | [TK]D-Fender | Ryushin : Yes, normal high-bias tape |
| 23:56.15 | Ryushin | I find that hard to believe. I had a heck of a nice tape deck, and it in no way sound better than my first cd player. |
| 23:56.36 | Strom_C | cassette isn't that fantastic - also, that doesn't take into account the fact that the music may have been remixed, levels messed with, passed through a crappy early 80s analog-to-digital converter, etc |
| 23:56.36 | Ryushin | Maybe the difference is, I didn't have a high quality machine recording to a cassette tape. |
| 23:56.41 | Ryushin | Only what they massed produced. |
| 23:56.52 | Strom_C | it also doesnt account for the listener's preconceived bias |
| 23:57.31 | [TK]D-Fender | Ryushin : Depends what you are listening to as well. If you are recording in digital form initially then its not getting any better. However digital loses a lot of frequencies "in-between".... |
| 23:57.39 | *** join/#asterisk apardo (n=apardo@87.217.146.172) |
| 23:59.42 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-712bd895009b4428) |
| 23:59.42 | *** mode/#asterisk [+o mogorman] by ChanServ |
